Author: jrose
Date: Fri Sep  5 15:11:35 2014
New Revision: 422684

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=422684
Log:
Dial API: Add a dial option to indicate the dialed channel will replace dialer

Adds an option to the dial API that marks an outgoing dial as replacing the 
dialing channel for the purpose of propagating accountcode. When it is used, 
AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of 
AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on the involved 
channels with ast_channel_req_accountcodes.

Review: https://reviewboard.asterisk.org/r/3968/

Modified:
    branches/13/include/asterisk/dial.h
    branches/13/main/dial.c

Modified: branches/13/include/asterisk/dial.h
URL: 
http://svnview.digium.com/svn/asterisk/branches/13/include/asterisk/dial.h?view=diff&rev=422684&r1=422683&r2=422684
==============================================================================
--- branches/13/include/asterisk/dial.h (original)
+++ branches/13/include/asterisk/dial.h Fri Sep  5 15:11:35 2014
@@ -45,6 +45,7 @@
        AST_DIAL_OPTION_MUSIC,                   /*!< Play music on hold 
instead of ringing to the calling channel */
        AST_DIAL_OPTION_DISABLE_CALL_FORWARDING, /*!< Disable call forwarding 
on channels */
        AST_DIAL_OPTION_PREDIAL,                 /*!< Execute a predial 
subroutine before dialing */
+       AST_DIAL_OPTION_DIAL_REPLACES_SELF,      /*!< The dial operation is a 
replacement for the requester */
        AST_DIAL_OPTION_MAX,                     /*!< End terminator -- must 
always remain last */
 };
 

Modified: branches/13/main/dial.c
URL: 
http://svnview.digium.com/svn/asterisk/branches/13/main/dial.c?view=diff&rev=422684&r1=422683&r2=422684
==============================================================================
--- branches/13/main/dial.c (original)
+++ branches/13/main/dial.c Fri Sep  5 15:11:35 2014
@@ -205,6 +205,7 @@
        { AST_DIAL_OPTION_MUSIC, music_enable, music_disable },                 
  /*!< Play music to the caller instead of ringing */
        { AST_DIAL_OPTION_DISABLE_CALL_FORWARDING, NULL, NULL },                
  /*!< Disable call forwarding on channels */
        { AST_DIAL_OPTION_PREDIAL, predial_enable, predial_disable },           
  /*!< Execute a subroutine on the outbound channels prior to dialing */
+       { AST_DIAL_OPTION_DIAL_REPLACES_SELF, NULL, NULL },                     
  /*!< The dial operation is a replacement for the requester */
        { AST_DIAL_OPTION_MAX, NULL, NULL },                                    
  /*!< Terminator of list */
 };
 
@@ -344,7 +345,11 @@
                
ast_connected_line_copy_from_caller(ast_channel_connected(channel->owner), 
ast_channel_caller(chan));
 
                ast_channel_language_set(channel->owner, 
ast_channel_language(chan));
-               ast_channel_req_accountcodes(channel->owner, chan, 
AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
+               if (channel->options[AST_DIAL_OPTION_DIAL_REPLACES_SELF]) {
+                       ast_channel_req_accountcodes(channel->owner, chan, 
AST_CHANNEL_REQUESTOR_REPLACEMENT);
+               } else {
+                       ast_channel_req_accountcodes(channel->owner, chan, 
AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
+               }
                if (ast_strlen_zero(ast_channel_musicclass(channel->owner)))
                        ast_channel_musicclass_set(channel->owner, 
ast_channel_musicclass(chan));
 


-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

svn-commits mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/svn-commits

Reply via email to