[asterisk-commits] branch oej/moduletest r13616 - in /team/oej/moduletest: ./ apps/ channels/

[asterisk-commits at lists.digium.com]

Author: oej
Date: Sun Mar 19 06:25:33 2006
New Revision: 13616

URL: http://svn.digium.com/view/asterisk?rev=13616&view=rev
Log:
Reset automerge

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

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

Propchange: team/oej/moduletest/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Mar 19 06:25:33 2006
@@ -1,1 +1,1 @@
-/branches/1.2:1-7496,7498-13301
+/branches/1.2:1-7496,7498-13614

Modified: team/oej/moduletest/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/oej/moduletest/apps/app_dial.c?rev=13616&r1=13615&r2=13616&view=diff
==============================================================================
--- team/oej/moduletest/apps/app_dial.c (original)
+++ team/oej/moduletest/apps/app_dial.c Sun Mar 19 06:25:33 2006
@@ -109,9 +109,10 @@
 "           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"
@@ -1420,6 +1421,7 @@
 			}
 			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/moduletest/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/moduletest/channels/chan_sip.c?rev=13616&r1=13615&r2=13616&view=diff
==============================================================================
--- team/oej/moduletest/channels/chan_sip.c (original)
+++ team/oej/moduletest/channels/chan_sip.c Sun Mar 19 06:25:33 2006
@@ -583,7 +583,7 @@
 #define sipdebug_config		ast_test_flag(&global_flags_page2, SIP_PAGE2_DEBUG_CONFIG)
 #define sipdebug_console	ast_test_flag(&global_flags_page2, SIP_PAGE2_DEBUG_CONSOLE)
 
-static int global_rtautoclear = 120;
+static int global_rtautoclear;
 
 /*! \brief sip_pvt: PVT structures are used for each SIP conversation, ie. a call  */
 static struct sip_pvt {
@@ -12347,6 +12347,7 @@
 	global_rtptimeout = 0;
 	global_rtpholdtimeout = 0;
 	global_rtpkeepalive = 0;
+	global_rtautoclear = 120;
 	pedanticsipchecking = 0;
 	global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
 	global_regattempts_max = 0;