[asterisk-commits] mmichelson: branch group/issue8824 r183826 - in /team/group/issue8824: channe...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Mar 23 17:06:09 CDT 2009
Author: mmichelson
Date: Mon Mar 23 17:06:05 2009
New Revision: 183826
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=183826
Log:
Address more of Russell's comments!!!!!!!!!1
Modified:
team/group/issue8824/channels/chan_sip.c
team/group/issue8824/funcs/func_connectedline.c
team/group/issue8824/funcs/func_redirecting.c
Modified: team/group/issue8824/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue8824/channels/chan_sip.c?view=diff&rev=183826&r1=183825&r2=183826
==============================================================================
--- team/group/issue8824/channels/chan_sip.c (original)
+++ team/group/issue8824/channels/chan_sip.c Mon Mar 23 17:06:05 2009
@@ -965,7 +965,7 @@
{ AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"}
};
-static enum AST_REDIRECTING_REASON sip_reason_str_to_code(const char *text)
+static enum AST_REDIRECTING_REASON sip_reason_str_to_code(const char *text)
{
enum AST_REDIRECTING_REASON ast = AST_REDIRECTING_REASON_UNKNOWN;
int i;
@@ -5102,7 +5102,7 @@
if (xmitres == XMIT_ERROR)
return -1;
p->invitestate = INV_CALLING;
-
+
/* Initialize auto-congest time */
AST_SCHED_REPLACE_UNREF(p->initid, sched, p->timer_b, auto_congest, p,
dialog_unref(_data, "dialog ptr dec when SCHED_REPLACE del op succeeded"),
@@ -5870,7 +5870,7 @@
ast_rtp_new_source(p->rtp);
p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE, FALSE);
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+ ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
}
p->lastrtptx = time(NULL);
res = ast_rtp_write(p->rtp, frame);
@@ -5888,7 +5888,7 @@
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE, FALSE);
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+ ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
}
p->lastrtptx = time(NULL);
res = ast_rtp_write(p->vrtp, frame);
@@ -5909,7 +5909,7 @@
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE, FALSE);
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+ ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
}
p->lastrtptx = time(NULL);
res = ast_rtp_write(p->trtp, frame);
@@ -6113,7 +6113,7 @@
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE, FALSE);
- ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+ ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
break;
}
res = -1;
@@ -8803,9 +8803,9 @@
}
respprep(&resp, p, msg, req);
- if (ast_test_flag(&p->flags[0], SIP_SENDRPID)
- && ast_test_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND)
- && (!strcasecmp(msg, "180 Ringing") || !strcasecmp(msg, "183 Session Progress"))) {
+ if (ast_test_flag(&p->flags[0], SIP_SENDRPID)
+ && ast_test_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND)
+ && (!strncasecmp(msg, "180", 3) || !strncasecmp(msg, "183", 3))) {
ast_clear_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
add_rpid(&resp, p);
}
@@ -9033,7 +9033,7 @@
* \pre if p->owner exists, it must be locked
* \brief Add Remote-Party-ID header to SIP message
*/
-static int add_rpid(struct sip_request *req, struct sip_pvt *p)
+static int add_rpid(struct sip_request *req, struct sip_pvt *p)
{
char tmp[256];
char *lid_num = NULL;
@@ -9042,7 +9042,7 @@
const char *fromdomain;
const char *privacy = NULL;
const char *screen = NULL;
- const char *anonymous_string = "\"Anonymous\" <anonymous at anonymous.invalid>";
+ const char *anonymous_string = "\"Anonymous\" <anonymous at anonymous.invalid>";
if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
return 0;
@@ -9062,7 +9062,7 @@
if (ast_test_flag(&p->flags[0], SIP_SENDRPID_PAI)) {
if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
- snprintf(tmp, sizeof(tmp), "%s", anonymous_string);
+ snprintf(tmp, sizeof(tmp), "%s", anonymous_string);
} else {
snprintf(tmp, sizeof(tmp), "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
}
@@ -9942,7 +9942,7 @@
if (p->owner && (p->owner->connected.id.number_presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
l = p->owner->connected.id.number;
- n = p->owner->connected.id.name;
+ n = p->owner->connected.id.name;
}
if (ast_strlen_zero(l))
l = default_callerid;
@@ -10707,7 +10707,7 @@
/*! \brief Notify peer that the connected line has changed */
static void update_connectedline(struct sip_pvt *p, const void *data, size_t datalen)
{
-
+
if (!ast_test_flag(&p->flags[0], SIP_SENDRPID))
return;
if (ast_strlen_zero(p->owner->connected.id.number))
@@ -10739,7 +10739,7 @@
} else {
if (ast_test_flag(&p->flags[1], SIP_PAGE2_RPID_IMMEDIATE)) {
struct sip_request resp;
-
+
if ((p->owner->_state == AST_STATE_RING) && !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT)) {
respprep(&resp, p, "180 Ringing", &p->initreq);
add_rpid(&resp, p);
@@ -12387,7 +12387,7 @@
cid_name = start;
start = ast_skip_blanks(end);
}
-
+
if (*start != '<')
return 0;
*start++ = '\0';
@@ -12541,7 +12541,7 @@
char tmp[256], *exten, *rexten, *rdomain, *rname = NULL;
char *params, *reason_param = NULL;
struct sip_request *req;
-
+
req = oreq ? oreq : &p->initreq;
ast_copy_string(tmp, get_header(req, "Diversion"), sizeof(tmp));
@@ -12572,7 +12572,7 @@
if ((reason_param = strcasestr(params, "reason="))) {
reason_param+=7;
/* Remove enclosing double-quotes */
- if (*reason_param == '"')
+ if (*reason_param == '"')
ast_strip_quoted(reason_param, "\"", "\"");
if (!ast_strlen_zero(reason_param)) {
sip_set_redirstr(p, reason_param);
@@ -12586,7 +12586,7 @@
rdomain = exten;
rexten = strsep(&rdomain, "@"); /* trim anything after @ */
- if (p->owner)
+ if (p->owner)
pbx_builtin_setvar_helper(p->owner, "__SIPRDNISDOMAIN", rdomain);
if (sip_debug_test_pvt(p))
@@ -13335,7 +13335,7 @@
ast_shrink_phone_number(tmp);
ast_string_field_set(p, cid_num, tmp);
}
- if (!ast_strlen_zero(peer->cid_name))
+ if (!ast_strlen_zero(peer->cid_name))
ast_string_field_set(p, cid_name, peer->cid_name);
if (peer->callingpres)
p->callingpres = peer->callingpres;
@@ -17072,7 +17072,7 @@
}
check_pendings(p);
break;
-
+
case 181: /* Call Is Being Forwarded */
if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
@@ -17120,9 +17120,9 @@
if (!reinvite)
/* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
/* For re-invites, we try to recover */
- ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
- }
-
+ ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
+ }
+
if (!req->ignore && p->owner && get_rpid(p, req)) {
/* Queue a connected line update */
ast_party_connected_line_init(&connected);
@@ -19771,7 +19771,7 @@
} else if (p->t38.state == T38_DISABLED) {
/* If this is not a re-invite or something to ignore - it's critical */
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
- transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE : TRUE, FALSE);
+ transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE : TRUE, FALSE);
}
p->invitestate = INV_TERMINATED;
@@ -19916,7 +19916,7 @@
ast_connected_line_update(current->chan2, &connected_caller);
} else {
/* Notify the first other party that they are connected to someone else assuming that target.chan1
- has progressed far enough through the dialplan to have it's called party information set. */
+ has progressed far enough through the dialplan to have it's called party information set. */
if (current->chan2) {
connected_caller = target.chan1->connected;
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
Modified: team/group/issue8824/funcs/func_connectedline.c
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue8824/funcs/func_connectedline.c?view=diff&rev=183826&r1=183825&r2=183826
==============================================================================
--- team/group/issue8824/funcs/func_connectedline.c (original)
+++ team/group/issue8824/funcs/func_connectedline.c Mon Mar 23 17:06:05 2009
@@ -137,7 +137,7 @@
default:
ast_log(LOG_ERROR, "Unknown connectedline option '%s'.\n", option);
return 0;
- } /* end switch */
+ }
}
else {
set_it = ast_connected_line_update;
@@ -231,7 +231,9 @@
static int load_module(void)
{
- return ast_custom_function_register(&connectedline_function);
+ return ast_custom_function_register(&connectedline_function)
+ ? AST_MODULE_LOAD_DECLINE
+ : AST_MODULE_LOAD_SUCCESS;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Connected Line dialplan function");
Modified: team/group/issue8824/funcs/func_redirecting.c
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue8824/funcs/func_redirecting.c?view=diff&rev=183826&r1=183825&r2=183826
==============================================================================
--- team/group/issue8824/funcs/func_redirecting.c (original)
+++ team/group/issue8824/funcs/func_redirecting.c Mon Mar 23 17:06:05 2009
@@ -145,7 +145,7 @@
}
return status;
-} /* end redirecting_id_read() */
+}
@@ -187,7 +187,7 @@
default:
ast_log(LOG_ERROR, "Unknown redirecting data type '%s'.\n", data);
break;
- } /* end switch */
+ }
} else if (!strncasecmp("to-", data, 3)) {
switch (redirecting_id_read(buf, len, data + 3, &chan->redirecting.to)) {
case ID_FIELD_VALID:
@@ -197,7 +197,7 @@
default:
ast_log(LOG_ERROR, "Unknown redirecting data type '%s'.\n", data);
break;
- } /* end switch */
+ }
} else if (!strncasecmp("pres", data, 4)) {
ast_copy_string(buf, ast_named_caller_presentation(chan->redirecting.from.number_presentation), len);
} else if (!strncasecmp("reason", data, 6)) {
@@ -211,7 +211,7 @@
ast_channel_unlock(chan);
return 0;
-} /* end redirecting_read() */
+}
@@ -286,7 +286,7 @@
}
return status;
-} /* end redirecting_id_write() */
+}
@@ -327,7 +327,7 @@
default:
ast_log(LOG_ERROR, "Unknown redirecting option '%s'.\n", option);
return 0;
- } /* end switch */
+ }
}
else {
set_it = ast_redirecting_update;
@@ -352,7 +352,7 @@
default:
ast_log(LOG_ERROR, "Unknown redirecting data type '%s'.\n", data);
break;
- } /* end switch */
+ }
} else if (!strncasecmp("to-", data, 3)) {
switch (redirecting_id_write(&redirecting.to, data + 3, value)) {
case ID_FIELD_VALID:
@@ -366,7 +366,7 @@
default:
ast_log(LOG_ERROR, "Unknown redirecting data type '%s'.\n", data);
break;
- } /* end switch */
+ }
} else if (!strncasecmp("pres", data, 4)) {
int pres;
@@ -419,7 +419,7 @@
}
return 0;
-} /* end redirecting_write() */
+}
@@ -444,7 +444,7 @@
static int unload_module(void)
{
return ast_custom_function_unregister(&redirecting_function);
-} /* end unload_module() */
+}
@@ -459,8 +459,10 @@
*/
static int load_module(void)
{
- return ast_custom_function_register(&redirecting_function);
-} /* end load_module() */
+ return ast_custom_function_register(&redirecting_function)
+ ? AST_MODULE_LOAD_DECLINE
+ : AST_MODULE_LOAD_SUCCESS;
+}
More information about the asterisk-commits
mailing list