[asterisk-commits] rmudgett: branch group/issue8824 r168329 - in /team/group/issue8824: ./ chann...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jan 9 17:48:39 CST 2009
Author: rmudgett
Date: Fri Jan 9 17:48:39 2009
New Revision: 168329
URL: http://svn.digium.com/view/asterisk?view=rev&rev=168329
Log:
Resolved conflicts and restarted automerge.
Merged revisions 168192-168193,168200,168265,168269-168270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r168192 | rmudgett | 2009-01-09 15:43:30 -0600 (Fri, 09 Jan 2009) | 10 lines
Merged revisions 168191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09 Jan 2009) | 3 lines
* Fix for JIRA AST-175/ABE-1757
* Miscellaneous doxygen comments added.
........
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r168193 | mmichelson | 2009-01-09 15:53:26 -0600 (Fri, 09 Jan 2009) | 21 lines
Merged revisions 168128 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan 2009) | 13 lines
Add check_via calls to more request handlers
INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests
were not checking the topmost Via to determine where
to send the response. Adding check_via calls to those
request handlers solves this.
(closes issue #13071)
Reported by: baron
Patches:
check_via.patch uploaded by baron (license 531)
Tested by: baron
........
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r168200 | russell | 2009-01-09 16:21:05 -0600 (Fri, 09 Jan 2009) | 10 lines
Merged revisions 168198 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09 Jan 2009) | 2 lines
Make this compile for mvanbaak
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r168265 | mvanbaak | 2009-01-09 17:04:46 -0600 (Fri, 09 Jan 2009) | 9 lines
Add a script to find out the correct settings for Asterisk behind NAT
(closes issue #13065)
Reported by: tzafrir
Patches:
sip_nat_settings uploaded by tzafrir (license 46)
sip_nat_settings_6 uploaded by mvanbaak (license 7)
Tested by: tzafrir, pabelanger, Dovid and moi
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r168269 | rmudgett | 2009-01-09 17:15:26 -0600 (Fri, 09 Jan 2009) | 1 line
Spacing change
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r168270 | kpfleming | 2009-01-09 17:16:08 -0600 (Fri, 09 Jan 2009) | 9 lines
Merged revisions 168267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan 2009) | 1 line
update to use new sound file packages that include license files
........
................
Added:
team/group/issue8824/contrib/scripts/sip_nat_settings
- copied unchanged from r168270, trunk/contrib/scripts/sip_nat_settings
Modified:
team/group/issue8824/ (props changed)
team/group/issue8824/CHANGES
team/group/issue8824/channels/chan_misdn.c
team/group/issue8824/channels/chan_sip.c
team/group/issue8824/res/res_musiconhold.c
team/group/issue8824/sounds/Makefile
Propchange: team/group/issue8824/
------------------------------------------------------------------------------
automerge = *
Propchange: team/group/issue8824/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/group/issue8824/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Fri Jan 9 17:48:39 2009
@@ -1,1 +1,1 @@
-/trunk:1-168172
+/trunk:1-168296
Modified: team/group/issue8824/CHANGES
URL: http://svn.digium.com/view/asterisk/team/group/issue8824/CHANGES?view=diff&rev=168329&r1=168328&r2=168329
==============================================================================
--- team/group/issue8824/CHANGES (original)
+++ team/group/issue8824/CHANGES Fri Jan 9 17:48:39 2009
@@ -153,6 +153,9 @@
timezones, especially if those daylight savings time ranges vary from your
machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
includes.
+ * The contrib/scripts/ directory now has a script called sip_nat_settings that will
+ give you the correct output for an asterisk box behind nat. It will give you the
+ externhost and localnet settings.
Asterisk Manager Interface
--------------------------
Modified: team/group/issue8824/channels/chan_misdn.c
URL: http://svn.digium.com/view/asterisk/team/group/issue8824/channels/chan_misdn.c?view=diff&rev=168329&r1=168328&r2=168329
==============================================================================
--- team/group/issue8824/channels/chan_misdn.c (original)
+++ team/group/issue8824/channels/chan_misdn.c Fri Jan 9 17:48:39 2009
@@ -333,7 +333,13 @@
* \brief Incoming call dialplan context identifier.
* \note The "context" string read in from /etc/asterisk/misdn.conf
*/
- char context[BUFFERSIZE];
+ char context[AST_MAX_CONTEXT];
+
+ /*!
+ * \brief The configured music-on-hold class to use for this call.
+ * \note The "musicclass" string read in from /etc/asterisk/misdn.conf
+ */
+ char mohinterpret[MAX_MUSICCLASS];
//int zero_read_cnt; /* Not used */
@@ -394,7 +400,7 @@
*/
struct timeval overlap_tv;
-// struct chan_list *peer; /* Not used */
+ //struct chan_list *peer; /* Not used */
/*!
* \brief Next channel call record in the list.
@@ -1262,7 +1268,7 @@
/* ******************************************************************* */
/*!
* \internal
- * \brief Add a confiured prefix to the given number.
+ * \brief Add a configured prefix to the given number.
*
* \param port Logical port number
* \param number_type Type-of-number passed in.
@@ -1802,7 +1808,7 @@
e->command = "misdn show config";
e->usage =
"Usage: misdn show config [<port> | description <config element> | descriptions [general|ports]]\n"
- " Use 0 for <port> to only print the general config.\n";
+ " Use 0 for <port> to only print the general config.\n";
return NULL;
case CLI_GENERATE:
return complete_show_config(a);
@@ -2744,7 +2750,6 @@
int port;
int hdlc = 0;
char lang[BUFFERSIZE + 1];
- char localmusicclass[BUFFERSIZE + 1];
char faxdetect[BUFFERSIZE + 1];
char buf[256];
char buf2[256];
@@ -2769,8 +2774,7 @@
misdn_cfg_get(port, MISDN_CFG_LANGUAGE, lang, sizeof(lang));
ast_string_field_set(ast, language, lang);
- misdn_cfg_get(port, MISDN_CFG_MUSICCLASS, localmusicclass, sizeof(localmusicclass));
- ast_string_field_set(ast, musicclass, localmusicclass);
+ misdn_cfg_get(port, MISDN_CFG_MUSICCLASS, ch->mohinterpret, sizeof(ch->mohinterpret));
misdn_cfg_get(port, MISDN_CFG_TXGAIN, &bc->txgain, sizeof(bc->txgain));
misdn_cfg_get(port, MISDN_CFG_RXGAIN, &bc->rxgain, sizeof(bc->rxgain));
@@ -3398,7 +3402,7 @@
start_bc_tones(p);
break;
case AST_CONTROL_HOLD:
- ast_moh_start(ast,data,ast->musicclass);
+ ast_moh_start(ast, data, p->mohinterpret);
chan_misdn_log(1, p->bc->port, " --> *\tHOLD pid:%d\n", p->bc->pid);
break;
case AST_CONTROL_UNHOLD:
Modified: team/group/issue8824/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/group/issue8824/channels/chan_sip.c?view=diff&rev=168329&r1=168328&r2=168329
==============================================================================
--- team/group/issue8824/channels/chan_sip.c (original)
+++ team/group/issue8824/channels/chan_sip.c Fri Jan 9 17:48:39 2009
@@ -15469,6 +15469,7 @@
unsigned int event;
const char *c = get_header(req, "Content-Type");
+ check_via(p, req);
/* Need to check the media/type */
if (!strcasecmp(c, "application/dtmf-relay") ||
!strcasecmp(c, "application/vnd.nortelnetworks.digits")) {
@@ -18036,6 +18037,7 @@
char *eventid = NULL;
char *sep;
+ check_via(p, req);
if( (sep = strchr(event, ';')) ) { /* XXX bug here - overwriting string ? */
*sep++ = '\0';
eventid = sep;
@@ -18195,6 +18197,7 @@
(this needs to be fixed in 1.4 as well)
*/
+ check_via(p, req);
if (p->lastinvite) {
/* if this is a request in an active dialog, just confirm that the dialog exists. */
transmit_response_with_allow(p, "200 OK", req, 0);
@@ -19618,6 +19621,7 @@
int res = 0;
+ check_via(p, req);
if (req->debug)
ast_verbose("Call %s got a SIP call transfer from %s: (REFER)!\n", p->callid, ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "callee" : "caller");
@@ -20157,6 +20161,7 @@
static int handle_request_message(struct sip_pvt *p, struct sip_request *req)
{
if (!req->ignore) {
+ check_via(p, req);
if (req->debug)
ast_verbose("Receiving message!\n");
receive_message(p, req);
Modified: team/group/issue8824/res/res_musiconhold.c
URL: http://svn.digium.com/view/asterisk/team/group/issue8824/res/res_musiconhold.c?view=diff&rev=168329&r1=168328&r2=168329
==============================================================================
--- team/group/issue8824/res/res_musiconhold.c (original)
+++ team/group/issue8824/res/res_musiconhold.c Fri Jan 9 17:48:39 2009
@@ -187,7 +187,7 @@
static int reload(void);
#define mohclass_ref(class) (ao2_ref((class), +1), class)
-#define mohclass_unref(class) (ao2_ref((class), -1), NULL)
+#define mohclass_unref(class) (ao2_ref((class), -1), (struct mohclass *) NULL)
static void moh_files_release(struct ast_channel *chan, void *data)
{
Modified: team/group/issue8824/sounds/Makefile
URL: http://svn.digium.com/view/asterisk/team/group/issue8824/sounds/Makefile?view=diff&rev=168329&r1=168328&r2=168329
==============================================================================
--- team/group/issue8824/sounds/Makefile (original)
+++ team/group/issue8824/sounds/Makefile Fri Jan 9 17:48:39 2009
@@ -17,7 +17,7 @@
SOUNDS_DIR:=$(DESTDIR)$(ASTDATADIR)/sounds
MOH_DIR:=$(DESTDIR)$(ASTDATADIR)/moh
-CORE_SOUNDS_VERSION:=1.4.13
+CORE_SOUNDS_VERSION:=1.4.14
EXTRA_SOUNDS_VERSION:=1.4.8
SOUNDS_URL:=http://downloads.digium.com/pub/telephony/sounds/releases
MCS:=$(subst -EN-,-en-,$(MENUSELECT_CORE_SOUNDS))
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