Restoring the old logic, since working around it and fixing it seemed too complicated.

- The SIP_OUTGOING flag indicates the direction of the last transaction in the dialog.
- The initreq stores the last request in the dialog, the request that opened the
  latest transaction.

Please now retry all the 1.4 bug reports with mixed to/from headers, tags etc
in ACK, BYE, CANCEL. Thanks!

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46398 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5d3f9f94d4dadcbdbc1845b16959ec7006b5f480..937ecfbc706d704f40f792e67c038314eb67f05b 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -701,7 +701,7 @@ struct sip_auth {
 #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 */
@@ -959,7 +959,8 @@ static struct sip_pvt {
        char lastmsg[256];                      /*!< Last Message sent/received */
        int amaflags;                           /*!< AMA Flags */
        int pendinginvite;                      /*!< Any pending invite ? (seqno of this) */
-       struct sip_request initreq;             /*!< Initial request that opened the SIP dialog */
+       struct sip_request initreq;             /*!< Request that opened the latest transaction
+                                                    within this SIP dialog */
        
        int maxtime;                            /*!< Max time for first response */
        int initid;                             /*!< Auto-congest ID if appropriate (scheduler) */
@@ -6339,6 +6340,7 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p)
        /* 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);
 }
 
@@ -10406,7 +10408,7 @@ static int sip_show_channel(int fd, int argc, char *argv[])
                                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);
@@ -12993,14 +12995,14 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                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)) {