[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

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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)

........

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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/
------------------------------------------------------------------------------
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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