[asterisk-commits] oej: trunk r338755 - /trunk/channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Sep 30 14:25:40 CDT 2011
Author: oej
Date: Fri Sep 30 14:25:36 2011
New Revision: 338755
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=338755
Log:
Formatting changes only
--Denna och nedanstående rader kommer inte med i loggmeddelandet--
M channels/chan_sip.c
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=338755&r1=338754&r2=338755
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Sep 30 14:25:36 2011
@@ -19921,10 +19921,11 @@
struct ast_party_connected_line connected;
struct ast_set_party_connected_line update_connected;
- if (reinvite)
+ if (reinvite) {
ast_debug(4, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
- else
+ } else {
ast_debug(4, "SIP response %d to standard invite\n", resp);
+ }
if (p->alreadygone) { /* This call is already gone */
ast_debug(1, "Got response on call that is already terminated: %s (ignoring)\n", p->callid);
@@ -19938,8 +19939,9 @@
/* RFC3261 says we must treat every 1xx response (but not 100)
that we don't recognize as if it was 183.
*/
- if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 181 && resp != 182 && resp != 183)
+ if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 181 && resp != 182 && resp != 183) {
resp = 183;
+ }
/* For INVITE, treat all 2XX responses as we would a 200 response */
if ((resp >= 200) && (resp < 300)) {
@@ -19947,16 +19949,19 @@
}
/* Any response between 100 and 199 is PROCEEDING */
- if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING)
+ if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING) {
p->invitestate = INV_PROCEEDING;
+ }
/* Final response, not 200 ? */
- if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA ))
+ if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA )) {
p->invitestate = INV_COMPLETED;
+ }
/* Final response, clear out pending invite */
- if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite)
+ if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite) {
p->pendinginvite = 0;
+ }
/* If this is a response to our initial INVITE, we need to set what we can use
* for this peer.
@@ -19968,15 +19973,17 @@
switch (resp) {
case 100: /* Trying */
case 101: /* Dialog establishment */
- if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
+ if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p)) {
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
+ }
check_pendings(p);
break;
case 180: /* 180 Ringing */
case 182: /* 182 Queued */
- if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
+ if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p)) {
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
+ }
if (!req->ignore && p->owner) {
if (get_rpid(p, req)) {
/* Queue a connected line update */
@@ -20005,8 +20012,9 @@
}
}
if (find_sdp(req)) {
- if (p->invitestate != INV_CANCELLED)
+ if (p->invitestate != INV_CANCELLED) {
p->invitestate = INV_EARLY_MEDIA;
+ }
res = process_sdp(p, req, SDP_T38_NONE);
if (!req->ignore && p->owner) {
/* Queue a progress frame only if we have SDP in 180 or 182 */
@@ -20037,8 +20045,9 @@
break;
case 183: /* Session progress */
- if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
+ if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) {
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
+ }
if (!req->ignore && p->owner) {
if (get_rpid(p, req)) {
/* Queue a connected line update */
@@ -20063,8 +20072,9 @@
sip_handle_cc(p, req, AST_CC_CCNR);
}
if (find_sdp(req)) {
- if (p->invitestate != INV_CANCELLED)
+ if (p->invitestate != INV_CANCELLED) {
p->invitestate = INV_EARLY_MEDIA;
+ }
res = process_sdp(p, req, SDP_T38_NONE);
if (!req->ignore && p->owner) {
/* Queue a progress frame */
@@ -20084,8 +20094,9 @@
break;
case 200: /* 200 OK on invite - someone's answering our call */
- if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
+ if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) {
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
+ }
p->authtries = 0;
if (find_sdp(req)) {
if ((res = process_sdp(p, req, SDP_T38_ACCEPT)) && !req->ignore)
@@ -20136,14 +20147,16 @@
update_call_counter(p, DEC_CALL_RINGING);
parse_ok_contact(p, req);
/* Save Record-Route for any later requests we make on this dialogue */
- if (!reinvite)
+ if (!reinvite) {
build_route(p, req, 1);
+ }
if(set_address_from_contact(p)) {
/* Bad contact - we don't know how to reach this device */
/* We need to ACK, but then send a bye */
- if (!p->route && !req->ignore)
+ if (!p->route && !req->ignore) {
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
+ }
}
}
@@ -20151,10 +20164,11 @@
if (!req->ignore && p->owner) {
if (!reinvite) {
ast_queue_control(p->owner, AST_CONTROL_ANSWER);
- if (sip_cfg.callevents)
+ if (sip_cfg.callevents) {
manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
"Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
p->owner->name, "SIP", p->owner->uniqueid, p->callid, p->fullcontact, p->peername);
+ }
} else { /* RE-invite */
if (p->t38.state == T38_DISABLED || p->t38.state == T38_REJECTED) {
ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
@@ -20166,8 +20180,9 @@
/* It's possible we're getting an 200 OK after we've tried to disconnect
by sending CANCEL */
/* First send ACK, then send bye */
- if (!req->ignore)
+ if (!req->ignore) {
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
+ }
}
/* Check for Session-Timers related headers */
@@ -20213,20 +20228,23 @@
case 401: /* Www auth */
/* First we ACK */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- if (p->options)
+ if (p->options) {
p->options->auth_type = resp;
+ }
/* Then we AUTH */
ast_string_field_set(p, theirtag, NULL); /* forget their old tag, so we don't match tags when getting response */
if (!req->ignore) {
- if (p->authtries < MAX_AUTHTRIES)
+ if (p->authtries < MAX_AUTHTRIES) {
p->invitestate = INV_CALLING;
+ }
if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", sip_get_header(&p->initreq, "From"));
pvt_set_needdestroy(p, "failed to authenticate on INVITE");
sip_alreadygone(p);
- if (p->owner)
+ if (p->owner) {
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+ }
}
}
break;
@@ -20257,8 +20275,9 @@
/* Could be REFER caused INVITE with replaces */
ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- if (p->owner)
+ if (p->owner) {
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+ }
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
break;
@@ -20272,8 +20291,9 @@
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
append_history(p, "Identity", "SIP identity is required. Not supported by Asterisk.");
ast_log(LOG_WARNING, "SIP identity required by proxy. SIP dialog '%s'. Giving up.\n", p->callid);
- if (p->owner)
+ if (p->owner) {
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+ }
break;
@@ -20300,18 +20320,21 @@
if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
change_t38_state(p, T38_REJECTED);
/* Try to reset RTP timers */
+ /* XXX Why is this commented away??? */
//ast_rtp_set_rtptimers_onhold(p->rtp);
/* Trigger a reinvite back to audio */
transmit_reinvite_with_sdp(p, FALSE, FALSE);
} else {
/* We can't set up this call, so give up */
- if (p->owner && !req->ignore)
+ if (p->owner && !req->ignore) {
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+ }
pvt_set_needdestroy(p, "received 488 response");
/* If there's no dialog to end, then mark p as already gone */
- if (!reinvite)
+ if (!reinvite) {
sip_alreadygone(p);
+ }
}
break;
case 491: /* Pending */
@@ -20342,12 +20365,14 @@
case 405: /* Not allowed */
case 501: /* Not implemented */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
- if (p->owner)
+ if (p->owner) {
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+ }
break;
}
- if (xmitres == XMIT_ERROR)
+ if (xmitres == XMIT_ERROR) {
ast_log(LOG_WARNING, "Could not transmit message in dialog %s\n", p->callid);
+ }
}
/* \brief Handle SIP response in NOTIFY transaction
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