[svn-commits] rizzo: branch rizzo/astobj2 r47948 -
/team/rizzo/astobj2/channels/chan_sip.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Wed Nov 22 15:31:39 MST 2006
Author: rizzo
Date: Wed Nov 22 16:31:39 2006
New Revision: 47948
URL: http://svn.digium.com/view/asterisk?view=rev&rev=47948
Log:
introduce a function record_history(p) to wrap testing
the SIP_NO_HISTORY flag.
Modified:
team/rizzo/astobj2/channels/chan_sip.c
Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=47948&r1=47947&r2=47948
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Wed Nov 22 16:31:39 2006
@@ -1009,6 +1009,11 @@
you know more) */
};
+static inline int record_history(const struct sip_pvt *p)
+{
+ return !ast_test_flag(&p->flags[0], SIP_NO_HISTORY);
+}
+
#ifdef USE_AO2 /* astobj2 implementation */
#include "asterisk/astobj2.h"
ao2_container *dialogs;
@@ -2240,7 +2245,7 @@
}
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 (record_history(p))
append_history(p, "SchedDestroy", "%d ms", ms);
if (p->autokillid > -1) /* cancel previous schedule, but keep the reference */
@@ -2377,7 +2382,7 @@
ast_inet_ntoa(dst->sin_addr),
ntohs(dst->sin_port), req->data);
}
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
+ if (record_history(p)) {
struct sip_request tmp;
parse_copy(&tmp, req);
append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"),
@@ -2408,7 +2413,7 @@
ast_inet_ntoa(dst->sin_addr),
ntohs(dst->sin_port), req->data);
}
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
+ if (record_history(p)) {
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);
@@ -3690,7 +3695,7 @@
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 (record_history(p)) {
if (p->rtp)
append_history(p, "RTCPaudio", "Quality:%s", audioqos);
if (p->vrtp)
@@ -4182,7 +4187,7 @@
for (v = i->chanvars ; v ; v = v->next)
pbx_builtin_setvar_helper(tmp,v->name,v->value);
- if (!ast_test_flag(&i->flags[0], SIP_NO_HISTORY))
+ if (record_history(i))
append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);
return tmp;
@@ -6786,7 +6791,7 @@
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 (record_history(p))
append_history(p, "ReInv", "Re-invite sent");
if (t38version)
add_t38_sdp(&req, p);
@@ -7517,7 +7522,7 @@
r->timeout = ast_sched_add(sched, global_reg_timeout*1000, sip_reg_timeout, r);
return 0; /* non fatal error */
}
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ if (record_history(p))
append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname);
/* Copy back Call-ID in case create_addr changed it */
ast_string_field_set(r, callid, p->callid);
@@ -12407,7 +12412,7 @@
ast_verbose("No authentication challenge, giving up for %s\n", r->hostname);
} else {
/* ok, log events, send the packet and be done */
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ if (record_history(p))
append_history(p, "RegistryAuth", "Try: %d", p->authtries);
if (sip_debug_test_pvt(p))
ast_verbose("Responding to challenge, registration to domain/host name %s\n", r->hostname);
@@ -13766,7 +13771,7 @@
if (option_debug)
ast_log(LOG_DEBUG, "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 (record_history(p)) /* This is a response, note what it was for */
append_history(p, "ReInv", "Re-invite received");
}
}
@@ -14571,18 +14576,18 @@
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
/* Get RTCP quality before end of call */
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) {
+ if (record_history(p) || p->owner) {
char *audioqos, *videoqos;
if (p->rtp) {
audioqos = ast_rtp_get_quality(p->rtp);
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ if (record_history(p))
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);
- if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
+ if (record_history(p))
append_history(p, "RTCPvideo", "Quality:%s", videoqos);
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
@@ -15268,7 +15273,7 @@
}
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 (record_history(p)) /* 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);
nounlock = 0;
@@ -17227,7 +17232,7 @@
}
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 (record_history(p))
append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
if (option_debug)
ast_log(LOG_DEBUG, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip));
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