[svn-commits] oej: branch 1.6.0 r128239 - in /branches/1.6.0: ./ channels/ configs/

[SVN commits to the Digium repositories]

Author: oej
Date: Sat Jul  5 15:49:40 2008
New Revision: 128239

URL: http://svn.digium.com/view/asterisk?view=rev&rev=128239
Log:
Merged revisions 128237 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

........
r128237 | oej | 2008-07-05 22:39:54 +0200 (Lör, 05 Jul 2008) | 2 lines

Make TCP disabled by default (it's considered experimental)

........

Modified:
    branches/1.6.0/   (props changed)
    branches/1.6.0/channels/chan_sip.c
    branches/1.6.0/configs/sip.conf.sample

Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=128239&r1=128238&r2=128239
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Sat Jul  5 15:49:40 2008
@@ -20285,7 +20285,9 @@
 	memset(&sip_tcp_desc.sin, 0, sizeof(sip_tcp_desc.sin));
 	memset(&sip_tls_desc.sin, 0, sizeof(sip_tls_desc.sin));
 
-	sip_tcp_desc.sin.sin_family = AF_INET;
+	/* sip_tcp_desc.sin.sin_family = AF_INET; Disabled, since TCP support in Asterisk is experimental */
+	sip_tcp_desc.sin.sin_family = 0;
+	default_tls_cfg.enabled = FALSE;
 
 	sip_tcp_desc.sin.sin_port = htons(STANDARD_SIP_PORT);
 	sip_tls_desc.sin.sin_port = htons(STANDARD_TLS_PORT);

Modified: branches/1.6.0/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/configs/sip.conf.sample?view=diff&rev=128239&r1=128238&r2=128239
==============================================================================
--- branches/1.6.0/configs/sip.conf.sample (original)
+++ branches/1.6.0/configs/sip.conf.sample Sat Jul  5 15:49:40 2008
@@ -75,9 +75,10 @@
 				; asterisk.conf, it defaults to that system name
 				; Realms MUST be globally unique according to RFC 3261
 				; Set this to your host name or domain name
-bindport=5060			; UDP Port to bind to (SIP standard port is 5060)
+bindport=5060			; UDP Port to bind to (SIP standard port for unencrypted UDP 
+				; and TCP sessions is 5060)
 				; bindport is the local UDP port that Asterisk will listen on
-bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
+bindaddr=0.0.0.0		; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
 
 ;
 ; Note that the TCP and TLS support for chan_sip is currently considered
@@ -85,7 +86,7 @@
 ; subject to change in any release.  If they are changed, the changes will
 ; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
 ;
-tcpenable=yes                   ; Enable server for incoming TCP connections (default is yes)
+tcpenable=no                    ; Enable server for incoming TCP connections (default is yes)
 tcpbindaddr=0.0.0.0	        ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
                                 ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)