[svn-commits] oej: trunk r46399 - in /trunk: ./ channels/chan_sip.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sun Oct 29 09:58:07 MST 2006
Author: oej
Date: Sun Oct 29 10:58:07 2006
New Revision: 46399
URL: http://svn.digium.com/view/asterisk?rev=46399&view=rev
Log:
- Fix the OUTGOING stuff (merge from 1.4)
- Make sure we UNREF authpeer when not needed
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://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=46399&r1=46398&r2=46399&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sun Oct 29 10:58:07 2006
@@ -702,7 +702,7 @@
#define SIP_USEREQPHONE (1 << 10) /*!< Add user=phone to numeric URI. Default off */
#define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */
#define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */
-#define SIP_OUTGOING (1 << 13) /*!< Is this an outgoing call? */
+#define SIP_OUTGOING (1 << 13) /*!< Direction of the last transaction in this dialog */
#define SIP_CAN_BYE (1 << 14) /*!< Can we send BYE on this dialog? */
#define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< Do not hangup at first ast_hangup */
#define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */
@@ -5569,16 +5569,6 @@
ast_log(LOG_DEBUG, "Strict routing enforced for session %s\n", p->callid);
}
-#ifdef SKREP
- /* Let's try to figure out the direction of this transaction within the dialog */
- /* If we're sending an ACK, we DID send the INVITE - which means outbound.
- INVITE's are outbound transactions, always
- */
- if (sipmethod == SIP_ACK || sipmethod == SIP_INVITE)
- is_outbound = TRUE;
- /* In other case's, let's follow the flow of the dialog */
-#endif
-
if (sipmethod == SIP_CANCEL)
c = p->initreq.rlPart2; /* Use original URI */
else if (sipmethod == SIP_ACK) {
@@ -6424,6 +6414,7 @@
/* Use this as the basis */
initialize_initreq(p, &req);
p->lastinvite = p->ocseq;
+ ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Change direction of this dialog */
return send_request(p, &req, XMIT_CRITICAL, p->ocseq);
}
@@ -10585,7 +10576,7 @@
ast_cli(fd, " * Subscription (type: %s)\n", subscription_type2str(cur->subscribed));
else
ast_cli(fd, " * SIP Call\n");
- ast_cli(fd, " Direction: %s\n", ast_test_flag(&cur->flags[0], SIP_OUTGOING)?"Outgoing":"Incoming");
+ ast_cli(fd, " Curr. trans. direction: %s\n", ast_test_flag(&cur->flags[0], SIP_OUTGOING) ? "Outgoing" : "Incoming");
ast_cli(fd, " Call-ID: %s\n", cur->callid);
ast_cli(fd, " Owner channel ID: %s\n", cur->owner ? cur->owner->name : "<none>");
ast_cli(fd, " Our Codec Capability: %d\n", cur->capability);
@@ -13171,14 +13162,14 @@
p->pendinginvite = seqno;
check_via(p, req);
+ copy_request(&p->initreq, req); /* Save this INVITE as the transaction basis */
if (!p->owner) { /* Not a re-invite */
- /* Use this as the basis */
- copy_request(&p->initreq, req);
if (debug)
ast_verbose("Using INVITE request as basis request - %s\n", p->callid);
append_history(p, "Invite", "New call: %s", p->callid);
parse_ok_contact(p, req);
} else { /* Re-invite on existing call */
+ ast_clear_flag(&p->flags[0], SIP_OUTGOING); /* This is now an inbound dialog */
/* Handle SDP here if we already have an owner */
if (find_sdp(req)) {
if (process_sdp(p, req)) {
@@ -14148,6 +14139,8 @@
if (!ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)) {
transmit_response(p, "403 Forbidden (policy)", req);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ if (authpeer)
+ ASTOBJ_UNREF(authpeer,sip_destroy_peer);
return 0;
}
@@ -14168,6 +14161,8 @@
if (gotdest) {
transmit_response(p, "404 Not Found", req);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ if (authpeer)
+ ASTOBJ_UNREF(authpeer,sip_destroy_peer);
return 0;
}
@@ -14176,6 +14171,8 @@
make_our_tag(p->tag, sizeof(p->tag));
if (!strcmp(event, "presence") || !strcmp(event, "dialog")) { /* Presence, RFC 3842 */
+ if (authpeer) /* We do not need the authpeer any more */
+ ASTOBJ_UNREF(authpeer,sip_destroy_peer);
/* Header from Xten Eye-beam Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/xpidf+xml */
/* Polycom phones only handle xpidf+xml, even if they say they can
@@ -14205,6 +14202,8 @@
if (option_debug > 1)
ast_log(LOG_DEBUG, "Received SIP mailbox subscription for unknown format: %s\n", accept);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ if (authpeer)
+ ASTOBJ_UNREF(authpeer,sip_destroy_peer);
return 0;
}
/* Looks like they actually want a mailbox status
@@ -14216,6 +14215,8 @@
transmit_response(p, "404 Not found (no mailbox)", req);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
ast_log(LOG_NOTICE, "Received SIP subscribe for peer without mailbox: %s\n", authpeer->name);
+ if (authpeer)
+ ASTOBJ_UNREF(authpeer,sip_destroy_peer);
return 0;
}
@@ -14225,14 +14226,18 @@
sip_destroy(authpeer->mwipvt);
authpeer->mwipvt = p; /* Link from peer to pvt */
p->relatedpeer = authpeer; /* Link from pvt to peer */
+ /* Do not release authpeer here */
} else { /* At this point, Asterisk does not understand the specified event */
transmit_response(p, "489 Bad Event", req);
if (option_debug > 1)
ast_log(LOG_DEBUG, "Received SIP subscribe for unknown event package: %s\n", event);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ if (authpeer)
+ ASTOBJ_UNREF(authpeer,sip_destroy_peer);
return 0;
}
+ /* Add subscription for extension state from the PBX core */
if (p->subscribed != MWI_NOTIFICATION && !resubscribe)
p->stateid = ast_extension_state_add(p->context, p->exten, cb_extensionstate, p);
@@ -14311,8 +14316,6 @@
if (!p->expiry)
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
}
- if (authpeer)
- ASTOBJ_UNREF(authpeer, sip_destroy_peer);
return 1;
}
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