[asterisk-commits] branch oej/managerstuff r13615 - in /team/oej/managerstuff: ./ apps/ channels/

[asterisk-commits at lists.digium.com]

Author: oej
Date: Sun Mar 19 06:06:01 2006
New Revision: 13615

URL: http://svn.digium.com/view/asterisk?rev=13615&view=rev
Log:
Reset automerge

Modified:
    team/oej/managerstuff/   (props changed)
    team/oej/managerstuff/apps/app_dial.c
    team/oej/managerstuff/channels/chan_sip.c

Propchange: team/oej/managerstuff/
------------------------------------------------------------------------------
    automerge = http://edvina.net/training/

Propchange: team/oej/managerstuff/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Mar 19 06:06:01 2006
@@ -1,1 +1,1 @@
-/branches/1.2:1-7496,7498-13281
+/branches/1.2:1-7496,7498-13614

Modified: team/oej/managerstuff/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/oej/managerstuff/apps/app_dial.c?rev=13615&r1=13614&r2=13615&view=diff
==============================================================================
--- team/oej/managerstuff/apps/app_dial.c (original)
+++ team/oej/managerstuff/apps/app_dial.c Sun Mar 19 06:06:01 2006
@@ -111,9 +111,10 @@
 "           other than the number assigned to the caller.\n"
 "    g    - Proceed with dialplan execution at the current extension if the\n"
 "           destination channel hangs up.\n"
-"    G(context^exten^pri) - If the call is answered, transfer both parties to\n"
-"           the specified priority. Optionally, an extension, or extension and\n"
-"           context may be specified. Otherwise, the current extension is used.\n"
+"    G(context^exten^pri) - If the call is answered, transfer the calling party to\n"
+"           the specified priority and the called party to the specified priority+1.\n"
+"           Optionally, an extension, or extension and context may be specified. \n"
+"           Otherwise, the current extension is used.\n"
 "    h    - Allow the called party to hang up by sending the '*' DTMF digit.\n"
 "    H    - Allow the calling party to hang up by hitting the '*' DTMF digit.\n"
 "    j    - Jump to priority n+101 if all of the requested channels were busy.\n"
@@ -1422,6 +1423,7 @@
 			}
 			ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
 			ast_parseable_goto(peer, opt_args[OPT_ARG_GOTO]);
+			peer->priority++;
 			ast_pbx_start(peer);
 			hanguptree(outgoing, NULL);
 			LOCAL_USER_REMOVE(u);

Modified: team/oej/managerstuff/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/managerstuff/channels/chan_sip.c?rev=13615&r1=13614&r2=13615&view=diff
==============================================================================
--- team/oej/managerstuff/channels/chan_sip.c (original)
+++ team/oej/managerstuff/channels/chan_sip.c Sun Mar 19 06:06:01 2006
@@ -380,7 +380,7 @@
 static char default_musicclass[MAX_MUSICCLASS];		/*!< Global music on hold class */
 
 /* Global settings only apply to the channel */
-static int global_rtautoclear = 120;
+static int global_rtautoclear;
 static int global_notifyringing;	/*!< Send notifications on ringing */
 static int srvlookup;			/*!< SRV Lookup on or off. Default is off, RFC behavior is on */
 static int pedanticsipchecking;		/*!< Extra checking ?  Default off */
@@ -12431,6 +12431,7 @@
 	global_rtptimeout = 0;
 	global_rtpholdtimeout = 0;
 	global_rtpkeepalive = 0;
+	global_rtautoclear = 120;
 	pedanticsipchecking = 0;
 	global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
 	global_regattempts_max = 0;