- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" and "tlsbindaddr".

Note: I don't think we can start properly without UDP port open, that needs to be tested.

- Removing "bindport" from configuration example, not needed to mention this any more

I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128525 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -21147,7 +21147,7 @@ static int reload_config(enum channelreloadreason reason)
 			if (ast_parse_arg(v->value, PARSE_INADDR, &stunaddr))
 				ast_log(LOG_WARNING, "Invalid STUN server address: %s\n", v->value);
 			externexpire = time(NULL);
-		} else if (!strcasecmp(v->name, "bindaddr")) {
+		} else if (!strcasecmp(v->name, "bindaddr") || !strcasecmp(v->name, "udpbindaddr")) {
 			if (ast_parse_arg(v->value, PARSE_INADDR, &bindaddr))
 				ast_log(LOG_WARNING, "Invalid address: %s\n", v->value);
 		} else if (!strcasecmp(v->name, "localnet")) {

configs/sip.conf.sample

@@ -75,11 +75,8 @@ allowoverlap=no			; Disable overlap dialing support. (Default is yes)
 				; 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 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 UDP listen socket to (0.0.0.0 binds to all)
-				; You can specify port here too, like 123.123.123.123:5080
+udpbindaddr=0.0.0.0		; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
+                                ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
 
 ;
 ; Note that the TCP and TLS support for chan_sip is currently considered