; ; 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 ; And to alter the To: or the From: header, you can additionally append ; the following to any of the above strings: ; [![touser[@todomain]][![fromuser][@fromdomain]]] ; ; ; 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 ; ; (Specifying only @todomain without touser will create an invalid SIP ; request.) ; ; Similarly, you can specify the From header as well, after a second ; exclamation mark: ; ; SIP/customer@mysipproxy!!customersupport@wearespindle.com ; ; 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=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) ;rtpbindaddr=172.16.42.1 ; IP address to bind RTP listen sock to (default is disabled). When ; disabled the udpbindaddr is used. ; 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_enabled = true ; Set to false to prevent chan_sip from listening to websockets. This ; is neeeded when using chan_sip and res_pjsip_transport_websockets on ; the same system. ;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. ; ; ----------------------- TLS settings ------------------------------------------------------------ ;tlscertfile= ; 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. If the ; file name ends in _rsa, for example "asterisk_rsa.pem", the files ; "asterisk_dsa.pem" and/or "asterisk_ecc.pem" are loaded ; (certificate, intermediates, private key), to support multiple ; algorithms for server authentication (RSA, DSA, ECDSA). If the chains ; are different, at least OpenSSL 1.0.2 is required. ; Default is to look for "asterisk.pem" in current directory ;tlsprivatekey= ; 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= ; 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= ; 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= ; 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 the general- ; purpose version-flexible SSL/TLS method (sslv23). ; With that, the actual protocol version used will ; be negotiated to the highest version mutually ; supported by Asterisk and the remote server, i.e. ; TLSv1.2. The supported protocols are listed at ; http://www.openssl.org/docs/ssl/SSL_CTX_new.html ; SSLv2 and SSLv3 are disabled within Asterisk. ; Your distribution might have changed that list ; further. ; ; -------------------------- 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= ; 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 = no ; Control when subscriptions get notified of ringing state. ; Specify 'no' to not send any ringing notifications. ; Specify 'yes' to always send ringing notifications (default). ; Specify 'notinuse' to only send ringing notifications for ; extensions that are not currently in use. This is useful as a ; visual indication of who is available to pick up an incoming call ;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=' 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 => fromuser:secret:username@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 ------------------------- ; 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=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. ; ; Note that this does not change the listen address for RTP, it only changes the ; advertised address in the SDP. The Asterisk RTP engine will still listen on ; the standard IP address. ; ; 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=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 RTP source IPs should be able to pass directmedia to ; each other. Note that directmedia ACLs are not a global ; setting, but must be defined per peer. ; (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. ;rtcp_mux=yes ; Enable support for RFC 5761 RTCP multiplexing which is required for ; WebRTC 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|) ; 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=[,] ; 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 = :@ ; auth = #@ ; 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 ; dtlsautogeneratecert ; 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. ; ; Note that all configuration options except dtlsenable can be set at the general level. ; If set they will be present on the user or peer unless overridden with a different value. ; ; 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 ; dtlsautogeneratecert = yes ; Enable ephemeral DTLS certificate generation. The default is 'no.' ; dtlscertfile = file ; Path to certificate file to present ; dtlsprivatekey = file ; Path to private key for certificate file ; dtlscipher = ; 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 @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 ;[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. ; PJSIP Configuration Samples and Quick Reference ; ; This file has several very basic configuration examples, to serve as a quick ; reference to jog your memory when you need to write up a new configuration. ; It is not intended to teach PJSIP configuration or serve as an exhaustive ; reference of options and potential scenarios. ; ; This file has two main sections. ; First, manually written examples to serve as a handy reference. ; Second, a list of all possible PJSIP config options by section. This is ; pulled from the XML config help. It only shows the synopsis for every item. ; If you want to see more detail please check the documentation sources ; mentioned at the top of this file. ; ============================================================================ ; NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE ; ; This file does not maintain the complete option documentation. ; ============================================================================ ; Documentation ; ; The official documentation is at http://wiki.asterisk.org ; You can read the XML configuration help via Asterisk command line with ; "config show help res_pjsip", then you can drill down through the various ; sections and their options. ; ;========!!!!!!!!!!!!!!!!!!! SECURITY NOTICE !!!!!!!!!!!!!!!!!!!!=========== ; ; At a minimum please read the file "README-SERIOUSLY.bestpractices.txt", ; located in the Asterisk source directory before starting Asterisk. ; Otherwise you risk allowing the security of the Asterisk system to be ; compromised. Beyond that please visit and read the security information on ; the wiki at: https://wiki.asterisk.org/wiki/x/EwFB ; ; A few basics to pay attention to: ; ; Anonymous Calls ; ; By default anonymous inbound calls via PJSIP are not allowed. If you want to ; route anonymous calls you'll need to define an endpoint named "anonymous". ; res_pjsip_endpoint_identifier_anonymous.so handles that functionality so it ; must be loaded. It is not recommended to accept anonymous calls. ; ; Access Control Lists ; ; See the example ACL configuration in this file. Read the configuration help ; for the section and all of its options. Look over the samples in acl.conf ; and documentation at https://wiki.asterisk.org/wiki/x/uA80AQ ; If possible, restrict access to only networks and addresses you trust. ; ; Dialplan Contexts ; ; When defining configuration (such as an endpoint) that links into ; dialplan configuration, be aware of what that dialplan does. It's easy to ; accidentally provide access to internal or outbound dialing extensions which ; could cost you severely. The "context=" line in endpoint configuration ; determines which dialplan context inbound calls will enter into. ; ;============================================================================= ; Overview of Configuration Section Types Used in the Examples ; ; * Transport "transport" ; * Configures res_pjsip transport layer interaction. ; * Endpoint "endpoint" ; * Configures core SIP functionality related to SIP endpoints. ; * Authentication "auth" ; * Stores inbound or outbound authentication credentials for use by trunks, ; endpoints, registrations. ; * Address of Record "aor" ; * Stores contact information for use by endpoints. ; * Endpoint Identification "identify" ; * Maps a host directly to an endpoint ; * Access Control List "acl" ; * Defines a permission list or references one stored in acl.conf ; * Registration "registration" ; * Contains information about an outbound SIP registration ; * Resource Lists ; * Contains information for configuring resource lists. ; * Phone Provisioning "phoneprov" ; * Contains information needed by res_phoneprov for autoprovisioning ; The following sections show example configurations for various scenarios. ; Most require a couple or more configuration types configured in concert. ;============================================================================= ; Naming of Configuration Sections ; ; Configuration section names are denoted with enclosing brackets, ; e.g. [6001] ; In most cases, you can name a section whatever makes sense to you. For example ; you might name a transport [transport-udp-nat] to help you remember how that ; section is being used. However, in some cases, ("endpoint" and "aor" types) ; the section name has a relationship to its function. ; ; Depending on the modules loaded, Asterisk can match SIP requests to an ; endpoint or aor in a few ways: ; ; 1) Match a section name for endpoint type sections to the username in the ; "From" header of inbound SIP requests. ; 2) Match a section name for aor type sections to the username in the "To" ; header of inbound SIP REGISTER requests. ; 3) With an identify type section configured, match an inbound SIP request of ; any type to an endpoint or aor based on the IP source address of the ; request. ; ; Note that sections can have the same name as long as their "type" options are ; set to different values. In most cases it makes sense to have associated ; configuration sections use the same name, as you'll see in the examples within ; this file. ;===============EXAMPLE TRANSPORTS============================================ ; ; A few examples for potential transport options. ; ; For the NAT transport example, be aware that the options starting with ; the prefix "external_" will only apply to communication with addresses ; outside the range set with "local_net=". ; ; You can have more than one of any type of transport, as long as it doesn't ; use the same resources (bind address, port, etc) as the others. ; Basic UDP transport ; ;[transport-udp] ;type=transport ;protocol=udp ;udp,tcp,tls,ws,wss,flow ;bind=0.0.0.0 ; ********** [transport-udp] type=transport protocol=udp ;udp,tcp,tls,ws,wss,flow bind=0.0.0.0 ; UDP transport behind NAT ;[transport-udp-nat] ;type=transport ;protocol=udp ;bind=0.0.0.0 ;local_net=192.0.2.0/24 ;external_media_address=203.0.113.1 ;external_signaling_address=203.0.113.1 ; Basic IPv6 UDP transport ; ;[transport-udp-ipv6] ;type=transport ;protocol=udp ;bind=:: ; Example IPv4 TLS transport ; ;[transport-tls] ;type=transport ;protocol=tls ;bind=0.0.0.0 ;cert_file=/path/mycert.crt ;priv_key_file=/path/mykey.key ;cipher=ADH-AES256-SHA,ADH-AES128-SHA ;method=tlsv1 ; Example flow transport ; ; A flow transport is used to reference a flow of signaling with a specific ; target. All endpoints or other objects that reference the transport will use ; the same underlying transport and can share runtime discovered transport ; configuration (such as service routes). The protocol in use will be determined ; based on the URI used to establish the connection. Currently only TCP and TLS ; are supported. ; ;[transport-flow] ;type=transport ;protocol=flow ;===============OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION============ ; ; This is a simple registration that works with some SIP trunking providers. ; You'll need to set up the auth example "mytrunk_auth" below to enable outbound ; authentication. Note that we "outbound_auth=" use for outbound authentication ; instead of "auth=", which is for inbound authentication. ; ; If you are registering to a server from behind NAT, be sure you assign a transport ; that is appropriately configured with NAT related settings. See the NAT transport example. ; ; "contact_user=" sets the SIP contact header's user portion of the SIP URI ; this will affect the extension reached in dialplan when the far end calls you at this ; registration. The default is 's'. ; ; If you would like to enable line support and have incoming calls related to this ; registration go to an endpoint automatically the "line" and "endpoint" options must ; be set. The "endpoint" option specifies what endpoint the incoming call should be ; associated with. ;[mytrunk] ;type=registration ;transport=transport-udp ;outbound_auth=mytrunk_auth ;server_uri=sip:sip.example.com ;client_uri=sip:1234567890@sip.example.com ;contact_user=1234567890 ;retry_interval=60 ;forbidden_retry_interval=600 ;expiration=3600 ;line=yes ;endpoint=mytrunk ;[mytrunk_auth] ;type=auth ;auth_type=userpass ;password=1234567890 ;username=1234567890 ;realm=sip.example.com ;===============ENDPOINT CONFIGURED AS A TRUNK, OUTBOUND AUTHENTICATION======= ; ; This is one way to configure an endpoint as a trunk. It is set up with ; "outbound_auth=" to enable authentication when dialing out through this ; endpoint. There is no inbound authentication set up since a provider will ; not normally authenticate when calling you. ; ; The identify configuration enables IP address matching against this endpoint. ; For calls from a trunking provider, the From user may be different every time, ; so we want to match against IP address instead of From user. ; ; If you want the provider of your trunk to know where to send your calls ; you'll need to use an outbound registration as in the example above this ; section. ; ; NAT ; ; At a basic level configure the endpoint with a transport that is set up ; with the appropriate NAT settings. There may be some additional settings you ; need here based on your NAT/Firewall scenario. Look to the CLI config help ; "config show help res_pjsip endpoint" or on the wiki for other NAT related ; options and configuration. We've included a few below. ; ; AOR ; ; Endpoints use one or more AOR sections to store their contact details. ; You can define multiple contact addresses in SIP URI format in multiple ; "contact=" entries. ; ;[mytrunk] ;type=endpoint ;transport=transport-udp ;context=from-external ;disallow=all ;allow=ulaw ;outbound_auth=mytrunk_auth ;aors=mytrunk ; ;A few NAT relevant options that may come in handy. ;force_rport=yes ;It's a good idea to read the configuration help for each ;direct_media=no ;of these options. ;ice_support=yes ;[mytrunk] ;type=aor ;contact=sip:198.51.100.1:5060 ;contact=sip:198.51.100.2:5060 ;[mytrunk] ;type=identify ;endpoint=mytrunk ;match=198.51.100.1 ;match=198.51.100.2 ;match=192.168.10.0:5061/24 ;=============ENDPOINT CONFIGURED AS A TRUNK, INBOUND AUTH AND REGISTRATION=== ; ; Here we are allowing a remote device to register to Asterisk and requiring ; that they authenticate for registration and calls. ; You'll note that this configuration is essentially the same as configuring ; an endpoint for use with a SIP phone. ;[7000] ;type=endpoint ;context=from-external ;disallow=all ;allow=ulaw ;transport=transport-udp ;auth=7000 ;aors=7000 ;[7000] ;type=auth ;auth_type=userpass ;password=7000 ;username=7000 ;[7000] ;type=aor ;max_contacts=1 ;===============ENDPOINT CONFIGURED FOR USE WITH A SIP PHONE================== ; ; This example includes the endpoint, auth and aor configurations. It ; requires inbound authentication and allows registration, as well as references ; a transport that you'll need to uncomment from the previous examples. ; ; Uncomment one of the transport lines to choose which transport you want. If ; not specified then the default transport chosen is the first compatible transport ; in the configuration file for the contact URL. ; ; Modify the "max_contacts=" line to change how many unique registrations to allow. ; ; Use the "contact=" line instead of max_contacts= if you want to statically ; define the location of the device. ; ; If using the TLS enabled transport, you may want the "media_encryption=sdes" ; option to additionally enable SRTP, though they are not mutually inclusive. ; ; If this endpoint were remote, and it was using a transport configured for NAT ; then you likely want to use "direct_media=no" to prevent audio issues. ;[6001] ;type=endpoint ;transport=transport-udp ;context=from-internal ;disallow=all ;allow=ulaw ;allow=gsm ;auth=6001 ;aors=6001 ; ; ********** ;[1000] ;type=endpoint ;transport=transport-udp ;context=from-internal ;disallow=all ;allow=ulaw ;allow=gsm ;uth=1000 ;aors=1000 ; ; A few more transports to pick from, and some related options below them. ; ;transport=transport-tls ;media_encryption=sdes ;transport=transport-udp-ipv6 ;transport=transport-udp-nat ;direct_media=no ; ; MWI related options ;aggregate_mwi=yes ;mailboxes=6001@default,7001@default ;mwi_from_user=6001 ; ; Extension and Device state options ; ;device_state_busy_at=1 ;allow_subscribe=yes ;sub_min_expiry=30 ; ; STIR/SHAKEN support. ; ;stir_shaken=no ;stir_shaken_profile=my_profile ;[6001] ;type=auth ;auth_type=userpass ;password=6001 ;username=6001 ; ********** ;[1000] ;type=auth ;auth_type=userpass ;password=1000fr ;username=1000 [1000] type=endpoint context=from-internal transport=transport-udp disallow=all allow=ulaw auth=1000auth aors=1000 [1000auth] type=auth auth_type=userpass username=1000 password=1000fr ;[6001] ;type=aor ;max_contacts=1 ;contact=sip:6001@192.0.2.1:5060 ; ********** [1000] type=aor max_contacts=1 ;contact=sip:6001@192.0.2.1:5060 [1001] type=endpoint transport=transport-udp context=from-internal disallow=all allow=ulaw auth=1001auth aors=1001 [1001auth] type=auth auth_type=userpass username=1001 password=1001fr [1001] type=aor max_contacts=5 [1003] type=endpoint context=from-internal transport=transport-udp disallow=all allow=ulaw auth=1003auth aors=1003 [1003auth] type=auth auth_type=userpass username=1003 password=1003fr [1003] type=aor max_contacts=25 [1004] type=endpoint context=from-internal disallow=all allow=ulaw transport=transport-udp auth=1004auth aors=1004 [1004auth] type=auth auth_type=userpass username=1004 password=1004fr [1004] type=aor max_contacts=5 [1008] type=endpoint context=from-internal transport=transport-udp disallow=all allow=ulaw auth=1008auth aors=1008 [1008auth] type=auth auth_type=userpass username=1008 password=1008fr [1008] type=aor max_contacts=3 [9999] type=endpoint context=from-internal transport=transport-udp disallow=all allow=ulaw auth=9999auth aors=9999 [9999auth] type=auth auth_type=userpass username=9999 password=9999fr [9999] type=aor max_contacts=5 ;contact=sip:6001@192.0.2.1:5060 [4000] ?type=endpoint context=from-internal transport=transport-udp disallow=all allow=ulaw ;===============ENDPOINT BEHIND NAT OR FIREWALL=============================== ; ; This example assumes your transport is configured with a public IP and the ; endpoint itself is behind NAT and maybe a firewall, rather than having ; Asterisk behind NAT. For the sake of simplicity, we'll assume a typical ; VOIP phone. The most important settings to configure are: ; ; * direct_media, to ensure Asterisk stays in the media path ; * rtp_symmetric and force_rport options to help the far-end NAT/firewall ; ; Depending on the settings of your remote SIP device or NAT/firewall device ; you may have to experiment with a combination of these settings. ; ; If both Asterisk and the remote phones are a behind NAT/firewall then you'll ; have to make sure to use a transport with appropriate settings (as in the ; transport-udp-nat example). ; ;[6002] ;type=endpoint ;transport=transport-udp ;context=from-internal ;disallow=all ;allow=ulaw ;auth=6002 ;aors=6002 ;direct_media=no ;rtp_symmetric=yes ;force_rport=yes ;rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port ;ice_support=yes ;This is specific to clients that support NAT traversal ;for media via ICE,STUN,TURN. See the wiki at: ;https://wiki.asterisk.org/wiki/x/D4FHAQ ;for a deeper explanation of this topic. ;[6002] ;type=auth ;auth_type=userpass ;password=6002 ;username=6002 ;[6002] ;type=aor ;max_contacts=2 ;============EXAMPLE ACL CONFIGURATION========================================== ; ; The ACL or Access Control List section defines a set of permissions to permit ; or deny access to various address or addresses. Alternatively it references an ; ACL configuration already set in acl.conf. ; ; The ACL configuration is independent of individual endpoint configuration and ; operates on all inbound SIP communication using res_pjsip. ; Reference an ACL defined in acl.conf. ; ;[acl] ;type=acl ;acl=example_named_acl1 ; Reference a contactacl specifically. ; ;[acl] ;type=acl ;contact_acl=example_contact_acl1 ; Define your own ACL here in pjsip.conf and ; permit or deny by IP address or range. ; ;[acl] ;type=acl ;deny=0.0.0.0/0.0.0.0 ;permit=209.16.236.0/24 ;deny=209.16.236.1 ; Restrict based on Contact Headers rather than IP. ; Define options multiple times for various addresses or use a comma-delimited string. ; ;[acl] ;type=acl ;contact_deny=0.0.0.0/0.0.0.0 ;contact_permit=209.16.236.0/24 ;contact_permit=209.16.236.1 ;contact_permit=209.16.236.2,209.16.236.3 ; Restrict based on Contact Headers rather than IP and use ; advanced syntax. Note the bang symbol used for "NOT", so we can deny ; 209.16.236.12/32 within the permit= statement. ; ;[acl] ;type=acl ;contact_deny=0.0.0.0/0.0.0.0 ;contact_permit=209.16.236.0 ;permit=209.16.236.0/24, !209.16.236.12/32 ;============EXAMPLE RLS CONFIGURATION========================================== ; ;Asterisk provides support for RFC 4662 Resource List Subscriptions. This allows ;for an endpoint to, through a single subscription, subscribe to the states of ;multiple resources. Resource lists are configured in pjsip.conf using the ;resource_list configuration object. Below is an example of a resource list that ;allows an endpoint to subscribe to the presence of alice, bob, and carol. ;[my_list] ;type=resource_list ;list_item=alice ;list_item=bob ;list_item=carol ;event=presence ;The "event" option in the resource list corresponds to the SIP event-package ;that the subscribed resources belong to. A resource list can only provide states ;for resources that belong to the same event-package. This means that you cannot ;create a list that is a combination of presence and message-summary resources, ;for instance. Any event-package that Asterisk supports can be used in a resource ;list (presence, dialog, and message-summary). Whenever support for a new event- ;package is added to Asterisk, support for that event-package in resource lists ;will automatically be supported. ;The "list_item" options indicate the names of resources to subscribe to. The ;way these are interpreted is event-package specific. For instance, with presence ;list_items, hints in the dialplan are looked up. With message-summary list_items, ;mailboxes are looked up using your installed voicemail provider (app_voicemail ;by default). Note that in the above example, the list_item options were given ;one per line. However, it is also permissible to provide multiple list_item ;options on a single line (e.g. list_item = alice,bob,carol). ;In addition to the options presented in the above configuration, there are two ;more configuration options that can be set. ; * full_state: dictates whether Asterisk should always send the states of ; all resources in the list at once. Defaults to "no". You should only set ; this to "yes" if you are interoperating with an endpoint that does not ; behave correctly when partial state notifications are sent to it. ; * notification_batch_interval: By default, Asterisk will send a NOTIFY request ; immediately when a resource changes state. This option causes Asterisk to ; start batching resource state changes for the specified number of milliseconds ; after a resource changes states. This way, if multiple resources change state ; within a brief interval, Asterisk can send a single NOTIFY request with all ; of the state changes reflected in it. ;There is a limitation to the size of resource lists in Asterisk. If a constructed ;notification from Asterisk will exceed 64000 bytes, then the message is deemed ;too large to send. If you find that you are seeing error messages about SIP ;NOTIFY requests being too large to send, consider breaking your lists into ;sub-lists. ;============EXAMPLE PHONEPROV CONFIGURATION================================ ; Before configuring provisioning here, see the documentation for res_phoneprov ; and configure phoneprov.conf appropriately. ; For each user to be autoprovisioned, a [phoneprov] configuration section ; must be created. At a minimum, the 'type', 'PROFILE' and 'MAC' variables must ; be set. All other variables are optional. ; Example: ;[1000] ;type=phoneprov ; must be specified as 'phoneprov' ;endpoint=1000 ; Required only if automatic setting of ; USERNAME, SECRET, DISPLAY_NAME and CALLERID ; are needed. ;PROFILE=digium ; required ;MAC=deadbeef4dad ; required ;SERVER=myserver.example.com ; A standard variable ;TIMEZONE=America/Denver ; A standard variable ;MYVAR=somevalue ; A user confdigured variable ; If the phoneprov sections have common variables, it is best to create a ; phoneprov template. The example below will produce the same configuration ; as the one specified above except that MYVAR will be overridden for ; the specific user. ; Example: ;[phoneprov_defaults](!) ;type=phoneprov ; must be specified as 'phoneprov' ;PROFILE=digium ; required ;SERVER=myserver.example.com ; A standard variable ;TIMEZONE=America/Denver ; A standard variable ;MYVAR=somevalue ; A user configured variable ;[1000](phoneprov_defaults) ;endpoint=1000 ; Required only if automatic setting of ; USERNAME, SECRET, DISPLAY_NAME and CALLERID ; are needed. ;MAC=deadbeef4dad ; required ;MYVAR=someOTHERvalue ; A user confdigured variable ; To have USERNAME and SECRET automatically set, the endpoint ; specified here must in turn have an outbound_auth section defined. ; Fuller example: ;[1000] ;type=endpoint ;outbound_auth=1000-auth ;callerid=My Name <8005551212> ;transport=transport-udp-nat ;[1000-auth] ;type=auth ;auth_type=userpass ;username=myname ;password=mysecret ;[phoneprov_defaults](!) ;type=phoneprov ; must be specified as 'phoneprov' ;PROFILE=someprofile ; required ;SERVER=myserver.example.com ; A standard variable ;TIMEZONE=America/Denver ; A standard variable ;MYVAR=somevalue ; A user configured variable ;[1000](phoneprov_defaults) ;endpoint=1000 ; Required only if automatic setting of ; USERNAME, SECRET, DISPLAY_NAME and CALLERID ; are needed. ;MAC=deadbeef4dad ; required ;MYVAR=someUSERvalue ; A user confdigured variable ;LABEL=1000 ; A standard variable ; The previous sections would produce a template substitution map as follows: ;MAC=deadbeef4dad ;added by pp1000 ;USERNAME=myname ;automatically added by 1000-auth username ;SECRET=mysecret ;automatically added by 1000-auth password ;PROFILE=someprofile ;added by defaults ;SERVER=myserver.example.com ;added by defaults ;SERVER_PORT=5060 ;added by defaults ;MYVAR=someUSERvalue ;added by defaults but overdidden by user ;CALLERID=8005551212 ;automatically added by 1000 callerid ;DISPLAY_NAME=My Name ;automatically added by 1000 callerid ;TIMEZONE=America/Denver ;added by defaults ;TZOFFSET=252100 ;automatically calculated by res_phoneprov ;DST_ENABLE=1 ;automatically calculated by res_phoneprov ;DST_START_MONTH=3 ;automatically calculated by res_phoneprov ;DST_START_MDAY=9 ;automatically calculated by res_phoneprov ;DST_START_HOUR=3 ;automatically calculated by res_phoneprov ;DST_END_MONTH=11 ;automatically calculated by res_phoneprov ;DST_END_MDAY=2 ;automatically calculated by res_phoneprov ;DST_END_HOUR=1 ;automatically calculated by res_phoneprov ;ENDPOINT_ID=1000 ;automatically added by this module ;AUTH_ID=1000-auth ;automatically added by this module ;TRANSPORT_ID=transport-udp-nat ;automatically added by this module ;LABEL=1000 ;added by user ; MODULE PROVIDING BELOW SECTION(S): res_pjsip ;==========================ENDPOINT SECTION OPTIONS========================= ;[endpoint] ; SYNOPSIS: Endpoint ;100rel=yes ; Allow support for RFC3262 provisional ACK tags (default: ; "yes") ;aggregate_mwi=yes ; (default: "yes") ;allow= ; Media Codec s to allow (default: "") ;allow_overlap=yes ; Enable RFC3578 overlap dialing support. (default: "yes") ;overlap_context=default ; Context to used for overlap dialing matches ; (default: same as context option) ;aors= ; AoR s to be used with the endpoint (default: "") ;auth= ; Authentication Object s associated with the endpoint (default: "") ;callerid= ; CallerID information for the endpoint (default: "") ;callerid_privacy=allowed_not_screened ; Default privacy level (default: "allowed_not_screened") ;callerid_tag= ; Internal id_tag for the endpoint (default: "") ;context=default ; Dialplan context for inbound sessions (default: ; "default") ;direct_media_glare_mitigation=none ; Mitigation of direct media re INVITE ; glare (default: "none") ;direct_media_method=invite ; Direct Media method type (default: "invite") ;trust_connected_line=yes ; Accept Connected Line updates from this endpoint ; (default: "yes") ;send_connected_line=yes ; Send Connected Line updates to this endpoint ; (default: "yes") ;connected_line_method=invite ; Connected line method type. ; When set to "invite", check the remote's ; Allow header and if UPDATE is allowed, send ; UPDATE instead of INVITE to avoid SDP ; renegotiation. If UPDATE is not Allowed, ; send INVITE. ; If set to "update", send UPDATE regardless ; of what the remote Allows. ; (default: "invite") ;direct_media=yes ; Determines whether media may flow directly between ; endpoints (default: "yes") ;disable_direct_media_on_nat=no ; Disable direct media session refreshes when ; NAT obstructs the media session (default: ; "no") ;disallow= ; Media Codec s to disallow (default: "") ;dtmf_mode=rfc4733 ; DTMF mode (default: "rfc4733") ;media_address= ; IP address used in SDP for media handling (default: "") ;bind_rtp_to_media_address= ; Bind the RTP session to the media_address. ; This causes all RTP packets to be sent from ; the specified address. (default: "no") ;force_rport=yes ; Force use of return port (default: "yes") ;ice_support=no ; Enable the ICE mechanism to help traverse NAT (default: "no") ;identify_by=username ; A comma-separated list of ways the Endpoint or AoR can be ; identified. ; "username": Identify by the From or To username and domain ; "auth_username": Identify by the Authorization username and realm ; "ip": Identify by the source IP address ; "header": Identify by a configured SIP header value. ; In the username and auth_username cases, if an exact match ; on both username and domain/realm fails, the match is ; retried with just the username. ; (default: "username,ip") ;redirect_method=user ; How redirects received from an endpoint are handled ; (default: "user") ;mailboxes= ; NOTIFY the endpoint when state changes for any of the specified mailboxes. ; Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state ; changes happen for any of the specified mailboxes. (default: "") ;voicemail_extension= ; The voicemail extension to send in the NOTIFY Message-Account header ; (default: global/default_voicemail_extension) ;mwi_subscribe_replaces_unsolicited=no ; An MWI subscribe will replace unsoliticed NOTIFYs ; (default: "no") ;moh_suggest=default ; Default Music On Hold class (default: "default") ;moh_passthrough=yes ; Pass Music On Hold through using SIP re-invites with sendonly ; when placing on hold and sendrecv when taking off hold ;outbound_auth= ; Authentication object used for outbound requests (default: ; "") ;outbound_proxy= ; Proxy through which to send requests, a full SIP URI ; must be provided (default: "") ;rewrite_contact=no ; Allow Contact header to be rewritten with the source ; IP address port (default: "no") ;rtp_symmetric=no ; Enforce that RTP must be symmetric (default: "no") ;send_diversion=yes ; Send the Diversion header conveying the diversion ; information to the called user agent (default: "yes") ;send_pai=no ; Send the P Asserted Identity header (default: "no") ;send_rpid=no ; Send the Remote Party ID header (default: "no") ;rpid_immediate=no ; Send connected line updates on unanswered incoming calls immediately. (default: "no") ;timers_min_se=90 ; Minimum session timers expiration period (default: ; "90") ;timers=yes ; Session timers for SIP packets (default: "yes") ;timers_sess_expires=1800 ; Maximum session timer expiration period ; (default: "1800") ;transport= ; Explicit transport configuration to use (default: "") ; This will force the endpoint to use the specified transport ; configuration to send SIP messages. You need to already know ; what kind of transport (UDP/TCP/IPv4/etc) the endpoint device ; will use. ;trust_id_inbound=no ; Accept identification information received from this ; endpoint (default: "no") ;trust_id_outbound=no ; Send private identification details to the endpoint ; (default: "no") ;type= ; Must be of type endpoint (default: "") ;use_ptime=no ; Use Endpoint s requested packetisation interval (default: ; "no") ;use_avpf=no ; Determines whether res_pjsip will use and enforce usage of ; AVPF for this endpoint (default: "no") ;media_encryption=no ; Determines whether res_pjsip will use and enforce ; usage of media encryption for this endpoint (default: ; "no") ;media_encryption_optimistic=no ; Use encryption if possible but don't fail the call ; if not possible. ;g726_non_standard=no ; When set to "yes" and an endpoint negotiates g.726 ; audio then g.726 for AAL2 packing order is used contrary ; to what is recommended in RFC3551. Note, 'g726aal2' also ; needs to be specified in the codec allow list ; (default: "no") ;inband_progress=no ; Determines whether chan_pjsip will indicate ringing ; using inband progress (default: "no") ;call_group= ; The numeric pickup groups for a channel (default: "") ;pickup_group= ; The numeric pickup groups that a channel can pickup (default: ; "") ;named_call_group= ; The named pickup groups for a channel (default: "") ;named_pickup_group= ; The named pickup groups that a channel can pickup ; (default: "") ;device_state_busy_at=0 ; The number of in use channels which will cause busy ; to be returned as device state (default: "0") ;t38_udptl=no ; Whether T 38 UDPTL support is enabled or not (default: "no") ;t38_udptl_ec=none ; T 38 UDPTL error correction method (default: "none") ;t38_udptl_maxdatagram=0 ; T 38 UDPTL maximum datagram size (default: ; "0") ;fax_detect=no ; Whether CNG tone detection is enabled (default: "no") ;fax_detect_timeout=30 ; How many seconds into a call before fax_detect is ; disabled for the call. ; Zero disables the timeout. ; (default: "0") ;t38_udptl_nat=no ; Whether NAT support is enabled on UDPTL sessions ; (default: "no") ;t38_bind_rtp_to_media_address= ; Bind the UDPTL session to the media_address. ; This causes all UDPTL packets to be sent from ; the specified address. (default: "no") ;tone_zone= ; Set which country s indications to use for channels created ; for this endpoint (default: "") ;language= ; Set the default language to use for channels created for this ; endpoint (default: "") ;one_touch_recording=no ; Determines whether one touch recording is allowed for ; this endpoint (default: "no") ;record_on_feature=automixmon ; The feature to enact when one touch recording ; is turned on (default: "automixmon") ;record_off_feature=automixmon ; The feature to enact when one touch recording ; is turned off (default: "automixmon") ;rtp_engine=asterisk ; Name of the RTP engine to use for channels created ; for this endpoint (default: "asterisk") ;allow_transfer=yes ; Determines whether SIP REFER transfers are allowed ; for this endpoint (default: "yes") ;sdp_owner=- ; String placed as the username portion of an SDP origin o line ; (default: "-") ;sdp_session=Asterisk ; String used for the SDP session s line (default: ; "Asterisk") ;tos_audio=0 ; DSCP TOS bits for audio streams (default: "0") ;tos_video=0 ; DSCP TOS bits for video streams (default: "0") ;cos_audio=0 ; Priority for audio streams (default: "0") ;cos_video=0 ; Priority for video streams (default: "0") ;allow_subscribe=yes ; Determines if endpoint is allowed to initiate ; subscriptions with Asterisk (default: "yes") ;sub_min_expiry=0 ; The minimum allowed expiry time for subscriptions ; initiated by the endpoint (default: "0") ;from_user= ; Username to use in From header for requests to this endpoint ; (default: "") ;mwi_from_user= ; Username to use in From header for unsolicited MWI NOTIFYs to ; this endpoint (default: "") ;from_domain= ; Domain to user in From header for requests to this endpoint ; (default: "") ;dtls_verify=no ; Verify that the provided peer certificate is valid (default: ; "no") ;dtls_rekey=0 ; Interval at which to renegotiate the TLS session and rekey ; the SRTP session (default: "0") ;dtls_auto_generate_cert= ; Enable ephemeral DTLS certificate generation (default: ; "no") ;dtls_cert_file= ; Path to certificate file to present to peer (default: ; "") ;dtls_private_key= ; Path to private key for certificate file (default: ; "") ;dtls_cipher= ; Cipher to use for DTLS negotiation (default: "") ;dtls_ca_file= ; Path to certificate authority certificate (default: "") ;dtls_ca_path= ; Path to a directory containing certificate authority ; certificates (default: "") ;dtls_setup= ; Whether we are willing to accept connections connect to the ; other party or both (default: "") ;dtls_fingerprint= ; Hash to use for the fingerprint placed into SDP ; (default: "SHA-256") ;srtp_tag_32=no ; Determines whether 32 byte tags should be used instead of 80 ; byte tags (default: "no") ;set_var= ; Variable set on a channel involving the endpoint. For multiple ; channel variables specify multiple 'set_var'(s) ;rtp_keepalive= ; Interval, in seconds, between comfort noise RTP packets if ; RTP is not flowing. This setting is useful for ensuring that ; holes in NATs and firewalls are kept open throughout a call. ;rtp_timeout= ; Hang up channel if RTP is not received for the specified ; number of seconds when the channel is off hold (default: ; "0" or not enabled) ;rtp_timeout_hold= ; Hang up channel if RTP is not received for the specified ; number of seconds when the channel is on hold (default: ; "0" or not enabled) ;contact_user= ; On outgoing requests, force the user portion of the Contact ; header to this value (default: "") ;incoming_call_offer_pref= ; Based on this setting, a joint list of ; preferred codecs between those received in an ; incoming SDP offer (remote), and those specified ; in the endpoint's "allow" parameter (local) ; is created and is passed to the Asterisk core. ; ; local - Include all codecs in the local list that ; are also in the remote list preserving the local ; order. (default). ; local_first - Include only the first codec in the ; local list that is also in the remote list. ; remote - Include all codecs in the remote list that ; are also in the local list preserving remote list ; order. ; remote_first - Include only the first codec in ; the remote list that is also in the local list. ;outgoing_call_offer_pref= ; Based on this setting, a joint list of ; preferred codecs between those received from the ; Asterisk core (remote), and those specified in ; the endpoint's "allow" parameter (local) is ; created and is used to create the outgoing SDP ; offer. ; ; local - Include all codecs in the local list that ; are also in the remote list preserving the local ; order. ; local_merge - Include all codecs in the local list ; preserving the local order. ; local_first - Include only the first codec in the ; local list. ; remote - Include all codecs in the remote list that ; are also in the local list preserving remote list ; order. ; remote_merge - Include all codecs in the local list ; preserving the remote list order. (default) ; remote_first - Include only the first codec in the ; remote list that is also in the local list. ;preferred_codec_only=no ; 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. ; default is no. ; NOTE: This option is deprecated in favor ; of incoming_call_offer_pref. Setting both ; options is unsupported. ;asymmetric_rtp_codec= ; Allow the sending and receiving codec to differ and ; not be automatically matched (default: "no") ;refer_blind_progress= ; Whether to notifies all the progress details on blind ; transfer (default: "yes"). The value "no" is useful ; for some SIP phones (Mitel/Aastra, Snom) which expect ; a sip/frag "200 OK" after REFER has been accepted. ;notify_early_inuse_ringing = ; Whether to notifies dialog-info 'early' ; on INUSE && RINGING state (default: "no"). ; The value "yes" is useful for some SIP phones ; (Cisco SPA) to be able to indicate and pick up ; ringing devices. ;max_audio_streams= ; The maximum number of allowed negotiated audio streams ; (default: 1) ;max_video_streams= ; The maximum number of allowed negotiated video streams ; (default: 1) ;webrtc= ; When set to "yes" this also enables the following values that are needed ; for webrtc: rtcp_mux, use_avpf, ice_support, and use_received_transport. ; The following configuration settings also get defaulted as follows: ; media_encryption=dtls ; dtls_verify=fingerprint ; dtls_setup=actpass ; A dtls_cert_file and a dtls_ca_file still need to be specified. ; Default for this option is "no" ;incoming_mwi_mailbox = ; Mailbox name to use when incoming MWI NOTIFYs are ; received. ; If an MWI NOTIFY is received FROM this endpoint, ; this mailbox will be used when notifying other modules ; of MWI status changes. If not set, incoming MWI ; NOTIFYs are ignored. ;follow_early_media_fork = ; On outgoing calls, if the UAS responds with ; different SDP attributes on subsequent 18X or 2XX ; responses (such as a port update) AND the To tag ; on the subsequent response is different than that ; on the previous one, follow it. This usually ; happens when the INVITE is forked to multiple UASs ; and more than 1 sends an SDP answer. ; This option must also be enabled in the system ; section. ; (default: yes) ;accept_multiple_sdp_answers = ; On outgoing calls, if the UAS responds with ; different SDP attributes on non-100rel 18X or 2XX ; responses (such as a port update) AND the To tag on ; the subsequent response is the same as that on the ; previous one, process it. This can happen when the ; UAS needs to change ports for some reason such as ; using a separate port for custom ringback. ; This option must also be enabled in the system ; section. ; (default: no) ;suppress_q850_reason_headers = ; Suppress Q.850 Reason headers for this endpoint. ; Some devices can't accept multiple Reason headers ; and get confused when both 'SIP' and 'Q.850' Reason ; headers are received. This option allows the ; 'Q.850' Reason header to be suppressed. ; (default: no) ;ignore_183_without_sdp = ; Do not forward 183 when it doesn't contain SDP. ; Certain SS7 internetworking scenarios can result in ; a 183 to be generated for reasons other than early ; media. Forwarding this 183 can cause loss of ; ringback tone. This flag emulates the behavior of ; chan_sip and prevents these 183 responses from ; being forwarded. ; (default: no) ;stir_shaken = ; If this is enabled, STIR/SHAKEN operations will be ; performed on this endpoint. This includes inbound ; and outbound INVITEs. On an inbound INVITE, Asterisk ; will check for an Identity header and attempt to ; verify the call. On an outbound INVITE, Asterisk will ; add an Identity header that others can use to verify ; calls from this endpoint. Additional configuration is ; done in stir_shaken.conf. ; The STIR_SHAKEN dialplan function must be used to get ; the verification results on inbound INVITEs. Nothing ; happens to the call if verification fails; it's up to ; you to determine what to do with the results. ; (default: no) ;stir_shaken_profile = ; If a profile is specified (defined in stir_shaken.conf), ; this endpoint will follow the rules defined there. ;allow_unauthenticated_options = ; By default, chan_pjsip will challenge an incoming ; OPTIONS request for authentication credentials just ; as it would an INVITE request. This is consistent ; with RFC 3261. ; There are many UAs that use an OPTIONS request as a ; "ping" and they expect a 200 response indicating that ; the remote party is up and running without a need to ; authenticate. ; Setting allow_unauthenticated_options to 'yes' will ; instruct chan_pjsip to skip the authentication step ; when it receives an OPTIONS request for this ; endpoint. ; There are security implications to enabling this ; setting as it can allow information disclosure to ; occur - specifically, if enabled, an external party ; could enumerate and find the endpoint name by ; sending OPTIONS requests and examining the ; responses. ; (default: no) ;geoloc_incoming_call_profile = ; This geolocation profile will be applied to all calls received ; by the channel driver from the remote endpoint before they're ; forwarded to the dialplan. ;geoloc_outgoing_call_profile = ; This geolocation profile will be applied to all calls received ; by the channel driver from the dialplan before they're forwarded ; the remote endpoint. ; ; send_aoc = ; This options turns on and off support for sending AOC to endpoints. ; AOC updates can be sent using the AOCMessage AMI action or come ; from PRI channels. ; (default: no) ;==========================AUTH SECTION OPTIONS========================= ;[auth] ; SYNOPSIS: Authentication type ; ; Note: Using the same auth section for inbound and outbound ; authentication is not recommended. There is a difference in ; meaning for an empty realm setting between inbound and outbound ; authentication uses. Look to the CLI config help ; "config show help res_pjsip auth realm" or on the wiki for the ; difference. ; ;auth_type=userpass ; Authentication type. May be ; "userpass" for plain text passwords or ; "md5" for pre-hashed credentials. ; (default: "userpass") ;nonce_lifetime=32 ; Lifetime of a nonce associated with this ; authentication config (default: "32") ;md5_cred= ; As an alternative to specifying a plain text password, ; you can hash the username, realm and password ; together one time and place the hash value here. ; The input to the hash function must be in the ; following format: ; :: ; For incoming authentication (asterisk is the UAS), ; the realm must match either the realm set in this object ; or the default set in in the "global" object. ; ; For outgoing authentication (asterisk is the UAC), ; the realm must match what the server will be sending ; in their WWW-Authenticate header. It can't be blank ; unless you expect the server to be sending a blank ; realm in the header. ; You can generate the hash with the following shell ; command: ; $ echo -n "myname:myrealm:mypassword" | md5sum ; Note the '-n'. You don't want a newline to be part ; of the hash. (default: "") ;password= ; PlainText password used for authentication (default: "") ;realm= ; For incoming authentication (asterisk is the UAS), ; this is the realm to be sent on WWW-Authenticate ; headers. If not specified, the global object's ; "default_realm" will be used. ; ; For outgoing authentication (asterisk is the UAC), this ; must either be the realm the server is expected to send, ; or left blank or contain a single '*' to automatically ; use the realm sent by the server. If you have multiple ; auth objects for an endpoint, the realm is also used to ; match the auth object to the realm the server sent. ; ; Using the same auth section for inbound and outbound ; authentication is not recommended. There is a difference in ; meaning for an empty realm setting between inbound and outbound ; authentication uses. ; (default: "") ;type= ; Must be auth (default: "") ;username= ; Username to use for account (default: "") ;==========================DOMAIN_ALIAS SECTION OPTIONS========================= ;[domain_alias] ; SYNOPSIS: Domain Alias ;type= ; Must be of type domain_alias (default: "") ;domain= ; Domain to be aliased (default: "") ;==========================TRANSPORT SECTION OPTIONS========================= ;[transport] ; SYNOPSIS: SIP Transport ; ;bind= ; IP Address and optional port to bind to for this transport (default: ; "") ; Note that for the Websocket transport the TLS configuration is configured ; in http.conf and is applied for all HTTPS traffic. ;ca_list_file= ; File containing a list of certificates to read TLS ONLY ; (default: "") ;ca_list_path= ; Path to directory containing certificates to read TLS ONLY. ; PJProject version 2.4 or higher is required for this option to ; be used. ; (default: "") ;cert_file= ; Certificate file for endpoint TLS ONLY ; Will read .crt or .pem file but only uses cert, ; a .key file must be specified via priv_key_file. ; Since PJProject version 2.5: If the file name ends in _rsa, ; for example "asterisk_rsa.pem", the files "asterisk_dsa.pem" ; and/or "asterisk_ecc.pem" are loaded (certificate, inter- ; mediates, private key), to support multiple algorithms for ; server authentication (RSA, DSA, ECDSA). If the chains are ; different, at least OpenSSL 1.0.2 is required. This option ; can be reloaded resulting in an updated certificate if the ; filename remains unchanged. ; (default: "") ;cipher= ; Preferred cryptography cipher names TLS ONLY (default: "") ;method= ; Method of SSL transport TLS ONLY (default: "") ;priv_key_file= ; Private key file TLS ONLY. This option can be reloaded ; resulting in an updated private key if the filename remains ; unchanged. ; (default: "") ;verify_client= ; Require verification of client certificate TLS ONLY (default: ; "") ;verify_server= ; Require verification of server certificate TLS ONLY (default: ; "") ;require_client_cert= ; Require client certificate TLS ONLY (default: "") ;domain= ; Domain the transport comes from (default: "") ;external_media_address= ; External IP address to use in RTP handling ; (default: "") ;external_signaling_address= ; External address for SIP signalling (default: ; "") ;external_signaling_port=0 ; External port for SIP signalling (default: ; "0") ;local_net= ; Network to consider local used for NAT purposes (default: "") ;password= ; Password required for transport (default: "") ;protocol=udp ; Protocol to use for SIP traffic (default: "udp") ;type= ; Must be of type transport (default: "") ;tos=0 ; Enable TOS for the signalling sent over this transport (default: "0") ;cos=0 ; Enable COS for the signalling sent over this transport (default: "0") ;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. ;allow_reload=no ; Although transports can now be reloaded, that may not be ; desirable because of the slight possibility of dropped ; calls. To make sure there are no unintentional drops, if ; this option is set to 'no' (the default) changes to the ; particular transport will be ignored. If set to 'yes', ; changes (if any) will be applied. ;symmetric_transport=no ; When a request from a dynamic contact comes in on a ; transport with this option set to 'yes', the transport ; name will be saved and used for subsequent outgoing ; requests like OPTIONS, NOTIFY and INVITE. It's saved ; as a contact uri parameter named 'x-ast-txp' and will ; display with the contact uri in CLI, AMI, and ARI ; output. On the outgoing request, if a transport ; wasn't explicitly set on the endpoint AND the request ; URI is not a hostname, the saved transport will be ; used and the 'x-ast-txp' parameter stripped from the ; outgoing packet. ;allow_wildcard_certs=no ; In conjunction with verify_server, if 'yes' allow use ; of wildcards, i.e. '*.' in certs for common, and ; subject alt names of type DNS for TLS transport ; types. Note, names must start with the wildcard. ; Partial wildcards, e.g. 'f*.example.com' and ; 'foo.*.com' are disallowed. As well, names only ; match against a single level meaning '*.example.com' ; matches 'foo.example.com', but not ; 'foo.bar.example.com'. Defaults to 'no'. ;==========================AOR SECTION OPTIONS========================= ;[aor] ; SYNOPSIS: The configuration for a location of an endpoint ;contact= ; Permanent contacts assigned to AoR (default: "") ;default_expiration=3600 ; Default expiration time in seconds for ; contacts that are dynamically bound to an AoR ; (default: "3600") ;mailboxes= ; Allow subscriptions for the specified mailbox(es) ; This option applies when an external entity subscribes to an AoR ; for Message Waiting Indications. (default: "") ;voicemail_extension= ; The voicemail extension to send in the NOTIFY Message-Account header ; (default: global/default_voicemail_extension) ;maximum_expiration=7200 ; Maximum time to keep an AoR (default: "7200") ;max_contacts=0 ; Maximum number of contacts that can bind to an AoR (default: ; "0") ;minimum_expiration=60 ; Minimum keep alive time for an AoR (default: "60") ;remove_existing=no ; Allow a registration to succeed by displacing any existing ; contacts that now exceed the max_contacts count. Any ; removed contacts are the next to expire. The behaviour is ; beneficial when rewrite_contact is enabled and max_contacts ; is greater than one. The removed contact is likely the old ; contact created by rewrite_contact that the device is ; refreshing. ; (default: "no") ;remove_unavailable=no ; If remove_existing is disabled, will allow a registration ; to succeed by removing only unavailable contacts when ; max_contacts is exceeded. This will reject a registration ; that exceeds max_contacts if no unavailable contacts are ; present to remove. If remove_existing is enabled, will ; prioritize removal of unavailable contacts before removing ; expiring soonest. This tames the behavior of remove_existing ; to only remove an available contact if an unavailable one is ; not present. ; (default: "no") ;type= ; Must be of type aor (default: "") ;qualify_frequency=0 ; Interval at which to qualify an AoR via OPTIONS requests. ; (default: "0") ;qualify_timeout=3.0 ; Qualify timeout in fractional seconds (default: "3.0") ;authenticate_qualify=no ; Authenticates a qualify request if needed ; (default: "no") ;outbound_proxy= ; Proxy through which to send OPTIONS requests, a full SIP URI ; must be provided (default: "") ;==========================SYSTEM SECTION OPTIONS========================= ;[system] ; SYNOPSIS: Options that apply to the SIP stack as well as other system-wide settings ;timer_t1=500 ; Set transaction timer T1 value milliseconds (default: "500") ;timer_b=32000 ; Set transaction timer B value milliseconds (default: "32000") ;compact_headers=no ; Use the short forms of common SIP header names ; (default: "no") ;threadpool_initial_size=0 ; Initial number of threads in the res_pjsip ; threadpool (default: "0") ;threadpool_auto_increment=5 ; The amount by which the number of threads is ; incremented when necessary (default: "5") ;threadpool_idle_timeout=60 ; Number of seconds before an idle thread ; should be disposed of (default: "60") ;threadpool_max_size=0 ; Maximum number of threads in the res_pjsip threadpool ; A value of 0 indicates no maximum (default: "0") ;disable_tcp_switch=yes ; Disable automatic switching from UDP to TCP transports ; if outgoing request is too large. ; See RFC 3261 section 18.1.1. ; Disabling this option has been known to cause interoperability ; issues, so disable at your own risk. ; (default: "yes") ;follow_early_media_fork = ; On outgoing calls, if the UAS responds with ; different SDP attributes on subsequent 18X or 2XX ; responses (such as a port update) AND the To tag ; on the subsequent response is different than that ; on the previous one, follow it. This usually ; happens when the INVITE is forked to multiple UASs ; and more than 1 sends an SDP answer. ; This option must also be enabled on endpoints that ; require this functionality. ; (default: yes) ;accept_multiple_sdp_answers = ; On outgoing calls, if the UAS responds with ; different SDP attributes on non-100rel 18X or 2XX ; responses (such as a port update) AND the To tag on ; the subsequent response is the same as that on the ; previous one, process it. This can happen when the ; UAS needs to change ports for some reason such as ; using a separate port for custom ringback. ; This option must also be enabled on endpoints that ; require this functionality. ; (default: no) ;disable_rport=no ; Disable the use of "rport" in outgoing requests. ;type= ; Must be of type system (default: "") ;==========================GLOBAL SECTION OPTIONS========================= ;[global] ; SYNOPSIS: Options that apply globally to all SIP communications ;max_forwards=70 ; Value used in Max Forwards header for SIP requests ; (default: "70") ;type= ; Must be of type global (default: "") ;user_agent=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 user_agent string. ;default_outbound_endpoint=default_outbound_endpoint ; Endpoint to use when ; sending an outbound ; request to a URI ; without a specified ; endpoint (default: "d ; efault_outbound_endpo ; int") ;debug=no ; Enable/Disable SIP debug logging. Valid options include yes|no ; or a host address (default: "no") ;keep_alive_interval=90 ; The interval (in seconds) at which to send (double CRLF) ; keep-alives on all active connection-oriented transports; ; for connection-less like UDP see qualify_frequency. ; (default: "90") ;contact_expiration_check_interval=30 ; The interval (in seconds) to check for expired contacts. ;disable_multi_domain=no ; Disable Multi Domain support. ; If disabled it can improve realtime performace by reducing ; number of database requsts ; (default: "no") ;endpoint_identifier_order=ip,username,anonymous ; The order by which endpoint identifiers are given priority. ; Currently, "ip", "header", "username", "auth_username" and "anonymous" ; are valid identifiers as registered by the res_pjsip_endpoint_identifier_* ; modules. Some modules like res_pjsip_endpoint_identifier_user register ; more than one identifier. Use the CLI command "pjsip show identifiers" ; to see the identifiers currently available. ; (default: ip,username,anonymous) ;max_initial_qualify_time=4 ; The maximum amount of time (in seconds) from ; startup that qualifies should be attempted on all ; contacts. If greater than the qualify_frequency ; for an aor, qualify_frequency will be used instead. ;regcontext=sipregistrations ; If regcontext is specified, Asterisk will dynamically ; create and destroy a NoOp priority 1 extension for a ; given endpoint who registers or unregisters with us. ; The extension added is the name of the endpoint. ;default_voicemail_extension=asterisk ; The voicemail extension to send in the NOTIFY Message-Account header ; if not set on endpoint or aor. ; (default: "") ; ; The following unidentified_request options are only used when "auth_username" ; matching is enabled in "endpoint_identifier_order". ; ;unidentified_request_count=5 ; The number of unidentified requests that can be ; received from a single IP address in ; unidentified_request_period seconds before a security ; event is generated. (default: 5) ;unidentified_request_period=5 ; See above. (default: 5 seconds) ;unidentified_request_prune_interval=30 ; The interval at which unidentified requests ; are check to see if they can be pruned. If they're ; older than twice the unidentified_request_period, ; they're pruned. ; ;default_from_user=asterisk ; When Asterisk generates an outgoing SIP request, the ; From header username will be set to this value if ; there is no better option (such as CallerID or ; endpoint/from_user) to be used ;default_realm=asterisk ; When Asterisk generates a challenge, the digest realm ; will be set to this value if there is no better option ; (such as auth/realm) to be used. ; Asterisk Task Processor Queue Size ; On heavy loaded system with DB storage you may need to increase ; taskprocessor queue. ; If the taskprocessor queue size reached high water level, ; the alert is triggered. ; If the alert is set the pjsip distibutor stops processing incoming ; requests until the alert is cleared. ; The alert is cleared when taskprocessor queue size drops to the ; low water clear level. ; The next options set taskprocessor queue levels for MWI. ;mwi_tps_queue_high=500 ; Taskprocessor high water alert trigger level. ;mwi_tps_queue_low=450 ; Taskprocessor low water clear alert level. ; The default is -1 for 90% of high water level. ; Unsolicited MWI ; If there are endpoints configured with unsolicited MWI ; then res_pjsip_mwi module tries to send MWI to all endpoints on startup. ;mwi_disable_initial_unsolicited=no ; Disable sending unsolicited mwi to all endpoints on startup. ; If disabled then unsolicited mwi will start processing ; on the endpoint's next contact update. ;ignore_uri_user_options=no ; Enable/Disable ignoring SIP URI user field options. ; If you have this option enabled and there are semicolons ; in the user field of a SIP URI then the field is truncated ; at the first semicolon. This effectively makes the semicolon ; a non-usable character for PJSIP endpoint names, extensions, ; and AORs. This can be useful for improving compatability with ; an ITSP that likes to use user options for whatever reason. ; Example: ; URI: "sip:1235557890;phone-context=national@x.x.x.x;user=phone" ; The user field is "1235557890;phone-context=national" ; Which becomes this: "1235557890" ; ; Note: The caller-id and redirecting number strings obtained ; from incoming SIP URI user fields are always truncated at the ; first semicolon. ;send_contact_status_on_update_registration=no ; Enable sending AMI ContactStatus ; event when a device refreshes its registration ; (default: "no") ;taskprocessor_overload_trigger=global ; Set the trigger the distributor will use to detect ; taskprocessor overloads. When triggered, the distributor ; will not accept any new requests until the overload has ; cleared. ; "global": (default) Any taskprocessor overload will trigger. ; "pjsip_only": Only pjsip taskprocessor overloads will trigger. ; "none": No overload detection will be performed. ; WARNING: The "none" and "pjsip_only" options should be used ; with extreme caution and only to mitigate specific issues. ; Under certain conditions they could make things worse. ;norefersub=yes ; Enable sending norefersub option tag in Supported header to advertise ; that the User Agent is capable of accepting a REFER request with ; creating an implicit subscription (see RFC 4488). ; (default: "yes") ;all_codecs_on_empty_reinvite=yes ; On reception of a re-INVITE without SDP Asterisk will send an SDP ; offer in the 200 OK response containing all configured codecs on the ; endpoint, instead of simply those that have already been negotiated. ; RFC 3261 specifies this as a SHOULD requirement. ; (default: "no") ;allow_sending_180_after_183=yes ; Allow Asterisk to send 180 Ringing to an endpoint ; after 183 Session Progress has been send. ; If disabled Asterisk will instead send only a ; 183 Session Progress to the endpoint. ; (default: "no") ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_acl ;==========================ACL SECTION OPTIONS========================= ;[acl] ; SYNOPSIS: Access Control List ;acl= ; List of IP ACL section names in acl conf (default: "") ;contact_acl= ; List of Contact ACL section names in acl conf (default: "") ;contact_deny= ; List of Contact header addresses to deny (default: "") ;contact_permit= ; List of Contact header addresses to permit (default: ; "") ;deny= ; List of IP addresses to deny access from (default: "") ;permit= ; List of IP addresses to permit access from (default: "") ;type= ; Must be of type acl (default: "") ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_outbound_registration ;==========================REGISTRATION SECTION OPTIONS========================= ;[registration] ; SYNOPSIS: The configuration for outbound registration ;auth_rejection_permanent=yes ; Determines whether failed authentication ; challenges are treated as permanent failures ; (default: "yes") ;client_uri= ; Client SIP URI used when attemping outbound registration ; (default: "") ;contact_user= ; Contact User to use in request (default: "") ;expiration=3600 ; Expiration time for registrations in seconds ; (default: "3600") ;max_retries=10 ; Maximum number of registration attempts (default: "10") ;outbound_auth= ; Authentication object to be used for outbound registrations ; (default: "") ;outbound_proxy= ; Proxy through which to send registrations, a full SIP URI ; must be provided (default: "") ;max_random_initial_delay=10 ; Maximum random delay for initial registrations (default: 10) ; Generally it is a good idea to space out registrations ; to not overload the system. If you have a small number ; of registrations and need them to register more quickly, ; you can reduce this to a lower value. ;retry_interval=60 ; Interval in seconds between retries if outbound ; registration is unsuccessful (default: "60") ;forbidden_retry_interval=0 ; Interval used when receiving a 403 Forbidden ; response (default: "0") ;fatal_retry_interval=0 ; Interval used when receiving a fatal response. ; (default: "0") A fatal response is any permanent ; failure (non-temporary 4xx, 5xx, 6xx) response ; received from the registrar. NOTE - if also set ; the 'forbidden_retry_interval' takes precedence ; over this one when a 403 is received. Also, if ; 'auth_rejection_permanent' equals 'yes' a 401 and ; 407 become subject to this retry interval. ;server_uri= ; SIP URI of the server to register against (default: "") ;transport= ; Transport used for outbound authentication (default: "") ;line= ; When enabled this option will cause a 'line' parameter to be ; added to the Contact header placed into the outgoing ; registration request. If the remote server sends a call ; this line parameter will be used to establish a relationship ; to the outbound registration, ultimately causing the ; configured endpoint to be used (default: "no") ;endpoint= ; When line support is enabled this configured endpoint name ; is used for incoming calls that are related to the outbound ; registration (default: "") ;type= ; Must be of type registration (default: "") ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_endpoint_identifier_ip ;==========================IDENTIFY SECTION OPTIONS========================= ;[identify] ; SYNOPSIS: Identifies endpoints via some criteria. ; ; NOTE: If multiple matching criteria are provided then an inbound request will ; be matched to the endpoint if it matches ANY of the criteria. ;endpoint= ; Name of endpoint identified (default: "") ;srv_lookups=yes ; Perform SRV lookups for provided hostnames. (default: yes) ;match= ; Comma separated list of IP addresses, networks, or hostnames to match ; against (default: "") ;match_header= ; SIP header with specified value to match against (default: "") ;type= ; Must be of type identify (default: "") ;========================PHONEPROV_USER SECTION OPTIONS======================= ;[phoneprov] ; SYNOPSIS: Contains variables for autoprovisioning each user ;endpoint= ; The endpoint from which to gather username, secret, etc. (default: "") ;PROFILE= ; The name of a profile configured in phoneprov.conf (default: "") ;MAC= ; The mac address for this user (default: "") ;OTHERVAR= ; Any other name value pair to be used in templates (default: "") ; Common variables include LINE, LINEKEYS, etc. ; See phoneprov.conf.sample for others. ;type= ; Must be of type phoneprov (default: "") ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_outbound_publish ;======================OUTBOUND_PUBLISH SECTION OPTIONS===================== ; See https://wiki.asterisk.org/wiki/display/AST/Publishing+Extension+State ; for more information. ;[outbound-publish] ;type=outbound-publish ; Must be of type 'outbound-publish'. ;expiration=3600 ; Expiration time for publications in seconds ;outbound_auth= ; Authentication object(s) to be used for outbound ; publishes. ; This is a comma-delimited list of auth sections ; defined in pjsip.conf used to respond to outbound ; authentication challenges. ; Using the same auth section for inbound and ; outbound authentication is not recommended. There ; is a difference in meaning for an empty realm ; setting between inbound and outbound authentication ; uses. See the auth realm description for details. ;outbound_proxy= ; SIP URI of the outbound proxy used to send ; publishes ;server_uri= ; SIP URI of the server and entity to publish to. ; This is the URI at which to find the entity and ; server to send the outbound PUBLISH to. ; This URI is used as the request URI of the outbound ; PUBLISH request from Asterisk. ;from_uri= ; SIP URI to use in the From header. ; This is the URI that will be placed into the From ; header of outgoing PUBLISH messages. If no URI is ; specified then the URI provided in server_uri will ; be used. ;to_uri= ; SIP URI to use in the To header. ; This is the URI that will be placed into the To ; header of outgoing PUBLISH messages. If no URI is ; specified then the URI provided in server_uri will ; be used. ;event= ; Event type of the PUBLISH. ;max_auth_attempts= ; Maximum number of authentication attempts before ; stopping the pub. ;transport= ; Transport used for outbound publish. ; A transport configured in pjsip.conf. As with other ; res_pjsip modules, this will use the first ; available transport of the appropriate type if ; unconfigured. ;multi_user=no ; Enable multi-user support (Asterisk 14+ only) ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_pubsub ;=============================RESOURCE-LIST=================================== ; See https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158 ; for more information. ;[resource_list] ;type=resource_list ; Must be of type 'resource_list'. ;event= ; The SIP event package that the list resource. ; belongs to. The SIP event package describes the ; types of resources that Asterisk reports the state ; of. ;list_item= ; The name of a resource to report state on. ; In general Asterisk looks up list items in the ; following way: ; 1. Check if the list item refers to another ; configured resource list. ; 2. Pass the name of the resource off to ; event-package-specific handlers to find the ; specified resource. ; The second part means that the way the list item ; is specified depends on what type of list this is. ; For instance, if you have the event set to ; presence, then list items should be in the form of ; dialplan_extension@dialplan_context. For ; message-summary, mailbox names should be listed. ;full_state=no ; Indicates if the entire list's state should be ; sent out. ; If this option is enabled, and a resource changes ; state, then Asterisk will construct a notification ; that contains the state of all resources in the ; list. If the option is disabled, Asterisk will ; construct a notification that only contains the ; states of resources that have changed. ; NOTE: Even with this option disabled, there are ; certain situations where Asterisk is forced to send ; a notification with the states of all resources in ; the list. When a subscriber renews or terminates ; its subscription to the list, Asterisk MUST send ; a full state notification. ;notification_batch_interval=0 ; Time Asterisk should wait, in milliseconds, ; before sending notifications. ;resource_display_name=no ; Indicates whether display name of resource ; or the resource name being reported. ; If this option is enabled, the Display Name ; will be reported as resource name. ; If the event set to presence or dialog, ; the HINT name will be set as the Display Name. ; For example: ; exten => 1234,hint,PJSIP/user1234(Alice) ; If enabled the resource name will be 'Alice'. ; If disabled the resource name will be '1234'. ; The message-summary is not supported yet. ;==========================INBOUND_PUBLICATION================================ ; See https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP ; for more information. ;[inbound-publication] ;type= ; Must be of type 'inbound-publication'. ;endpoint= ; Optional name of an endpoint that is only allowed ; to publish to this resource. ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_publish_asterisk ;==========================ASTERISK_PUBLICATION=============================== ; See https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP ; for more information. ;[asterisk-publication] ;type=asterisk-publication ; Must be of type 'asterisk-publication'. ;devicestate_publish= ; Optional name of a publish item that can be used ; to publish a req. ;mailboxstate_publish= ; Optional name of a publish item that can be used ; to publish a req. ;device_state=no ; Whether we should permit incoming device state ; events. ;device_state_filter= ; Optional regular expression used to filter what ; devices we accept events for. ;mailbox_state=no ; Whether we should permit incoming mailbox state ; events. ;mailbox_state_filter= ; Optional regular expression used to filter what ; mailboxes we accept events for. ;================================TEL URIs===================================== ; ; Asterisk has TEL URI support, but with limited scope. Support is only for ; TEL URIs present in traffic from a remote party. Asterisk does not generate ; any TEL URIs of its own. ; ; Currently, the allowed request types are INVITE, ACK, BYE, and CANCEL. Any ; other request type that contains a TEL URI will behave as it did before. ; TEL URIs are allowed in the request, From, and To headers. ; ; You can match a TEL URI From header by IP, header, or auth_username.