[asterisk-commits] russell: branch 1.6.2 r234012 - in /branches/1.6.2: ./ CHANGES

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Dec 9 17:17:28 CST 2009


Author: russell
Date: Wed Dec  9 17:17:25 2009
New Revision: 234012

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=234012
Log:
Merged revisions 234008 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

........
  r234008 | russell | 2009-12-09 17:13:28 -0600 (Wed, 09 Dec 2009) | 5 lines
  
  Fix up the faxdetect entry in CHANGES.
  
  This feature was listed as a 1.6.2 feature, even though it's in all 1.6.X
  versions.  The description of the feature was also no longer accurate.
........

Modified:
    branches/1.6.2/   (props changed)
    branches/1.6.2/CHANGES

Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.2/CHANGES
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/CHANGES?view=diff&rev=234012&r1=234011&r2=234012
==============================================================================
--- branches/1.6.2/CHANGES (original)
+++ branches/1.6.2/CHANGES Wed Dec  9 17:17:25 2009
@@ -25,9 +25,6 @@
    remote services. For backwards compatibility, "secret" still has the
    same function as before, but now you can configure both a remote secret and a
    local secret for mutual authentication.
- * Added a new 'faxdetect=yes|no' configuration option to sip.conf.  When this
-   option is enabled, a SIP channel will go to the fax extension (if it exists)
-   after T38 is negotiated.  This option is disabled by default.
  * If the channel variable  ATTENDED_TRANSFER_COMPLETE_SOUND is set, 
    the sound will be played to the target of an attended transfer
  * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
@@ -530,6 +527,10 @@
 
 SIP changes
 -----------
+ * Added a new 'faxdetect=yes|no' configuration option to sip.conf.  When this
+    option is enabled, Asterisk will watch for a CNG tone in the incoming audio
+    for a received call.  If it is detected, the channel will jump to the 
+    'fax' extension in the dialplan.
   * Improved NAT and STUN support.
      chan_sip now can use port numbers in bindaddr, externip and externhost
      options, as well as contact a STUN server to detect its external address




More information about the asterisk-commits mailing list