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Merge of several needed fixes for 1.8-digiumphones

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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


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8-digiumphones@362042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Matthew Jordan
2012-04-12 18:47:16 +00:00
parent dc52cb3031
commit 149442bf11
2 changed files with 18 additions and 10 deletions
Download Patch File Download Diff File Expand all files Collapse all files

23
channels/chan_sip.c
View File

@@ -12238,7 +12238,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init,
/* 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 @@ static void update_connectedline(struct sip_pvt *p, const void *data, size_t dat
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 @@ static void check_pendings(struct sip_pvt *p)
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 @@ static void check_pendings(struct sip_pvt *p)
}
/* 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 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
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 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
}
/* 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)) {

5
main/features.c
View File

@@ -7314,8 +7314,6 @@ int ast_do_pickup(struct ast_channel *chan, struct ast_channel *target)
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 @@ int ast_do_pickup(struct ast_channel *chan, struct ast_channel *target)
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 @@ pickup_failed:
if (!ast_channel_datastore_remove(target, ds_pickup)) {
ast_datastore_free(ds_pickup);
}
ast_party_connected_line_free(&connected_caller);
return res;
}