Merged revisions 55914 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

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r55914 | file | 2007-02-21 12:18:19 -0500 (Wed, 21 Feb 2007) | 2 lines

Add a flag that indicates whether a SIP dialog is an outgoing call or not. SIP_OUTGOING originally did it but it was repurposed to the direction of the last transaction, which can cause update_call_counter to falsely decrease the wrong counters. (please don't hurt me oej) (issue #8943 reported by mdu113)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55915 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 21709d6..6ec8897 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -796,9 +796,10 @@ struct sip_auth {
 #define SIP_PAGE2_CALL_ONHOLD_INACTIVE (1 << 24)       /*!< 24: Inactive  */
 #define SIP_PAGE2_RFC2833_COMPENSATE    (1 << 25)      /*!< 25: ???? */
 #define SIP_PAGE2_BUGGY_MWI            (1 << 26)       /*!< 26: Buggy CISCO MWI fix */
-#define SIP_PAGE2_NOTEXT               (1 << 27)       /*!< 26: Text not supported  */
-#define SIP_PAGE2_TEXTSUPPORT          (1 << 28)       /*!< 27: Global text enable */
-#define SIP_PAGE2_DEBUG_TEXT           (1 << 29)       /*!< 28: Global text debug */
+#define SIP_PAGE2_NOTEXT               (1 << 27)       /*!< 27: Text not supported  */
+#define SIP_PAGE2_TEXTSUPPORT          (1 << 28)       /*!< 28: Global text enable */
+#define SIP_PAGE2_DEBUG_TEXT           (1 << 29)       /*!< 29: Global text debug */
+#define SIP_PAGE2_OUTGOING_CALL         (1 << 30)       /*!< 30: Is this an outgoing call? */
 
 #define SIP_PAGE2_FLAGS_TO_COPY \
        (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
@@ -3296,7 +3297,7 @@ static int update_call_counter(struct sip_pvt *fup, int event)
 {
        char name[256];
        int *inuse = NULL, *call_limit = NULL, *inringing = NULL;
-       int outgoing = ast_test_flag(&fup->flags[0], SIP_OUTGOING);
+       int outgoing = ast_test_flag(&fup->flags[1], SIP_PAGE2_OUTGOING_CALL);
        struct sip_user *u = NULL;
        struct sip_peer *p = NULL;
 
@@ -15962,6 +15963,8 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
                return NULL;
        }
 
+       ast_set_flag(&p->flags[1], SIP_PAGE2_OUTGOING_CALL);
+
        if (!(p->options = ast_calloc(1, sizeof(*p->options)))) {
                sip_destroy(p);
                ast_log(LOG_ERROR, "Unable to build option SIP data structure - Out of memory\n");