Merged revisions 285710 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.6.2

........
  r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines
  
  Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent.
  
  (closes issue #16903)
  Reported by: Nick_Lewis
  Patches: 
        pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
  Tested by: Nick_Lewis
........


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285711 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 37 insertions, 18 deletions

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 01fd29b00..320895669 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -144,6 +144,8 @@ allowoverlap=no                 ; Disable overlap dialing support. (Default is y
 ;  d) Listen on the IPv4 and IPv6 wildcards.  Example: bindaddr=::
 ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
 ; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
+; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
+;  IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
 ;
 ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
 ; for TLS).
@@ -213,7 +215,7 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;pedantic=yes                   ; Enable checking of tags in headers,
                                 ; international character conversions in URIs
                                 ; and multiline formatted headers for strict
-                                ; SIP compatibility (defaults to "no")
+                                ; SIP compatibility (defaults to "yes")
 
 ; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
 ;tos_sip=cs3                    ; Sets TOS for SIP packets.
@@ -253,6 +255,18 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; Message-Account in the MWI notify message
                                 ; defaults to "asterisk"
 
+; Codec negotiation
+;
+; When Asterisk is receiving a call, the codec will initially be set to the
+; first codec in the allowed codecs defined for the user receiving the call
+; that the caller also indicates that it supports. But, after the caller
+; starts sending RTP, Asterisk will switch to using whatever codec the caller
+; is sending.
+;
+; When Asterisk is placing a call, the codec used will be the first codec in
+; the allowed codecs that the callee indicates that it supports. Asterisk will
+; *not* switch to whatever codec the callee is sending.
+;
 ;preferred_codec_only=yes       ; Respond to a SIP invite with the single most preferred codec
                                 ; rather than advertising all joint codec capabilities. This
                                 ; limits the other side's codec choice to exactly what we prefer.
@@ -354,6 +368,10 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; instead of letting the requester know whether there was
                                 ; a matching user or peer for their request.  This reduces
                                 ; the ability of an attacker to scan for valid SIP usernames.
+                                ; This option is set to "yes" by default.
+
+;auth_options_requests = yes    ; Enabling this option will authenticate OPTIONS requests just like
+                                ; INVITE requests are.  By default this option is disabled.
 
 ;g726nonstandard = yes          ; If the peer negotiates G726-32 audio, use AAL2 packing
                                 ; order instead of RFC3551 packing order (this is required
@@ -364,6 +382,10 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;outboundproxy=proxy.provider.domain:8080       ; send outbound signaling to this proxy, not directly to the devices
 ;outboundproxy=proxy.provider.domain,force      ; Send ALL outbound signalling to proxy, ignoring route: headers
 ;outboundproxy=tls://proxy.provider.domain      ; same as '=proxy.provider.domain' except we try to connect with tls
+;outboundproxy=192.0.2.1                        ; IPv4 address literal (default port is 5060)
+;outboundproxy=2001:db8::1                      ; IPv6 address literal (default port is 5060)
+;outboundproxy=192.168.0.2.1:5062               ; IPv4 address literal with explicit port
+;outboundproxy=[2001:db8::1]:5062               ; IPv6 address literal with explicit port
 ;                                               ; (could also be tcp,udp) - defining transports on the proxy line only
 ;                                               ; applies for the global proxy, otherwise use the transport= option
 ;matchexternaddrlocally = yes     ; Only substitute the externaddr or externhost setting if it matches
@@ -720,31 +742,18 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;        externhost=foo.dyndns.net       ; refreshed periodically
 ;        externrefresh=180               ; change the refresh interval
 ;
-;   c. "stunaddr = stun.server[:port]" queries the STUN server specified
-;      as an argument to obtain the external address/port.
-;      Queries are also sent periodically every "externrefresh" seconds
-;      (as a side effect, sending the query also acts as a keepalive for
-;      the state entry on the nat box):
-;
-;        stunaddr = foo.stun.com:3478
-;        externrefresh = 15
-;
-;      NOTE: STUN is only implemented for IPv4.
-;
 ;   Note that at the moment all these mechanism work only for the SIP socket.
-;   The IP address discovered with externaddr/externhost/STUN is reused for
+;   The IP address discovered with externaddr/externhost is reused for
 ;   media sessions as well, but the port numbers are not remapped so you
 ;   may still experience problems.
 ;
 ; NOTE 1: in some cases, NAT boxes will use different port numbers in
 ; the internal<->external mapping. In these cases, the "externaddr" and
-; "externhost" might not help you configure addresses properly, and you
-; really need to use STUN.
+; "externhost" might not help you configure addresses properly.
 ;
 ; NOTE 2: when using "externaddr" or "externhost", the address part is
-; also used as the external address for media sessions. Even if you 
-; use "stunaddr", STUN queries will be sent only from the SIP port,
-; not from media sockets. Thus, the port information in the SDP may be wrong!
+; also used as the external address for media sessions. Thus, the port
+; information in the SDP may be wrong!
 ;
 ; In addition to the above, Asterisk has an additional "nat" parameter to
 ; address NAT-related issues in incoming SIP or media sessions.
@@ -776,6 +785,16 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ; can not be set per-user or per-peer.
 ;
 ; media_address = 172.16.42.1
+;
+; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
+; perceived external network address has changed.  When the stun_monitor is installed and
+; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
+; of network change has occurred. By default this option is enabled, but only takes effect once
+; res_stun_monitor is configured.  If res_stun_monitor is enabled and you wish to not
+; generate all outbound registrations on a network change, use the option below to disable
+; this feature.
+;
+; subscribe_network_change_event = yes ; on by default
 
 ;----------------------------------- MEDIA HANDLING --------------------------------
 ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's