[asterisk-commits] lmadsen: branch 1.4 r255503 - in /branches/1.4: apps/ configs/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Mar 31 12:43:02 CDT 2010


Author: lmadsen
Date: Wed Mar 31 12:42:58 2010
New Revision: 255503

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=255503
Log:
Add documentation clarifying when 't' and 'T' can be used.

(closes issue #17021)
Reported by: kovzol
Tested by: lmadsen, kovzol, davidw, ebroad

Modified:
    branches/1.4/apps/app_dial.c
    branches/1.4/configs/sip.conf.sample

Modified: branches/1.4/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/apps/app_dial.c?view=diff&rev=255503&r1=255502&r2=255503
==============================================================================
--- branches/1.4/apps/app_dial.c (original)
+++ branches/1.4/apps/app_dial.c Wed Mar 31 12:42:58 2010
@@ -203,10 +203,12 @@
 "           answered the call.\n"  	
 "    t    - Allow the called party to transfer the calling party by sending the\n"
 "           DTMF sequence defined in the blindxfer setting in the featuremap section\n"
-"           of features.conf.\n"
+"           of features.conf. This setting does not perform policy enforcement on\n"
+"           transfers initiated by other methods.\n"
 "    T    - Allow the calling party to transfer the called party by sending the\n"
 "           DTMF sequence defined in the blindxfer setting in the featuremap section\n"
-"           of features.conf.\n"
+"           of features.conf. This setting does not perform policy enforcement on\n"
+"           transfers initiated by other methods.\n"
 "    w    - Allow the called party to enable recording of the call by sending\n"
 "           the DTMF sequence defined in the automon setting in the featuremap section\n"
 "           of features.conf.\n"

Modified: branches/1.4/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/configs/sip.conf.sample?view=diff&rev=255503&r1=255502&r2=255503
==============================================================================
--- branches/1.4/configs/sip.conf.sample (original)
+++ branches/1.4/configs/sip.conf.sample Wed Mar 31 12:42:58 2010
@@ -47,7 +47,8 @@
 ;allowguest=no                  ; Allow or reject guest calls (default is yes)
 allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
 ;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
-                                ; Default is enabled
+                                ; Default is enabled. The Dial() options 't' and 'T' are not
+                                ; related as to whether SIP transfers are allowed or not.
 ;realm=mydomain.tld             ; Realm for digest authentication
                                 ; defaults to "asterisk". If you set a system name in
                                 ; asterisk.conf, it defaults to that system name




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