[asterisk-commits] jamesgolovich: branch group/sip-tcptls r92265 - /team/group/sip-tcptls/doc/

[SVN commits to the Asterisk project]

Author: jamesgolovich
Date: Tue Dec 11 00:05:26 2007
New Revision: 92265

URL: http://svn.digium.com/view/asterisk?view=rev&rev=92265
Log:
Update siptls.txt to include that asterisk is a viable sip tcp/tls 
client/server

Modified:
    team/group/sip-tcptls/doc/siptls.txt

Modified: team/group/sip-tcptls/doc/siptls.txt
URL: http://svn.digium.com/view/asterisk/team/group/sip-tcptls/doc/siptls.txt?view=diff&rev=92265&r1=92264&r2=92265
==============================================================================
--- team/group/sip-tcptls/doc/siptls.txt (original)
+++ team/group/sip-tcptls/doc/siptls.txt Tue Dec 11 00:05:26 2007
@@ -7,12 +7,13 @@
 you must install a copy of your CA on the client.
 
 So far this code has been test with:
+Asterisk as client and server (TLS and TCP)
 Polycom Soundpoint IP Phones (TLS and TCP)
 	Polycom phones require that the host (ip or hostname) that is
 	configured match the 'common name' in the certificate
-
 Minisip Softphone (TLS and TCP)
 Cisco IOS Gateways (TCP only)
+
 
 sip.conf options
 ----------------