Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv4608/channels
Modified Files:
chan_sip.c
Log Message:
use symbolic constants for RTP method flags, and add debugging output to
sip_alloc to indicate when RTP is/is not allocated (bug #3986)
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.720
retrieving revision 1.721
diff -u -d -r1.720 -r1.721
--- chan_sip.c 3 May 2005 02:16:45 -0000 1.720
+++ chan_sip.c 3 May 2005 02:32:50 -0000 1.721
@@ -131,26 +131,28 @@
#define SIP_PUBLISH 14
#define SIP_RESPONSE 100
+#define RTP 1
+#define NO_RTP 0
const struct cfsip_methods {
int id;
int need_rtp; /* when this is the 'primary' use for a pvt
structure, does it need RTP? */
char *text;
} sip_methods[] = {
- { 0, 1, "-UNKNOWN-" },
- { SIP_REGISTER, 0, "REGISTER" },
- { SIP_OPTIONS, 0, "OPTIONS" },
- { SIP_NOTIFY, 0, "NOTIFY" },
- { SIP_INVITE, 1, "INVITE" },
- { SIP_ACK, 0, "ACK" },
- { SIP_PRACK, 0, "PRACK" },
- { SIP_BYE, 0, "BYE" },
- { SIP_REFER, 0, "REFER" },
- { SIP_SUBSCRIBE, 0, "SUBSCRIBE" },
- { SIP_MESSAGE, 0, "MESSAGE" },
- { SIP_UPDATE, 0, "UPDATE" },
- { SIP_INFO, 0, "INFO" },
- { SIP_CANCEL, 0, "CANCEL" },
- { SIP_PUBLISH, 0, "PUBLISH" }
+ { 0, RTP, "-UNKNOWN-" },
+ { SIP_REGISTER, NO_RTP, "REGISTER" },
+ { SIP_OPTIONS, NO_RTP, "OPTIONS" },
+ { SIP_NOTIFY, NO_RTP, "NOTIFY" },
+ { SIP_INVITE, RTP, "INVITE" },
+ { SIP_ACK, NO_RTP, "ACK" },
+ { SIP_PRACK, NO_RTP, "PRACK" },
+ { SIP_BYE, NO_RTP, "BYE" },
+ { SIP_REFER, NO_RTP, "REFER" },
+ { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE" },
+ { SIP_MESSAGE, NO_RTP, "MESSAGE" },
+ { SIP_UPDATE, NO_RTP, "UPDATE" },
+ { SIP_INFO, NO_RTP, "INFO" },
+ { SIP_CANCEL, NO_RTP, "CANCEL" },
+ { SIP_PUBLISH, NO_RTP, "PUBLISH" }
};
@@ -2571,17 +2573,17 @@
if (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833)
p->noncodeccapability |= AST_RTP_DTMF;
strcpy(p->context, default_context);
- /* Add to list */
+ /* Add to active dialog list */
ast_mutex_lock(&iflock);
p->next = iflist;
iflist = p;
ast_mutex_unlock(&iflock);
if (option_debug)
- ast_log(LOG_DEBUG, "Allocating new SIP call for %s\n", callid);
+ ast_log(LOG_DEBUG, "Allocating new SIP dialog for %s - %s
(%s)\n", callid ? callid : "(No Call-ID)", sip_methods[intended_method].text,
p->rtp ? "With RTP" : "No RTP");
return p;
}
-/*--- find_call: Connect incoming SIP message to current call or create new
call structure */
+/*--- find_call: Connect incoming SIP message to current dialog or create new
dialog structure */
/* Called by handle_request ,sipsock_read */
static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in
*sin, const int intended_method)
{
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