Fix small sip conf issues (bug #3296)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4731 65c4cc65-6c06-0410-ace0-fbb531ad65f3

configs/sip.conf.sample

@@ -1,5 +1,5 @@
 ;
-; SIP Configuration for Asterisk
+; SIP Configuration example for Asterisk
 ;
 ; Syntax for specifying a SIP device in extensions.conf is
 ; SIP/devicename where devicename is defined in a section below.
@@ -52,7 +52,7 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 
 ;disallow=all			; First disallow all codecs
 ;allow=ulaw			; Allow codecs in order of preference
-;allow=ilbc			; Note: codec order is respected only in [general]
+;allow=ilbc			; 
 ;musicclass=default		; Sets the default music on hold class for all SIP calls
 				; This may also be set for individual users/peers
 ;language=en			; Default language setting for all users/peers
@@ -67,22 +67,16 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 				; use 'never' to never use in-band signalling, even in cases
 				; where some buggy devices might not render it
 ;useragent=Asterisk PBX		; Allows you to change the user agent string
-;nat=no				; NAT settings 
-                                ; yes = Always ignore info and assume NAT
-                                ; no = Use NAT mode only according to RFC3581 
-                                ; never = Never attempt NAT mode or RFC3581 support
-				; route = Assume NAT, don't send rport 
-				; (work around more UNIDEN bugs)
 ;promiscredir = no      	; If yes, allows 302 or REDIR to non-local SIP address
 	                       	; Note that promiscredir when redirects are made to the
        	                	; local system will cause loops since SIP is incapable
+       	                	; of performing a "hairpin" call.
 ;usereqphone = no		; If yes, ";user=phone" is added to uri that contains
 				; a valid phone number
-       	                	; of performing a "hairpin" call.
 ;dtmfmode = rfc2833		; Set default dtmfmode for sending DTMF. Default: rfc2833
 				; Other options: 
 				; info : SIP INFO messages
-				; inband : Inband audio
+				; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
 
 ;compactheaders = yes		; send compact sip headers.
 
@@ -125,26 +119,43 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
   
 ;registertimeout=20		; retry registration calls every 20 seconds (default)
 
+;---------------------------------------------- NAT SUPPORT ------------------------
+; The externip, externhost and localnet settings are used if you use Asterisk behind
+; a NAT device to communicate with services on the outside.
+
 ;externip = 200.201.202.203	; Address that we're going to put in outbound SIP messages
 				; if we're behind a NAT
 
 				; The externip and localnet is used
 				; when registering and communicating with other proxies
 				; that we're registered with
-				; You may add multiple local networks.  A reasonable set of defaults
-				; are:
 ;externhost=foo.dyndns.net	; Alternatively you can specify an 
 				; external host, and Asterisk will 
 				; perform DNS queries periodically.  Not
 				; recommended for production 
 				; environments!  Use externip instead
 ;externrefresh=10		; How often to refresh externhost if 
-				; usedl
+				; used
+				; You may add multiple local networks.  A reasonable set of defaults
+				; are:
 ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
 ;localnet=10.0.0.0/255.0.0.0	; Also RFC1918
 ;localnet=172.16.0.0/12		; Another RFC1918 with CIDR notation
 ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
 
+; The nat= setting is used when Asterisk is on a public IP, communicating with
+; devices hidden behind a NAT device (broadband router).
+; If you have one-way audio problems, you usually have problems with your NAT 
+; configuration or your firewalls support of SIP+RTP ports.
+; You configure Asterisk choice of RTP ports for incoming audio in rtp.conf
+;
+;nat=no				; Global NAT settings  (Affects all peers and users)
+                                ; yes = Always ignore info and assume NAT
+                                ; no = Use NAT mode only according to RFC3581 
+                                ; never = Never attempt NAT mode or RFC3581 support
+				; route = Assume NAT, don't send rport 
+				; (work around more UNIDEN bugs)
+
 ;-----------------------------------------------------------------------------------
 ; Users and peers have different settings available. Friends have all settings,
 ; since a friend is both a peer and a user
@@ -191,23 +202,41 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 
 ;[sip_proxy]
 ; For incoming calls only. Example: FWD (Free World Dialup)
-;type=user
+; We match on IP address of the proxy for incoming calls 
+; since we can not match on username (caller id)
+;type=peer
 ;context=from-fwd
+;host=fwd.pulver.com
 
 ;[sip_proxy-out]
 ;type=peer          		; we only want to call out, not be called
 ;secret=guessit
 ;username=yourusername		; Authentication user for outbound proxies
 ;fromuser=yourusername		; Many SIP providers require this!
+;fromdomain=provider.sip.domain	
 ;host=box.provider.com
 ;usereqphone=yes		; This provider requires ";user=phone" on URI
 
+;------------------------------------------------------------------------------
+; Definitions of locally connected SIP phones
+;
+; type = user	a device that calls us
+; type = peer	a device we place calls to
+; type = friend two configurations (peer+user) in one
+;
+; For local phones, type=friend works most of the time
+;
+; If you have one-way audio, you propably have NAT problems. 
+; If Asterisk is on a public IP, and the phone is inside of a NAT device
+; you will need to configure nat option for those phones.
+; Also, turn on qualify=yes to keep the nat session open
+
 ;[grandstream1]
-;type=friend 			; either "friend" (peer+user), "peer" or "user"
+;type=friend 			
-;context=from-sip
+;context=from-sip		; Where to start in the dialplan when this phone calls
-;fromuser=grandstream1		; overrides the callerid, e.g. required by FWD
+;callerid=John Doe <1234>	; Full caller ID, to override the phones config
-;callerid=John Doe <1234>
 ;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
 ;canreinvite=yes		; allow RTP voice traffic to bypass Asterisk
 ;dtmfmode=info			; either RFC2833 or INFO for the BudgeTone
@@ -223,13 +252,12 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 
 
 ;[xlite1]
-;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
+; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
-;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
+; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
 ;type=friend
 ;regexten=1234			; When they register, create extension 1234
-;username=xlite1
 ;callerid="Jane Smith" <5678>
-;host=dynamic
+;host=dynamic			; This device needs to register
 ;nat=yes			; X-Lite is behind a NAT router
 ;canreinvite=no			; Typically set to NO if behind NAT
 ;disallow=all
@@ -247,11 +275,10 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ;dtmfmode=inband		; Choices are inband, rfc2833, or info
 ;defaultip=192.168.0.59		; IP used until peer registers
 ;username=snom			; Username to use in INVITE until peer registers
-;mailbox=1234,2345		; Mailboxes for message waiting indicator
+;mailbox=1234@context,2345      ; Mailbox(-es) for message waiting indicator
 ;restrictcid=yes		; To have the callerid restriced -> sent as ANI
 ;disallow=all
 ;allow=ulaw			; dtmfmode=inband only works with ulaw or alaw!
-;mailbox=1234@context,2345      ; Mailbox(-es) for message waiting indicator
 
 
 ;[polycom]
@@ -271,7 +298,7 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ;username=pingtel
 ;secret=blah
 ;host=dynamic
-;insecure=yes			; To match a peer based by IP address only and not peer
+;insecure=yes			; To match a peer based by IP address only and not peer name
 ;insecure=very			; To allow registered hosts to call without re-authenticating
 ;qualify=1000			; Consider it down if it's 1 second to reply
 				; Helps with NAT session
@@ -286,7 +313,7 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ;secret=blah
 ;qualify=200			; Qualify peer is no more than 200ms away
 ;nat=yes			; This phone may be natted
-				; Send SIP and RTP to  IP address that packet is 
+				; 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
 ;canreinvite=no			; Asterisk by default tries to redirect the
@@ -294,18 +321,6 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 				; the caller to the callee.  Some devices do not
 				; support this (especially if one of them is 
 				; behind a NAT).
-;defaultip=192.168.0.4
+;defaultip=192.168.0.4		; IP address to use until registration
+;username=goran			; Username to use when calling this device before registration
 
-;[cisco2]
-;type=friend
-;username=cisco2
-;fromuser=markster		; Specify user to put in "from" instead of callerid
-;fromdomain=yourdomain.com	; Specify domain to put in "from" instead of callerid
-				; fromuser and fromdomain are used when Asterisk
-				; places calls to this account.  It is not used for
-				; calls from this account.
-;secret=blah
-;host=dynamic
-;defaultip=192.168.0.4
-;amaflags=default		; Choices are default, omit, billing, documentation
-;accountcode=markster		; Users may be associated with an accountcode to ease billing