643 lines
40 KiB
Text
643 lines
40 KiB
Text
|
; 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
|