[svn-commits] mjordan: branch 1.8-digiumphones r362042 - in /branches/1.8-digiumphones: ./ ...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Apr 12 13:47:20 CDT 2012
Author: mjordan
Date: Thu Apr 12 13:47:16 2012
New Revision: 362042
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=362042
Log:
Merge of several needed fixes for 1.8-digiumphones
This merges fixes for the following issues into the 1.8-digiumphones branch:
* ASTERISK-19355 - Call transfer with consultation frequently fails in cross-
linked Asterisk scenario (directmedia & sendrpid active)
* ASTERISK 19365 - Remote SIP Call legs are frequently not released in a
cross-linked Asterisk scenario (directmedia & sendrpid)
* ASTERISK-19183 - Sporadically missing connectedline event to caller channel
in directed pickup app
Modified:
branches/1.8-digiumphones/ (props changed)
branches/1.8-digiumphones/channels/chan_sip.c
branches/1.8-digiumphones/main/features.c
Propchange: branches/1.8-digiumphones/
------------------------------------------------------------------------------
svn:mergeinfo = /branches/1.8:357665,358162,359656,359706,359979,360086,360884
Modified: branches/1.8-digiumphones/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8-digiumphones/channels/chan_sip.c?view=diff&rev=362042&r1=362041&r2=362042
==============================================================================
--- branches/1.8-digiumphones/channels/chan_sip.c (original)
+++ branches/1.8-digiumphones/channels/chan_sip.c Thu Apr 12 13:47:16 2012
@@ -12238,7 +12238,7 @@
/* If init=1, we should not generate a new branch. If it's 0, we need a new branch. */
reqprep(&req, p, sipmethod, 0, init ? 0 : 1);
}
-
+
if (p->options && p->options->auth) {
add_header(&req, p->options->authheader, p->options->auth);
}
@@ -13060,7 +13060,7 @@
if (p->owner->_state == AST_STATE_UP || ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
struct sip_request req;
- if (p->invitestate == INV_CONFIRMED || p->invitestate == INV_TERMINATED) {
+ if (!p->pendinginvite && (p->invitestate == INV_CONFIRMED || p->invitestate == INV_TERMINATED)) {
reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1);
add_header(&req, "Allow", ALLOWED_METHODS);
@@ -20048,6 +20048,10 @@
if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA) {
p->invitestate = INV_CANCELLED;
transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
+ /* If the cancel occurred on an initial invite, cancel the pending BYE */
+ if (!ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
+ ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);
+ }
/* Actually don't destroy us yet, wait for the 487 on our original
INVITE, but do set an autodestruct just in case we never get it. */
} else {
@@ -20061,8 +20065,8 @@
}
/* Perhaps there is an SD change INVITE outstanding */
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
- }
- ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);
+ ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);
+ }
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) {
/* if we can't REINVITE, hold it for later */
@@ -20224,7 +20228,7 @@
int outgoing = ast_test_flag(&p->flags[0], SIP_OUTGOING);
int res = 0;
int xmitres = 0;
- int reinvite = (p->owner && p->owner->_state == AST_STATE_UP);
+ int reinvite = ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
char *p_hdrval;
int rtn;
struct ast_party_connected_line connected;
@@ -20414,10 +20418,11 @@
p->authtries = 0;
if (find_sdp(req)) {
if ((res = process_sdp(p, req, SDP_T38_ACCEPT)) && !req->ignore)
- if (!reinvite)
+ if (!reinvite) {
/* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
/* For re-invites, we try to recover */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
+ }
ast_rtp_instance_activate(p->rtp);
}
@@ -20461,9 +20466,9 @@
update_call_counter(p, DEC_CALL_RINGING);
parse_ok_contact(p, req);
/* Save Record-Route for any later requests we make on this dialogue */
- if (!reinvite)
+ if (!reinvite) {
build_route(p, req, 1, resp);
-
+ }
if(set_address_from_contact(p)) {
/* Bad contact - we don't know how to reach this device */
/* We need to ACK, but then send a bye */
@@ -20611,6 +20616,7 @@
update_call_counter(p, DEC_CALL_LIMIT);
append_history(p, "Hangup", "Got 487 on CANCEL request from us on call without owner. Killing this dialog.");
}
+ check_pendings(p);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
break;
case 415: /* Unsupported media type */
@@ -21308,8 +21314,9 @@
}
/* If this is a NOTIFY for a subscription clear the flag that indicates that we have a NOTIFY pending */
- if (!p->owner && sipmethod == SIP_NOTIFY && p->pendinginvite)
+ if (!p->owner && sipmethod == SIP_NOTIFY && p->pendinginvite) {
p->pendinginvite = 0;
+ }
/* Get their tag if we haven't already */
if (ast_strlen_zero(p->theirtag) || (resp >= 200)) {
Modified: branches/1.8-digiumphones/main/features.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8-digiumphones/main/features.c?view=diff&rev=362042&r1=362041&r2=362042
==============================================================================
--- branches/1.8-digiumphones/main/features.c (original)
+++ branches/1.8-digiumphones/main/features.c Thu Apr 12 13:47:16 2012
@@ -7314,8 +7314,6 @@
ast_connected_line_copy_from_caller(&connected_caller, &chan->caller);
ast_channel_unlock(chan);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
- ast_channel_queue_connected_line_update(chan, &connected_caller, NULL);
- ast_party_connected_line_free(&connected_caller);
ast_cel_report_event(target, AST_CEL_PICKUP, NULL, NULL, chan);
@@ -7329,6 +7327,8 @@
goto pickup_failed;
}
+ ast_channel_queue_connected_line_update(chan, &connected_caller, NULL);
+
/* setting this flag to generate a reason header in the cancel message to the ringing channel */
ast_set_flag(chan, AST_FLAG_ANSWERED_ELSEWHERE);
@@ -7353,6 +7353,7 @@
if (!ast_channel_datastore_remove(target, ds_pickup)) {
ast_datastore_free(ds_pickup);
}
+ ast_party_connected_line_free(&connected_caller);
return res;
}
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