[svn-commits] trunk r36177 - in /trunk: channels/chan_sip.c doc/extconfig.txt doc/realtime.txt

[svn-commits at lists.digium.com]

Author: oej
Date: Tue Jun 27 12:30:14 2006
New Revision: 36177

URL: http://svn.digium.com/view/asterisk?rev=36177&view=rev
Log:
Inspired by issue 6742, but solved in a different way.
(Yes, I like the system name setting)

Modified:
    trunk/channels/chan_sip.c
    trunk/doc/extconfig.txt
    trunk/doc/realtime.txt

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=36177&r1=36176&r2=36177&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Jun 27 12:30:14 2006
@@ -2175,12 +2175,20 @@
 	return 0;	
 }
 
-/*! \brief Update peer object in realtime storage */
+/*! \brief Update peer object in realtime storage 
+	If the Asterisk system name is set in asterisk.conf, we will use
+	that name and store that in the "regserver" field in the sippeers
+	table to facilitate multi-server setups.
+*/
 static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey)
 {
 	char port[10];
 	char ipaddr[20];
 	char regseconds[20];
+
+	char *sysname = ast_config_AST_SYSTEM_NAME;
+	char *syslabel = NULL;
+
 	time_t nowtime = time(NULL) + expirey;
 	const char *fc = fullcontact ? "fullcontact" : NULL;
 	
@@ -2188,9 +2196,14 @@
 	ast_inet_ntoa(ipaddr, sizeof(ipaddr), sin->sin_addr);
 	snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port));
 	
+	if (ast_strlen_zero(sysname))	/* No system name, disable this */
+		sysname = NULL;
+	else
+		syslabel = "regserver";
+
 	ast_update_realtime("sippeers", "name", peername, "ipaddr", ipaddr,
 		"port", port, "regseconds", regseconds,
-		"username", username, fc, fullcontact, NULL); /* note fc _can_ be NULL */
+		"username", username, fc, fullcontact, syslabel, sysname, NULL); /* note fc _can_ be NULL */
 }
 
 /*! \brief Automatically add peer extension to dial plan */
@@ -7141,7 +7154,7 @@
 {
 	if (!ast_test_flag(&global_flags[1], SIP_PAGE2_IGNOREREGEXPIRE)) {
 		if (ast_test_flag(&peer->flags[1], SIP_PAGE2_RT_FROMCONTACT))
-			ast_update_realtime("sippeers", "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "", "regseconds", "0", "username", "", NULL);
+			ast_update_realtime("sippeers", "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "", "regseconds", "0", "username", "", "regserver", "", NULL);
 		else 
 			ast_db_del("SIP/Registry", peer->name);
 	}

Modified: trunk/doc/extconfig.txt
URL: http://svn.digium.com/view/asterisk/trunk/doc/extconfig.txt?rev=36177&r1=36176&r2=36177&view=diff
==============================================================================
--- trunk/doc/extconfig.txt (original)
+++ trunk/doc/extconfig.txt Tue Jun 27 12:30:14 2006
@@ -54,6 +54,12 @@
 
 in order to store appropriate parameters required for SIP.
 
+In addition to this, if you add a field named "regserver" to the
+SIP peers table and have the system name set in asterisk.conf, 
+Asterisk will store the system name that the user registered on in 
+the database. This can be used to direct calls to go through the server 
+that holds the registration (for NAT traversal purposes).
+
 A Voicemail table would look more like this:
 
 +----------+---------+----------+----------+-----------+---------------+

Modified: trunk/doc/realtime.txt
URL: http://svn.digium.com/view/asterisk/trunk/doc/realtime.txt?rev=36177&r1=36176&r2=36177&view=diff
==============================================================================
--- trunk/doc/realtime.txt (original)
+++ trunk/doc/realtime.txt Tue Jun 27 12:30:14 2006
@@ -49,9 +49,8 @@
 most of the functionality works the same way for realtime friends as for
 the ones in static configuration.
 
-There is some work to create a solution for Realtime SIP devices that
-loads from database and stays in memory for the duration of a call or
-a registration, but that work is not integrated into Asterisk yet.
+With caching, the device stays in memory for a specified time. More 
+information about this is to be found in the sip.conf sample file.
 
 * New function in the dial plan: The Realtime Switch
 ----------------------------------------------------
@@ -94,8 +93,11 @@
 Defined well-known family names are:
 
 * sippeers, sipusers	SIP peers and users
-* iaxfriends		IAX2 peers
+* iaxpeers, iaxusers	IAX2 peers and users
 * voicemail		Voicemail accounts 
+* queues		Queues
+* queue_members		Queue members
+* extensions		Realtime extensions (switch)
 
 There is documentation of the SQL database in the file
 doc/extconfig.txt in your Asterisk source code tree.