Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity

Fix small sip conf issues (bug #3296)

Browse Source 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.2-netsec
Mark Spencer 21 years ago
parent adc9e59e0b
commit b705f19938
1 changed files with 66 additions and 51 deletions
Show Stats Download Patch File Download Diff File

95
configs/sip.conf.sample
Unescape View File

@ -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"
;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
@ -223,13 +252,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;[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
;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

Write Preview
Loading…
Cancel
Save