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https://github.com/asterisk/asterisk.git
synced 2026-07-06 06:43:06 -07:00
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
+23
-19
@@ -778,7 +778,7 @@ struct sip_auth {
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#define SIP_PROG_INBAND_NEVER (0 << 25)
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#define SIP_PROG_INBAND_NO (1 << 25)
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#define SIP_PROG_INBAND_YES (2 << 25)
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#define SIP_NO_HISTORY (1 << 27) /*!< D: Suppress recording request/response history */
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#define __SIP_NO_HISTORY (1 << 27) /*!< D: Suppress recording request/response history */
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#define SIP_CALL_LIMIT (1 << 28) /*!< D: Call limit enforced for this call */
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#define SIP_SENDRPID (1 << 29) /*!< DP: Remote Party-ID Support */
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#define SIP_INC_COUNT (1 << 30) /*!< D: Did this dialog increment the counter of in-use calls? */
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@@ -983,6 +983,10 @@ struct sip_pvt {
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ast_group_t pickupgroup; /*!< Pickup group */
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int lastinvite; /*!< Last Cseq of invite */
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struct ast_flags flags[2]; /*!< SIP_ flags */
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/* boolean or small integers that don't belong in flags */
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char do_history; /*!< Set if we want to record history */
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int timer_t1; /*!< SIP timer T1, ms rtt */
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unsigned int sipoptions; /*!< Supported SIP options on the other end */
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struct ast_codec_pref prefs; /*!< codec prefs */
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@@ -2310,7 +2314,7 @@ static void sip_scheddestroy(struct sip_pvt *p, int ms)
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}
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if (sip_debug_test_pvt(p))
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ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
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if (p->do_history)
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append_history(p, "SchedDestroy", "%d ms", ms);
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if (p->autokillid > -1)
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@@ -2445,7 +2449,7 @@ static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmitty
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ast_inet_ntoa(dst->sin_addr),
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ntohs(dst->sin_port), req->data);
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}
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
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if (p->do_history) {
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struct sip_request tmp;
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parse_copy(&tmp, req);
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append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"),
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@@ -2478,7 +2482,7 @@ static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittyp
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else
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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);
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}
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
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if (p->do_history) {
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struct sip_request tmp;
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parse_copy(&tmp, req);
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append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
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@@ -3871,7 +3875,7 @@ static int sip_hangup(struct ast_channel *ast)
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transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
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/* Get RTCP quality before end of call */
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
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if (p->do_history) {
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if (p->rtp)
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append_history(p, "RTCPaudio", "Quality:%s", audioqos);
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if (p->vrtp)
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@@ -4404,7 +4408,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
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tmp = NULL;
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}
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if (!ast_test_flag(&i->flags[0], SIP_NO_HISTORY))
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if (i->do_history)
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append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);
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/* Inform manager user about new channel and their SIP call ID */
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@@ -4736,7 +4740,7 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
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ast_copy_flags(&p->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
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ast_copy_flags(&p->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
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ast_set2_flag(&p->flags[0], !recordhistory, SIP_NO_HISTORY);
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p->do_history = recordhistory;
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p->branch = ast_random();
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make_our_tag(p->tag, sizeof(p->tag));
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@@ -6324,7 +6328,7 @@ static int temp_pvt_init(void *data)
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{
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struct sip_pvt *p = data;
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ast_set_flag(&p->flags[0], SIP_NO_HISTORY);
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p->do_history = 0; /* XXX do we need it ? isn't already all 0 ? */
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return ast_string_field_init(p, 512);
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}
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@@ -7156,7 +7160,7 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version)
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add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
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if (sipdebug)
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add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
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if (p->do_history)
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append_history(p, "ReInv", "Re-invite sent");
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if (t38version)
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add_t38_sdp(&req, p);
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@@ -7794,7 +7798,7 @@ static int sip_reregister(void *data)
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if (!r)
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return 0;
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if (r->call && !ast_test_flag(&r->call->flags[0], SIP_NO_HISTORY))
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if (r->call && r->call->do_history)
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append_history(r->call, "RegistryRenew", "Account: %s@%s", r->username, r->hostname);
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/* Since registry's are only added/removed by the the monitor thread, this
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may be overkill to reference/dereference at all here */
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@@ -7896,7 +7900,7 @@ static int transmit_register(struct sip_registry *r, int sipmethod, const char *
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ast_log(LOG_WARNING, "Unable to allocate registration transaction (memory or socket error)\n");
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return 0;
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}
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
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if (p->do_history)
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append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname);
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p->outboundproxy = obproxy_get(p, NULL);
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@@ -12005,7 +12009,7 @@ static int do_register_auth(struct sip_pvt *p, struct sip_request *req, enum sip
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/* No old challenge */
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return -1;
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}
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
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if (p->do_history)
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append_history(p, "RegistryAuth", "Try: %d", p->authtries);
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if (sip_debug_test_pvt(p) && p->registry)
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ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname);
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@@ -14320,7 +14324,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
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p->jointcapability = p->capability;
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ast_debug(1, "Hm.... No sdp for the moment\n");
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}
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) /* This is a response, note what it was for */
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if (p->do_history) /* This is a response, note what it was for */
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append_history(p, "ReInv", "Re-invite received");
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}
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} else if (debug)
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@@ -15194,25 +15198,25 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
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sip_alreadygone(p);
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/* Get RTCP quality before end of call */
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) {
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if (p->do_history || p->owner) {
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char *audioqos, *videoqos, *textqos;
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if (p->rtp) {
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audioqos = ast_rtp_get_quality(p->rtp, NULL);
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
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if (p->do_history)
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append_history(p, "RTCPaudio", "Quality:%s", audioqos);
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if (p->owner)
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pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
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}
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if (p->vrtp) {
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videoqos = ast_rtp_get_quality(p->vrtp, NULL);
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
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if (p->do_history)
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append_history(p, "RTCPvideo", "Quality:%s", videoqos);
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if (p->owner)
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pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
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}
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if (p->trtp) {
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textqos = ast_rtp_get_quality(p->trtp, NULL);
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
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if (p->do_history)
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append_history(p, "RTCPtext", "Quality:%s", textqos);
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if (p->owner)
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pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", textqos);
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@@ -15890,7 +15894,7 @@ static int sipsock_read(int *id, int fd, short events, void *ignore)
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}
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p->recv = sin;
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) /* This is a request or response, note what it was for */
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if (p->do_history) /* This is a request or response, note what it was for */
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append_history(p, "Rx", "%s / %s / %s", req.data, get_header(&req, "CSeq"), req.rlPart2);
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if (!lockretry) {
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@@ -18035,7 +18039,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
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}
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if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
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if (chan->_state != AST_STATE_UP) { /* We are in early state */
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
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if (p->do_history)
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append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
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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));
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} else if (!p->pendinginvite) { /* We are up, and have no outstanding invite */
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