[asterisk-commits] may: branch may/ooh323_ipv6_direct_rtp r349927 - in /team/may/ooh323_ipv6_dir...

[SVN commits to the Asterisk project]

Author: may
Date: Fri Jan  6 15:13:13 2012
New Revision: 349927

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=349927
Log:
fix blobs

Modified:
    team/may/ooh323_ipv6_direct_rtp/addons/ooh323c/src/ooh245.c
    team/may/ooh323_ipv6_direct_rtp/configs/ooh323.conf.sample

Modified: team/may/ooh323_ipv6_direct_rtp/addons/ooh323c/src/ooh245.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_ipv6_direct_rtp/addons/ooh323c/src/ooh245.c?view=diff&rev=349927&r1=349926&r2=349927
==============================================================================
--- team/may/ooh323_ipv6_direct_rtp/addons/ooh323c/src/ooh245.c (original)
+++ team/may/ooh323_ipv6_direct_rtp/addons/ooh323c/src/ooh245.c Fri Jan  6 15:13:13 2012
@@ -3264,7 +3264,7 @@
       OOTRACEDBGC3("Empty TCS found.  (%s, %s)\n",
                     call->callType, call->callToken);
 
-      ooH245AcknowledgeTerminalCapabilitySet(call);   
+      ooH245AcknowledgeTerminalCapabilitySet(call);
       call->remoteTermCapSeqNo = tcs->sequenceNumber;
 
 /* close all transmit chans */

Modified: team/may/ooh323_ipv6_direct_rtp/configs/ooh323.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_ipv6_direct_rtp/configs/ooh323.conf.sample?view=diff&rev=349927&r1=349926&r2=349927
==============================================================================
--- team/may/ooh323_ipv6_direct_rtp/configs/ooh323.conf.sample (original)
+++ team/may/ooh323_ipv6_direct_rtp/configs/ooh323.conf.sample Fri Jan  6 15:13:13 2012
@@ -27,13 +27,13 @@
 ;       OOH323/exten/peer OR OOH323/exten at ip
 ;
 ; Domain name resolution is not yet supported.
-; 
+;
 ; When a H.323 user calls into asterisk, his H323ID is matched with the profile
 ; name and context is determined to route the call
 ;
-; The channel driver will register all global aliases and aliases defined in 
+; The channel driver will register all global aliases and aliases defined in
 ; peer profiles with the gatekeeper, if one exists. So, that when someone
-; outside our pbx (non-user) calls an extension, gatekeeper will route that 
+; outside our pbx (non-user) calls an extension, gatekeeper will route that
 ; call to our asterisk box, from where it will be routed as per dial plan.
 
 
@@ -47,9 +47,9 @@
 ;The dotted IP address asterisk should listen on for incoming H323
 ;connections
 ;Default - tries to find out local ip address on it's own
-bindaddr=0.0.0.0    
-
-;This parameter indicates whether channel driver should register with 
+bindaddr=0.0.0.0
+
+;This parameter indicates whether channel driver should register with
 ;gatekeeper as a gateway or an endpoint.
 ;Default - no
 ;gateway=no
@@ -65,7 +65,7 @@
 
 ;H323-ID to be used for asterisk server
 ;Default - Asterisk PBX
-h323id=ObjSysAsterisk 
+h323id=ObjSysAsterisk
 e164=100
 
 ;CallerID to use for calls
@@ -128,7 +128,7 @@
 ; CNG tone or an incoming T.38 RequestMode packet
 ;
 ; yes - enable both detection (CNG & T.38)
-; no - disable both 
+; no - disable both
 ; cng - enable CNG detection (default)
 ; t38 - enable T.38 request detection
 ;
@@ -137,7 +137,7 @@
 ; User/peer/friend definitions:
 ; User config options                    Peer config options
 ; ------------------                     -------------------
-; context                            
+; context
 ; disallow                               disallow
 ; allow                                  allow
 ; accountcode                            accountcode
@@ -174,7 +174,7 @@
 context=context1
 disallow=all
 allow=gsm
-allow=ulaw    
+allow=ulaw