Enhancements to connected line and redirecting work.

From reviewboard:

Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.

First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.

Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.

Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.

Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.

Review: https://reviewboard.asterisk.org/r/652/


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263541 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 7 insertions, 0 deletions

diff --git a/channels/misdn_config.c b/channels/misdn_config.c
index 82d3547f7..e14f2adee 100644
--- a/channels/misdn_config.c
+++ b/channels/misdn_config.c
@@ -134,6 +134,13 @@ static const struct misdn_cfg_spec port_spec[] = {
 		"Sets the musiconhold class." },
 	{ "callerid", MISDN_CFG_CALLERID, MISDN_CTYPE_STR, "", NONE,
 		"Set the outgoing caller id to the value." },
+	{ "incoming_cid_tag", MISDN_CFG_INCOMING_CALLERID_TAG, MISDN_CTYPE_STR, "", NONE,
+		"Set the incoming caller id string tag to the value." },
+	{ "append_msn_to_cid_tag", MISDN_CFG_APPEND_MSN_TO_CALLERID_TAG, MISDN_CTYPE_BOOL, "no", NONE,
+		"Automatically appends incoming or outgoing MSN to the incoming caller\n"
+		"\tid string tag. An underscore '_' is used as delimiter. Incoming calls\n"
+		"\twill have the dialed number appended, and outgoing calls will have the\n"
+		"\tcaller number appended to the tag." },
 	{ "method", MISDN_CFG_METHOD, MISDN_CTYPE_STR, "standard", NONE,
 		"Set the method to use for channel selection:\n"
 		"\t  standard     - Use the first free channel starting from the lowest number.\n"