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authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2010-03-31 17:49:07 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2010-03-31 17:49:07 +0000
commitf6797cb4bded2e9c81d97203e75a98402a7dc904 (patch)
tree14ee932255684662cbee90613aefa6ed20f72cfc
parent2e322b39295a49a14e1d223326ef9fa592f603c8 (diff)
Merged revisions 255504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31 Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T' can be used. (closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@255505 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--apps/app_dial.c6
-rw-r--r--configs/sip.conf.sample3
2 files changed, 6 insertions, 3 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c
index 44acc35f8..a12880aae 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -309,11 +309,13 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
</option>
<option name="t">
<para>Allow the called party to transfer the calling party by sending the
- DTMF sequence defined in <filename>features.conf</filename>.</para>
+ DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
+ transfers initiated by other methods.</para>
</option>
<option name="T">
<para>Allow the calling party to transfer the called party by sending the
- DTMF sequence defined in <filename>features.conf</filename>.</para>
+ DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
+ transfers initiated by other methods.</para>
</option>
<option name="U" argsep="^">
<argument name="x" required="true">
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 3d3c6555e..760b1fa53 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -92,7 +92,8 @@ context=default ; Default context for incoming calls
; instead of the From: field.
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