;
;videosupport=yes ; in the this section to get any video support at all. on
; You can turn it off on a per peer basis if the general
; video support is enabled, but you can't enable it for
; one peer only without enabling in the general section.
;maxcallbitrate=384 ; Videosupport and maxcallbitrate is settable4 kb/s)
; for peers and users as well
;callevents=no ; performs events (e.g. hold)n sip ua
;alwaysauthreject = yes ; for any reason, always reject with '401 Unauthorized',
; instead of letting the requester know whether there was
; a matching user or peer for their request
; order instead of RFC3551 packing order (this is required
;g726nonstandard = yes ; for Sipura and Grandstream ATAs, among others). This is
; contrary to the RFC3551 specification, the peer _should_
; be negotiating AAL2-G726-32 instead :-(
; your localnet setting. Unless you have some sort of strange network
;matchexterniplocally = yes ; setup you will not need to enable this.ost setting if it matches
;
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given peer who registers or unregisters with
; us and have a "regexten=" configuration item.
; Multiple contexts may be specified by separating them with '&'. The
; actual extension is the 'regexten' parameter of the registering peer or its
; (provider).
;
; host is either a host name defined in DNS or the name of a section defined
; below.
;
; Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_p; 0 = continue forever, hammering the other server
; until it accepts the registration
; ; Default is 0 tries, continue forever
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
;----------------------------------------- NAT SUPPORT ------------------------
; The externip, externhost and localnet settings are used if you use Asterisk
; messages if we're behind a NAT
; behind a NAT device to communicate with services on the outside.
; The externip and localnet is used
; Tip 1: Avoid assigning host; when registering and communicating with other proxies
; that we're registered with
;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
;externhost=foo.dyndns.net ; external host, and Asterisk will
; (instead of type=fri; perform DNS queries periodically. Nots
; recommended for production
; environments! Use externip instead
;externrefresh=1020 ; usedoften to refresh externhost ifseconds (default)
; You may add multiple local networks. A reasonable
;registerattempts=10 ; set of defaults are:on attempts before we give up
;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
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; SIP channel. Defaults to "no". An enabled jitterbuffer willa
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.
; channel. Defaults to "no".
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; channel. Two implementations are currently available - "fixed"
; jbimpl = fixed ; (with size always equals to jbmaxsize) and "adaptive" (witha SIP
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
[authentication]
; Global credentials for outbound calls, i.e. when a proxy challenges your
; Asterisk server for authentication. These credentials override
; any credentials in peer/register definition if realm is matched.
;
; This way, Asterisk can authenticate for outbound calls to other
; realms. We match realm on the proxy challenge and pick an set of
; credentials from this list
; Syntax:
; auth = <user>:<secret>@<realm>
; auth = <user>#<md5secret>@<realm>
; Example:
;auth=mark:topsecret@digium.com
;
; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm
;------------------------------------------------------------------------------
; Users and peers have different settings available. Friends have all settings,
; since a friend is both a peer and a user
;
; User config options: Peer configuration:
; -------------------- -------------------
;port=80 ; The port number we want to connect to on the remote side
; Also used as "defaultport" in combination with "defaultip" settings
;------------------------------------------------------------------------------
; Definitions of locally connected SIP devices
;
; type = user a device that authenticates to us by "from" field to place calls
; type = peer a device we place calls to or that calls us and we match by host
; type = friend two configurations (peer+user) in one
;
; For device names, we recommend using only a-z, numerics (0-9) and underscore
;
; For local phones, type=friend ; on incoming calls to Asterisk
;host=192.168.0.23 ; No registration allowedivate IP address
;nat=nou have one-way audio, you; there is not NAT between phone and Asterisk
;canreinvite=yes on a public IP,; allow RTP voice traffic to bypass Asterisk
; from the phone to asterisk
;dtmfmode=infod to configure nat; 1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
;call-limit=1on qualify=yes to k; memory only 1 outgoing call and 1 incoming call at a time
; This will affect your subscriptions as well.
; There is no combined call counter for a "friend"
; so there's currently no way in sip.conf to limit
;[grandstream1] ; to one inbound or outbound call per phone. Use
; the group counters in the dial plan for that.
;type=friend ;
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"one calls
;disallow=alln Doe <1234> ; listed with allow= does NOT matter!use allow=
;allow=alaw ; Note: In user sections the order of codecs
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
; See doc/callingpres.txt for more information
;allow=g729 ; Pass-thru only unless g729 license obtained
;callingpres=allowed_passed_screen ; Set caller ID presentation
;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;secret=blahpoly
;host=fwd.pulver.com
;[sip_proxy-out]
;type=peer ; we only want to call out, not be called
; You can have several "domain" settings
;allowexternaldomains=no ; Default is yes and REFER to non-local domains
; 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
; yes = Always ignore info and assume NAT
; devices hidden behind a NAT de; no = Use NAT mode only according to RFC3581 (;rport)
; never = Never attempt NAT mode or RFC3581 support
; audio problems, you usually ha; route = Assume NAT, don't send rportn or your
; (work around more UNIDEN bugs)
; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
; ports for incoming audio in rtp.conf
;----------------------------------- MEDIA HANDLING --------------------------------
;
; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
;nat=no ; Global NAT settings (Affects all peers and users)
; no reason for Asterisk to stay in the media path, the media will be redirected.
; RTP media stream (audio) to go directly from
; This does not really work with; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
; clients on the inside of a NAT; The default setting is YES. If you have all clients
; behind a NAT, or for some other reason wants Asterisk to
; ; stay in the audio path, you may want to turn this off.
;canreinvite=yes ; In Asterisk 1.4 this setting also affect direct RTP
; at call setup (a new feature in 1.4 - setting up the
; call directly between the endpoints instead of sending
; a re-INVITE).
; the call directly with media peer-2-peer without re-invites.
;directrtpsetup=yes ; Will not work for video and cases where the callee sendsup
; RTP payloads and fmtp headers in the 200 OK that does not match the
; callers INVITE. This will also fail if canreinvite is enabled when
; the device is actually behind NAT.
; (reinvite) but only when the peer where the media is being
;canreinvite=nonat ; sent is known to not be behind a NAT (as the RTP core can
; determine it based on the apparent IP address the media
; arrives from).
; instead of INVITE. This can be combined with 'nonat', as
;canreinvite=update ; 'canreinvite=update,nonat'. It implies 'yes'.th redirection,
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
;
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
;disallow=all ; listed with allow= does NOT matter!use allow=
;allow=alaw ; Note: In user sections the order of codecs
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
; See doc/callingpres.txt for more information
;allow=g729 ; Pass-thru only unless g729 license obtained
;callingpres=allowed_passed_screen ; Set caller ID presentation
[1111]
; 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=1111 ; When they register, create extension 1234
context=demo1
callerid="vovan"
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=ulaw
allow=alaw
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
[aster4]
; 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
t
context=demo1
c
host=192.168.7.4
h
qualify=1
q
;nat=yes ; X-Lite is behind a NAT router
;
canreinvite=no ; Typically set to NO if behind NAT
c
disallow=all
d
allow=gsm ; GSM consumes far less bandwidth than ulaw
a
allow=ulaw
allow=alaw
; subscribes for mailbox notification
;vmexten=voicemail ; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to "asterisk"
;type=friend ; Friends place calls and receive calls
;disallow=all
;context=from-sip ; Context for incoming calls from this user
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;secret=blah
;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
;language=de ; Use German prompts for this user
;host=dynamic ; This peer register with us
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59 ; IP used until peer registers
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
;subscribemwi=yes ; Only send notifications if this phone