[asterisk-commits] branch oej/disable-ol-and-sub r13614 - in
/team/oej/disable-ol-and-sub: ./ ap...
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Sun Mar 19 05:03:24 MST 2006
Author: oej
Date: Sun Mar 19 06:03:20 2006
New Revision: 13614
URL: http://svn.digium.com/view/asterisk?rev=13614&view=rev
Log:
reset automerge
Modified:
team/oej/disable-ol-and-sub/ (props changed)
team/oej/disable-ol-and-sub/apps/app_dial.c
team/oej/disable-ol-and-sub/channels/chan_sip.c
team/oej/disable-ol-and-sub/configs/sip.conf.sample
Propchange: team/oej/disable-ol-and-sub/
------------------------------------------------------------------------------
automerge = http://edvina.net/training/
Propchange: team/oej/disable-ol-and-sub/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.
Propchange: team/oej/disable-ol-and-sub/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Mar 19 06:03:20 2006
@@ -1,1 +1,1 @@
-/trunk:1-13521
+/trunk:1-13587
Modified: team/oej/disable-ol-and-sub/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/oej/disable-ol-and-sub/apps/app_dial.c?rev=13614&r1=13613&r2=13614&view=diff
==============================================================================
--- team/oej/disable-ol-and-sub/apps/app_dial.c (original)
+++ team/oej/disable-ol-and-sub/apps/app_dial.c Sun Mar 19 06:03:20 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/disable-ol-and-sub/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/disable-ol-and-sub/channels/chan_sip.c?rev=13614&r1=13613&r2=13614&view=diff
==============================================================================
--- team/oej/disable-ol-and-sub/channels/chan_sip.c (original)
+++ team/oej/disable-ol-and-sub/channels/chan_sip.c Sun Mar 19 06:03:20 2006
@@ -391,7 +391,7 @@
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 */
@@ -412,7 +412,7 @@
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 int global_callevents; /*!< Whether we send manager events or not */
@@ -622,7 +622,6 @@
#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 {
@@ -1679,15 +1678,15 @@
{
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);
}
}
}
@@ -1709,7 +1708,7 @@
ast_sched_del(sched, peer->expire);
if (peer->pokeexpire > -1)
ast_sched_del(sched, peer->pokeexpire);
- register_peer_exten(peer, 0);
+ register_peer_exten(peer, FALSE);
ast_free_ha(peer->ha);
if (ast_test_flag((&peer->flags_page2), SIP_PAGE2_SELFDESTRUCT))
apeerobjs--;
@@ -5806,7 +5805,7 @@
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)) {
@@ -5885,7 +5884,7 @@
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 */
@@ -8009,7 +8008,6 @@
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";
@@ -8125,6 +8123,8 @@
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) {
@@ -8200,6 +8200,8 @@
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);
@@ -8351,7 +8353,7 @@
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");
@@ -12486,7 +12488,7 @@
/* 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;
@@ -12505,6 +12507,7 @@
global_rtptimeout = 0;
global_rtpholdtimeout = 0;
global_rtpkeepalive = 0;
+ global_rtautoclear = 120;
global_allowsubscribe = TRUE; /* Global flag, default = TRUE */
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 */
@@ -12600,10 +12603,10 @@
} 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/disable-ol-and-sub/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/team/oej/disable-ol-and-sub/configs/sip.conf.sample?rev=13614&r1=13613&r2=13614&view=diff
==============================================================================
--- team/oej/disable-ol-and-sub/configs/sip.conf.sample (original)
+++ team/oej/disable-ol-and-sub/configs/sip.conf.sample Sun Mar 19 06:03:20 2006
@@ -207,7 +207,7 @@
;----------------------------------------- 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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