Asterisk-sip.conf
Overview
Asterisk sip.conf 문서 설명.
Basic
sip.conf 는 크게 4가지 섹션으로 이루어져 있다.
[general] context=unauthenticated ; default context for incoming calls allowguest=no ; disable unauthenticated calls srvlookup=no ; disable DNS SRV record lookup on outbound calls ; (unless you have a reliable DNS connection, in which case yes) udpbindaddr=0.0.0.0 ; listen for UDP requests on all interfaces tcpenable=no ; disable TCP support [office-phone](!) ; create a template for our devices type=friend ; the channel driver will match on username first, ; IP second context=LocalSets ; this is where calls from the device will enter ; the dialplan host=dynamic ; the device will register with asterisk nat=force_rport,comedia ; assume device is behind NAT ; *** NAT stands for Network Address Translation, ; which allows multiple internal devices to share an ; external IP address. dtmfmode=auto ; accept touch-tones from the devices, negotiated ; automatically disallow=all ; reset which voice codecs this device will accept or offer allow=g722 ; audio codecs to accept from, and request to, the device allow=ulaw ; in the order we prefer allow=alaw ; define a device name and use the office-phone template [0000FFFF0001](office-phone) secret=4VQ96sg6ROc ; a unique password for this device -- ; DON'T USE THE PASSWORD WE'VE USED IN THIS EXAMPLE! ; define another device name using the same template [0000FFFF0002](office-phone) secret=sKAw7GCTtcA ; a unique password for this device -- ; DON'T USE THE PASSWORD WE'VE USED IN THIS EXAMPLE!
[general]
[general] 섹션은 가장 기본이 되는 섹션으로 일반적인 설정 옵션과 프로토콜의 동작 방식 등을 설정하게 된다. 그리고 기본값으로 지정될 값들을 설정하기도 한다.
예를 들어, 위의 설정 예를 보면, 기본 context가 unauthenticated로 되어 있는 것을 볼 수 있다. 이는 콜이 인입되면 기본적으로 실행되는 다이얼 플랜을 지칭한다. 다음으로 allowgeust 옵션은 인증되지 않은 콜들을 허용하지 않겠다는 옵션 설정이다.
srvlookup 옵션은 Asterisk 동작시, DNS를 참조하여 동작하겠다는 뜻이며, 만약 DNS 가 설정되어 있지 않다면 반드시 no로 설정해야 한다.
udpbinaddr 옵션은 Listen 할, UDP IP 주소를 설정하는 옵션이다. 0.0.0.0 으로 설정시, 사용 가능한 모든 인터페이스를 Listen 하게 된다.
tcpenable 옵션은 TCP 를 통한 Request 에도 응답할 수 있도록 해준다.
[office-phone](!)
섹션 이름 [office-phone] 뒤에 붙는 (!)의 의미는 해당 섹션을 템플릿으로 사용하겠다는 의미이다. 템플릿을 사용하게 되면 불필요하게 반복되는 설정을 없앨 수 있으며, 오타와 같은 위험에서도 벗어날 수 있게 된다. 한번 템플릿에서 설정된 내용은 다시금 템플릿을 참조하는 곳에서 같은 옵션이지만 다른 값을 설정할 수도 있다. 이런 경우, 대부분의 경우에는 덮어쓰기가 되어버려, 가장 나중에 설정된 내용이 입력되게 된다. 단, type, allow, disallow 와 같은 몇가지 항목들은 예외이다.
type
type 항목은 Asterisk 로, 콜이 인입되었을 때, 어떤 부분을 기준으로 콜을 매칭시키는지를 결정한다.
- peer : Source IP 주소와 Port number 를 기준으로 매칭한다.
- user : From header 의 Username 을 기준으로 매칭한다. sip.conf 파일에서 [] 안에 같은 이름이 있다면 해당 항목을 매칭시킨다.
- friend : peer 와 user 모두를 이용하여 매칭한다.
context
현재 설정된 device 에서 요청이 전송되면, context 항목에 설정된 dialplan 이 작동하게 된다.
host
host 항목은 설정된 device 의 IP 주소를 지정한다. 만약 고정된 IP 주소가 아닌, 동적으로 변하는 IP 주소라면, "dynamic" 으로 설정하면 된다. 만약 특정 IP 주소를 입력한다면, 오직 설정된 IP 주소에서만 해당 device 가 작동하게 될 것이다. host 에 고정 아이피 주소를 입력함으로써 얻는 이점은, 한번 고정 IP 로 입력된 device 에 대해서는 asterisk 로 더이상의 registraion 을 하지 않아도 된다는 점이다.
nat
엄밀히 말하면, nat 설정은 NAT(Network address translation) 네트워크 뒤쪽에 있는 device 에만 필요한 설정이다. 왜냐하면 SIP 프로토콜에는 메시지 속에 IP 주소를 포함하고 있기 때문이다. 만약 device 가 사설망에서 동작하게 된다면, SIP 메시지에는 사설 네트워크 주소를 포함하게 될 것이다.
dtmfmode
dtmfmode 옵션은 해당 device 에서 사용하는 DTMF(touch-tone) 방식을 지정하는 옵션이다. 4가지 방식의 DTMF 옵션이 가능하다(info, inband, rfc2833, auto). info 는 SIP 프로토콜의 INFO 메소드를 사용하는 방식이고, inband 는 inband audio tone, rfc2833 는 out-of-band 방식(RFC2833)을 위한 옵션이다. auto 는 Asterisk 에서 자동으로 DTMF 방식을 설정하게 하는 옵션이다(auto 로 설정할 경우, 가능하다면 rfc2833 방식을 사용하게 된다).
disallow/allow
마지막 두가지 옵션(disallow/allow) 은 어느 audio codec 을 허용하지/말지를 설정하는 옵션이다. disallow=all 설정을 제일 먼저 설정함으로써, 이전에 설정되어있는 codec 옵션 내용([general] 섹션이나 다른 default 설정 에서 설정되어 있는)을 초기화하고, 어떤 codec 들을 허용할지를 설정할 수 있다. 설정 예제에서는 ulaw 와 alaw 를 설정했는데, 대부분은 ulaw 를 사용한다(만약, 미국이나 캐나다가 아닌 곳에서 사용한다면 대부분 alaw 를 사용하게 된다).
General section
SIP Configuration - general The [general] section of sip.conf includes the following variables:
General
context=public ; Default context for incoming calls. Defaults to 'default'
;allowguest=no ; 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.
;recordonfeature=automixmon ; Default feature to use when receiving 'Record: on' header
; from an INFO message. Defaults to 'automon'. Works with
; dynamic features. Feature must be usable on requesting
; channel for it to work. Setting this value to a blank
; will disable it.
;recordofffeature=automixmon ; Default feature to use when receiving 'Record: off' header
; from an INFO message. Defaults to 'automon'. Works with
; dynamic features. Feature must be usable on requesting
; channel for it to work. Setting this value to a blank
; will disable it.
; With the current situation, you can do one of four things:
; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1
; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
; c) Listen on the IPv4 wildcard. Example: bindaddr=0.0.0.0
; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=::
; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
;
; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
; for TLS).
; IPv4 example: bindaddr=0.0.0.0:5062
; IPv6 example: bindaddr=[::]:5062
;
; The address family of the bound UDP address is used to determine how Asterisk performs
; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
; however, that Asterisk ignores all records except the first one. In case d), when both A
; and AAAA records are available, either an A or AAAA record will be first, and which one
; depends on the operating system. On systems using glibc, AAAA records are given
; priority.
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)
; When a dialog is started with another SIP endpoint, the other endpoint
; should include an Allow header telling us what SIP methods the endpoint
; implements. However, some endpoints either do not include an Allow header
; or lie about what methods they implement. In the former case, Asterisk
; makes the assumption that the endpoint supports all known SIP methods.
; If you know that your SIP endpoint does not provide support for a specific
; method, then you may provide a comma-separated list of methods that your
; endpoint does not implement in the disallowed_methods option. Note that
; if your endpoint is truthful with its Allow header, then there is no need
; to set this option. This option may be set in the general section or may
; be set per endpoint. If this option is set both in the general section and
; in a peer section, then the peer setting completely overrides the general
; setting (i.e. the result is *not* the union of the two options).
;
; Note also that while Asterisk currently will parse an Allow header to learn
; what methods an endpoint supports, the only actual use for this currently
; is for determining if Asterisk may send connected line UPDATE requests and
; MESSAGE requests. Its use may be expanded in the future.
;
; disallowed_methods = UPDATE
;
; Note that the TCP and TLS support for chan_sip is currently considered
; experimental. Since it is new, all of the related configuration options are
; subject to change in any release. If they are changed, the changes will
; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
;
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)
;websocket_write_timeout = 100 ; Default write timeout to set on websocket transports.
; This value may need to be adjusted for connections where
; Asterisk must write a substantial amount of data and the
; receiving clients are slow to process the received information.
; Value is in milliseconds; default is 100 ms.
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 (seconds)
;minexpiry=60 ; Minimum length of registrations (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;submaxexpiry=3600 ; Maximum allowed time of incoming subscriptions (seconds), default: maxexpiry
;subminexpiry=60 ; Minimum length of subscriptions, default: minexpiry
;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
;keepalive=60 ; Interval at which keepalive packets should be sent to a peer
; Valid options are yes (60 seconds), no, or the number of seconds.
; Default: 0
;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=ulaw ; Allow codecs in order of preference
;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
; for framing options
;autoframing=yes ; Set packetization based on the remote endpoint's (ptime)
; preferences. Defaults to no.
;
; 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
;tonezone=se ; Default tonezone 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.
;trust_id_outbound = no ; Controls whether or not we trust this peer with private identity
; information (when the remote party has callingpres=prohib or equivalent).
; no - RPID/PAI headers will not be included for private peer information
; yes - RPID/PAI headers will include the private peer information. Privacy
; requirements will be indicated in a Privacy header for sendrpid=pai
; legacy - RPID/PAI will be included for private peer information. In the
; case of sendrpid=pai, private data that would be included in them
; will be anonymized. For sendrpid=rpid, private data may be included
; but the remote party's domain will be anonymized. The way legacy
; behaves may violate RFC-3325, but it follows historic behavior.
; This option is set to 'legacy' by default
;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=no ; 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: no
;useragent=Asterisk 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]
;textsupport=no ; Support for ITU-T T.140 realtime text.
; The default value is "no".
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
; Videosupport and maxcallbitrate is settable
; for peers and users as well
;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.
;accept_outofcall_message = no ; Disable this option to reject all MESSAGE requests outside of a
; call. By default, this option is enabled. When enabled, MESSAGE
; requests are passed in to the dialplan.
;outofcall_message_context = messages ; Context all out of dialog msgs are sent to. When this
; option is not set, the context used during peer matching
; is used. This option can be defined at both the peer and
; global level.
;auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests.
; By default this option is enabled. However, it can be disabled
; should an application desire to not load the Asterisk server with
; doing authentication and implement end to end security in the
; message body.
;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 :-(
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
;outboundproxy=192.0.2.1 ; IPv4 address literal (default port is 5060)
;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060)
;outboundproxy=192.168.0.2.1:5062 ; IPv4 address literal with explicit port
;outboundproxy=[2001:db8::1]:5062 ; IPv6 address literal with explicit port
; ; (could also be tcp,udp) - defining transports on the proxy line only
; ; applies for the global proxy, otherwise use the transport= option
;supportpath=yes ; This activates parsing and handling of Path header as defined in RFC 3327. This enables
; Asterisk to route outgoing out-of-dialog requests via a set of proxies by using a pre-loaded
; route-set defined by the Path headers in the REGISTER request.
; NOTE: There are multiple things to consider with this setting:
; * As this influences routing of SIP requests make sure to not trust Path headers provided
; by the user's SIP client (the proxy in front of Asterisk should remove existing user
; provided Path headers).
; * When a peer has both a path and outboundproxy set, the path will be added to Route: header
; but routing to next hop is done using the outboundproxy.
; * If set globally, not only will all peers use the Path header, but outbound REGISTER
; requests from Asterisk will add path to the Supported header.
;rtsavepath=yes ; If using dynamic realtime, store the path headers
;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches
; your localnet setting. Unless you have some sort of strange network
; setup you will not need to enable this.
;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.
;contactacl=named_acl_example ; Use named ACLs defined in acl.conf
;rtp_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.
;send_diversion=no ; Default "yes" ; Asterisk normally sends Diversion headers with certain SIP
; invites to relay data about forwarded calls. If this option
; is disabled, Asterisk won't send Diversion headers unless
; they are added manually.
; 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.
; When the Transfer() application sends a REFER SIP message, extra headers specified in
; the dialplan by way of SIPAddHeader are sent out with that message. 1.8 and earlier did not
; add the extra headers. To revert to 1.8- behavior, call SIPRemoveHeader with no arguments
; before calling Transfer() to remove all additional headers from the channel. The setting
; below is for transitional compatibility only.
;
;refer_addheaders=yes ; on by default
;autocreatepeer=no ; Allow any UAC not explicitly defined to register
; WITHOUT AUTHENTICATION. Enabling this options poses a high
; potential security risk and should be avoided unless the
; server is behind a trusted firewall.
; If set to "yes", then peers created in this fashion
; are purged during SIP reloads.
; When set to "persist", the peers created in this fashion
; are not purged during SIP reloads.
;
- allowsubscribe = yes|no
- Allow or Ignore Subscribe requests
- allow = <codec>
- Allow codecs in order of preference (Use DISALLOW=ALL first, before allowing other codecs)
- disallow = all
- Disallow all codecs (global configuration)
- Asterisk sip allowexternaldomains = yes|no
- Enable/Disable INVITE and REFER to non-local domains. Default yes. (New in v1.2.x).
- allowguest = yes|no
- Allow or reject guest calls. Default is yes. (this can also be set to 'osp' if asterisk was compiled with OSP support). (New in v1.2.x).
- Asterisk sip allowoverlap = yes|no
- Enable/disable overlap dialing support.
- Default yes (Overlap dial provides for a longer time-out period between digits, also called the inter-digit timer.
- With overlap dial set to off, the gateway expects to receive the digits one right after the other coming in to this line with very little delay between digits.
- With overlap dial set to on, then the device waits up to about 2 seconds between digits).
- autocreatepeer = yes|no
- If set, anyone will be able to log in as a peer (with no check of credentials; useful for operation with SER).
- Default no.
- Asterisk sip autodomain = yes|no
- Enable/disable Asterisk's ability to add local hostnames and local IP address to the domain list.
- externip or externhost are also taken into the domain list. Default no. (New in v1.2.x).
- bindaddr = IP_Address
- IP Address to bind to (listen on).
- Default 0.0.0.0 (all network interfaces).
- bindport = Number
- UDP Port to bind to (listen on).
- Used to be port in Asterisk v1.0.x. Default 5060.
- callerid = <string>
- Caller ID information used when nothing else is available.
- Defaults to asterisk. (The ability to override the default appears to available in Asterisk 1.0.9. Unsure about other versions.)
- canreinvite = update|yes|no|nonat (global setting)
- For some reason this defaults to yes, so beware...
- Asterisk sip checkmwi = Number
- Global interval (in seconds) between mailbox checks.
- Default 10 seconds. (New in v1.2.x)
- Asterisk sip compactheaders = yes|no
- Indicates Asterisk should send compact (i.e. abbreviated) headers in the SIP messages.
- Default no. (New in v1.2.x)
- context = <contextname>
- This is the default context and is used when a endpoint has no context property.
- The context in section of an endpoint is used to route calls from that endpoint to the wanted destination.
- The context body is located in extensions.conf.
- defaultexpiry= Number
- Default duration (in seconds) of incoming/outgoing registration. Default 120 seconds.
- domain = domains
- Comma separated list of domains which Asterisk is responsible for. (New in Asterisk 1.2.x)
- dtmfmode = inband|info|rfc2833 (global setting)
- Default rfc2833. Warning: inband very high CPU load.
- dumphistory = yes|no
- Enable support for dumping of SIP conversation's transaction history to LOG_DEBUG. Default no. (New in v1.2.x)
- externip = IP_Address or a hostname
- Address that we're going to put in SIP messages if we're behind a NAT.
- If a hostname is used as the value, then the IP address associated with the hostname is looked up only once during the reading of the sip.conf.
- If you want support for a hostname associated with a dynamic IP address, use Asterisk sip externhost.
- externhost = hostname.tld
- (New in Asterisk 1.2.x)
- externrefresh = Number
- Specify how often (in seconds) a hostname DNS lookup should be performed for the value entered in 'externhost'. Default 10 seconds. (New in Asterisk 1.2.x).
- ignoreregexpire = yes|no
- Indicates whether to use Contact information about a peer even if the information is stale because it has reached its expiration time.
- Default no. (New in v1.2.x)
- jbenable = yes|no
- Enables the use of a jitterbuffer on the receiving side of a SIP channel. (Added in Version 1.4)
- jbforce = yes|no
- Forces the use of a jitterbuffer on the receive side of a SIP channel. Defaults to "no". (Added in Version 1.4)
- jbmaxsize = Number
- Max length of the jitterbuffer in milliseconds. (Added in Version 1.4)
- jbresyncthreshold = Number
- 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. (Added in Version 1.4)
- jbimpl = fixed|adaptive
- 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. (Added in Version 1.4)
- jblog = no|yes
- Enables jitterbuffer frame logging. Defaults to "no". (Added in Version 1.4)
- language = <string>
- Default language used by any Playback()/Background().
- limitonpeers = yes|no
- If set to yes use only the peer call counter for both incoming and outgoing calls (ref. hints, subscriptions, BLF; added in 1.4)
- localnet = NetAddress/Netmask
- local network and mask.
- fromdomain= <domain>
- Sets default From: domain in SIP messages when acting as a SIP ua (client)
- insecure = very|yes|no|invite|port
- Specifies how to handle connections with peers.
- Default no (authenticate all connections). invite and port added in v1.2.x, yes and very removed in v1.6.x, possible to use multiple options separated by commas from v1.4.x
- maxexpiry = Number
- Max duration (in seconds) of incoming registration we allow. Default 3600 seconds.
- musicclass = value
- one of the classes specified in musiconhold.conf
- musdiconhold = value
- same as musicclass
- nat = yes|no
- Please note that as of Asterisk 1.0.x nat can now have the values: yes|no|never|route. Default no which really means to use rfc3581 techniques.
- notifymimetype = mediatype/subtype
- Allow overriding of mime type in MWI NOTIFY used in voicemail online messages. Valid MIME types can be found here. Default application/simple-message-summary. (New in v1.2.x).
- notifyringing = yes|no
- Notify subscription on RINGING state. Default yes. (New in v1.2.x).
- outboundproxy = IP_address or DNS SRV name (excluding the _sip._udp prefix)
- SRV name, hostname, or IP address of the outbound SIP Proxy. (New in v1.2.x).
- outboundproxyport = Number
- UDP port number for the Outbound SIP Proxy. (New in v1.2.x).
- pedantic = yes|no
- Enable slow, pedantic checking of Call-ID:s, multiline SIP headers and URI-encoded headers. Default no (in Asterisk 1.8 default yes).
- port = <portno>
- Default SIP port of peer. (this is not the port for Asterisk to listen. See bindport).
- progressinband = never|no|yes
- If we should generate in-band ringing always. Default never.
- promiscredir= yes|no
- Allows support for 302 Redirects; (Note: will redirect them all to the local extension returned in Contact rather than to that extension at the destination). Default no.
- qualify = yes|no|milliseconds
- Check if client is reachable. If yes, the checks occur every 60 seconds. Default no.
- realm = my realm
- (Change authentication realm from asterisk (default) to your own. Requires Asterisk v1.x)
- recordhistory = yes|no
- Enable logging of SIP conversation's transaction history. Default no. (New in v1.2.x).
- regcontext = context
- Default context to use for SIP REGISTER replies from the SIP Registrar.
- register => <username>:<password>:[authid]@<sip client/peer id in sip.conf>/<contact>
- SIP provider를 등록한다.
- 자세한 내용은 Outbound sip registration 항목을 참조하자.
- registerattempts = Number
- Number of SIP REGISTER messages to send to a SIP Registrar before giving up. Default 0 (no limit). (New in v1.2.x).
- registertimeout = Number
- Number of seconds to wait for a response from a SIP Registrar before classifying the SIP REGISTER has timed out. Default 20 seconds. (New in v1.2.x).
- relaxdtmf = yes|no
- Default no.
- rtautoclear = yes|no|number
- Auto-Expire friends created on the fly. If yes the autoexpire will be in 120 seconds. Default yes. (New in v1.2.x). Buggy up to 1.4.19, see bug 12707
- rtcachefriends = yes|no
- Cache realtime friends by adding them to the internal list just like friends added from the config file. Default no. (New in v1.2.x). Buggy up to 1.4.19, see bug 12707
- rtsavesysname = yes|no
- If set will write the value of asterisk.conf (options) systemname to the sip peer table in the field "regserver". Useful for multi-server systems. (New in v1.?)
- rtpholdtimeout = Number
- Max number of seconds of inactivity before terminating a call on hold. Default 0 (no limit). (New in v1.2.x).
- rtpkeepalive = Number
- Number of seconds, when a RTP Keepalive packet will be sent if no other RTP traffic on that connection. Default 0 (no RTP Keepalive). (New in v1.2.x).
- rtptimeout = Number
- Number of seconds, to wait for RTP traffic before classify the connection as discontinued. Default 0 (no RTP timeout). (New in v1.2.x).
- rtupdate = yes|no
- Send registry updates to the database when using Realtime support. Default yes. (New in v1.2.x).
- sendrpid = yes|no
- If a Remote-Party-ID SIP header should be sent. Default no.
- sipdebug = yes|no
- Default setting for whether SIP debug is enabled upon loading of the sip.conf. Default no. (New in v1.2.x).
- srvlookup = yes|no
- Enable DNS SRV lookups on calls. Default yes. (Default is no prior to v1.4.14)
- tos = <value>
- Set IP QoS parameters for outgoing media streams (numeric values are also accepted, like tos=184 )
- trustrpid = yes|no
- If Remote-Party-ID SIP header should be trusted. Default no.
- useclientcode = yes|no
- If yes, then the Call Originator as stated in the CDR will be changed to whatever is specified in a X-ClientCode SIP Header. Default no. (New in v1.2.x)
- usereqphone = yes|no
- Indicates whether to add a ";user=phone" to the URI. Default no. (New in v1.2.x)
- useragent = <string>
- Allow the SIP header "User-Agent" to be customized. Default asterisk.
- videosupport = yes|no
- Turn on support for SIP video (peer specific setting added in SVN Dec 21 2005, bug 5427. Default no.
- vmexten = <string>
- Dialplan extension to reach mailbox. Default asterisk. (New in v1.2.x)
- callevents = yes|no
- Set to yes to receive events on AMI when a call is put on/off hold.
- disallowed_methods=
- (1.8.x) When a dialog is started with another SIP endpoint, the other endpoint should include an Allow header telling us what SIP methods the endpoint implements.
- However, some endpoints either do not include and Allow header or lie about what methods they implement.
- In the former case, Asterisk makes the assumption that the endpoint supports all known SIP methods.
- If you know that your SIP endpoint does not provide support for a specific method, then you may provide a comma-separated list of methods that your endpoint does not implement in the disallowed_methods option.
- Note that if your endpoint is truthful with its Allow header, then there is no need to set this option. This option may be set in the general section or may be set per endpoint.
- If this option is set both in the general section and in a peer section, then the peer setting completely overrides the general setting (i.e. the result is *not* the union of the two options).
- Note also that while Asterisk currently will parse an Allow header to learn what methods an endpoint supports, the only actual use for this currently is for determining if Asterisk may send connected line UPDATE requests.
- Its use may be expanded in the future.
- preferred_codec_only= (1.8.x)
- 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.
- engine= (1.8.x)
- RTP engine to use when communicating with the device
TLS settings
;------------------------ TLS settings ------------------------------------------------------------
;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections
; The certificates must be sorted starting with the subject's certificate
; and followed by intermediate CA certificates if applicable.
; Default is to look for "asterisk.pem" in current directory
;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
; If no tlsprivatekey is specified, tlscertfile is searched for
; for both public and private key.
;tlscafile=</path/to/certificate>
; If the server your connecting to uses a self signed certificate
; you should have their certificate installed here so the code can
; verify the authenticity of their certificate.
;tlscapath=</path/to/ca/dir>
; A directory full of CA certificates. The files must be named with
; the CA subject name hash value.
; (see man SSL_CTX_load_verify_locations for more info)
;tlsdontverifyserver=[yes|no]
; If set to yes, don't verify the servers certificate when acting as
; a client. If you don't have the server's CA certificate you can
; set this and it will connect without requiring tlscafile to be set.
; Default is no.
;tlscipher=<SSL cipher string>
; A string specifying which SSL ciphers to use or not use
; A list of valid SSL cipher strings can be found at:
; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
;
;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
; Specify protocol for outbound client connections.
; If left unspecified, the default is sslv2.
SIP timers
;--------------------------- 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
;--------------------------- 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 (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'. ; uac - Default to the caller initially refreshing when possible ; uas - Default to the callee initially refreshing when possible ; ; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other ; endpoint's preference for who will handle refreshes. Asterisk will never override the ; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint ; fighting over who sends the refreshes. This holds true for the initiation of session ; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or ; whether Asterisk is currently the refresher or not. ; ;session-timers=originate ;session-expires=600 ;session-minse=90 ;session-refresher=uac
SIP DEBUGGING
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration.
; NOTE: You cannot use the CLI to turn it off. You'll
; need to edit this and reload the config.
;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)
;--------------------------- 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.
- callcounter : SIP Channel 채널에서 Call counter 를 활성화 할지 여부를 설정한다. 만약 Queue 를 사용한다면 반드시 yes로 설정해주어야 한다.
T.38 FAX SUPPORT
;----------------------------------------- T.38 FAX SUPPORT ---------------------------------- ; ; This setting is available in the [general] section as well as in device configurations. ; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off. ; ; t38pt_udptl = yes ; Enables T.38 with FEC error correction. ; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction. ; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction. ; t38pt_udptl = yes,none ; Enables T.38 with no error correction. ; ; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that ; is based on an incorrect interpretation of the T.38 recommendation, and results in failures ; because Asterisk does not believe it can send T.38 packets of a reasonable size to that ; endpoint (Cisco media gateways are one example of this situation). In these cases, during a ; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL ; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you ; can set an override (globally, or on a per-device basis) to make Asterisk ignore the ; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead. ; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option, ; like this: ; ; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides ; ; the other endpoint's provided value to assume we can ; ; send 400 byte T.38 FAX packets to it. ; ; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists) ; based one or more events being detected. The events that can be detected are an incoming ; CNG tone or an incoming T.38 re-INVITE request. ; ; faxdetect = yes ; Default 'no', 'yes' enables both CNG and T.38 detection ; faxdetect = cng ; Enables only CNG detection ; faxdetect = t38 ; Enables only T.38 detection
OUTBOUND SIP REGISTRATIONS
;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
;
;
;
; domain is either
; - domain in DNS
; - host name in DNS
; - the name of a peer defined below or in realtime
; The domain is where you register your username, so your SIP uri you are registering to
; is username@domain
;
; If no extension is given, the 's' extension is used. The extension needs to
; be defined in extensions.conf to be able to accept calls from this SIP proxy
; (provider).
;
; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
; this is equivalent to having the following line in the general section:
;
; register => username:secret@host/callbackextension
;
; and more readable because you don't have to write the parameters in two places
; (note that the "port" is ignored - this is a bug that should be fixed).
;
; Note that a register= line doesn't mean that we will match the incoming call in any
; other way than described above. If you want to control where the call enters your
; dialplan, which context, you want to define a peer with the hostname of the provider's
; server. If the provider has multiple servers to place calls to your system, you need
; a peer for each server.
;
; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
; contain a port number. Since the logical separator between a host and port number is a
; ':' character, and this character is already used to separate between the optional "secret"
; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
; they are blank. See the third example below for an illustration.
;
;
; Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
; connect to local extension 1234 in extensions.conf, default context,
; unless you configure a [sip_proxy] section below, and configure a
; context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
; Tip 2: Use separate inbound and outbound sections for SIP providers
; (instead of type=friend) if you have calls in both directions
;
;register => 3456@mydomain:5082::@mysipprovider.com
;
; Note that in this example, the optional authuser and secret portions have
; been left blank because we have specified a port in the user section
;
;register => tls://username:xxxxxx@sip-tls-proxy.example.org
;
; The 'transport' part defaults to 'udp' but may also be 'tcp' or 'tls'.
; Using 'udp://' explicitly is also useful in case the username part
; contains a '/' ('user/name').
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
; 0 = continue forever, hammering the other server
; until it accepts the registration
; Default is 0 tries, continue forever
;register_retry_403=yes ; Treat 403 responses to registrations as if they were
; 401 responses and continue retrying according to normal
; retry rules.
OUTBOUND MWI SUBSCRIPTIONS
;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS ------------------------- ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval ; by other phones. At this time, you can only subscribe using UDP as the transport. ; Format for the mwi register statement is: ; mwi => user[:secret[:authuser]]@host[:port]/mailbox ; ; Examples: ;mwi => 1234:password@mysipprovider.com/1234 ;mwi => 1234:password@myportprovider.com:6969/1234 ;mwi => 1234:password:authuser@myauthprovider.com/1234 ;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234 ; ; MWI received will be stored in the 1234 mailbox of the SIP_Remote context. ; It can be used by other phones by following the below: ; mailbox=1234@SIP_Remote
NAT SUPPORT
;----------------------------------------- NAT SUPPORT ------------------------ ; ; WARNING: SIP operation behind a NAT is tricky and you really need ; to read and understand well the following section. ; ; When Asterisk is behind a NAT device, the "local" address (and port) that ; a socket is bound to has different values when seen from the inside or ; from the outside of the NATted network. Unfortunately this address must ; be communicated to the outside (e.g. in SIP and SDP messages), and in ; order to determine the correct value Asterisk needs to know: ; ; + whether it is talking to someone "inside" or "outside" of the NATted network. ; This is configured by assigning the "localnet" parameter with a list ; of network addresses that are considered "inside" of the NATted network. ; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY. ; Multiple entries are allowed, e.g. a reasonable set is the following: ; ; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses ; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 ; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation ; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network ; ; + the "externally visible" address and port number to be used when talking ; to a host outside the NAT. This information is derived by one of the ; following (mutually exclusive) config file parameters: ; ; a. "externaddr = hostname[:port]" specifies a static address[:port] to ; be used in SIP and SDP messages. ; The hostname is looked up only once, when [re]loading sip.conf . ; If a port number is not present, use the port specified in the "udpbindaddr" ; (which is not guaranteed to work correctly, because a NAT box might remap the ; port number as well as the address). ; This approach can be useful if you have a NAT device where you can ; configure the mapping statically. Examples: ; ; externaddr = 12.34.56.78 ; use this address. ; externaddr = 12.34.56.78:9900 ; use this address and port. ; externaddr = mynat.my.org:12600 ; Public address of my nat box. ; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT. ; ; externtcpport will default to the externaddr or externhost port if either one is set. ; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT. ; ; externtlsport port will default to the RFC designated port of 5061. ; ; b. "externhost = hostname[:port]" is similar to "externaddr" except ; that the hostname is looked up every "externrefresh" seconds ; (default 10s). This can be useful when your NAT device lets you choose ; the port mapping, but the IP address is dynamic. ; Beware, you might suffer from service disruption when the name server ; resolution fails. Examples: ; ; externhost=foo.dyndns.net ; refreshed periodically ; externrefresh=180 ; change the refresh interval ; ; Note that at the moment all these mechanism work only for the SIP socket. ; The IP address discovered with externaddr/externhost is reused for ; media sessions as well, but the port numbers are not remapped so you ; may still experience problems. ; ; NOTE 1: in some cases, NAT boxes will use different port numbers in ; the internal<->external mapping. In these cases, the "externaddr" and ; "externhost" might not help you configure addresses properly. ; ; NOTE 2: when using "externaddr" or "externhost", the address part is ; also used as the external address for media sessions. Thus, the port ; information in the SDP may be wrong! ; ; In addition to the above, Asterisk has an additional "nat" parameter to ; address NAT-related issues in incoming SIP or media sessions. ; In particular, depending on the 'nat= ' settings described below, Asterisk ; may override the address/port information specified in the SIP/SDP messages, ; and use the information (sender address) supplied by the network stack instead. ; However, this is only useful if the external traffic can reach us. ; The following settings are allowed (both globally and in individual sections): ; ; nat = no ; Do no special NAT handling other than RFC3581 ; nat = force_rport ; Pretend there was an rport parameter even if there wasn't ; nat = comedia ; Send media to the port Asterisk received it from regardless ; ; of where the SDP says to send it. ; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default) ; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT ; ; The nat settings can be combined. For example, to set both force_rport and comedia ; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no', ; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings ; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then ; the non-auto option will be ignored. ; ; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send ; SIP responses to it via the source IP and port from which the request originated ; instead of the address/port listed in the top-most Via header. This is useful if a ; client knows that it is behind a NAT and therefore cannot guess from what address/port ; its request will be sent. Asterisk will always honor the 'rport' parameter if it is ; sent. The force_rport setting causes Asterisk to always send responses back to the ; address/port from which it received requests; even if the other side doesn't support ; adding the 'rport' parameter. ; ; 'comedia RTP handling' refers to the technique of sending RTP to the port that the ; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is ; only partially related to RFC 4145 which was referred to as COMEDIA while it was in ; draft form. This method is used to accomodate endpoints that may be located behind ; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to ; for their media streams is not the actual address/port that will be used on the nearer ; side of the NAT. ; ; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from ; the nat setting in a peer definition, then the peer username will be discoverable ; by outside parties as Asterisk will respond to different ports for defined and ; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE ; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the ; other, then valid peers with settings differing from those in the general section will ; be discoverable. ; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by ; RFC 4961; Asterisk will always send RTP packets from the same port number it expects ; to receive them on. ; ; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using ; the media_address configuration option. This is only applicable to the general section and ; can not be set per-user or per-peer. ; ; media_address = 172.16.42.1 ; ; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the ; perceived external network address has changed. When the stun_monitor is installed and ; configured, chan_sip will renew all outbound registrations when the monitor detects any sort ; of network change has occurred. By default this option is enabled, but only takes effect once ; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not ; generate all outbound registrations on a network change, use the option below to disable ; this feature. ; ; subscribe_network_change_event = yes ; on by default ; ; ICE/STUN/TURN usage can be enabled globally or on a per-peer basis using the icesupport ; configuration option. When set to yes ICE support is enabled. When set to no it is disabled. ; It is disabled by default. ; ; icesupport = yes
MEDIA HANDLING
;----------------------------------- 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'.
;directmedia=outgoing ; When sending directmedia reinvites, do not send an immediate
; reinvite on an incoming call leg. This option is useful when
; peered with another SIP user agent that is known to send
; immediate direct media reinvites upon call establishment. Setting
; the option in this situation helps to prevent potential glares.
; Setting this option 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.
;directmediaacl=acl_example ; Use named ACLs defined in acl.conf
;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.
;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80
;
;avpf=yes ; Enable inter-operability with media streams using the AVPF RTP profile.
; This will cause all offers and answers to use AVPF (or SAVPF). This
; option may be specified at the global or peer scope.
;force_avp=yes ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for
; media streams when appropriate, even if a DTLS stream is present.
REALTIME SUPPORT
;----------------------------------------- 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
;----------------------------------------- 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
;------------------------------ 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
;------------------------------ 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".
Peer section
SIP configurations - peers and clients These variables can be configured for each SIP peer definition:
(If not specified, the configuration variable can be used for both type=peer and type=user.)
- accountcode = <string>
- Users may be associated with an accountcode. See Asterisk billing
- allow = <codec>
- Allow codecs in order of preference (Use DISALLOW=ALL first, before allowing other codecs)
- disallow = all
- Disallow all codecs for this peer or user definition.
- allowguest = yes|no
- Allow or reject guest calls (default is yes, this can also be set to 'osp' if asterisk was compiled with OSP support). (New in v1.2.x).
- amaflags
- Categorization for CDR records. Choices are default, omit, billing, documentation. See Asterisk billing
- astdb
- Appears to insert a value in the Asterisk database. See example below.
- auth = <authname>
- Value assigned to the Digest username= SIP header.
- callerid = <string>
- Caller ID information used when nothing else is available. Defaults to asterisk.
- busylevel = number
- Number of simultaneous calls until user/peer is busy
- call-limit = number
- Number of simultaneous calls through this user/peer.
- callgroup = num1,num2-num3
- Defines call groups for calls to this device.
- callingpres = number|descriptive_text
- Set Caller-ID presentation on a call.
- Valid descriptive values are: allowed_not_screened, allowed_passed_screen, allowed_failed_screen, allowed, prohib_not_screened, prohib_passed_screen, prohib_failed_screen, prohib, and unavailable.
- See SetCallerPres for more information.
- Default allowed_not_screened.
- canreinvite = update|yes|no|nonat
- Client 가 SIP re-invite 를 할 수 있는지 없는지를 설정한다.
- 기본값은 yes 이다.
- cid_number = <string>
- On incoming (through this peer) calls sets the outbound $CALLERID(num) to <string>. (New in v.1.4.x)
- context = <context_name>
- If type=user, the Context for the inbound call from this SIP user definition.
- If type=peer, the Context in the dialplan for outbound calls from this SIP peer definition.
- If type=friend the context used for both inbound and outbound calls through the SIP entities definition.
- If no type=user entry matches an inbound call, then a type=peer or type=friend will match if the hostname or IP address defined in host= matches.
- defaultip = Dotted.Quad.IP.Addr
- host=dynamic 지정시 사용하게 될 기본 IP address 를 설정한다.
- Client 가 아직 Register 하지 않았을 경우 사용된다.
- Valid only for type=peer.
- defaultuser
- 사용자 id를 설정한다. username 항목은 사용하지 않는다.
- directrtpsetup = yes|no
- canreinvite 옵션과 비슷하다. 하지만 SIP Proxy 와 비슷하게 별도의 미디어 처리과정 없이 바로 콜을 넘겨준다는 점이 다르다.
- dtmfmode = inband|info|rfc2833
- DTMF 시그널 처리 방식을 설정한다.
- 기본값은 rfc2833 이다.
- inband 방식 지정시, 많은 량의 CPU 파워가 필요하다.
- fromuser = <from_ID>
- Specify user to put in "from" instead of $CALLERID(number) (overrides the callerid) when placing calls _to_ peer (another SIP proxy).
- Valid only for type=peer.
- "from" 에 $CALLERID(number) 대신에 여기에 설정된 ID를 입력한다. type=peer 일때만 유효하다.
- fromdomain = <domain>
- Sets default From: domain in SIP messages when placing calls _to_ peer.
- Valid only when in [general] section or type=peer.
- SIP 헤더에서의 Default from: domain 을 설정한다. type=peer 일때만 유효하다.
- fullcontact = <sip:uri_contact>
- SIP URI contact for realtime peer. Valid only for realtime peers.
- fullname = "Full Name"
- Sets outbound $CALLERID(name). (New in v1.4.x)
- host = dynamic|hostname|IPAddr
- Client 접속 방식을 지정한다.
- 지정된 IP 주소 혹은 도메인 네임, 혹은 동적 IP 등록 가능 여부를 설정할 수 있다.
- incominglimit and outgoinglimit = Number
- Limits for number of simultaneous active calls for a SIP client. Valid only for type=peer.
- insecure = very|yes|no|invite|port
- Specifies how to handle connections with peers. Default no (authenticate all connections).
- Invite and port added in v1.2.x, yes and very removed in v1.6.x, possible to use multiple options separated by commas from v1.4.x
- ipaddr
- Dotted Quad IP address of the peer. Valid only for realtime peers.
- language
- A language code defined in indications.conf - defines language for prompts
- mailbox = mailbox
- Voicemail extension (for message waiting indications). Valid only for type=peer. Edit: also valid for type=friend (verified with 1.4.22.1).
- md5secret
- MD5-Hash of "<user>:==SIP_realm==:<secret>" (can be used instead of secret).
- Default for authenticating to an Asterisk server when SIP realm is not explicitly declared is "<user>:asterisk:<secret>".
- musicclass
- one of the classes specified in musiconhold.conf
- musiconhold
- Set class of musiconhold on calls from SIP phone.
- Calls to the phone require SetMusicOnHold cmd of higher priority (lower numerical value of priority) than Dial cmd in dialplan in order to set this class for the call.
- Calls have the MusicOnHold class set on a per call basis, not on a per phone basis, and making a call through any extension specifying SetMusicOnHold will override this value for the call.
- subscribemwi
- Instructs Asterisk to not send NOTIFY messages for message waiting indication (added in v1.4)
- name = <name>
- Name of the realtime peer. If the peer is an actual phone then this will generally be the extension number of that phone.
- On some softphones this field corresponds to the "username" field/option in the softphone's settings. Valid only for realtime peers.
- nat = yes|no
- This variable changes the behaviour of Asterisk for clients behind a firewall.
- This does not solve the problem if Asterisk is behind the firewall and the client on the outside.
- Please note that Asterisk 1.0.x nat can now have the values: yes|no|never|route; Asterisk 1.8 can have the values: yes|no|force_rport|comedia.
- Default no which really means to use rfc3581 techniques.
- outboundproxy = IP_address or DNS SRV name (excluding the _sip._udp prefix)
- SRV name, hostname, or IP address of the outbound SIP Proxy. Valid only in [general] section and type=peer. (New in v1.2.x).
- permit, deny,mask
- IP address and network restriction
- pickupgroup
- Group that can pickup fellow workers' calls using *8 and the Pickup() application on the *8 extension
- port
- SIP port of the client
- progressinband = never|no|yes
- If we should generate in-band ringing always.
- Default never.
- promiscredir = yes|no
- Allows support for 302 Redirects;
- (Note: will redirect them all to the local extension returned in Contact rather than to that extension at the destination).
- Default no.
- qualify = yes|no|milliseconds
- Client 의 연결 여부를 확인한다.
- Yes 설정시, "sip show peers" 명령에서 Status 부분에서 연결 여부를 확인할 수 있다.
- 연결 확인 주기는 qualifyfreq 항목을 참조한다. 만약 숫자를 설정하게 되면, millisecond 단위로 확인 주기가 설정된다.
- 설정 전
Name/username Host Dyn Forcerport Comedia ACL Port Status Description sip_test_01/sip_test_ 192.168.0.56 D Auto (No) No 55500 Unmonitored 4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 3 offline]
- 설정 후
Name/username Host Dyn Forcerport Comedia ACL Port Status Description sip_test_01/sip_test_ 192.168.0.56 D Auto (No) No 64129 OK (179 ms) 4 sip peers [Monitored: 1 online, 3 offline Unmonitored: 0 online, 0 offline]
- regexten =
- None
- regseconds = seconds
- Number of seconds between SIP REGISTER. Valid only for realtime peer entries.
- restrictcid
- (yes/no) To have the callerid restricted -> sent as ANI; use this to hide the caller ID.
- This does not seem to work. This variable has been deprecated as of v1.2.x.
- rtpkeepalive = seconds
- Number of seconds, when a RTP Keepalive packet will be sent if no other RTP traffic on that connection.
- Default 0 (no RTP Keepalive). Valid only in [general] section and type=peer.
- rtptimeout = seconds
- Terminate call if x seconds of no RTP activity when we're not on hold. Valid only in [general] section and type=peer.
- rtpholdtimeout = seconds
- Terminate call if x seconds of no RTP activity when we're on hold (must be larger than rtptimeout). Valid only in [general] section and type=peer.
- secret
- If Asterisk is acting as a SIP Server, then this SIP client must login with this Password (A shared secret).
- If Asterisk is acting as a SIP client to a remote SIP server that requires SIP INVITE authentication, then this field is used to authenticate SIP INVITEs that Asterisk sends to the remote SIP server.
- Asterisk 1.6.2.x: Changed the secret parameter to remotesecret.
- sendrpid = yes|no
- If a Remote-Party-ID SIP header should be sent. Default no.
- setvar = variable=value
- Channel variable to be set for all calls from this peer/user.
- subscribecontext = <context_name>
- Set a specific context for SIP SUBSCRIBE requests
- trunkname
- Indicates this peer definition is for a SIP trunk. As a result, the $CALLERID(name) will start off blank and requires the dialplan to set the $CALLERID(name). (New in v1.6.x)
- trustrpid = yes|no
- If Remote-Party-ID SIP header should be trusted. Default no.
- type = user|peer|friend
- Relationship to client - outbound provider or full client?
- peer : Match incoming requests to a configuration entry using the source IP address and port number.
- user : Match incoming requests to a configuration entry using the username in the From header of the SIP request. This name is matched to a section in sip.conf with the same name in square brackets.
- friend : This enables matching rules for both peer and user. This is the setting most commonly used for SIP phones.
- useclientcode = yes|no
- If yes, then the Call Originator as stated in the CDR will be changed to whatever is specified in a X-ClientCode SIP Header. Default no. (New in v1.2.x)
- usereqphone = yes|no
- Indicates whether to add a ";user=phone" to the URI. Default no. Valid only in [general] and type=peer.
- username = <username[@realm]>
- If Asterisk is accepting SIP INVITE requests from a remote SIP client, this field specifies the user name for authentication. (Contrast with fromuser.) Also, for peers that
- register with Asterisk, this username is used in INVITEs until we have a registration.
- vmexten = <string>
- Dialplan extension to reach mailbox. Default asterisk. Valid only in [general] or type=peer.
Notes
Asterisk 1.6 and later support SIP over TCP. Before that it only supports SIP over UDP. Asterisk 1.8 comes with IPv6 support. For Grandstream phones: set dtmfmode=info Asterisk uses the incoming RTP Stream as a timing source for sending its outgoing Stream. If the incoming stream is interrupted due to silence suppression then musiconhold will be choppy. So in conclusion, you cannot use silence suppression. Make sure ALL SIP phones have disabled silence suppression. There is a solution for the silence suppression problem, see bug 5374 for details.
Outbound sip registration
Asterisk can register as a SIP user agent to a SIP proxy (provider) Format for the register statement is:
register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
domain is either
- domain in DNS
- host name in DNS
- the name of a peer defined below or in realtime
The domain is where you register your username, so your SIP uri you are registering to is username@domain
If no extension is given, the 's' extension is used. The extension needs to be defined in extensions.conf to be able to accept calls from this SIP proxy (provider).
A similar effect can be achieved by adding a "callbackextension" option in a peer section. this is equivalent to having the following line in the general section:
register => username:secret@host/callbackextension
and more readable because you don't have to write the parameters in two places (note that the "port" is ignored - this is a bug that should be fixed).
Note that a register= line doesn't mean that we will match the incoming call in any other way than described above. If you want to control where the call enters your dialplan, which context, you want to define a peer with the hostname of the provider's server. If the provider has multiple servers to place calls to your system, you need a peer for each server.
Beginning with Asterisk version 1.6.2, the "user" portion of the register line may contain a port number. Since the logical separator between a host and port number is a ':' character, and this character is already used to separate between the optional "secret" and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if they are blank. See the third example below for an illustration.
Examples:
register => 1234:password@mysipprovider.com
This will pass incoming calls to the 's' extension
register => 2345:password@sip_proxy/1234
Register 2345 at sip provider 'sip_proxy'. Calls from this provider connect to local extension 1234 in extensions.conf, default context, unless you configure a [sip_proxy] section below, and configure a context. Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] Tip 2: Use separate inbound and outbound sections for SIP providers
(instead of type=friend) if you have calls in both directions
register => 3456@mydomain:5082::@mysipprovider.com
Note that in this example, the optional authuser and secret portions have been left blank because we have specified a port in the user section
register => tls://username:xxxxxx@sip-tls-proxy.example.org
The 'transport' part defaults to 'udp' but may also be 'tcp', 'tls', 'ws', or 'wss'. Using 'udp://' explicitly is also useful in case the username part contains a '/' ('user/name').
register => 1234:password@example.com/trunk_01
register => 1235:password@example.com/trunk_01
register => 1236:password@example.com/trunk_01
register => 1237:password@example.com/trunk_01
registertimeout=20 ; retry registration calls every 20 seconds (default)
registerattempts=10 ; Number of registration attempts before we give up
; 0 = continue forever, hammering the other server
; until it accepts the registration
; Default is 0 tries, continue forever
register_retry_403=yes ; Treat 403 responses to registrations as if they were
; 401 responses and continue retrying according to normal
; retry rules.
Example
inbound 설정
- inbound example
outbound 설정
자체 트렁크를 가지고 있고, 트렁크를 사용하여 발신을 하고 싶다면 다음의 설정을 해주어야 한다.
- outbound example 1
[mysipprovider-out] type=peer secret=password username=2345 host=sipserver.mysipprovider.com fromuser=2345 fromdomain=fwd.pulver.com canreinvite=no insecure=invite,port qualify=yes nat=yes context=from-mysipprovider ; this section will be defined in extensions.conf
- outbound example 2
[myprovider] type = peer host = your.provider.tld defaultuser = username secret = password ; Most providers won't authenticate when they send calls to you, ; so you need this line to just accept their calls. insecure = invite dtmfmode = rfc2833 disallow = all allow = ulaw
External link
References
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