[Asterisk-cvs] asterisk/configs sip.conf.sample,1.52,1.53
markster at lists.digium.com
markster at lists.digium.com
Sun Jan 9 12:01:20 CST 2005
Update of /usr/cvsroot/asterisk/configs
In directory mongoose.digium.com:/tmp/cvs-serv25903/configs
Modified Files:
sip.conf.sample
Log Message:
Fix small sip conf issues (bug #3296)
Index: sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.52
retrieving revision 1.53
diff -u -d -r1.52 -r1.53
--- sip.conf.sample 5 Jan 2005 15:15:12 -0000 1.52
+++ sip.conf.sample 9 Jan 2005 18:05:41 -0000 1.53
@@ -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.
@@ -22,8 +22,8 @@
[general]
context=default ; Default context for incoming calls
-;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
- ; if asterisk was compiled with OSP support.
+;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
+ ; if asterisk was compiled with OSP support.
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;realm=mydomain.tld ; Realm for digest authentication
@@ -42,8 +42,8 @@
;pedantic=yes ; Enable slow, pedantic checking for Pingtel
; and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
-;tos=184 ; Set IP QoS to either a keyword or numeric val
-;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
+;tos=184 ; Set IP QoS to either a keyword or numeric val
+;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600 ; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
@@ -52,7 +52,7 @@
;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 @@
; 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.
@@ -121,30 +115,47 @@
; unless you configure a [sip_proxy] section below, and configure a context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
; Tip 2: Use separate type=peer and type=user sections for SIP providers
-; (instead of type=friend) if you have calls in both directions
+; (instead of type=friend) if you have calls in both directions
;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
@@ -191,29 +202,47 @@
;[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"
-;context=from-sip
-;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD
-;callerid=John Doe <1234>
+;type=friend
+;context=from-sip ; Where to start in the dialplan when this phone calls
+;callerid=John Doe <1234> ; Full caller ID, to override the phones config
;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
;incominglimit=1 ; permit only 1 outgoing call at a time
; from the phone to asterisk
-;mailbox=1234 at default ; mailbox 1234 in voicemail context "default"
+;mailbox=1234 at default ; mailbox 1234 in voicemail context "default"
;disallow=all ; need to disallow=all before we can use allow=
;allow=ulaw ; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
@@ -223,17 +252,16 @@
;[xlite1]
-;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
-;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
+; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
+; 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
+;regexten=1234 ; When they register, create extension 1234
;callerid="Jane Smith" <5678>
-;host=dynamic
-;nat=yes ; X-Lite is behind a NAT router
-;canreinvite=no ; Typically set to NO if behind NAT
+;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
-;allow=gsm ; GSM consumes far less bandwidth than ulaw
+;allow=gsm ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
@@ -247,11 +275,10 @@
;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 at 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 at context,2345 ; Mailbox(-es) for message waiting indicator
+;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;[polycom]
@@ -271,7 +298,7 @@
;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 @@
;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 @@
; 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
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