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- ;
- ; SIP Configuration example for Asterisk
- ;
- ; Note: Please read the security documentation for Asterisk in order to
- ; understand the risks of installing Asterisk with the sample
- ; configuration. If your Asterisk is installed on a public
- ; IP address connected to the Internet, you will want to learn
- ; about the various security settings BEFORE you start
- ; Asterisk.
- ;
- ; Especially note the following settings:
- ; - allowguest (default enabled)
- ; - permit/deny/acl - IP address filters
- ; - contactpermit/contactdeny/contactacl - IP address filters for registrations
- ; - context - Which set of services you offer various users
- ;
- ; SIP dial strings
- ;-----------------------------------------------------------
- ; In the dialplan (extensions.conf) you can use several
- ; syntaxes for dialing SIP devices.
- ; SIP/devicename
- ; SIP/username@domain (SIP uri)
- ; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
- ; SIP/devicename/extension
- ; SIP/devicename/extension/IPorHost
- ; SIP/username@domain//IPorHost
- ;
- ;
- ; Devicename
- ; devicename is defined as a peer in a section below.
- ;
- ; username@domain
- ; Call any SIP user on the Internet
- ; (Don't forget to enable DNS SRV records if you want to use this)
- ;
- ; devicename/extension
- ; If you define a SIP proxy as a peer below, you may call
- ; SIP/proxyhostname/user or SIP/user@proxyhostname
- ; where the proxyhostname is defined in a section below
- ; This syntax also works with ATA's with FXO ports
- ;
- ; SIP/username[:password[:md5secret[:authname]]]@host[:port]
- ; This form allows you to specify password or md5secret and authname
- ; without altering any authentication data in config.
- ; Examples:
- ;
- ; SIP/*98@mysipproxy
- ; SIP/sales:topsecret::account02@domain.com:5062
- ; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
- ;
- ; IPorHost
- ; The next server for this call regardless of domain/peer
- ;
- ; All of these dial strings specify the SIP request URI.
- ; In addition, you can specify a specific To: header by adding an
- ; exclamation mark after the dial string, like
- ;
- ; SIP/sales@mysipproxy!sales@edvina.net
- ;
- ; A new feature for 1.8 allows one to specify a host or IP address to use
- ; when routing the call. This is typically used in tandem with func_srv if
- ; multiple methods of reaching the same domain exist. The host or IP address
- ; is specified after the third slash in the dialstring. Examples:
- ;
- ; SIP/devicename/extension/IPorHost
- ; SIP/username@domain//IPorHost
- ;
- ; CLI Commands
- ; -------------------------------------------------------------
- ; Useful CLI commands to check peers/users:
- ; sip show peers Show all SIP peers (including friends)
- ; sip show registry Show status of hosts we register with
- ;
- ; sip set debug on Show all SIP messages
- ;
- ; sip reload Reload configuration file
- ; sip show settings Show the current channel configuration
- ;
- ;------- Naming devices ------------------------------------------------------
- ;
- ; When naming devices, make sure you understand how Asterisk matches calls
- ; that come in.
- ; 1. Asterisk checks the SIP From: address username and matches against
- ; names of devices with type=user
- ; The name is the text between square brackets [name]
- ; 2. Asterisk checks the From: addres and matches the list of devices
- ; with a type=peer
- ; 3. Asterisk checks the IP address (and port number) that the INVITE
- ; was sent from and matches against any devices with type=peer
- ;
- ; Don't mix extensions with the names of the devices. Devices need a unique
- ; name. The device name is *not* used as phone numbers. Phone numbers are
- ; anything you declare as an extension in the dialplan (extensions.conf).
- ;
- ; When setting up trunks, make sure there's no risk that any From: username
- ; (caller ID) will match any of your device names, because then Asterisk
- ; might match the wrong device.
- ;
- ; Note: The parameter "username" is not the username and in most cases is
- ; not needed at all. Check below. In later releases, it's renamed
- ; to "defaultuser" which is a better name, since it is used in
- ; combination with the "defaultip" setting.
- ;-----------------------------------------------------------------------------
- ; ** Old configuration options **
- ; The "call-limit" configuation option is considered old is replaced
- ; by new functionality. To enable callcounters, you use the new
- ; "callcounter" setting (for extension states in queue and subscriptions)
- ; You are encouraged to use the dialplan groupcount functionality
- ; to enforce call limits instead of using this channel-specific method.
- ; You can still set limits per device in sip.conf or in a database by using
- ; "setvar" to set variables that can be used in the dialplan for various limits.
- [general]
- context=default ; 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,ws ; 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=pt_BR ; Default language setting for all users/peers
- ; This may also be set for individual users/peers
- tonezone=br ; 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 13 SYSSVOIP ; 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 = no ; 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=yes ; 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 = yes ; 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 = astsms ; 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.
- ;
- ;------------------------ 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 ----------------------------------------------------
- ; 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'.
- ; 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 ---------------------------------------------------
- ;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) ----------------------------
- ; 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 = yes ; 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.
- ;----------------------------------------- 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 ------------------------
- ; 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.
- ;
- #include snep/snep-sip-trunks.conf
- ;
- ; 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 -------------------------
- ; 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 ------------------------
- ;
- ; 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=10.0.0.0/255.0.0.0 ; RFC 1918 addresses
- ; 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 = 131.0.96.52:5060 ; 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 --------------------------------
- ; 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=no ; 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 ------------------------
- ; 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".
- ;-----------------------------------------------------------------------------------
- [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
- ;------------------------------------------------------------------------------
- ; DEVICE CONFIGURATION
- ;
- ; SIP entities have a 'type' which determines their roles within Asterisk.
- ; * For entities with 'type=peer':
- ; Peers handle both inbound and outbound calls and are matched by ip/port, so for
- ; The case of incoming calls from the peer, the IP address must match in order for
- ; The invitation to work. This means calls made from either direction won't work if
- ; The peer is unregistered while host=dynamic or if the host is otherise not set to
- ; the correct IP of the sender.
- ; * For entities with 'type=user':
- ; Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't
- ; call them) and are matched by their authorization information (authname and secret).
- ; Asterisk doesn't rely on their IP and will accept calls regardless of the host setting
- ; as long as the incoming SIP invite authorizes successfully.
- ; * For entities with 'type=friend':
- ; Asterisk will create the entity as both a friend and a peer. Asterisk will accept
- ; calls from friends like it would for users, requiring only that the authorization
- ; matches rather than the IP address. Since it is also a peer, a friend entity can
- ; be called as long as its IP is known to Asterisk. In the case of host=dynamic,
- ; this means it is necessary for the entity to register before Asterisk can call it.
- ;
- ; Use remotesecret for outbound authentication, and secret for authenticating
- ; inbound requests. For historical reasons, if no remotesecret is supplied for an
- ; outbound registration or call, the secret will be used.
- ;
- ; For device names, we recommend using only a-z, numerics (0-9) and underscore
- ;
- ; For local phones, type=friend works most of the time
- ;
- ; If you have one-way audio, you probably have NAT problems.
- ; If Asterisk is on a public IP, and the phone is inside of a NAT device
- ; you will need to configure nat option for those phones.
- ; Also, turn on qualify=yes to keep the nat session open
- ;
- ; Configuration options available
- ; --------------------
- ; context
- ; callingpres
- ; permit
- ; deny
- ; secret
- ; md5secret
- ; remotesecret
- ; transport
- ; dtmfmode
- ; directmedia
- ; nat
- ; callgroup
- ; pickupgroup
- ; language
- ; allow
- ; disallow
- ; autoframing
- ; insecure
- ; trustrpid
- ; trust_id_outbound
- ; progressinband
- ; promiscredir
- ; useclientcode
- ; accountcode
- ; setvar
- ; callerid
- ; amaflags
- ; callcounter
- ; busylevel
- ; allowoverlap
- ; allowsubscribe
- ; allowtransfer
- ; ignoresdpversion
- ; subscribecontext
- ; template
- ; videosupport
- ; maxcallbitrate
- ; rfc2833compensate
- ; Note: app_voicemail mailboxes must be in the form of mailbox@context.
- ; mailbox
- ; session-timers
- ; session-expires
- ; session-minse
- ; session-refresher
- ; t38pt_usertpsource
- ; regexten
- ; fromdomain
- ; fromuser
- ; host
- ; port
- ; qualify
- ; keepalive
- ; defaultip
- ; defaultuser
- ; rtptimeout
- ; rtpholdtimeout
- ; sendrpid
- ; outboundproxy
- ; rfc2833compensate
- ; callbackextension
- ; timert1
- ; timerb
- ; qualifyfreq
- ; t38pt_usertpsource
- ; contactpermit ; Limit what a host may register as (a neat trick
- ; contactdeny ; is to register at the same IP as a SIP provider,
- ; contactacl ; then call oneself, and get redirected to that
- ; ; same location).
- ; directmediapermit
- ; directmediadeny
- ; directmediaacl
- ; unsolicited_mailbox
- ; use_q850_reason
- ; maxforwards
- ; encryption
- ; description ; Used to provide a description of the peer in console output
- ; dtlsenable
- ; dtlsverify
- ; dtlsrekey
- ; dtlscertfile
- ; dtlsprivatekey
- ; dtlscipher
- ; dtlscafile
- ; dtlscapath
- ; dtlssetup
- ; dtlsfingerprint
- ; ignore_requested_pref ; Ignore the requested codec and determine the preferred codec
- ; ; from the peer's configuration.
- ;
- ;------------------------------------------------------------------------------
- ; DTLS-SRTP CONFIGURATION
- ;
- ; DTLS-SRTP support is available if the underlying RTP engine in use supports it.
- ;
- ; dtlsenable = yes ; Enable or disable DTLS-SRTP support
- ; dtlsverify = yes ; Verify that provided peer certificate and fingerprint are valid
- ; ; A value of 'yes' will perform both certificate and fingerprint verification
- ; ; A value of 'no' will perform no certificate or fingerprint verification
- ; ; A value of 'fingerprint' will perform ONLY fingerprint verification
- ; ; A value of 'certificate' will perform ONLY certficiate verification
- ; dtlsrekey = 60 ; Interval at which to renegotiate the TLS session and rekey the SRTP session
- ; ; If this is not set or the value provided is 0 rekeying will be disabled
- ; dtlscertfile = file ; Path to certificate file to present
- ; dtlsprivatekey = file ; Path to private key for certificate file
- ; dtlscipher = <SSL cipher string> ; Cipher to use for TLS negotiation
- ; ; A list of valid SSL cipher strings can be found at:
- ; ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
- ; dtlscafile = file ; Path to certificate authority certificate
- ; dtlscapath = path ; Path to a directory containing certificate authority certificates
- ; dtlssetup = actpass ; Whether we are willing to accept connections, connect to the other party, or both.
- ; ; Valid options are active (we want to connect to the other party), passive (we want to
- ; ; accept connections only), and actpass (we will do both). This value will be used in
- ; ; the outgoing SDP when offering and for incoming SDP offers when the remote party sends
- ; ; actpass
- ; dtlsfingerprint = sha-1 ; The hash to use for the fingerprint in SDP (valid options are sha-1 and sha-256)
- ;[sip_proxy]
- ; For incoming calls only. Example: FWD (Free World Dialup)
- ; We match on IP address of the proxy for incoming calls
- ; since we can not match on username (caller id)
- ;type=peer
- ;context=from-fwd
- ;host=fwd.pulver.com
- ;[sip_proxy-out]
- ;type=peer ; we only want to call out, not be called
- ;remotesecret=guessit ; Our password to their service
- ;defaultuser=yourusername ; Authentication user for outbound proxies
- ;fromuser=yourusername ; Many SIP providers require this!
- ;fromdomain=provider.sip.domain
- ;host=box.provider.com
- ;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
- ; ; accept both tcp and udp. The default transport type is only used for
- ; ; outbound messages until a Registration takes place. During the
- ; ; peer Registration the transport type may change to another supported
- ; ; type if the peer requests so.
- ;usereqphone=yes ; This provider requires ";user=phone" on URI
- ;callcounter=yes ; Enable call counter
- ;busylevel=2 ; Signal busy at 2 or more calls
- ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
- ;port=80 ; The port number we want to connect to on the remote side
- ; Also used as "defaultport" in combination with "defaultip" settings
- ;--- sample definition for a provider
- ;[provider1]
- ;type=peer
- ;host=sip.provider1.com
- ;fromuser=4015552299 ; how your provider knows you
- ;remotesecret=youwillneverguessit ; The password we use to authenticate to them
- ;secret=gissadetdu ; The password they use to contact us
- ;callbackextension=123 ; Register with this server and require calls coming back to this extension
- ;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
- ; ; accept both tcp and udp. Default is udp. The first transport
- ; ; listed will always be used for outgoing connections.
- ;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
- ; ; message count will be stored in the configured virtual mailbox. It can be used
- ; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the
- ; ; mailbox.
- ;
- ; Because you might have a large number of similar sections, it is generally
- ; convenient to use templates for the common parameters, and add them
- ; the the various sections. Examples are below, and we can even leave
- ; the templates uncommented as they will not harm:
- [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
- ; Or, more simply:
- ;allow=!all,ilbc,g729,gsm,g723,ulaw
- [ulaw-phone](!) ; and another one for ulaw-only
- disallow=all
- allow=ulaw
- ; Again, more simply:
- ;allow=!all,ulaw
- ; and finally instantiate a few phones
- ;
- ; [2133](natted-phone,my-codecs)
- ; secret = peekaboo
- ; [2134](natted-phone,ulaw-phone)
- ; secret = not_very_secret
- ; [2136](public-phone,ulaw-phone)
- ; secret = not_very_secret_either
- ; ...
- ;
- ; Standard configurations not using templates look like this:
- ;
- ;[grandstream1]
- ;type=friend
- ;context=from-sip ; Where to start in the dialplan when this phone calls
- ;recordonfeature=dynamicfeature1 ; Feature to use when INFO with Record: on is received.
- ;recordofffeature=dynamicfeature2 ; Feature to use when INFO with Record: off is received.
- ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
- ; on incoming calls to Asterisk
- ;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'.
- ;host=192.168.0.23 ; we have a static but private IP address
- ; No registration allowed
- ;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
- ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
- ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
- ; from the phone to asterisk (deprecated)
- ; 1 for the explicit peer, 1 for the explicit user,
- ; remember that a friend equals 1 peer and 1 user in
- ; memory
- ; There is no combined call counter for a "friend"
- ; so there's currently no way in sip.conf to limit
- ; to one inbound or outbound call per phone. Use
- ; the group counters in the dial plan for that.
- ;
- ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
- ;disallow=all ; need to disallow=all before we can use allow=
- ;allow=ulaw ; Note: In user sections the order of codecs
- ; listed with allow= does NOT matter!
- ;allow=alaw
- ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
- ;allow=g729 ; Pass-thru only unless g729 license obtained
- ;callingpres=allowed_passed_screen ; Set caller ID presentation
- ; See function CALLERPRES documentation for possible
- ; values.
- ;[xlite1]
- ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
- ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
- ;type=friend
- ;regexten=1234 ; When they register, create extension 1234
- ;callerid="Jane Smith" <5678>
- ;host=dynamic ; This device needs to register
- ;directmedia=no ; Typically set to NO if behind NAT
- ;disallow=all
- ;allow=gsm ; GSM consumes far less bandwidth than ulaw
- ;allow=ulaw
- ;allow=alaw
- ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
- ;registertrying=yes ; Send a 100 Trying when the device registers.
- ;[snom]
- ;type=friend ; Friends place calls and receive calls
- ;context=from-sip ; Context for incoming calls from this user
- ;secret=blah
- ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
- ;language=de ; Use German prompts for this user
- ;host=dynamic ; This peer register with us
- ;dtmfmode=inband ; Choices are inband, rfc2833, or info
- ;defaultip=192.168.0.59 ; IP used until peer registers
- ;mailbox=1234@context,2345@context ; Mailbox(-es) for message waiting indicator
- ;subscribemwi=yes ; Only send notifications if this phone
- ; subscribes for mailbox notification
- ;vmexten=voicemail ; dialplan extension to reach mailbox
- ; sets the Message-Account in the MWI notify message
- ; defaults to global vmexten which defaults to "asterisk"
- ;disallow=all
- ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
- ;[polycom]
- ;type=friend ; Friends place calls and receive calls
- ;context=from-sip ; Context for incoming calls from this user
- ;secret=blahpoly
- ;host=dynamic ; This peer register with us
- ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
- ;defaultuser=polly ; Username to use in INVITE until peer registers
- ;defaultip=192.168.40.123
- ; Normally you do NOT need to set this parameter
- ;disallow=all
- ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
- ;progressinband=no ; Polycom phones don't work properly with "never"
- ;[pingtel]
- ;type=friend
- ;secret=blah
- ;host=dynamic
- ;insecure=port ; Allow matching of peer by IP address without
- ; matching port number
- ;insecure=invite ; Do not require authentication of incoming INVITEs
- ;insecure=port,invite ; (both)
- ;qualify=1000 ; Consider it down if it's 1 second to reply
- ; Helps with NAT session
- ; qualify=yes uses default value
- ;qualifyfreq=60 ; Qualification: How often to check for the
- ; host to be up in seconds
- ; Set to low value if you use low timeout for
- ; NAT of UDP sessions
- ;
- ; Call group and Pickup group should be in the range from 0 to 63
- ;
- ;callgroup=1,3-4 ; We are in caller groups 1,3,4
- ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
- ;namedcallgroup=engineering,sales,netgroup,protgroup ; We are in named call groups engineering,sales,netgroup,protgroup
- ;namedpickupgroup=sales ; We can do call pick-p for named call group sales
- ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
- ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
- ;permit=192.168.0.60/255.255.255.0
- ;permit=192.168.0.60/24 ; we can also use CIDR notation for subnet masks
- ;permit=2001:db8::/32 ; IPv6 ACLs can be specified if desired. IPv6 ACLs
- ; apply only to IPv6 addresses, and IPv4 ACLs apply
- ; only to IPv4 addresses.
- ;acl=named_acl_example ; Use named ACLs defined in acl.conf
- ;[cisco1]
- ;type=friend
- ;secret=blah
- ;qualify=200 ; Qualify peer is no more than 200ms away
- ;host=dynamic ; This device registers with us
- ;directmedia=no ; Asterisk by default tries to redirect the
- ; RTP media stream (audio) to go directly from
- ; the caller to the callee. Some devices do not
- ; support this (especially if one of them is
- ; behind a NAT).
- ;defaultip=192.168.0.4 ; IP address to use until registration
- ;defaultuser=goran ; Username to use when calling this device before registration
- ; Normally you do NOT need to set this parameter
- ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
- ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
- ; cause the given audio file to
- ; be played upon completion of
- ; an attended transfer to the
- ; target of the transfer.
- ;[pre14-asterisk]
- ;type=friend
- ;secret=digium
- ;host=dynamic
- ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
- ; You must have this turned on or DTMF reception will work improperly.
- ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
- ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
- ; external IP address of the remote device. If port forwarding is done at the client side
- ; then UDPTL will flow to the remote device.
- #include snep/snep-sip.conf
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