[svn-commits] dvossel: trunk r247915 - /trunk/channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Feb 19 11:40:30 CST 2010
Author: dvossel
Date: Fri Feb 19 11:40:26 2010
New Revision: 247915
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=247915
Log:
handle_request_invite revise comment, fix coding guideline issues
I'm working with this code right now trying to analyze a deadlock.
This change is just to clean up a few things before I make a more
complex patch.
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=247915&r1=247914&r2=247915
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Feb 19 11:40:26 2010
@@ -18835,7 +18835,7 @@
char exten[AST_MAX_EXTENSION];
char context[AST_MAX_CONTEXT];
} pickup = {
- .exten = "",
+ .exten = "",
};
st_ref = SESSION_TIMER_REFRESHER_AUTO;
@@ -18988,7 +18988,7 @@
If it's not in early mode, 486 Busy.
*/
-
+
/* Skip leading whitespace */
replace_id = ast_skip_blanks(replace_id);
@@ -19036,13 +19036,12 @@
}
}
+ /* This locks both refer_call pvt and refer_call pvt's owner!!!*/
if (!error && ast_strlen_zero(pickup.exten) && (p->refer->refer_call = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
error = 1;
}
-
- /* At this point, bot the pvt and the owner of the call to be replaced is locked */
/* The matched call is the call from the transferer to Asterisk .
We want to bridge the bridged part of the call to the
@@ -19136,7 +19135,6 @@
} else if (debug)
ast_verbose("Ignoring this INVITE request\n");
-
if (!p->lastinvite && !req->ignore && !p->owner) {
/* This is a new invite */
/* Handle authentication if this is our first invite */
@@ -19155,7 +19153,7 @@
ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", get_header(req, "From"));
transmit_response_reliable(p, "403 Forbidden", req);
}
- p->invitestate = INV_COMPLETED;
+ p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_string_field_set(p, theirtag, NULL);
return 0;
@@ -19172,7 +19170,7 @@
if (process_sdp(p, req, SDP_T38_INITIATE)) {
/* Unacceptable codecs */
transmit_response_reliable(p, "488 Not acceptable here", req);
- p->invitestate = INV_COMPLETED;
+ p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_debug(1, "No compatible codecs for this SIP call.\n");
return -1;
@@ -19206,7 +19204,7 @@
ast_log(LOG_NOTICE, "Failed to place call for device %s, too many calls\n", p->username);
transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- p->invitestate = INV_COMPLETED;
+ p->invitestate = INV_COMPLETED;
}
return 0;
}
@@ -19225,14 +19223,14 @@
transmit_response_reliable(p, "484 Address Incomplete", req);
else {
char *decoded_exten = ast_strdupa(p->exten);
-
+
transmit_response_reliable(p, "404 Not Found", req);
ast_uri_decode(decoded_exten);
ast_log(LOG_NOTICE, "Call from '%s' to extension"
" '%s' rejected because extension not found.\n",
S_OR(p->username, p->peername), decoded_exten);
}
- p->invitestate = INV_COMPLETED;
+ p->invitestate = INV_COMPLETED;
update_call_counter(p, DEC_CALL_LIMIT);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return 0;
@@ -19242,7 +19240,7 @@
/* Basically for calling to IP/Host name only */
if (ast_strlen_zero(p->exten))
ast_string_field_set(p, exten, "s");
- /* Initialize our tag */
+ /* Initialize our tag */
make_our_tag(p->tag, sizeof(p->tag));
/* First invitation - create the channel */
@@ -19304,9 +19302,9 @@
if (!ast_strlen_zero(p_uac_min_se)) {
rtn = parse_minse(p_uac_min_se, &uac_min_se);
if (rtn != 0) {
- transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req);
- p->invitestate = INV_COMPLETED;
- if (!p->lastinvite) {
+ transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req);
+ p->invitestate = INV_COMPLETED;
+ if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
return -1;
@@ -19409,7 +19407,7 @@
if (!req->ignore && p)
p->lastinvite = seqno;
- if (replace_id) { /* Attended transfer or call pickup - we're the target */
+ if (replace_id) { /* Attended transfer or call pickup - we're the target */
if (!ast_strlen_zero(pickup.exten)) {
append_history(p, "Xfer", "INVITE/Replace received");
@@ -19821,7 +19819,7 @@
pvt_set_needdestroy(p, "outside of dialog");
}
return 0;
- }
+ }
/* Check if transfer is allowed from this device */
@@ -19834,7 +19832,7 @@
}
if (!req->ignore && ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
- /* Already have a pending REFER */
+ /* Already have a pending REFER */
transmit_response(p, "491 Request pending", req);
append_history(p, "Xfer", "Refer failed. Request pending.");
return 0;
@@ -19890,7 +19888,7 @@
p->refer->localtransfer = 1;
} else if (sipdebug)
ast_debug(3, "This SIP transfer is to a remote SIP extension (remote domain %s)\n", p->refer->refer_to_domain);
-
+
/* Is this a repeat of a current request? Ignore it */
/* Don't know what else to do right now. */
if (req->ignore)
@@ -19949,7 +19947,7 @@
ast_queue_control(current.chan1, AST_CONTROL_UNHOLD);
}
- ast_set_flag(&p->flags[0], SIP_GOTREFER);
+ ast_set_flag(&p->flags[0], SIP_GOTREFER);
/* Attended transfer: Find all call legs and bridge transferee with target*/
if (p->refer->attendedtransfer) {
@@ -19968,7 +19966,7 @@
*nounlock = 1;
ast_channel_unlock(current.chan1);
copy_request(¤t.req, req);
- ast_clear_flag(&p->flags[0], SIP_GOTREFER);
+ ast_clear_flag(&p->flags[0], SIP_GOTREFER);
p->refer->status = REFER_200OK;
append_history(p, "Xfer", "REFER to call parking.");
ast_manager_event_multichan(EVENT_FLAG_CALL, "Transfer", 2, chans, "TransferMethod: SIP\r\nTransferType: Blind\r\nChannel: %s\r\nUniqueid: %s\r\nSIP-Callid: %s\r\nTargetChannel: %s\r\nTargetUniqueid: %s\r\nTransferExten: %s\r\nTransfer2Parking: Yes\r\n",
@@ -20022,7 +20020,7 @@
/* FAKE ringing if not attended transfer */
if (!p->refer->attendedtransfer)
transmit_notify_with_sipfrag(p, seqno, "183 Ringing", FALSE);
-
+
/* For blind transfer, this will lead to a new call */
/* For attended transfer to remote host, this will lead to
a new SIP call with a replaces header, if the dial plan allows it
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