[asterisk-commits] branch oej/test-this-branch r13617 - in /team/oej/test-this-branch: ./ apps/ ...

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Sun Mar 19 05:30:11 MST 2006


Author: oej
Date: Sun Mar 19 06:30:06 2006
New Revision: 13617

URL: http://svn.digium.com/view/asterisk?rev=13617&view=rev
Log:
reset automerge.

Modified:
    team/oej/test-this-branch/   (props changed)
    team/oej/test-this-branch/apps/app_dial.c
    team/oej/test-this-branch/channels/chan_sip.c
    team/oej/test-this-branch/configs/sip.conf.sample

Propchange: team/oej/test-this-branch/
------------------------------------------------------------------------------
    automerge = http://edvina.net/training/

Propchange: team/oej/test-this-branch/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.

Propchange: team/oej/test-this-branch/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Mar 19 06:30:06 2006
@@ -1,1 +1,1 @@
-/trunk:1-13536
+/trunk:1-13616

Modified: team/oej/test-this-branch/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/apps/app_dial.c?rev=13617&r1=13616&r2=13617&view=diff
==============================================================================
--- team/oej/test-this-branch/apps/app_dial.c (original)
+++ team/oej/test-this-branch/apps/app_dial.c Sun Mar 19 06:30:06 2006
@@ -112,9 +112,10 @@
 "           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"
@@ -1425,6 +1426,7 @@
 			}
 			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/test-this-branch/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/channels/chan_sip.c?rev=13617&r1=13616&r2=13617&view=diff
==============================================================================
--- team/oej/test-this-branch/channels/chan_sip.c (original)
+++ team/oej/test-this-branch/channels/chan_sip.c Sun Mar 19 06:30:06 2006
@@ -417,7 +417,7 @@
 static struct ast_codec_pref default_prefs;		/*!< Default codec prefs */
 
 /* 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 */
@@ -439,7 +439,7 @@
 static int recordhistory;		/*!< Record SIP history. Off by default */
 static int dumphistory;			/*!< Dump history to verbose before destroying SIP dialog */
 static char global_realm[MAXHOSTNAMELEN]; 		/*!< Default realm */
-static char regcontext[AST_MAX_CONTEXT];		/*!< Context for auto-extensions */
+static char global_regcontext[AST_MAX_CONTEXT];		/*!< Context for auto-extensions */
 static char global_useragent[AST_MAX_EXTENSION];	/*!< Useragent for the SIP channel */
 static int allow_external_domains;	/*!< Accept calls to external SIP domains? */
 static char default_parkinglot[AST_MAX_EXTENSION];	/*!< Default parking lot */
@@ -703,7 +703,6 @@
 #define sipdebug		ast_test_flag(&global_flags_page2, SIP_PAGE2_DEBUG)
 #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)
-
 
 /*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe  */
 static struct sip_pvt {
@@ -1771,15 +1770,15 @@
 {
 	char multi[256];
 	char *stringp, *ext;
-	if (!ast_strlen_zero(regcontext)) {
+	if (!ast_strlen_zero(global_regcontext)) {
 		ast_copy_string(multi, ast_strlen_zero(peer->regexten) ? peer->name : peer->regexten, sizeof(multi));
 		stringp = multi;
 		while((ext = strsep(&stringp, "&"))) {
 			if (onoff)
-				ast_add_extension(regcontext, 1, ext, 1, NULL, NULL, "Noop",
+				ast_add_extension(global_regcontext, 1, ext, 1, NULL, NULL, "Noop",
 						  ast_strdup(peer->name), free, "SIP");
 			else
-				ast_context_remove_extension(regcontext, ext, 1, NULL);
+				ast_context_remove_extension(global_regcontext, ext, 1, NULL);
 		}
 	}
 }
@@ -1797,6 +1796,7 @@
 		ast_variables_destroy(device->chanvars);
 		device->chanvars = NULL;
 	}
+<<<<<<< .working
 	if (device->expire > -1)
 		ast_sched_del(sched, device->expire);
 	if (device->pokeexpire > -1)
@@ -6440,7 +6440,7 @@
 	destroy_association(peer);
 	
 	manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
-	register_peer_exten(peer, 0);
+	register_peer_exten(peer, FALSE);
 	peer->expire = -1;
 	ast_device_state_changed("SIP/%s", peer->name);
 	if (ast_test_flag((&peer->flags_page2), SIP_PAGE2_SELFDESTRUCT) || ast_test_flag((&peer->flags_page2), SIP_PAGE2_RTAUTOCLEAR)) {
@@ -6519,7 +6519,7 @@
 	if (peer->expire > -1)
 		ast_sched_del(sched, peer->expire);
 	peer->expire = ast_sched_add(sched, (expiry + 10) * 1000, expire_register, peer);
-	register_peer_exten(peer, 1);
+	register_peer_exten(peer, TRUE);
 }
 
 /*! \brief Parse contact header for 200 OK on INVITE */
@@ -8715,7 +8715,6 @@
 
 static char mandescr_show_peer[] = 
 "Description: Show one SIP peer with details on current status.\n"
-"  The XML format is under development, feedback welcome! /oej\n"
 "Variables: \n"
 "  Peer: <name>           The peer name you want to check.\n"
 "  ActionID: <id>	  Optional action ID for this AMI transaction.\n";
@@ -8837,6 +8836,8 @@
 		ast_cli(fd, "  ToHost       : %s\n", peer->tohost);
 		ast_cli(fd, "  Addr->IP     : %s Port %d\n",  peer->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "(Unspecified)", ntohs(peer->addr.sin_port));
 		ast_cli(fd, "  Defaddr->IP  : %s Port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
+		if (!ast_strlen_zero(global_regcontext))
+			ast_cli(fd, "  Reg. exten   : %s\n", peer->regexten);
 		ast_cli(fd, "  Def. Username: %s\n", peer->username);
 		ast_cli(fd, "  SIP Options  : ");
 		if (peer->sipoptions) {
@@ -8917,6 +8918,8 @@
 		ast_cli(fd, "Address-IP: %s\r\nAddress-Port: %d\r\n",  peer->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "", ntohs(peer->addr.sin_port));
 		ast_cli(fd, "Default-addr-IP: %s\r\nDefault-addr-port: %d\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
 		ast_cli(fd, "Default-Username: %s\r\n", peer->username);
+		if (!ast_strlen_zero(global_regcontext))
+			ast_cli(fd, "RegExtension: %s\r\n", peer->regexten);
 		ast_cli(fd, "Codecs: ");
 		ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability);
 		ast_cli(fd, "%s\r\n", codec_buf);
@@ -9068,7 +9071,7 @@
 	ast_cli(fd, "  Realm. auth:            %s\n", authl ? "Yes": "No");
 	ast_cli(fd, "  User Agent:             %s\n", global_useragent);
 	ast_cli(fd, "  MWI checking interval:  %d secs\n", global_mwitime);
-	ast_cli(fd, "  Reg. context:           %s\n", ast_strlen_zero(regcontext) ? "(not set)" : regcontext);
+	ast_cli(fd, "  Reg. context:           %s\n", ast_strlen_zero(global_regcontext) ? "(not set)" : global_regcontext);
 	ast_cli(fd, "  Caller ID:              %s\n", default_callerid);
 	ast_cli(fd, "  From: Domain:           %s\n", default_fromdomain);
 	ast_cli(fd, "  Record SIP history:     %s\n", recordhistory ? "On" : "Off");
@@ -13449,7 +13452,7 @@
 
 	/* Reset channel settings to default before re-configuring */
 	allow_external_domains = DEFAULT_ALLOW_EXT_DOM;				/* Allow external invites */
-	regcontext[0] = '\0';
+	global_regcontext[0] = '\0';
 	expiry = DEFAULT_EXPIRY;
 	global_notifyringing = DEFAULT_NOTIFYRINGING;
 	global_allowsubscribe = TRUE;
@@ -13472,6 +13475,7 @@
 	ast_set_flag(&global_flags_page2, SIP_PAGE2_ALLOWSUBSCRIBE);	/* Default for peers, users: TRUE */
 	ast_set_flag(&global_flags_page2, SIP_PAGE2_ALLOWOVERLAP);	/* Default for peers, users: TRUE */
 
+	global_rtautoclear = 120;
 	ast_set_flag(&global_flags_page2, SIP_PAGE2_RTUPDATE);
 
 	/* Initialize some reasonable defaults at SIP reload (used both for channel and as default for peers and users */
@@ -13579,10 +13583,10 @@
 		} else if (!strcasecmp(v->name, "language")) {
 			ast_copy_string(default_language, v->value, sizeof(default_language));
 		} else if (!strcasecmp(v->name, "regcontext")) {
-			ast_copy_string(regcontext, v->value, sizeof(regcontext));
+			ast_copy_string(global_regcontext, v->value, sizeof(global_regcontext));
 			/* Create context if it doesn't exist already */
-			if (!ast_context_find(regcontext))
-				ast_context_create(NULL, regcontext, "SIP");
+			if (!ast_context_find(global_regcontext))
+				ast_context_create(NULL, global_regcontext, "SIP");
 		} else if (!strcasecmp(v->name, "callerid")) {
 			ast_copy_string(default_callerid, v->value, sizeof(default_callerid));
 		} else if (!strcasecmp(v->name, "fromdomain")) {

Modified: team/oej/test-this-branch/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/configs/sip.conf.sample?rev=13617&r1=13616&r2=13617&view=diff
==============================================================================
--- team/oej/test-this-branch/configs/sip.conf.sample (original)
+++ team/oej/test-this-branch/configs/sip.conf.sample Sun Mar 19 06:30:06 2006
@@ -222,7 +222,7 @@
 
 ;----------------------------------------- REALTIME SUPPORT ------------------------
 ; For additional information on ARA, the Asterisk Realtime Architecture,
-; please read README.realtime and README.extconfig in the /doc directory of the
+; please read realtime.txt and extconfig.txt in the /doc directory of the
 ; source code.
 ;
 ;rtcachefriends=yes		; Cache realtime friends by adding them to the internal list



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