[asterisk-commits] rizzo: trunk r76231 - /trunk/channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jul 20 22:53:18 CDT 2007
Author: rizzo
Date: Fri Jul 20 22:53:17 2007
New Revision: 76231
URL: http://svn.digium.com/view/asterisk?view=rev&rev=76231
Log:
use explicit variables for things that don't need to
be stored in ast_flags. First victim is 'SIP_NO_HISTORY'
replaced by a 'do_history' field in the sip_pvt structure.
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=76231&r1=76230&r2=76231
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Jul 20 22:53:17 2007
@@ -778,7 +778,7 @@
#define SIP_PROG_INBAND_NEVER (0 << 25)
#define SIP_PROG_INBAND_NO (1 << 25)
#define SIP_PROG_INBAND_YES (2 << 25)
-#define SIP_NO_HISTORY (1 << 27) /*!< D: Suppress recording request/response history */
+#define __SIP_NO_HISTORY (1 << 27) /*!< D: Suppress recording request/response history */
#define SIP_CALL_LIMIT (1 << 28) /*!< D: Call limit enforced for this call */
#define SIP_SENDRPID (1 << 29) /*!< DP: Remote Party-ID Support */
#define SIP_INC_COUNT (1 << 30) /*!< D: Did this dialog increment the counter of in-use calls? */
@@ -983,6 +983,10 @@
ast_group_t pickupgroup; /*!< Pickup group */
int lastinvite; /*!< Last Cseq of invite */
struct ast_flags flags[2]; /*!< SIP_ flags */
+
+ /* boolean or small integers that don't belong in flags */
+ char do_history; /*!< Set if we want to record history */
+
int timer_t1; /*!< SIP timer T1, ms rtt */
unsigned int sipoptions; /*!< Supported SIP options on the other end */
struct ast_codec_pref prefs; /*!< codec prefs */
@@ -2310,7 +2314,7 @@
}
if (sip_debug_test_pvt(p))
ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ if (p->do_history)
append_history(p, "SchedDestroy", "%d ms", ms);
if (p->autokillid > -1)
@@ -2445,7 +2449,7 @@
ast_inet_ntoa(dst->sin_addr),
ntohs(dst->sin_port), req->data);
}
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
+ if (p->do_history) {
struct sip_request tmp;
parse_copy(&tmp, req);
append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"),
@@ -2478,7 +2482,7 @@
else
ast_verbose("%sTransmitting (no NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->sa.sin_addr), ntohs(p->sa.sin_port), req->data);
}
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
+ if (p->do_history) {
struct sip_request tmp;
parse_copy(&tmp, req);
append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
@@ -3871,7 +3875,7 @@
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
/* Get RTCP quality before end of call */
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
+ if (p->do_history) {
if (p->rtp)
append_history(p, "RTCPaudio", "Quality:%s", audioqos);
if (p->vrtp)
@@ -4404,7 +4408,7 @@
tmp = NULL;
}
- if (!ast_test_flag(&i->flags[0], SIP_NO_HISTORY))
+ if (i->do_history)
append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);
/* Inform manager user about new channel and their SIP call ID */
@@ -4736,7 +4740,7 @@
ast_copy_flags(&p->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&p->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
- ast_set2_flag(&p->flags[0], !recordhistory, SIP_NO_HISTORY);
+ p->do_history = recordhistory;
p->branch = ast_random();
make_our_tag(p->tag, sizeof(p->tag));
@@ -6324,7 +6328,7 @@
{
struct sip_pvt *p = data;
- ast_set_flag(&p->flags[0], SIP_NO_HISTORY);
+ p->do_history = 0; /* XXX do we need it ? isn't already all 0 ? */
return ast_string_field_init(p, 512);
}
@@ -7156,7 +7160,7 @@
add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
if (sipdebug)
add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ if (p->do_history)
append_history(p, "ReInv", "Re-invite sent");
if (t38version)
add_t38_sdp(&req, p);
@@ -7794,7 +7798,7 @@
if (!r)
return 0;
- if (r->call && !ast_test_flag(&r->call->flags[0], SIP_NO_HISTORY))
+ if (r->call && r->call->do_history)
append_history(r->call, "RegistryRenew", "Account: %s@%s", r->username, r->hostname);
/* Since registry's are only added/removed by the the monitor thread, this
may be overkill to reference/dereference at all here */
@@ -7896,7 +7900,7 @@
ast_log(LOG_WARNING, "Unable to allocate registration transaction (memory or socket error)\n");
return 0;
}
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ if (p->do_history)
append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname);
p->outboundproxy = obproxy_get(p, NULL);
@@ -12005,7 +12009,7 @@
/* No old challenge */
return -1;
}
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ if (p->do_history)
append_history(p, "RegistryAuth", "Try: %d", p->authtries);
if (sip_debug_test_pvt(p) && p->registry)
ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname);
@@ -14320,7 +14324,7 @@
p->jointcapability = p->capability;
ast_debug(1, "Hm.... No sdp for the moment\n");
}
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) /* This is a response, note what it was for */
+ if (p->do_history) /* This is a response, note what it was for */
append_history(p, "ReInv", "Re-invite received");
}
} else if (debug)
@@ -15194,25 +15198,25 @@
sip_alreadygone(p);
/* Get RTCP quality before end of call */
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) {
+ if (p->do_history || p->owner) {
char *audioqos, *videoqos, *textqos;
if (p->rtp) {
audioqos = ast_rtp_get_quality(p->rtp, NULL);
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ if (p->do_history)
append_history(p, "RTCPaudio", "Quality:%s", audioqos);
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
}
if (p->vrtp) {
videoqos = ast_rtp_get_quality(p->vrtp, NULL);
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ if (p->do_history)
append_history(p, "RTCPvideo", "Quality:%s", videoqos);
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
}
if (p->trtp) {
textqos = ast_rtp_get_quality(p->trtp, NULL);
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ if (p->do_history)
append_history(p, "RTCPtext", "Quality:%s", textqos);
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", textqos);
@@ -15890,7 +15894,7 @@
}
p->recv = sin;
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) /* This is a request or response, note what it was for */
+ if (p->do_history) /* This is a request or response, note what it was for */
append_history(p, "Rx", "%s / %s / %s", req.data, get_header(&req, "CSeq"), req.rlPart2);
if (!lockretry) {
@@ -18035,7 +18039,7 @@
}
if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
if (chan->_state != AST_STATE_UP) { /* We are in early state */
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ if (p->do_history)
append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
ast_debug(1, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip.sin_addr));
} else if (!p->pendinginvite) { /* We are up, and have no outstanding invite */
More information about the asterisk-commits
mailing list