[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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