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+; 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