#egrep -v "^\;|^\ *$" /etc/asterisk/sip.conf
[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
; Default is enabled
; defaults to "asterisk". If you set a system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
; bindport is the local UDP port that Asterisk will listen on
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
; If configured, Asterisk will only allow
; INVITE and REFER to non-local domains
; Use "sip show domains" to list local domains
; international character conversions in URIs
; and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
; and subscriptions (seconds)
; Defaults to 100 ms
; fully. Enable this option to not get error messages
; when sending MWI to phones with this bug.
; Message-Account in the MWI notify message
; defaults to "asterisk"
; This may also be set for individual users/peers
; use 'never' to never use in-band signalling, even in cases
; where some buggy devices might not render it
; Valid values: yes, no, never Default: never
; Note that promiscredir when redirects are made to the
; local system will cause loops since Asterisk is incapable
; of performing a "hairpin" call.
; a valid phone number
; Other options:
; info : SIP INFO messages
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise
; in the this section to get any video support at all.
; 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.
; Videosupport and maxcallbitrate is settable
; for peers and users as well
; performs events (e.g. hold)
; 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
; 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
; setup you will not need to enable this.
; on the audio channel
; when we're not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
; on the audio channel
; when we're on hold (must be > rtptimeout)
; (default is off - zero)
; the moment the channel loads this configuration
; (see sip history / sip no history)
; SIP history is output to the DEBUG logging channel
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
; Turning on notifyringing and notifyhold will add a lot
; more database transactions if you are using realtime.
; status notification when you are using type=friend
; Inbound calls, that really apply to the user part
; of a friend will now be added to and compared with
; the peer limit instead of applying two call limits,
; one for the peer and one for the user.
; "sip show inuse" will only show active calls on
; the peer side of a "type=friend" object if this
; setting is turned on.
; 0 = continue forever, hammering the other server
; until it accepts the registration
; Default is 0 tries, continue forever
; 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
; external host, and Asterisk will
; perform DNS queries periodically. Not
; recommended for production
; environments! Use externip instead
; used
; You may add multiple local networks. A reasonable
; set of defaults are:
; yes = Always ignore info and assume NAT
; no = Use NAT mode only according to RFC3581 (;rport)
; never = Never attempt NAT mode or RFC3581 support
; route = Assume NAT, don't send rport
; (work around more UNIDEN bugs)
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind 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.
; 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.
; Will not work for video and cases where the callee sends
; 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
; 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,nonat'. It implies 'yes'.
; just like friends added from the config file only on a
; as-needed basis? (yes|no)
; Default= no
; If set to yes, when a SIP UA registers successfully, the ip address,
; the origination port, the registration period, and the username of
; the UA will be set to database via realtime.
; If not present, defaults to 'yes'.
; as if it had just registered? (yes|no|<seconds>)
; If set to yes, when the registration expires, the friend will
; vanish from the configuration until requested again. If set
; to an integer, friends expire within this number of seconds
; instead of the registration interval.
;
; For non-realtime peers, when their registration expires, the
; information will _not_ be removed from memory or the Asterisk database
; if you attempt to place a call to the peer, the existing information
; will be used in spite of it having expired
;
; For realtime peers, when the peer is retrieved from realtime storage,
; the registration information will be used regardless of whether
; it has expired or not; if it expires while the realtime peer
; is still in memory (due to caching or other reasons), the
; information will not be removed from realtime storage
; Add domain and configure incoming context
; for external calls to this domain
; You can have several "domain" settings
; Default is yes
; name and local IP to domain list.
; non-peers, use your primary domain "identity"
; for From: headers instead of just your IP
; address. This is to be polite and
; it may be a mandatory requirement for some
; destinations which do not have a prior
; account relationship with your server.
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; 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".
; 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.
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
[authentication]
; Call-limits will not be enforced on real-time peers,
; since they are not stored in-memory
; Also used as "defaultport" in combination with "defaultip" settings
; on incoming calls to Asterisk
; No registration allowed
; from the phone to asterisk
; 1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
; memory
; 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
; to one inbound or outbound call per phone. Use
; the group counters in the dial plan for that.
;
; listed with allow= does NOT matter!
; See doc/callingpres.txt for more information
[xlite1]
type=friend
regexten=2222
callerid="cherep" <1234>
host=dynamic ; This device needs to register
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
; subscribes for mailbox notification
; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to "asterisk"
; Normally you do NOT need to set this parameter
; matching port number
; Helps with NAT session
; qualify=yes uses default value
; Send SIP and RTP to the IP address that packet is
; received from instead of trusting SIP headers
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
; behind a NAT).
; Normally you do NOT need to set this parameter
; You must have this turned on or DTMF reception will work improperly.
; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
; external IP address of the remote device. If port forwarding is done at the client side
; then UDPTL will flow to the remote device.