[asterisk-commits] murf: branch murf/AEL-blockcomments r41504 - in
/team/murf/AEL-blockcomments:...
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Wed Aug 30 18:34:19 MST 2006
Author: murf
Date: Wed Aug 30 20:34:18 2006
New Revision: 41504
URL: http://svn.digium.com/view/asterisk?rev=41504&view=rev
Log:
Merged revisions 41409,41429,41433-41437,41456-41457,41475 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r41409 | file | 2006-08-30 12:02:53 -0600 (Wed, 30 Aug 2006) | 10 lines
Merged revisions 41390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r41390 | file | 2006-08-30 13:58:31 -0400 (Wed, 30 Aug 2006) | 2 lines
Properly handle an ETIMEDOUT result from pthread_cond_timedwait (issue #7318 reported by arkadia)
........
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r41429 | russell | 2006-08-30 13:02:34 -0600 (Wed, 30 Aug 2006) | 13 lines
Blocked revisions 41411 via svnmerge
........
r41411 | russell | 2006-08-30 14:59:44 -0400 (Wed, 30 Aug 2006) | 6 lines
Restore original functionality of 1.2 in places where ANI was not set, but was
changed to be set. The original change was done to ensure that the behavior of
the "callerid" option in each channel driver was consistent, but it caused an
unexpected behavior change of CDR records for users, so this change is being
reverted in 1.2. (issue #7695)
........
................
r41433 | oej | 2006-08-30 13:07:21 -0600 (Wed, 30 Aug 2006) | 2 lines
Issue #7572 (Boesl) - hangup channel that get buggy 487 response (imported from 1.2)
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r41434 | russell | 2006-08-30 13:07:59 -0600 (Wed, 30 Aug 2006) | 2 lines
add a note about behavior of the "clid" field in the CDR
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r41435 | oej | 2006-08-30 13:16:53 -0600 (Wed, 30 Aug 2006) | 3 lines
Mark ALERT_INFO as deprecated. This can now be done with the sipaddheader() application and
does not need special code in chan_sip any more.
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r41436 | oej | 2006-08-30 13:20:29 -0600 (Wed, 30 Aug 2006) | 2 lines
Note to myself: Remember Russell's note: Always compile first...
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r41437 | oej | 2006-08-30 13:51:31 -0600 (Wed, 30 Aug 2006) | 3 lines
Don't add headers to an uninitialized eq (from issue 7694 garyhai, but not
a resolution to that bug report)
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r41456 | oej | 2006-08-30 14:20:52 -0600 (Wed, 30 Aug 2006) | 2 lines
update docs
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r41457 | oej | 2006-08-30 14:24:32 -0600 (Wed, 30 Aug 2006) | 2 lines
Why check bridgepeer twice?
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r41475 | kpfleming | 2006-08-30 15:44:05 -0600 (Wed, 30 Aug 2006) | 2 lines
change default setting for autofallthrough
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Modified:
team/murf/AEL-blockcomments/ (props changed)
team/murf/AEL-blockcomments/UPGRADE.txt
team/murf/AEL-blockcomments/channels/chan_sip.c
team/murf/AEL-blockcomments/configs/extensions.conf.sample
team/murf/AEL-blockcomments/doc/channelvariables.txt
team/murf/AEL-blockcomments/main/pbx.c
Propchange: team/murf/AEL-blockcomments/
------------------------------------------------------------------------------
Binary property 'branch-1.2-blocked' - no diff available.
Propchange: team/murf/AEL-blockcomments/
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Binary property 'branch-1.2-merged' - no diff available.
Propchange: team/murf/AEL-blockcomments/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Wed Aug 30 20:34:18 2006
@@ -1,1 +1,1 @@
-/trunk:1-41396
+/trunk:1-41503
Modified: team/murf/AEL-blockcomments/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/team/murf/AEL-blockcomments/UPGRADE.txt?rev=41504&r1=41503&r2=41504&view=diff
==============================================================================
--- team/murf/AEL-blockcomments/UPGRADE.txt (original)
+++ team/murf/AEL-blockcomments/UPGRADE.txt Wed Aug 30 20:34:18 2006
@@ -96,6 +96,11 @@
not set, it uses the transferee variable. If not set in any channel, it will
attempt to use the last non macro context. If not possible, it will default
to the current context.
+
+* The autofallthrough setting introduced in Asterisk 1.2 now defaults to 'yes';
+ if your dialplan relies on the ability to 'run off the end' of an extension
+ and wait for a new extension without using WaitExten() to accomplish that,
+ you will need set autofallthrough to 'no' in your extensions.conf file.
Command Line Interface:
@@ -302,6 +307,10 @@
in coming versions of Asterisk. Please use the dialplan function
SIPCHANINFO(useragent) instead.
+* The ALERT_INFO dialplan variable is deprecated and will be removed
+ in coming versions of Asterisk. Please use the dialplan application
+ sipaddheader() to add the "Alert-Info" header to the outbound invite.
+
The Zap channel:
* Support for MFC/R2 has been removed, as it has not been functional for some
@@ -383,3 +392,13 @@
server puts a phone on a different server on hold, the remote server will be
responsible for playing the hold music to its local phone that was put on
hold instead of the far end server across the network playing the music.
+
+CDR Records:
+
+* The behavior of the "clid" field of the CDR has always been that it will
+ contain the callerid ANI if it is set, or the callerid number if ANI was not
+ set. When using the "callerid" option for various channel drivers, some
+ would set ANI and some would not. This has been cleared up so that all
+ channel drivers set ANI. If you would like to change the callerid number
+ on the channel from the dialplan and have that change also show up in the
+ CDR, then you *must* set CALLERID(ANI) as well as CALLERID(num).
Modified: team/murf/AEL-blockcomments/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/murf/AEL-blockcomments/channels/chan_sip.c?rev=41504&r1=41503&r2=41504&view=diff
==============================================================================
--- team/murf/AEL-blockcomments/channels/chan_sip.c (original)
+++ team/murf/AEL-blockcomments/channels/chan_sip.c Wed Aug 30 20:34:18 2006
@@ -81,6 +81,10 @@
* The PBX issues a hangup on both incoming and outgoing calls through
* the sip_hangup() function
*
+ * \par Deprecated stuff
+ * This is deprecated and will be removed after the 1.4 release
+ * - the SIPUSERAGENT dialplan variable
+ * - the ALERT_INFO dialplan variable
*/
@@ -4922,8 +4926,6 @@
ast_set_write_format(p->owner, p->owner->writeformat);
}
- bridgepeer = ast_bridged_channel(p->owner);
-
/* Turn on/off music on hold if we are holding/unholding */
if ((bridgepeer = ast_bridged_channel(p->owner))) {
if (sin.sin_addr.s_addr && !sendonly) {
@@ -7061,8 +7063,6 @@
else
snprintf(referto, sizeof(referto), "<sip:%s>", dest);
- add_header(&req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
-
/* save in case we get 407 challenge */
sip_refer_allocate(p);
ast_copy_string(p->refer->refer_to, referto, sizeof(p->refer->refer_to));
@@ -7070,6 +7070,8 @@
p->refer->status = REFER_SENT; /* Set refer status */
reqprep(&req, p, SIP_REFER, 0, 1);
+ add_header(&req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
+
add_header(&req, "Refer-To", referto);
add_header(&req, "Allow", ALLOWED_METHODS);
add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
@@ -11827,6 +11829,8 @@
break;
case 487: /* Response on INVITE that has been CANCELled */
/* channel now destroyed - dec the inUse counter */
+ if (owner)
+ ast_queue_hangup(p->owner);
update_call_counter(p, DEC_CALL_LIMIT);
break;
case 482: /*
Modified: team/murf/AEL-blockcomments/configs/extensions.conf.sample
URL: http://svn.digium.com/view/asterisk/team/murf/AEL-blockcomments/configs/extensions.conf.sample?rev=41504&r1=41503&r2=41504&view=diff
==============================================================================
--- team/murf/AEL-blockcomments/configs/extensions.conf.sample (original)
+++ team/murf/AEL-blockcomments/configs/extensions.conf.sample Wed Aug 30 20:34:18 2006
@@ -27,13 +27,13 @@
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
-; or HANGUP depending on Asterisk's best guess (strongly recommended).
+; or HANGUP depending on Asterisk's best guess. This is the default.
;
; If autofallthrough is not set, then if an extension runs out of
-; things to do, asterisk will wait for a new extension to be dialed
+; things to do, Asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
-autofallthrough=yes
+;autofallthrough=no
;
; If clearglobalvars is set, global variables will be cleared
; and reparsed on an extensions reload, or Asterisk reload.
Modified: team/murf/AEL-blockcomments/doc/channelvariables.txt
URL: http://svn.digium.com/view/asterisk/team/murf/AEL-blockcomments/doc/channelvariables.txt?rev=41504&r1=41503&r2=41504&view=diff
==============================================================================
--- team/murf/AEL-blockcomments/doc/channelvariables.txt (original)
+++ team/murf/AEL-blockcomments/doc/channelvariables.txt Wed Aug 30 20:34:18 2006
@@ -710,7 +710,7 @@
---------------------------------------------------------
${SIPCALLID} * SIP Call-ID: header verbatim (for logging or CDR matching)
${SIPDOMAIN} * SIP destination domain of an inbound call (if appropriate)
-${SIPUSERAGENT} * SIP user agent
+${SIPUSERAGENT} * SIP user agent (deprecated)
${SIPURI} * SIP uri
${SIP_CODEC} Set the SIP codec for a call
${SIP_URI_OPTIONS} * additional options to add to the URI for an outgoing call
Modified: team/murf/AEL-blockcomments/main/pbx.c
URL: http://svn.digium.com/view/asterisk/team/murf/AEL-blockcomments/main/pbx.c?rev=41504&r1=41503&r2=41504&view=diff
==============================================================================
--- team/murf/AEL-blockcomments/main/pbx.c (original)
+++ team/murf/AEL-blockcomments/main/pbx.c Wed Aug 30 20:34:18 2006
@@ -240,7 +240,7 @@
AST_MUTEX_DEFINE_STATIC(globalslock);
static struct varshead globals = AST_LIST_HEAD_NOLOCK_INIT_VALUE;
-static int autofallthrough = 0;
+static int autofallthrough = 1;
AST_MUTEX_DEFINE_STATIC(maxcalllock);
static int countcalls = 0;
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