; SIP Configuration for Asterisk

[general]
context=default                 ; Default context for incoming calls
allowguest=yes                   ; Allow or reject guest calls (default is yes)
				; If your Asterisk is connected to the Internet
				; and you have allowguest=yes
				; you want to check which services you offer everyone
				; out there, by enabling them in the default context (see below).
;match_auth_username=yes        ; if available, match user entry using the
                                ; 'username' field from the authentication line
                                ; instead of the From: field.
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
;allowoverlap=yes               ; Enable RFC3578 overlap dialing support.
                                ; Can use the Incomplete application to collect the
                                ; needed digits from an ambiguous dialplan match.
;allowoverlap=dtmf              ; Enable overlap dialing support using DTMF delivery
                                ; methods (inband, RFC2833, SIP INFO) in the early
                                ; media phase.  Uses the Incomplete application to
                                ; collect the needed digits.
;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
                                ; Default is enabled. The Dial() options 't' and 'T' are not
                                ; related as to whether SIP transfers are allowed or not.
;realm=mydomain.tld             ; Realm for digest authentication
                                ; 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
;domainsasrealm=no              ; Use domains list as realms
                                ; You can serve multiple Realms specifying several
                                ; 'domain=...' directives (see below). 
                                ; In this case Realm will be based on request 'From'/'To' header
                                ; and should match one of domain names.
                                ; Otherwise default 'realm=...' will be used.


udpbindaddr=0.0.0.0             ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
                                ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)


                                tcpenable=no                    ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
                                ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)

;tlsenable=no                   ; Enable server for incoming TLS (secure) connections (default is no)
;tlsbindaddr=0.0.0.0            ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
                                ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
                                ; Remember that the IP address must match the common name (hostname) in the
                                ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
                                ; For details how to construct a certificate for SIP see 
                                ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs

;tcpauthtimeout = 30            ; tcpauthtimeout specifies the maximum number
				; of seconds a client has to authenticate.  If
				; the client does not authenticate beofre this
				; timeout expires, the client will be
                                ; disconnected. (default: 30 seconds)

;tcpauthlimit = 100             ; tcpauthlimit specifies the maximum number of
				; unauthenticated sessions that will be allowed
                                ; to connect at any given time. (default: 100)

transport=udp                   ; Set the default transports.  The order determines the primary default transport.
                                ; If tcpenable=no and the transport set is tcp, we will fallback to UDP.

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
                                ; Specifying a port in a SIP peer definition or
                                ; when dialing outbound calls will supress SRV
                                ; lookups for that peer or call.

;pedantic=yes                   ; Enable checking of tags in headers,
                                ; international character conversions in URIs
                                ; and multiline formatted headers for strict
                                ; SIP compatibility (defaults to "yes")

; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
;tos_sip=cs3                    ; Sets TOS for SIP packets.
;tos_audio=ef                   ; Sets TOS for RTP audio packets.
;tos_video=af41                 ; Sets TOS for RTP video packets.
;tos_text=af41                  ; Sets TOS for RTP text packets.

;cos_sip=3                      ; Sets 802.1p priority for SIP packets.
;cos_audio=5                    ; Sets 802.1p priority for RTP audio packets.
;cos_video=4                    ; Sets 802.1p priority for RTP video packets.
;cos_text=3                     ; Sets 802.1p priority for RTP text packets.

;maxexpiry=3600                 ; Maximum allowed time of incoming registrations
                                ; and subscriptions (seconds)
;minexpiry=60                   ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120              ; Default length of incoming/outgoing registration
;mwiexpiry=3600                 ; Expiry time for outgoing MWI subscriptions
;maxforwards=70			; Setting for the SIP Max-Forwards: header (loop prevention)
				; Default value is 70
;qualifyfreq=60                 ; Qualification: How often to check for the host to be up in seconds
				; and reported in milliseconds with sip show settings.
                                ; Set to low value if you use low timeout for NAT of UDP sessions
				; Default: 60
;qualifygap=100			; Number of milliseconds between each group of peers being qualified
				; Default: 100
;qualifypeers=1			; Number of peers in a group to be qualified at the same time
				; Default: 1
;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
;buggymwi=no                    ; Cisco SIP firmware doesn't support the MWI RFC
                                ; fully. Enable this option to not get error messages
                                ; when sending MWI to phones with this bug.
;mwi_from=asterisk              ; When sending MWI NOTIFY requests, use this setting in
                                ; the From: header as the "name" portion. Also fill the
			        ; "user" portion of the URI in the From: header with this
			        ; value if no fromuser is set
			        ; Default: empty
;vmexten=voicemail              ; dialplan extension to reach mailbox sets the
                                ; Message-Account in the MWI notify message
                                ; defaults to "asterisk"

; Codec negotiation
;
; When Asterisk is receiving a call, the codec will initially be set to the
; first codec in the allowed codecs defined for the user receiving the call
; that the caller also indicates that it supports. But, after the caller
; starts sending RTP, Asterisk will switch to using whatever codec the caller
; is sending.
;
; When Asterisk is placing a call, the codec used will be the first codec in
; the allowed codecs that the callee indicates that it supports. Asterisk will
; *not* switch to whatever codec the callee is sending.
;
preferred_codec_only=yes       ; Respond to a SIP invite with the single most preferred codec
                                ; rather than advertising all joint codec capabilities. This
                                ; limits the other side's codec choice to exactly what we prefer.

;disallow=all                   ; First disallow all codecs
allow=alaw
allow=ulaw                     ; Allow codecs in order of preference
allow=ilbc                     ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
				; for framing options
;
; This option specifies a preference for which music on hold class this channel
; should listen to when put on hold if the music class has not been set on the
; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
; channel putting this one on hold did not suggest a music class.
;
; This option may be specified globally, or on a per-user or per-peer basis.
;
;mohinterpret=default
;
; This option specifies which music on hold class to suggest to the peer channel
; when this channel places the peer on hold. It may be specified globally or on
; a per-user or per-peer basis.
;
;mohsuggest=default
;
;parkinglot=plaza               ; Sets the default parking lot for call parking
                                ; This may also be set for individual users/peers
                                ; Parkinglots are configured in features.conf
;language=en                    ; Default language setting for all users/peers
                                ; This may also be set for individual users/peers
;relaxdtmf=yes                  ; Relax dtmf handling
;trustrpid = no                 ; If Remote-Party-ID should be trusted
;sendrpid = yes                 ; If Remote-Party-ID should be sent (defaults to no)
;sendrpid = rpid                ; Use the "Remote-Party-ID" header
                                ; to send the identity of the remote party
                                ; This is identical to sendrpid=yes
;sendrpid = pai                 ; Use the "P-Asserted-Identity" header
                                ; to send the identity of the remote party
;rpid_update = no               ; In certain cases, the only method by which a connected line
                                ; change may be immediately transmitted is with a SIP UPDATE request.
                                ; If communicating with another Asterisk server, and you wish to be able
                                ; transmit such UPDATE messages to it, then you must enable this option.
                                ; Otherwise, we will have to wait until we can send a reinvite to
                                ; transmit the information.
;prematuremedia=no              ; Some ISDN links send empty media frames before 
                                ; the call is in ringing or progress state. The SIP 
                                ; channel will then send 183 indicating early media
                                ; which will be empty - thus users get no ring signal.
                                ; Setting this to "yes" will stop any media before we have
                                ; call progress (meaning the SIP channel will not send 183 Session
                                ; Progress for early media). Default is "yes". Also make sure that
                                ; the SIP peer is configured with progressinband=never. 
                                ;
                                ; In order for "noanswer" applications to work, you need to run
                                ; the progress() application in the priority before the app.

;progressinband=never           ; If we should generate in-band ringing always
                                ; 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
useragent=N39 Door PBX          ; Allows you to change the user agent string
                                ; The default user agent string also contains the Asterisk
                                ; version. If you don't want to expose this, change the
                                ; useragent string.
;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 Asterisk is incapable
                                ; of performing a "hairpin" call.
;usereqphone = no               ; If yes, ";user=phone" is added to uri that contains
                                ; a valid phone number
;dtmfmode = rfc2833             ; Set default dtmfmode for sending DTMF. Default: rfc2833
                                ; Other options:
                                ; info : SIP INFO messages (application/dtmf-relay)
                                ; shortinfo : SIP INFO messages (application/dtmf)
                                ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
                                ; auto : Use rfc2833 if offered, inband otherwise

;compactheaders = yes           ; send compact sip headers.
;
;videosupport=yes               ; Turn on support for SIP video. You need to turn this
                                ; on in 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.
                                ; If you set videosupport to "always", then RTP ports will
                                ; always be set up for video, even on clients that don't
                                ; support it.  This assists callfile-derived calls and
                                ; certain transferred calls to use always use video when
                                ; available. [yes|NO|always]

;maxcallbitrate=384             ; Maximum bitrate for video calls (default 384 kb/s)
                                ; Videosupport and maxcallbitrate is settable
                                ; for peers and users as well
;callevents=no                  ; generate manager events when sip ua
                                ; performs events (e.g. hold)
;authfailureevents=no           ; generate manager "peerstatus" events when peer can't
                                ; authenticate with Asterisk. Peerstatus will be "rejected".
alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to be rejected,
                                ; for any reason, always reject with an identical response
                                ; equivalent to valid username and invalid password/hash
                                ; instead of letting the requester know whether there was
                                ; a matching user or peer for their request.  This reduces
                                ; the ability of an attacker to scan for valid SIP usernames.
                                ; This option is set to "yes" by default.

;auth_options_requests = yes    ; Enabling this option will authenticate OPTIONS requests just like
                                ; INVITE requests are.  By default this option is disabled.

;g726nonstandard = yes          ; If the peer negotiates G726-32 audio, use AAL2 packing
                                ; 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 :-(
;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from registering
                                ; as any IP address used for staticly defined
                                ; hosts.  This helps avoid the configuration
                                ; error of allowing your users to register at
                                ; the same address as a SIP provider.

;contactdeny=0.0.0.0/0.0.0.0           ; Use contactpermit and contactdeny to
;contactpermit=172.16.0.0/255.255.0.0  ; restrict at what IPs your users may
                                       ; register their phones.
contactpermit=172.23.48.0/255.255.252.0


;engine=asterisk                ; RTP engine to use when communicating with the device

;
; 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
; name if 'regexten' is not provided.  If more than one context is provided,
; the context must be specified within regexten by appending the desired
; context after '@'.  More than one regexten may be supplied if they are
; separated by '&'.  Patterns may be used in regexten.
;
;regcontext=sipregistrations
;regextenonqualify=yes          ; Default "no"
                                ; If you have qualify on and the peer becomes unreachable
                                ; this setting will enforce inactivation of the regexten
                                ; extension for the peer
;legacy_useroption_parsing=yes	; Default "no"      ; If you have this option enabled and there are semicolons
                                                    ; in the user field of a sip URI, the field be truncated
                                                    ; at the first semicolon seen. This effectively makes
                                                    ; semicolon a non-usable character for peer names, extensions,
                                                    ; and maybe other, less tested things.  This can be useful
                                                    ; for improving compatability with devices that like to use
                                                    ; user options for whatever reason.  The behavior is similar to
                                                    ; how SIP URI's were typically handled in 1.6.2, hence the name.

; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
; in square brackets.  For example, the caller id value 555.5555 becomes 5555555
; when this option is enabled.  Disabling this option results in no modification
; of the caller id value, which is necessary when the caller id represents something
; that must be preserved.  This option can only be used in the [general] section.
; By default this option is on.
;
;shrinkcallerid=yes     ; on by default


;use_q850_reason = no ; Default "no"
                      ; Set to yes add Reason header and use Reason header if it is available.

;--------------------------- SIP timers ----------------------------------------------------
; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms or the measured run-trip time between
; Asterisk and the device if you have qualify=yes for the device.
;
;t1min=100                      ; Minimum roundtrip time for messages to monitored hosts
                                ; Defaults to 100 ms
;timert1=500                    ; Default T1 timer
                                ; Defaults to 500 ms or the measured round-trip
                                ; time to a peer (qualify=yes).
;timerb=32000                   ; Call setup timer. If a provisional response is not received
                                ; in this amount of time, the call will autocongest
                                ; Defaults to 64*timert1

;--------------------------- RTP timers ----------------------------------------------------
; These timers are currently used for both audio and video streams. The RTP timeouts
; are only applied to the audio channel.
; The settings are settable in the global section as well as per device
;
;rtptimeout=60                  ; Terminate call if 60 seconds of no RTP or RTCP activity
                                ; 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.
;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP or RTCP activity
                                ; on the audio channel
                                ; when we're on hold (must be > rtptimeout)
;rtpkeepalive=<secs>            ; Send keepalives in the RTP stream to keep NAT open
                                ; (default is off - zero)

;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
; This mechanism can detect and reclaim SIP channels that do not terminate through normal
; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
; The operation of Session-Timers is driven by the following configuration parameters:
;
; * session-timers    - Session-Timers feature operates in the following three modes:
;                            originate : Request and run session-timers always
;                            accept    : Run session-timers only when requested by other UA
;                            refuse    : Do not run session timers in any case
;                       The default mode of operation is 'accept'.
; * session-expires   - Maximum session refresh interval in seconds. Defaults to 1800 secs.
; * session-minse     - Minimum session refresh interval in seconds. Defualts to 90 secs.
; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
;
;session-timers=originate
;session-expires=600
;session-minse=90
;session-refresher=uas
;
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes                 ; Turn on SIP debugging by default, from
                                ; the moment the channel loads this configuration
;recordhistory=yes              ; Record SIP history by default
                                ; (see sip history / sip no history)
;dumphistory=yes                ; Dump SIP history at end of SIP dialogue
                                ; SIP history is output to the DEBUG logging channel


;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
; You can subscribe to the status of extensions with a "hint" priority
; (See extensions.conf.sample for examples)
; chan_sip support two major formats for notifications: dialog-info and SIMPLE
;
; You will get more detailed reports (busy etc) if you have a call counter enabled
; for a device.
;
; If you set the busylevel, we will indicate busy when we have a number of calls that
; matches the busylevel treshold.
;
; For queues, you will need this level of detail in status reporting, regardless
; if you use SIP subscriptions. Queues and manager use the same internal interface
; for reading status information.
;
; Note: Subscriptions does not work if you have a realtime dialplan and use the
; realtime switch.
;
;allowsubscribe=no              ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default     ; Set a specific context for SUBSCRIBE requests
                                ; Useful to limit subscriptions to local extensions
                                ; Settable per peer/user also
;notifyringing = no             ; Control whether subscriptions already INUSE get sent
                                ; RINGING when another call is sent (default: yes)
;notifyhold = yes               ; Notify subscriptions on HOLD state (default: no)
                                ; Turning on notifyringing and notifyhold will add a lot
                                ; more database transactions if you are using realtime.
;notifycid = yes                ; Control whether caller ID information is sent along with
                                ; dialog-info+xml notifications (supported by snom phones).
                                ; Note that this feature will only work properly when the
                                ; incoming call is using the same extension and context that
                                ; is being used as the hint for the called extension.  This means
                                ; that it won't work when using subscribecontext for your sip
                                ; user or peer (if subscribecontext is different than context).
                                ; This is also limited to a single caller, meaning that if an
                                ; extension is ringing because multiple calls are incoming,
                                ; only one will be used as the source of caller ID.  Specify
                                ; 'ignore-context' to ignore the called context when looking
                                ; for the caller's channel.  The default value is 'no.' Setting
                                ; notifycid to 'ignore-context' also causes call-pickups attempted
                                ; via SNOM's NOTIFY mechanism to set the context for the call pickup
                                ; to PICKUPMARK.
;callcounter = yes              ; Enable call counters on devices. This can be set per
                                ; device too.

;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)

register => {{ gatekeeper_sip_registration }}/s
;
;     This will pass incoming calls to the 's' extension

;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
; no reason for Asterisk to stay in the media path, the media will be redirected.
; This does not really work well in the case where Asterisk is outside and the
; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
;
;directmedia=yes                ; Asterisk by default tries to redirect the
                                ; RTP media stream 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 want Asterisk to
                                ; stay in the audio path, you may want to turn this off.

                                ; 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).

                                ; Additionally this option does not disable all reINVITE operations.
                                ; It only controls Asterisk generating reINVITEs for the specific
                                ; purpose of setting up a direct media path. If a reINVITE is
                                ; needed to switch a media stream to inactive (when placed on
                                ; hold) or to T.38, it will still be done, regardless of this 
                                ; setting. Note that direct T.38 is not supported.

;directmedia=nonat              ; An additional option is to allow media path redirection
                                ; (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).

;directmedia=update             ; Yet a third option... use UPDATE for media path redirection,
                                ; instead of INVITE. This can be combined with 'nonat', as
                                ; 'directmedia=update,nonat'. It implies 'yes'.

;directrtpsetup=yes             ; Enable the new experimental direct RTP setup. This sets up
                                ; 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 directmedia is enabled when
                                ; the device is actually behind NAT.

;directmediadeny=0.0.0.0/0      ; Use directmediapermit and directmediadeny to restrict 
;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
                                ; (There is no default setting, this is just an example)
                                ; Use this if some of your phones are on IP addresses that
                                ; can not reach each other directly. This way you can force 
                                ; RTP to always flow through asterisk in such cases.

;ignoresdpversion=yes           ; By default, Asterisk will honor the session version
                                ; number in SDP packets and will only modify the SDP
                                ; session if the version number changes. This option will
                                ; force asterisk to ignore the SDP session version number
                                ; and treat all SDP data as new data.  This is required
                                ; for devices that send us non standard SDP packets
                                ; (observed with Microsoft OCS). By default this option is
                                ; off.

;sdpsession=Asterisk PBX        ; Allows you to change the SDP session name string, (s=)
                                ; Like the useragent parameter, the default user agent string
                                ; also contains the Asterisk version.
;sdpowner=root                  ; Allows you to change the username field in the SDP owner string, (o=)
                                ; This field MUST NOT contain spaces
;encryption=no                  ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
                                ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
                                ; the peer does not support SRTP. Defaults to no.

;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
;
;rtcachefriends=yes             ; Cache realtime friends by adding them to the internal list
                                ; just like friends added from the config file only on a
                                ; as-needed basis? (yes|no)

;rtsavesysname=yes              ; Save systemname in realtime database at registration
                                ; Default= no

;rtupdate=yes                   ; Send registry updates to database using realtime? (yes|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'. Note: realtime peers will
                                ; probably not function across reloads in the way that you expect, if
                                ; you turn this option off.
;rtautoclear=yes                ; Auto-Expire friends created on the fly on the same schedule
                                ; 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.

;ignoreregexpire=yes            ; Enabling this setting has two functions:
                                ;
                                ; 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

;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
; domains, each of which can direct the call to a specific context if desired.
; By default, all domains are accepted and sent to the default context or the
; context associated with the user/peer placing the call.
; REGISTER to non-local domains will be automatically denied if a domain
; list is configured.
;
; Domains can be specified using:
; domain=<domain>[,<context>]
; Examples:
; domain=myasterisk.dom
; domain=customer.com,customer-context
;
; In addition, all the 'default' domains associated with a server should be
; added if incoming request filtering is desired.
; autodomain=yes
;
; To disallow requests for domains not serviced by this server:
; allowexternaldomains=no

;domain=mydomain.tld,mydomain-incoming
                                ; Add domain and configure incoming context
                                ; for external calls to this domain
;domain=1.2.3.4                 ; Add IP address as local domain
                                ; You can have several "domain" settings
;allowexternaldomains=no        ; Disable INVITE and REFER to non-local domains
                                ; Default is yes
;autodomain=yes                 ; Turn this on to have Asterisk add local host
                                ; name and local IP to domain list.

; fromdomain=mydomain.tld       ; When making outbound SIP INVITEs to
                                ; 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.

;------------------------------ Advice of Charge CONFIGURATION --------------------------
; snom_aoc_enabled = yes;     ; This options turns on and off support for sending AOC-D and
                              ; AOC-E to snom endpoints.  This option can be used both in the
                              ; peer and global scope.  The default for this option is off.


;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
                              ; 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.

; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
                              ; channel. Defaults to "no".

; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.

; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
                              ; 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.

; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
                              ; 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.

; jbtargetextra = 40          ; This option only affects the jb when 'jbimpl = adaptive' is set.
                              ; The option represents the number of milliseconds by which the new jitter buffer
                              ; will pad its size. the default is 40, so without modification, the new
                              ; jitter buffer will set its size to the jitter value plus 40 milliseconds.
                              ; increasing this value may help if your network normally has low jitter,
                              ; but occasionally has spikes.

; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".

;----------------------------- SIP_CAUSE reporting ---------------------------------
; storesipcause = no          ; This option causes chan_sip to set the
			      ; HASH(SIP_CAUSE,<channel name>) channel variable
			      ; to the value of the last sip response.
			      ; WARNING: enabling this option carries a
			      ; significant performance burden. It should only
			      ; be used in low call volume situations. This
                              ; option 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


[basic-options](!)                ; a template
        dtmfmode=rfc2833
        context=from-office
        type=friend

[natted-phone](!,basic-options)   ; another template inheriting basic-options
        directmedia=no
        host=dynamic

[public-phone](!,basic-options)   ; another template inheriting basic-options
        directmedia=yes

[my-codecs](!)                    ; a template for my preferred codecs
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw

[ulaw-phone](!)                   ; and another one for ulaw-only
        disallow=all
        allow=ulaw