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Add a missing part of the connected line work into trunk.

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Part of the work done for connected line was to add an optional
argument to the 'f' option to allow for the connected party information
of the outgoing channel to be set to the argument provided. This was
overlooked during the merge of the work to trunk and is being added
back now. The CHANGES file has also been updated to note this change.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
certified/1.8.6
Mark Michelson 17 years ago
parent 40275f8410
commit c9d1ffcae8
2 changed files with 29 additions and 8 deletions
Show Stats Download Patch File Download Diff File

4
CHANGES
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@ -127,6 +127,10 @@ Applications
* Page has a new option 'A(x)' which will playback an announcement simultaneously
to all paged phones (and optionally excluding the caller's one using the new
option 'n') before the call is bridged.
* The 'f' option to Dial has been augmented to take an optional argument. If no
argument is provided, the 'f' option works as it always has. If an argument is
provided, then the connected party information of all outgoing channels created
during the Dial will be set to the argument passed to the 'f' option.
Dialplan Functions
------------------

33
apps/app_dial.c
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@ -132,10 +132,13 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
<para>Execute the <literal>h</literal> extension for peer after the call ends</para>
</option>
<option name="f">
<para>Force the callerid of the <emphasis>calling</emphasis> channel to be set as the
extension associated with the channel using a dialplan <literal>hint</literal>.
<argument name="x" required="false" />
<para>If <replaceable>x</replaceable> is not provided, force the callerid of the <emphasis>calling</emphasis>
channel to be set as the extension associated with the channel using a dialplan <literal>hint</literal>.
For example, some PSTNs do not allow CallerID to be set to anything
other than the number assigned to the caller.</para>
other than the number assigned to the caller. If <replaceable>x</replaceable> is provided, though, then
this option behaves quite differently. Any outgoing channel created will have its connected party information
set to <replaceable>x</replaceable></para>
</option>
<option name="F" argsep="^">
<argument name="context" required="false" />
@ -546,6 +549,7 @@ enum {
OPT_ARG_DURATION_STOP,
OPT_ARG_OPERMODE,
OPT_ARG_SCREEN_NOINTRO,
OPT_ARG_FORCECLID,
/* note: this entry _MUST_ be the last one in the enum */
OPT_ARG_ARRAY_SIZE,
};
@ -558,7 +562,7 @@ AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
AST_APP_OPTION('d', OPT_DTMF_EXIT),
AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
AST_APP_OPTION('e', OPT_PEER_H),
AST_APP_OPTION('f', OPT_FORCECLID),
AST_APP_OPTION_ARG('f', OPT_FORCECLID, OPT_ARG_FORCECLID),
AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
AST_APP_OPTION('g', OPT_GO_ON),
AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
@ -1623,7 +1627,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
struct cause_args num = { chan, 0, 0, 0 };
int cause;
char numsubst[256];
char cidname[AST_MAX_EXTENSION] = "";
char *cid_num = NULL, *cid_name = NULL;
struct ast_bridge_config config = { { 0, } };
struct timeval calldurationlimit = { 0, };
@ -1718,6 +1722,8 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
goto done;
}
if (ast_test_flag64(&opts, OPT_FORCECLID) && !ast_strlen_zero(opt_args[OPT_ARG_FORCECLID]))
ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &cid_name, &cid_num);
if (ast_test_flag64(&opts, OPT_RESETCDR) && chan->cdr)
ast_cdr_reset(chan->cdr, NULL);
if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
@ -1748,7 +1754,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
}
ast_channel_unlock(chan);
ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_IGNORE_CONNECTEDLINE |
OPT_CANCEL_TIMEOUT | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB);
OPT_CANCEL_TIMEOUT | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB | OPT_FORCECLID);
/* loop through the list of dial destinations */
rest = args.peers;
@ -1894,8 +1900,18 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
}
ast_set_flag64(tmp, DIAL_NOCONNECTEDLINE);
}
ast_connected_line_copy_from_caller(&tc->connected, &chan->cid);
if (ast_test_flag64(peerflags, OPT_FORCECLID) && !ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
struct ast_party_connected_line connected;
ast_party_connected_line_set_init(&connected, &tmp->chan->connected);
connected.id.number = cid_num;
connected.id.name = cid_name;
connected.id.number_presentation = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
ast_channel_set_connected_line(tmp->chan, &connected);
} else {
ast_connected_line_copy_from_caller(&tmp->chan->connected, &chan->cid);
}
S_REPLACE(tc->cid.cid_rdnis, ast_strdup(chan->cid.cid_rdnis));
ast_party_redirecting_copy(&tc->redirecting, &chan->redirecting);
@ -1958,6 +1974,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
ast_verb(3, "Called %s\n", numsubst);
ast_channel_unlock(chan);
if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
char cidname[AST_MAX_EXTENSION];
ast_set_callerid(tc, tmpexten, get_cid_name(cidname, sizeof(cidname), chan), NULL);
}
}

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