[svn-commits] eliel: branch group/appdocsxml r152687 - in /team/group/appdocsxml: ./ apps/ ...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Oct 29 19:32:53 CDT 2008
Author: eliel
Date: Wed Oct 29 19:32:52 2008
New Revision: 152687
URL: http://svn.digium.com/view/asterisk?view=rev&rev=152687
Log:
Resolve automerge conflicts.
Modified:
team/group/appdocsxml/ (props changed)
team/group/appdocsxml/apps/app_dial.c
team/group/appdocsxml/apps/app_directory.c
team/group/appdocsxml/apps/app_queue.c
team/group/appdocsxml/apps/app_stack.c
team/group/appdocsxml/apps/app_transfer.c
team/group/appdocsxml/apps/app_voicemail.c
team/group/appdocsxml/channels/chan_dahdi.c
team/group/appdocsxml/channels/chan_local.c
team/group/appdocsxml/channels/chan_mgcp.c
team/group/appdocsxml/channels/chan_sip.c
team/group/appdocsxml/configs/features.conf.sample
team/group/appdocsxml/configs/phoneprov.conf.sample
team/group/appdocsxml/include/asterisk/features.h
team/group/appdocsxml/include/asterisk/pbx.h
team/group/appdocsxml/main/features.c
Propchange: team/group/appdocsxml/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/group/appdocsxml/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Wed Oct 29 19:32:52 2008
@@ -1,1 +1,1 @@
-/trunk:1-152069
+/trunk:1-152686
Modified: team/group/appdocsxml/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/apps/app_dial.c?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/apps/app_dial.c (original)
+++ team/group/appdocsxml/apps/app_dial.c Wed Oct 29 19:32:52 2008
@@ -1,7 +1,7 @@
/*
* Asterisk -- An open source telephony toolkit.
*
- * Copyright (C) 1999 - 2006, Digium, Inc.
+ * Copyright (C) 1999 - 2008, Digium, Inc.
*
* Mark Spencer <markster at digium.com>
*
@@ -189,15 +189,18 @@
</variable>
<variable name="LIMIT_TIMEOUT_FILE">
<value name="filename"/>
- <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached. If not set, the time remaining will be announced.</para>
+ <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
+ If not set, the time remaining will be announced.</para>
</variable>
<variable name="LIMIT_CONNECT_FILE">
<value name="filename"/>
- <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins. If not set, the time remaining will be announced.</para>
+ <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
+ If not set, the time remaining will be announced.</para>
</variable>
<variable name="LIMIT_WARNING_FILE">
<value name="filename"/>
- <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
+ <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
+ a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
</variable>
</variablelist>
</option>
@@ -383,7 +386,7 @@
If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,
however, the variable will be unset after use.</para>
-
+
<para>This application sets the following channel variables:</para>
<variablelist>
<variable name="DIALEDTIME">
@@ -1499,6 +1502,13 @@
struct ast_dial_features *caller_features;
int fulldial = 0, num_dialed = 0;
+ /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
+ pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
+ pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
+ pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
+ pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
+ pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
+
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
@@ -2002,16 +2012,17 @@
struct ast_app *theapp;
const char *gosub_result;
char *gosub_args, *gosub_argstart;
-
- res = ast_autoservice_start(chan);
- if (res) {
+ int res9 = -1;
+
+ res9 = ast_autoservice_start(chan);
+ if (res9) {
ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
- res = -1;
+ res9 = -1;
}
theapp = pbx_findapp("Gosub");
- if (theapp && !res) {
+ if (theapp && !res9) {
replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
/* Set where we came from */
@@ -2029,50 +2040,50 @@
}
if (gosub_args) {
- res = pbx_exec(peer, theapp, gosub_args);
+ res9 = pbx_exec(peer, theapp, gosub_args);
ast_pbx_run(peer);
ast_free(gosub_args);
if (option_debug)
- ast_log(LOG_DEBUG, "Gosub exited with status %d\n", res);
+ ast_log(LOG_DEBUG, "Gosub exited with status %d\n", res9);
} else
ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
- res = 0;
- } else if (!res) {
+ res9 = 0;
+ } else if (!res9) {
ast_log(LOG_ERROR, "Could not find application Gosub\n");
- res = -1;
+ res9 = -1;
}
if (ast_autoservice_stop(chan) < 0) {
ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
- res = -1;
+ res9 = -1;
}
ast_channel_lock(peer);
- if (!res && (gosub_result = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
+ if (!res9 && (gosub_result = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
char *gosub_transfer_dest;
if (!strcasecmp(gosub_result, "BUSY")) {
ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
ast_set_flag64(peerflags, OPT_GO_ON);
- res = -1;
+ res9 = -1;
} else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
ast_set_flag64(peerflags, OPT_GO_ON);
- res = -1;
+ res9 = -1;
} else if (!strcasecmp(gosub_result, "CONTINUE")) {
/* hangup peer and keep chan alive assuming the macro has changed
the context / exten / priority or perhaps
the next priority in the current exten is desired.
*/
ast_set_flag64(peerflags, OPT_GO_ON);
- res = -1;
+ res9 = -1;
} else if (!strcasecmp(gosub_result, "ABORT")) {
/* Hangup both ends unless the caller has the g flag */
- res = -1;
+ res9 = -1;
} else if (!strncasecmp(gosub_result, "GOTO:", 5) && (gosub_transfer_dest = ast_strdupa(gosub_result + 5))) {
- res = -1;
+ res9 = -1;
/* perform a transfer to a new extension */
if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
replace_macro_delimiter(gosub_transfer_dest);
@@ -2161,32 +2172,34 @@
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - start_time));
pbx_builtin_setvar_helper(chan, "DIALEDTIME", toast);
- if (ast_test_flag64(&opts, OPT_PEER_H)) {
+ if (res != AST_PBX_NO_HANGUP_PEER_PARKED && ast_test_flag64(&opts, OPT_PEER_H)) {
ast_log(LOG_NOTICE, "PEER context: %s; PEER exten: %s; PEER priority: %d\n",
peer->context, peer->exten, peer->priority);
}
-
- strcpy(peer->context, chan->context);
-
- if (ast_test_flag64(&opts, OPT_PEER_H) && ast_exists_extension(peer, peer->context, "h", 1, peer->cid.cid_num)) {
+ if (res != AST_PBX_NO_HANGUP_PEER_PARKED)
+ strcpy(peer->context, chan->context);
+
+ if (res != AST_PBX_NO_HANGUP_PEER_PARKED && ast_test_flag64(&opts, OPT_PEER_H) && ast_exists_extension(peer, peer->context, "h", 1, peer->cid.cid_num)) {
int autoloopflag;
int found;
+ int res9;
+
strcpy(peer->exten, "h");
peer->priority = 1;
autoloopflag = ast_test_flag(peer, AST_FLAG_IN_AUTOLOOP); /* save value to restore at the end */
ast_set_flag(peer, AST_FLAG_IN_AUTOLOOP);
- while ((res = ast_spawn_extension(peer, peer->context, peer->exten, peer->priority, peer->cid.cid_num, &found, 1)) == 0)
+ while ((res9 = ast_spawn_extension(peer, peer->context, peer->exten, peer->priority, peer->cid.cid_num, &found, 1)) == 0)
peer->priority++;
- if (found && res) {
+ if (found && res9) {
/* Something bad happened, or a hangup has been requested. */
ast_debug(1, "Spawn extension (%s,%s,%d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
ast_verb(2, "Spawn extension (%s, %s, %d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
}
ast_set2_flag(peer, autoloopflag, AST_FLAG_IN_AUTOLOOP); /* set it back the way it was */
}
- if (res != AST_PBX_NO_HANGUP_PEER) {
+ if (res != AST_PBX_NO_HANGUP_PEER && res != AST_PBX_NO_HANGUP_PEER_PARKED) {
if (!ast_check_hangup(peer) && ast_test_flag64(&opts, OPT_CALLEE_GO_ON) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GO_ON])) {
replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GO_ON]);
ast_parseable_goto(peer, opt_args[OPT_ARG_CALLEE_GO_ON]);
@@ -2199,23 +2212,28 @@
}
}
out:
- if (moh) {
- moh = 0;
- ast_moh_stop(chan);
- } else if (sentringing) {
- sentringing = 0;
- ast_indicate(chan, -1);
- }
- ast_channel_early_bridge(chan, NULL);
- hanguptree(outgoing, NULL, 0); /* In this case, there's no answer anywhere */
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
- senddialendevent(chan, pa.status);
- ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
-
- if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_KEEPALIVE) && (res != AST_PBX_INCOMPLETE)) {
- if (calldurationlimit)
- memset(&chan->whentohangup, 0, sizeof(chan->whentohangup));
- res = 0;
+ /* cleaning up chan is not a good idea here if AST_PBX_KEEPALIVE
+ is returned; chan will get the love it needs from another
+ thread */
+ if (res != AST_PBX_KEEPALIVE) {
+ if (moh) {
+ moh = 0;
+ ast_moh_stop(chan);
+ } else if (sentringing) {
+ sentringing = 0;
+ ast_indicate(chan, -1);
+ }
+ ast_channel_early_bridge(chan, NULL);
+ hanguptree(outgoing, NULL, 0); /* In this case, there's no answer anywhere */
+ pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
+ senddialendevent(chan, pa.status);
+ ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
+
+ if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_KEEPALIVE) && (res != AST_PBX_INCOMPLETE)) {
+ if (calldurationlimit)
+ memset(&chan->whentohangup, 0, sizeof(chan->whentohangup));
+ res = 0;
+ }
}
done:
Modified: team/group/appdocsxml/apps/app_directory.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/apps/app_directory.c?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/apps/app_directory.c (original)
+++ team/group/appdocsxml/apps/app_directory.c Wed Oct 29 19:32:52 2008
@@ -159,6 +159,10 @@
{
char digit;
+ if (ast_strlen_zero(text)) {
+ return -1;
+ }
+
while (*template) {
digit = toupper(*text++);
switch (digit) {
Modified: team/group/appdocsxml/apps/app_queue.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/apps/app_queue.c?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/apps/app_queue.c (original)
+++ team/group/appdocsxml/apps/app_queue.c Wed Oct 29 19:32:52 2008
@@ -3993,21 +3993,24 @@
/* If the queue member did an attended transfer, then the TRANSFER already was logged in the queue_log
* when the masquerade occurred. These other "ending" queue_log messages are unnecessary
*/
- if (!attended_transfer_occurred(qe->chan)) {
+ if (bridge != AST_PBX_KEEPALIVE && !attended_transfer_occurred(qe->chan)) {
struct ast_datastore *transfer_ds;
if (strcasecmp(oldcontext, qe->chan->context) || strcasecmp(oldexten, qe->chan->exten)) {
ast_queue_log(queuename, qe->chan->uniqueid, member->membername, "TRANSFER", "%s|%s|%ld|%ld|%d",
qe->chan->exten, qe->chan->context, (long) (callstart - qe->start),
(long) (time(NULL) - callstart), qe->opos);
- send_agent_complete(qe, queuename, peer, member, callstart, vars, sizeof(vars), TRANSFER);
+ if (bridge != AST_PBX_NO_HANGUP_PEER && bridge != AST_PBX_NO_HANGUP_PEER_PARKED)
+ send_agent_complete(qe, queuename, peer, member, callstart, vars, sizeof(vars), TRANSFER);
} else if (ast_check_hangup(qe->chan)) {
ast_queue_log(queuename, qe->chan->uniqueid, member->membername, "COMPLETECALLER", "%ld|%ld|%d",
(long) (callstart - qe->start), (long) (time(NULL) - callstart), qe->opos);
- send_agent_complete(qe, queuename, peer, member, callstart, vars, sizeof(vars), CALLER);
+ if (bridge != AST_PBX_NO_HANGUP_PEER && bridge != AST_PBX_NO_HANGUP_PEER_PARKED)
+ send_agent_complete(qe, queuename, peer, member, callstart, vars, sizeof(vars), CALLER);
} else {
ast_queue_log(queuename, qe->chan->uniqueid, member->membername, "COMPLETEAGENT", "%ld|%ld|%d",
(long) (callstart - qe->start), (long) (time(NULL) - callstart), qe->opos);
- send_agent_complete(qe, queuename, peer, member, callstart, vars, sizeof(vars), AGENT);
+ if (bridge != AST_PBX_NO_HANGUP_PEER && bridge != AST_PBX_NO_HANGUP_PEER_PARKED)
+ send_agent_complete(qe, queuename, peer, member, callstart, vars, sizeof(vars), AGENT);
}
ast_channel_lock(qe->chan);
transfer_ds = ast_channel_datastore_find(qe->chan, &queue_transfer_info, NULL);
@@ -4019,7 +4022,7 @@
update_queue(qe->parent, member, callcompletedinsl);
}
- if (bridge != AST_PBX_NO_HANGUP_PEER)
+ if (bridge != AST_PBX_NO_HANGUP_PEER && bridge != AST_PBX_NO_HANGUP_PEER_PARKED)
ast_hangup(peer);
res = bridge ? bridge : 1;
ao2_ref(member, -1);
Modified: team/group/appdocsxml/apps/app_stack.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/apps/app_stack.c?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/apps/app_stack.c (original)
+++ team/group/appdocsxml/apps/app_stack.c Wed Oct 29 19:32:52 2008
@@ -118,8 +118,6 @@
static void gosub_release_frame(struct ast_channel *chan, struct gosub_stack_frame *frame)
{
- unsigned char i;
- char argname[15];
struct ast_var_t *vardata;
/* If chan is not defined, then we're calling it as part of gosub_free,
@@ -128,14 +126,6 @@
* that frame, so that we re-expose the variables from the previous frame
* that were hidden by this one.
*/
- if (chan) {
- for (i = 1; i <= frame->arguments && i != 0; i++) {
- snprintf(argname, sizeof(argname), "ARG%hhd", i);
- pbx_builtin_setvar_helper(chan, argname, NULL);
- }
- }
-
- /* Delete local variables */
while ((vardata = AST_LIST_REMOVE_HEAD(&frame->varshead, entries))) {
if (chan)
pbx_builtin_setvar_helper(chan, ast_var_name(vardata), NULL);
@@ -295,6 +285,8 @@
frame_set_var(chan, newframe, argname, args2.argval[i]);
ast_debug(1, "Setting '%s' to '%s'\n", argname, args2.argval[i]);
}
+ snprintf(argname, sizeof(argname), "%d", args2.argc);
+ frame_set_var(chan, newframe, "ARGC", argname);
/* And finally, save our return address */
oldlist = stack_store->data;
Modified: team/group/appdocsxml/apps/app_transfer.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/apps/app_transfer.c?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/apps/app_transfer.c (original)
+++ team/group/appdocsxml/apps/app_transfer.c Wed Oct 29 19:32:52 2008
@@ -42,12 +42,9 @@
Transfer caller to remote extension.
</synopsis>
<syntax>
- <parameter name="destination" required="true" argsep="/">
+ <parameter name="dest" required="true" argsep="/">
<argument name="Tech" />
- <argument name="dest" required="true" />
- </parameter>
- <parameter name="options">
- <para>No options available.</para>
+ <argument name="destination" required="true" />
</parameter>
</syntax>
<description>
@@ -88,20 +85,16 @@
char *parse;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(dest);
- AST_APP_ARG(options);
);
if (ast_strlen_zero((char *)data)) {
- ast_log(LOG_WARNING, "Transfer requires an argument ([Tech/]destination[,options])\n");
+ ast_log(LOG_WARNING, "Transfer requires an argument ([Tech/]destination)\n");
pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE");
return 0;
} else
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
-
- if (args.options) {
- }
dest = args.dest;
Modified: team/group/appdocsxml/apps/app_voicemail.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/apps/app_voicemail.c?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/apps/app_voicemail.c (original)
+++ team/group/appdocsxml/apps/app_voicemail.c Wed Oct 29 19:32:52 2008
@@ -3889,7 +3889,7 @@
}
fprintf(p, "%s %s <%s>" ENDL, first_line ? "From:" : "", passdata, who);
} else {
- fprintf(p, "From: %s <%s>" ENDL, quote(passdata, passdata2, len_passdata2), who);
+ fprintf(p, "From: %s <%s>" ENDL, quote(passdata2, passdata, len_passdata), who);
}
ast_channel_free(ast);
} else {
Modified: team/group/appdocsxml/channels/chan_dahdi.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/channels/chan_dahdi.c?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/channels/chan_dahdi.c (original)
+++ team/group/appdocsxml/channels/chan_dahdi.c Wed Oct 29 19:32:52 2008
@@ -13763,8 +13763,6 @@
confp->chan.buf_no = numbufs;
if (!strcasecmp(policy, "full")) {
confp->chan.buf_policy = DAHDI_POLICY_WHEN_FULL;
- } else if (!strcasecmp(policy, "half")) {
- confp->chan.buf_policy = DAHDI_POLICY_IMMEDIATE /*HALF_FULL*/;
} else if (!strcasecmp(policy, "immediate")) {
confp->chan.buf_policy = DAHDI_POLICY_IMMEDIATE;
} else {
Modified: team/group/appdocsxml/channels/chan_local.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/channels/chan_local.c?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/channels/chan_local.c (original)
+++ team/group/appdocsxml/channels/chan_local.c Wed Oct 29 19:32:52 2008
@@ -505,11 +505,15 @@
* Note that cid_num and cid_name aren't passed in the ast_channel_alloc
* call, so it's done here instead.
*/
+ p->chan->cid.cid_dnid = ast_strdup(p->owner->cid.cid_dnid);
p->chan->cid.cid_num = ast_strdup(p->owner->cid.cid_num);
p->chan->cid.cid_name = ast_strdup(p->owner->cid.cid_name);
p->chan->cid.cid_rdnis = ast_strdup(p->owner->cid.cid_rdnis);
p->chan->cid.cid_ani = ast_strdup(p->owner->cid.cid_ani);
p->chan->cid.cid_pres = p->owner->cid.cid_pres;
+ p->chan->cid.cid_ani2 = p->owner->cid.cid_ani2;
+ p->chan->cid.cid_ton = p->owner->cid.cid_ton;
+ p->chan->cid.cid_tns = p->owner->cid.cid_tns;
ast_string_field_set(p->chan, language, p->owner->language);
ast_string_field_set(p->chan, accountcode, p->owner->accountcode);
ast_cdr_update(p->chan);
Modified: team/group/appdocsxml/channels/chan_mgcp.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/channels/chan_mgcp.c?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/channels/chan_mgcp.c (original)
+++ team/group/appdocsxml/channels/chan_mgcp.c Wed Oct 29 19:32:52 2008
@@ -3431,8 +3431,9 @@
ast_verb(1, "Reloading MGCP\n");
reload_config(1);
/* Add an I/O event to our UDP socket */
- if (mgcpsock > -1)
+ if (mgcpsock > -1 && !mgcpsock_read_id) {
mgcpsock_read_id = ast_io_add(io, mgcpsock, mgcpsock_read, AST_IO_IN, NULL);
+ }
}
/* Check for interfaces needing to be killed */
Modified: team/group/appdocsxml/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/channels/chan_sip.c?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/channels/chan_sip.c (original)
+++ team/group/appdocsxml/channels/chan_sip.c Wed Oct 29 19:32:52 2008
@@ -7264,7 +7264,7 @@
struct sockaddr_in peer;
ast_rtp_get_peer(p->rtp, &peer);
if (peer.sin_addr.s_addr) {
- memcpy(&sin.sin_addr, &peer.sin_addr, sizeof(&sin.sin_addr));
+ memcpy(&sin.sin_addr, &peer.sin_addr, sizeof(sin.sin_addr));
if (debug) {
ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_inet_ntoa(sin.sin_addr));
}
Modified: team/group/appdocsxml/configs/features.conf.sample
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/configs/features.conf.sample?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/configs/features.conf.sample (original)
+++ team/group/appdocsxml/configs/features.conf.sample Wed Oct 29 19:32:52 2008
@@ -52,12 +52,12 @@
[featuremap]
-;blindxfer => #1 ; Blind transfer (default is #)
-;disconnect => *0 ; Disconnect (default is *)
-;automon => *1 ; One Touch Record a.k.a. Touch Monitor
-;atxfer => *2 ; Attended transfer
-;parkcall => #72 ; Park call (one step parking)
-;automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor
+;blindxfer => #1 ; Blind transfer (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
+;disconnect => *0 ; Disconnect (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
+;automon => *1 ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
+;atxfer => *2 ; Attended transfer -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
+;parkcall => #72 ; Park call (one step parking) -- Make sure to set the K and/or K option in the Dial() app call!
+;automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!
[applicationmap]
; Note that the DYNAMIC_FEATURES channel variable must be set to use the features
Modified: team/group/appdocsxml/configs/phoneprov.conf.sample
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/configs/phoneprov.conf.sample?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/configs/phoneprov.conf.sample (original)
+++ team/group/appdocsxml/configs/phoneprov.conf.sample Wed Oct 29 19:32:52 2008
@@ -53,6 +53,46 @@
static_file => sip.ver,plain/text
static_file => sip.cfg
static_file => custom.cfg
+static_file => 2201-06642-001.bootrom.ld,application/octet-stream
+static_file => 2201-06642-001.sip.ld,application/octet-stream
+static_file => 2345-11000-001.bootrom.ld,application/octet-stream
+static_file => 2345-11300-001.bootrom.ld,application/octet-stream
+static_file => 2345-11300-010.bootrom.ld,application/octet-stream
+static_file => 2345-11300-010.sip.ld,application/octet-stream
+static_file => 2345-11402-001.bootrom.ld,application/octet-stream
+static_file => 2345-11402-001.sip.ld,application/octet-stream
+static_file => 2345-11500-001.bootrom.ld,application/octet-stream
+static_file => 2345-11500-010.bootrom.ld,application/octet-stream
+static_file => 2345-11500-020.bootrom.ld,application/octet-stream
+static_file => 2345-11500-030.bootrom.ld,application/octet-stream
+static_file => 2345-11500-030.sip.ld,application/octet-stream
+static_file => 2345-11500-040.bootrom.ld,application/octet-stream
+static_file => 2345-11500-040.sip.ld,application/octet-stream
+static_file => 2345-11600-001.bootrom.ld,application/octet-stream
+static_file => 2345-11600-001.sip.ld,application/octet-stream
+static_file => 2345-11605-001.bootrom.ld,application/octet-stream
+static_file => 2345-11605-001.sip.ld,application/octet-stream
+static_file => 2345-12200-001.bootrom.ld,application/octet-stream
+static_file => 2345-12200-001.sip.ld,application/octet-stream
+static_file => 2345-12200-002.bootrom.ld,application/octet-stream
+static_file => 2345-12200-002.sip.ld,application/octet-stream
+static_file => 2345-12200-004.bootrom.ld,application/octet-stream
+static_file => 2345-12200-004.sip.ld,application/octet-stream
+static_file => 2345-12200-005.bootrom.ld,application/octet-stream
+static_file => 2345-12200-005.sip.ld,application/octet-stream
+static_file => 2345-12500-001.bootrom.ld,application/octet-stream
+static_file => 2345-12500-001.sip.ld,application/octet-stream
+static_file => 2345-12560-001.bootrom.ld,application/octet-stream
+static_file => 2345-12560-001.sip.ld,application/octet-stream
+static_file => 2345-12600-001.bootrom.ld,application/octet-stream
+static_file => 2345-12600-001.sip.ld,application/octet-stream
+static_file => 2345-12670-001.bootrom.ld,application/octet-stream
+static_file => 2345-12670-001.sip.ld,application/octet-stream
+static_file => 3111-15600-001.bootrom.ld,application/octet-stream
+static_file => 3111-15600-001.sip.ld,application/octet-stream
+static_file => 3111-40000-001.bootrom.ld,application/octet-stream
+static_file => 3111-40000-001.sip.ld,application/octet-stream
+
${MAC}.cfg => 000000000000.cfg ; Dynamically generated files.
${MAC}-phone.cfg => 000000000000-phone.cfg ; (relative to AST_DATA_DIR/phoneprov)
config/${MAC} => polycom.xml ; Dynamic Filename => template file
Modified: team/group/appdocsxml/include/asterisk/features.h
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/include/asterisk/features.h?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/include/asterisk/features.h (original)
+++ team/group/appdocsxml/include/asterisk/features.h Wed Oct 29 19:32:52 2008
@@ -61,14 +61,15 @@
};
-#define AST_FEATURE_RETURN_HANGUP -1
-#define AST_FEATURE_RETURN_SUCCESSBREAK 0
-#define AST_FEATURE_RETURN_PBX_KEEPALIVE AST_PBX_KEEPALIVE
-#define AST_FEATURE_RETURN_NO_HANGUP_PEER AST_PBX_NO_HANGUP_PEER
-#define AST_FEATURE_RETURN_PASSDIGITS 21
-#define AST_FEATURE_RETURN_STOREDIGITS 22
-#define AST_FEATURE_RETURN_SUCCESS 23
-#define AST_FEATURE_RETURN_KEEPTRYING 24
+#define AST_FEATURE_RETURN_HANGUP -1
+#define AST_FEATURE_RETURN_SUCCESSBREAK 0
+#define AST_FEATURE_RETURN_PBX_KEEPALIVE AST_PBX_KEEPALIVE
+#define AST_FEATURE_RETURN_NO_HANGUP_PEER AST_PBX_NO_HANGUP_PEER
+#define AST_FEATURE_RETURN_NO_HANGUP_PEER_PARKED AST_PBX_NO_HANGUP_PEER_PARKED
+#define AST_FEATURE_RETURN_PASSDIGITS 21
+#define AST_FEATURE_RETURN_STOREDIGITS 22
+#define AST_FEATURE_RETURN_SUCCESS 23
+#define AST_FEATURE_RETURN_KEEPTRYING 24
/*!
Modified: team/group/appdocsxml/include/asterisk/pbx.h
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/include/asterisk/pbx.h?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/include/asterisk/pbx.h (original)
+++ team/group/appdocsxml/include/asterisk/pbx.h Wed Oct 29 19:32:52 2008
@@ -38,12 +38,13 @@
#define AST_PBX_REPLACE 1
/*! \brief Special return values from applications to the PBX { */
-#define AST_PBX_HANGUP -1 /*!< Jump to the 'h' exten */
-#define AST_PBX_OK 0 /*!< No errors */
-#define AST_PBX_ERROR 1 /*!< Jump to the 'e' exten */
-#define AST_PBX_KEEPALIVE 10 /*!< Destroy the thread, but don't hang up the channel */
-#define AST_PBX_NO_HANGUP_PEER 11
-#define AST_PBX_INCOMPLETE 12 /*!< Return to PBX matching, allowing more digits for the extension */
+#define AST_PBX_HANGUP -1 /*!< Jump to the 'h' exten */
+#define AST_PBX_OK 0 /*!< No errors */
+#define AST_PBX_ERROR 1 /*!< Jump to the 'e' exten */
+#define AST_PBX_KEEPALIVE 10 /*!< Destroy the thread, but don't hang up the channel */
+#define AST_PBX_NO_HANGUP_PEER 11 /*!< The peer has been involved in a transfer */
+#define AST_PBX_INCOMPLETE 12 /*!< Return to PBX matching, allowing more digits for the extension */
+#define AST_PBX_NO_HANGUP_PEER_PARKED 13 /*!< Don't touch the peer channel - it was sent to the parking lot and might be gone by now */
/*! } */
#define PRIORITY_HINT -1 /*!< Special Priority for a hint */
Modified: team/group/appdocsxml/main/features.c
URL: http://svn.digium.com/view/asterisk/team/group/appdocsxml/main/features.c?view=diff&rev=152687&r1=152686&r2=152687
==============================================================================
--- team/group/appdocsxml/main/features.c (original)
+++ team/group/appdocsxml/main/features.c Wed Oct 29 19:32:52 2008
@@ -836,7 +836,7 @@
res = ast_park_call(parkee, parker, 0, NULL);
if (!res) {
if (sense == FEATURE_SENSE_CHAN) {
- res = AST_PBX_NO_HANGUP_PEER;
+ res = AST_PBX_NO_HANGUP_PEER_PARKED;
} else {
res = AST_PBX_KEEPALIVE;
}
@@ -1186,7 +1186,7 @@
the thread dies -- We have to be careful now though. We are responsible for
hanging up the channel, else it will never be hung up! */
- return (transferer == peer) ? AST_PBX_KEEPALIVE : AST_PBX_NO_HANGUP_PEER;
+ return (transferer == peer) ? AST_PBX_KEEPALIVE : AST_PBX_NO_HANGUP_PEER_PARKED;
} else {
ast_log(LOG_WARNING, "Unable to park call %s\n", transferee->name);
}
@@ -1789,6 +1789,8 @@
}
else if (res == AST_PBX_NO_HANGUP_PEER)
return AST_FEATURE_RETURN_NO_HANGUP_PEER;
+ else if (res == AST_PBX_NO_HANGUP_PEER_PARKED)
+ return AST_FEATURE_RETURN_NO_HANGUP_PEER_PARKED;
else if (res)
return AST_FEATURE_RETURN_SUCCESSBREAK;
@@ -2436,10 +2438,12 @@
}
before_you_go:
- new_chan_cdr = pick_unlocked_cdr(chan->cdr); /* the proper chan cdr, if there are forked cdrs */
- new_peer_cdr = pick_unlocked_cdr(peer->cdr); /* the proper chan cdr, if there are forked cdrs */
-
- if (!ast_test_flag(&(config->features_caller),AST_FEATURE_NO_H_EXTEN) && ast_exists_extension(chan, chan->context, "h", 1, chan->cid.cid_num)) {
+
+ /* run the hangup exten on the chan object IFF it was NOT involved in a parking situation
+ * if it were, then chan belongs to a different thread now, and might have been hung up long
+ * ago.
+ */
+ if (res != AST_PBX_KEEPALIVE && !ast_test_flag(&(config->features_caller),AST_FEATURE_NO_H_EXTEN) && ast_exists_extension(chan, chan->context, "h", 1, chan->cid.cid_num)) {
struct ast_cdr *swapper;
char savelastapp[AST_MAX_EXTENSION];
char savelastdata[AST_MAX_EXTENSION];
@@ -2483,39 +2487,57 @@
ast_copy_string(bridge_cdr->lastdata, savelastdata, sizeof(bridge_cdr->lastdata));
}
- /* obey the NoCDR() wishes. */
- if (new_chan_cdr && ast_test_flag(new_chan_cdr, AST_CDR_FLAG_POST_DISABLED) && new_peer_cdr && !ast_test_flag(new_peer_cdr, AST_CDR_FLAG_POST_DISABLED))
- ast_set_flag(new_peer_cdr, AST_CDR_FLAG_POST_DISABLED); /* DISABLED is viral-- it will propagate across a bridge */
- if (!new_chan_cdr || (new_chan_cdr && !ast_test_flag(new_chan_cdr, AST_CDR_FLAG_POST_DISABLED))) {
- struct ast_channel *chan_ptr = NULL;
-
- ast_cdr_end(bridge_cdr);
-
- ast_cdr_detach(bridge_cdr);
-
- /* do a specialized reset on the beginning channel
- CDR's, if they still exist, so as not to mess up
- issues in future bridges;
-
- Here are the rules of the game:
- 1. The chan and peer channel pointers will not change
- during the life of the bridge.
- 2. But, in transfers, the channel names will change.
- between the time the bridge is started, and the
- time the channel ends.
- Usually, when a channel changes names, it will
- also change CDR pointers.
- 3. Usually, only one of the two channels (chan or peer)
- will change names.
- 4. Usually, if a channel changes names during a bridge,
- it is because of a transfer. Usually, in these situations,
- it is normal to see 2 bridges running simultaneously, and
- it is not unusual to see the two channels that change
- swapped between bridges.
- 5. After a bridge occurs, we have 2 or 3 channels' CDRs
- to attend to; if the chan or peer changed names,
- we have the before and after attached CDR's.
- */
+ /* obey the NoCDR() wishes. -- move the DISABLED flag to the bridge CDR if it was set on the channel during the bridge... */
+ if (res != AST_PBX_KEEPALIVE) {
+ new_chan_cdr = pick_unlocked_cdr(chan->cdr); /* the proper chan cdr, if there are forked cdrs */
+ if (new_chan_cdr && ast_test_flag(new_chan_cdr, AST_CDR_FLAG_POST_DISABLED))
+ ast_set_flag(bridge_cdr, AST_CDR_FLAG_POST_DISABLED);
+ }
+
+ /* we can post the bridge CDR at this point */
+ ast_cdr_end(bridge_cdr);
+ ast_cdr_detach(bridge_cdr);
+
+ /* do a specialized reset on the beginning channel
+ CDR's, if they still exist, so as not to mess up
+ issues in future bridges;
+
+ Here are the rules of the game:
+ 1. The chan and peer channel pointers will not change
+ during the life of the bridge.
+ 2. But, in transfers, the channel names will change.
+ between the time the bridge is started, and the
+ time the channel ends.
+ Usually, when a channel changes names, it will
+ also change CDR pointers.
+ 3. Usually, only one of the two channels (chan or peer)
+ will change names.
+ 4. Usually, if a channel changes names during a bridge,
+ it is because of a transfer. Usually, in these situations,
+ it is normal to see 2 bridges running simultaneously, and
+ it is not unusual to see the two channels that change
+ swapped between bridges.
+ 5. After a bridge occurs, we have 2 or 3 channels' CDRs
+ to attend to; if the chan or peer changed names,
+ we have the before and after attached CDR's.
+ 6. Parking has to be accounted for in the code:
+ a. Parking will cause ast_bridge_call to return
+ either AST_PBX_NO_HANGUP_PEER or AST_PBX_NO_HANGUP_PEER_PARKED;
+ in the latter case, peer is (most likely) a bad
+ pointer, you can no longer deref it. If it does still
+ exist, it is under another's thread control, and
+ could be destroyed at any time.
+ b. The same applies to AST_PBX_KEEPALIVE, in which
+ case, the chan ptr cannot be used, as another thread
+ owns it and may have destroyed the channel.
+ c. In the former case, you need to check peer to see if it
+ still exists before you deref it, and obtain a lock.
+ d. In neither case should you do an ast_hangup(peer).
+ e. Do not overwrite the result code from ast_bridge_call.
+ */
+
+ if (res != AST_PBX_KEEPALIVE && new_chan_cdr) {
+ struct ast_channel *chan_ptr = NULL;
if (strcasecmp(orig_channame, chan->name) != 0) {
/* old channel */
@@ -2538,28 +2560,35 @@
} else {
ast_cdr_specialized_reset(chan_cdr,0); /* nothing changed, reset the chan_cdr */
}
- if (strcasecmp(orig_peername, peer->name) != 0) {
- /* old channel */
- chan_ptr = ast_get_channel_by_name_locked(orig_peername);
- if (chan_ptr) {
- if (!ast_bridged_channel(chan_ptr)) {
- struct ast_cdr *cur;
- for (cur = chan_ptr->cdr; cur; cur = cur->next) {
+ }
+
+ if (res != AST_PBX_NO_HANGUP_PEER_PARKED) { /* if the peer was involved in a park, don't even touch it; it's probably gone */
+ struct ast_channel *chan_ptr = NULL;
+ new_peer_cdr = pick_unlocked_cdr(peer->cdr); /* the proper chan cdr, if there are forked cdrs */
+ if (new_chan_cdr && ast_test_flag(new_chan_cdr, AST_CDR_FLAG_POST_DISABLED) && new_peer_cdr && !ast_test_flag(new_peer_cdr, AST_CDR_FLAG_POST_DISABLED))
+ ast_set_flag(new_peer_cdr, AST_CDR_FLAG_POST_DISABLED); /* DISABLED is viral-- it will propagate across a bridge */
+ if (strcasecmp(orig_peername, peer->name) != 0) {
+ /* old channel */
+ chan_ptr = ast_get_channel_by_name_locked(orig_peername);
+ if (chan_ptr) {
+ if (!ast_bridged_channel(chan_ptr)) {
+ struct ast_cdr *cur;
+ for (cur = chan_ptr->cdr; cur; cur = cur->next) {
if (cur == peer_cdr) {
- break;
- }
- }
- if (cur)
- ast_cdr_specialized_reset(peer_cdr,0);
- }
- ast_channel_unlock(chan_ptr);
- }
- /* new channel */
- ast_cdr_specialized_reset(new_peer_cdr,0);
- } else {
- ast_cdr_specialized_reset(peer_cdr,0); /* nothing changed, reset the peer_cdr */
- }
- }
+ break;
+ }
+ }
+ if (cur)
+ ast_cdr_specialized_reset(peer_cdr,0);
+ }
+ ast_channel_unlock(chan_ptr);
+ }
+ /* new channel */
+ ast_cdr_specialized_reset(new_peer_cdr,0);
+ } else {
+ ast_cdr_specialized_reset(peer_cdr,0); /* nothing changed, reset the peer_cdr */
+ }
+ }
return res;
}
@@ -3014,7 +3043,7 @@
ast_cdr_setdestchan(chan->cdr, peer->name);
/* Simulate the PBX hanging up */
- if (res != AST_PBX_NO_HANGUP_PEER)
+ if (res != AST_PBX_NO_HANGUP_PEER && res != AST_PBX_NO_HANGUP_PEER_PARKED)
ast_hangup(peer);
return res;
} else {
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