===
======================================================================
+------------------------------------------------------------------------------
+--- Functionality changes since Asterisk 1.6.2.20 -------------
+------------------------------------------------------------------------------
+
+SIP Changes
+-----------
+ * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
+ now defaults to yes. It is very important that phones requiring nat=no be
+ specifically set as such instead of relying on the default setting. If at all
+ possible, all devices should have nat settings configured in the general section as
+ opposed to configuring nat per-device.
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
------------------------------------------------------------------------------
}
} else if (!strcasecmp(v->name, "nat")) {
ast_set_flag(&mask[0], SIP_NAT);
- ast_clear_flag(&flags[0], SIP_NAT);
- if (!strcasecmp(v->value, "never"))
- ast_set_flag(&flags[0], SIP_NAT_NEVER);
- else if (!strcasecmp(v->value, "route"))
- ast_set_flag(&flags[0], SIP_NAT_ROUTE);
- else if (ast_true(v->value))
- ast_set_flag(&flags[0], SIP_NAT_ALWAYS);
- else
- ast_set_flag(&flags[0], SIP_NAT_RFC3581);
+ ast_set_flag(&flags[0], SIP_NAT_ALWAYS);
+ if (!strcasecmp(v->value, "never")) {
+ ast_set_flags_to(&flags[0], SIP_NAT, SIP_NAT_NEVER);
+ } else if (!strcasecmp(v->value, "route")) {
+ ast_set_flags_to(&flags[0], SIP_NAT, SIP_NAT_ROUTE);
+ } else if (ast_false(v->value)) {
+ ast_set_flags_to(&flags[0], SIP_NAT, SIP_NAT_RFC3581);
+ }
} else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
ast_clear_flag(&flags[0], SIP_REINVITE);
return 0;
}
+static void display_nat_warning(const char *cat, int reason, struct ast_flags *flags) {
+ int global_nat, specific_nat;
+
+ if (reason == CHANNEL_MODULE_LOAD && (specific_nat = ast_test_flag(&flags[0], SIP_NAT)) != (global_nat = ast_test_flag(&global_flags[0], SIP_NAT))) {
+ ast_log(LOG_WARNING, "!!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make\n");
+ ast_log(LOG_WARNING, "!!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users\n");
+ ast_log(LOG_WARNING, "!!! will be sent to a different port than replies for an existing peer/user. If at all possible,\n");
+ ast_log(LOG_WARNING, "!!! use the global 'nat' setting and do not set 'nat' per peer/user.\n");
+ ast_log(LOG_WARNING, "!!! (config category='%s' global='%s' peer/user='%s')\n", cat, nat2str(global_nat), nat2str(specific_nat));
+ }
+}
+
/*! \brief Re-read SIP.conf config file
\note This function reloads all config data, except for
active peers (with registrations). They will only
ast_copy_string(default_mohinterpret, DEFAULT_MOHINTERPRET, sizeof(default_mohinterpret));
ast_copy_string(default_mohsuggest, DEFAULT_MOHSUGGEST, sizeof(default_mohsuggest));
ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
- ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
- ast_set_flag(&global_flags[0], SIP_NAT_RFC3581); /*!< NAT support if requested by device with rport */
- ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA); /*!< Allow re-invites */
+ ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
+ ast_set_flag(&global_flags[0], SIP_NAT_RFC3581); /*!< NAT support if requested by device with rport */
+ ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA); /*!< Allow re-invites */
+ ast_set_flag(&global_flags[0], SIP_NAT_ALWAYS); /*!< Default to nat=yes */
ast_set_flag(&global_flags[1], SIP_PAGE2_FORWARD_LOOP_DETECTED); /*!< Set up call forward on 482 Loop Detected */
/* Debugging settings, always default to off */
}
peer = build_peer(cat, ast_variable_browse(cfg, cat), NULL, 0, 0);
if (peer) {
+ display_nat_warning(cat, reason, &peer->flags[0]);
ao2_t_link(peers, peer, "link peer into peers table");
if ((peer->type & SIP_TYPE_PEER) && peer->addr.sin_addr.s_addr) {
ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
; The following settings are allowed (both globally and in individual sections):
;
; nat = no ; default. Use NAT mode only according to RFC3581 (;rport)
-; nat = yes ; Always ignore info and assume NAT
+; nat = yes ; Always ignore info and assume NAT (default)
; nat = never ; Never attempt NAT mode or RFC3581 support
; nat = route ; route = Assume NAT, don't send rport
; ; (work around more UNIDEN bugs)
+;
+; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
+; the nat setting in a peer definition, then the peer username will be discoverable
+; by outside parties as Asterisk will respond to different ports for defined and
+; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
+; GENERAL SECTION. Specifically, if nat=route or nat=yes in one section and nat=no or
+; nat=never in the other, then valid users with settings differing from those in the
+; general section will be discoverable.
;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
type=friend
[natted-phone](!,basic-options) ; another template inheriting basic-options
- nat=yes
directmedia=no
host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
- nat=no
directmedia=yes
[my-codecs](!) ; a template for my preferred codecs
; on incoming calls to Asterisk
;host=192.168.0.23 ; we have a static but private IP address
; No registration allowed
-;nat=no ; there is not NAT between phone and Asterisk
;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
;regexten=1234 ; When they register, create extension 1234
;callerid="Jane Smith" <5678>
;host=dynamic ; This device needs to register
-;nat=yes ; X-Lite is behind a NAT router
;directmedia=no ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;type=friend
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
-;nat=yes ; This phone may be natted
- ; Send SIP and RTP to the IP address that packet is
- ; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;directmedia=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
projects / thirdparty / asterisk.git / commitdiff
