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-rw-r--r--configs/sip.conf.sample15
1 files changed, 8 insertions, 7 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 3c77a88be..cb492161e 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -824,6 +824,14 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; for their media streams is not actual port number that will be used on the nearer
; side of the NAT.
;
+; 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=force_rport in one section and nat=no in the
+; other, then valid users with settings differing from those in the general section will
+; be discoverable.
+;
; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
; to receive them on.
@@ -1212,12 +1220,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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
@@ -1257,7 +1263,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'.
;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
@@ -1287,7 +1292,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;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
@@ -1361,9 +1365,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;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