; ; SIP Configuration example for Asterisk ; ; Note: Please read the security documentation for Asterisk in order to ; understand the risks of installing Asterisk with the sample ; configuration. If your Asterisk is installed on a public ; IP address connected to the Internet, you will want to learn ; about the various security settings BEFORE you start ; Asterisk. ; ; Especially note the following settings: ; - allowguest (default enabled) ; - permit/deny/acl - IP address filters ; - contactpermit/contactdeny/contactacl - IP address filters for registrations ; - context - Which set of services you offer various users ; ; SIP dial strings ;----------------------------------------------------------- ; In the dialplan (extensions.conf) you can use several ; syntaxes for dialing SIP devices. ; SIP/devicename ; SIP/username@domain (SIP uri) ; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] ; SIP/devicename/extension ; SIP/devicename/extension/IPorHost ; SIP/username@domain//IPorHost ; ; ; Devicename ; devicename is defined as a peer in a section below. ; ; username@domain ; Call any SIP user on the Internet ; (Don't forget to enable DNS SRV records if you want to use this) ; ; devicename/extension ; If you define a SIP proxy as a peer below, you may call ; SIP/proxyhostname/user or SIP/user@proxyhostname ; where the proxyhostname is defined in a section below ; This syntax also works with ATA's with FXO ports ; ; SIP/username[:password[:md5secret[:authname]]]@host[:port] ; This form allows you to specify password or md5secret and authname ; without altering any authentication data in config. ; Examples: ; ; SIP/*98@mysipproxy ; SIP/sales:topsecret::email@example.com:5062 ; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:firstname.lastname@example.org ; ; 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@email@example.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) ; 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) 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 ; ; 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. ;prematuremedia=no ; Some ISDN links send empty media frames before ; the call is in ringing or progress state. The SIP ; channel will then send 183 indicating early media ; which will be empty - thus users get no ring signal. ; Setting this to "yes" will stop any media before we have ; call progress (meaning the SIP channel will not send 183 Session ; Progress for early media). Default is "yes". Also make sure that ; the SIP peer is configured with progressinband=never. ; ; In order for "noanswer" applications to work, you need to run ; the progress() application in the priority before the app. ;progressinband=never ; If we should generate in-band ringing always ; use 'never' to never use in-band signalling, even in cases ; where some buggy devices might not render it ; Valid values: yes, no, never Default: never ;useragent=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] ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s) ; Videosupport and maxcallbitrate is settable ; for peers and users as well ;callevents=no ; generate manager events when sip ua ; performs events (e.g. hold) ;authfailureevents=no ; generate manager "peerstatus" events when peer can't ; authenticate with Asterisk. Peerstatus will be "rejected". ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected, ; for any reason, always reject with an identical response ; equivalent to valid username and invalid password/hash ; instead of letting the requester know whether there was ; a matching user or peer for their request. This reduces ; the ability of an attacker to scan for valid SIP usernames. ; This option is set to "yes" by default. ;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like ; INVITE requests are. By default this option is disabled. ;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 ;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 ;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 ;autocreatepeers=no ; Allow any not exsplicitly defined here UAC 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. ; When enabled by setting to "yes", the autocreated peers are ; pruned immediately when the "sip reload" command is issued ; through CLI. When enabled by setting to "persist", the auto- ; created peers survive the "sip reload" command. ; ;------------------------ TLS settings ------------------------------------------------------------ ;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem format only) to use for TLS connections ; default is to look for "asterisk.pem" in current directory ;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections. ; If no tlsprivatekey is specified, tlscertfile is searched for ; for both public and private key. ;tlscafile=</path/to/certificate> ; If the server your connecting to uses a self signed certificate ; you should have their certificate installed here so the code can ; verify the authenticity of their certificate. ;tlscapath=</path/to/ca/dir> ; A directory full of CA certificates. The files must be named with ; the CA subject name hash value. ; (see man SSL_CTX_load_verify_locations for more info) ;tlsdontverifyserver=[yes|no] ; If set to yes, don't verify the servers certificate when acting as ; a client. If you don't have the server's CA certificate you can ; set this and it will connect without requiring tlscafile to be set. ; Default is no. ;tlscipher=<SSL cipher string> ; A string specifying which SSL ciphers to use or not use ; A list of valid SSL cipher strings can be found at: ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS ; ;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2. ; Specify protocol for outbound client connections. ; If left unspecified, the default is sslv2. ; ;--------------------------- SIP timers ---------------------------------------------------- ; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. ; ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;timert1=500 ; Default T1 timer ; Defaults to 500 ms or the measured round-trip ; time to a peer (qualify=yes). ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 ;--------------------------- RTP timers ---------------------------------------------------- ; These timers are currently used for both audio and video streams. The RTP timeouts ; are only applied to the audio channel. ; The settings are settable in the global section as well as per device ; ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity ; on the audio channel ; when we're not on hold. This is to be able to hangup ; a call in the case of a phone disappearing from the net, ; like a powerloss or grandma tripping over a cable. ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity ; on the audio channel ; when we're on hold (must be > rtptimeout) ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open ; (default is off - zero) ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------ ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions. ; This mechanism can detect and reclaim SIP channels that do not terminate through normal ; signaling procedures. Session-Timers can be configured globally or at a user/peer level. ; The operation of Session-Timers is driven by the following configuration parameters: ; ; * session-timers - Session-Timers feature operates in the following three modes: ; originate : Request and run session-timers always ; accept : Run session-timers only when requested by other UA ; refuse : Do not run session timers in any case ; The default mode of operation is 'accept'. ; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs. ; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs. ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'. ; uac - Default to the caller initially refreshing when possible ; uas - Default to the callee initially refreshing when possible ; ; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other ; endpoint's preference for who will handle refreshes. Asterisk will never override the ; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint ; fighting over who sends the refreshes. This holds true for the initiation of session ; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or ; whether Asterisk is currently the refresher or not. ; ;session-timers=originate ;session-expires=600 ;session-minse=90 ;session-refresher=uac ; ;--------------------------- SIP DEBUGGING --------------------------------------------------- ;sipdebug = yes ; Turn on SIP debugging by default, from ; the moment the channel loads this configuration ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;dumphistory=yes ; Dump SIP history at end of SIP dialogue ; SIP history is output to the DEBUG logging channel ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ---------------------------- ; You can subscribe to the status of extensions with a "hint" priority ; (See extensions.conf.sample for examples) ; chan_sip support two major formats for notifications: dialog-info and SIMPLE ; ; You will get more detailed reports (busy etc) if you have a call counter enabled ; for a device. ; ; If you set the busylevel, we will indicate busy when we have a number of calls that ; matches the busylevel treshold. ; ; For queues, you will need this level of detail in status reporting, regardless ; if you use SIP subscriptions. Queues and manager use the same internal interface ; for reading status information. ; ; Note: Subscriptions does not work if you have a realtime dialplan and use the ; realtime switch. ; ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes) ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests ; Useful to limit subscriptions to local extensions ; Settable per peer/user also ;notifyringing = no ; Control whether subscriptions already INUSE get sent ; RINGING when another call is sent (default: yes) ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no) ; Turning on notifyringing and notifyhold will add a lot ; more database transactions if you are using realtime. ;notifycid = yes ; Control whether caller ID information is sent along with ; dialog-info+xml notifications (supported by snom phones). ; Note that this feature will only work properly when the ; incoming call is using the same extension and context that ; is being used as the hint for the called extension. This means ; that it won't work when using subscribecontext for your sip ; user or peer (if subscribecontext is different than context). ; This is also limited to a single caller, meaning that if an ; extension is ringing because multiple calls are incoming, ; only one will be used as the source of caller ID. Specify ; 'ignore-context' to ignore the called context when looking ; for the caller's channel. The default value is 'no.' Setting ; notifycid to 'ignore-context' also causes call-pickups attempted ; via SNOM's NOTIFY mechanism to set the context for the call pickup ; to PICKUPMARK. ;callcounter = yes ; Enable call counters on devices. This can be set per ; device too. ;----------------------------------------- T.38 FAX SUPPORT ---------------------------------- ; ; This setting is available in the [general] section as well as in device configurations. ; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off. ; ; t38pt_udptl = yes ; Enables T.38 with FEC error correction. ; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction. ; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction. ; t38pt_udptl = yes,none ; Enables T.38 with no error correction. ; ; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that ; is based on an incorrect interpretation of the T.38 recommendation, and results in failures ; because Asterisk does not believe it can send T.38 packets of a reasonable size to that ; endpoint (Cisco media gateways are one example of this situation). In these cases, during a ; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL ; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you ; can set an override (globally, or on a per-device basis) to make Asterisk ignore the ; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead. ; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option, ; like this: ; ; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides ; ; the other endpoint's provided value to assume we can ; ; send 400 byte T.38 FAX packets to it. ; ; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists) ; based one or more events being detected. The events that can be detected are an incoming ; CNG tone or an incoming T.38 re-INVITE request. ; ; faxdetect = yes ; Default 'no', 'yes' enables both CNG and T.38 detection ; faxdetect = cng ; Enables only CNG detection ; faxdetect = t38 ; Enables only T.38 detection ; ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------ ; Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: ; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] ; ; ; ; domain is either ; - domain in DNS ; - host name in DNS ; - the name of a peer defined below or in realtime ; The domain is where you register your username, so your SIP uri you are registering to ; is username@domain ; ; If no extension is given, the 's' extension is used. The extension needs to ; be defined in extensions.conf to be able to accept calls from this SIP proxy ; (provider). ; ; A similar effect can be achieved by adding a "callbackextension" option in a peer section. ; this is equivalent to having the following line in the general section: ; ; register => username:secret@host/callbackextension ; ; and more readable because you don't have to write the parameters in two places ; (note that the "port" is ignored - this is a bug that should be fixed). ; ; Note that a register= line doesn't mean that we will match the incoming call in any ; other way than described above. If you want to control where the call enters your ; dialplan, which context, you want to define a peer with the hostname of the provider's ; server. If the provider has multiple servers to place calls to your system, you need ; a peer for each server. ; ; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may ; contain a port number. Since the logical separator between a host and port number is a ; ':' character, and this character is already used to separate between the optional "secret" ; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish ; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if ; they are blank. See the third example below for an illustration. ; ; ; Examples: ; ;register => 1234:firstname.lastname@example.org ; ; 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:email@example.com ; ; The 'transport' part defaults to 'udp' but may also be 'tcp', 'tls', 'ws', or 'wss'. ; Using 'udp://' explicitly is also useful in case the username part ; contains a '/' ('user/name'). ;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 ;----------------------------------------- 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:firstname.lastname@example.org/1234 ;mwi => 1234:email@example.com:6969/1234 ;mwi => 1234:password:firstname.lastname@example.org/1234 ;mwi => 1234:password:email@example.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 = 18.104.22.168 ; use this address. ; externaddr = 22.214.171.124:9900 ; use this address and port. ; externaddr = mynat.my.org:12600 ; Public address of my nat box. ; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT. ; ; externtcpport will default to the externaddr or externhost port if either one is set. ; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT. ; ; externtlsport port will default to the RFC designated port of 5061. ; ; b. "externhost = hostname[:port]" is similar to "externaddr" except ; that the hostname is looked up every "externrefresh" seconds ; (default 10s). This can be useful when your NAT device lets you choose ; the port mapping, but the IP address is dynamic. ; Beware, you might suffer from service disruption when the name server ; resolution fails. Examples: ; ; externhost=foo.dyndns.net ; refreshed periodically ; externrefresh=180 ; change the refresh interval ; ; Note that at the moment all these mechanism work only for the SIP socket. ; The IP address discovered with externaddr/externhost is reused for ; media sessions as well, but the port numbers are not remapped so you ; may still experience problems. ; ; NOTE 1: in some cases, NAT boxes will use different port numbers in ; the internal<->external mapping. In these cases, the "externaddr" and ; "externhost" might not help you configure addresses properly. ; ; NOTE 2: when using "externaddr" or "externhost", the address part is ; also used as the external address for media sessions. Thus, the port ; information in the SDP may be wrong! ; ; In addition to the above, Asterisk has an additional "nat" parameter to ; address NAT-related issues in incoming SIP or media sessions. ; In particular, depending on the 'nat= ' settings described below, Asterisk ; may override the address/port information specified in the SIP/SDP messages, ; and use the information (sender address) supplied by the network stack instead. ; However, this is only useful if the external traffic can reach us. ; The following settings are allowed (both globally and in individual sections): ; ; nat = no ; Do no special NAT handling other than RFC3581 ; nat = force_rport ; Pretend there was an rport parameter even if there wasn't ; nat = comedia ; Send media to the port Asterisk received it from regardless ; ; of where the SDP says to send it. ; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default) ; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT ; ; The nat settings can be combined. For example, to set both force_rport and comedia ; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no', ; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings ; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then ; the non-auto option will be ignored. ; ; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send ; SIP responses to it via the source IP and port from which the request originated ; instead of the address/port listed in the top-most Via header. This is useful if a ; client knows that it is behind a NAT and therefore cannot guess from what address/port ; its request will be sent. Asterisk will always honor the 'rport' parameter if it is ; sent. The force_rport setting causes Asterisk to always send responses back to the ; address/port from which it received requests; even if the other side doesn't support ; adding the 'rport' parameter. ; ; 'comedia RTP handling' refers to the technique of sending RTP to the port that the ; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is ; only partially related to RFC 4145 which was referred to as COMEDIA while it was in ; draft form. This method is used to accomodate endpoints that may be located behind ; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to ; for their media streams is not the actual address/port that will be used on the nearer ; side of the NAT. ; ; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from ; the nat setting in a peer definition, then the peer username will be discoverable ; by outside parties as Asterisk will respond to different ports for defined and ; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE ; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the ; other, then valid peers with settings differing from those in the general section will ; be discoverable. ; ; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by ; RFC 4961; Asterisk will always send RTP packets from the same port number it expects ; to receive them on. ; ; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using ; the media_address configuration option. This is only applicable to the general section and ; can not be set per-user or per-peer. ; ; media_address = 172.16.42.1 ; ; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the ; perceived external network address has changed. When the stun_monitor is installed and ; configured, chan_sip will renew all outbound registrations when the monitor detects any sort ; of network change has occurred. By default this option is enabled, but only takes effect once ; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not ; generate all outbound registrations on a network change, use the option below to disable ; this feature. ; ; subscribe_network_change_event = yes ; on by default ; ; ICE/STUN/TURN usage can be disabled 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. ; ; icesupport = no ;----------------------------------- MEDIA HANDLING -------------------------------- ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's ; no reason for Asterisk to stay in the media path, the media will be redirected. ; This does not really work well in the case where Asterisk is outside and the ; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat. ; ;directmedia=yes ; Asterisk by default tries to redirect the ; RTP media stream to go directly from ; the caller to the callee. Some devices do not ; support this (especially if one of them is behind a NAT). ; The default setting is YES. If you have all clients ; behind a NAT, or for some other reason want Asterisk to ; stay in the audio path, you may want to turn this off. ; This setting also affect direct RTP ; at call setup (a new feature in 1.4 - setting up the ; call directly between the endpoints instead of sending ; a re-INVITE). ; Additionally this option does not disable all reINVITE operations. ; It only controls Asterisk generating reINVITEs for the specific ; purpose of setting up a direct media path. If a reINVITE is ; needed to switch a media stream to inactive (when placed on ; hold) or to T.38, it will still be done, regardless of this ; setting. Note that direct T.38 is not supported. ;directmedia=nonat ; An additional option is to allow media path redirection ; (reinvite) but only when the peer where the media is being ; sent is known to not be behind a NAT (as the RTP core can ; determine it based on the apparent IP address the media ; arrives from). ;directmedia=update ; Yet a third option... use UPDATE for media path redirection, ; instead of INVITE. This can be combined with 'nonat', as ; 'directmedia=update,nonat'. It implies 'yes'. ;directmedia=outgoing ; When sending directmedia reinvites, do not send an immediate ; reinvite on an incoming call leg. This option is useful when ; peered with another SIP user agent that is known to send ; immediate direct media reinvites upon call establishment. Setting ; the option in this situation helps to prevent potential glares. ; Setting this option implies 'yes'. ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up ; the call directly with media peer-2-peer without re-invites. ; Will not work for video and cases where the callee sends ; RTP payloads and fmtp headers in the 200 OK that does not match the ; callers INVITE. This will also fail if directmedia is enabled when ; the device is actually behind NAT. ;directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict ;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other ; (There is no default setting, this is just an example) ; Use this if some of your phones are on IP addresses that ; can not reach each other directly. This way you can force ; RTP to always flow through asterisk in such cases. ;directmediaacl=acl_example ; Use named ACLs defined in acl.conf ;ignoresdpversion=yes ; By default, Asterisk will honor the session version ; number in SDP packets and will only modify the SDP ; session if the version number changes. This option will ; force asterisk to ignore the SDP session version number ; and treat all SDP data as new data. This is required ; for devices that send us non standard SDP packets ; (observed with Microsoft OCS). By default this option is ; off. ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=) ; Like the useragent parameter, the default user agent string ; also contains the Asterisk version. ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=) ; This field MUST NOT contain spaces ;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media) ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if ; the peer does not support SRTP. Defaults to no. ;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80 ; ;avpf=yes ; Enable inter-operability with media streams using the AVPF RTP profile. ; This will cause all offers and answers to use AVPF (or SAVPF). This ; option may be specified at the global or peer scope. ;----------------------------------------- REALTIME SUPPORT ------------------------ ; For additional information on ARA, the Asterisk Realtime Architecture, ; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration ; ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) ;rtsavesysname=yes ; Save systemname in realtime database at registration ; Default= no ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) ; If set to yes, when a SIP UA registers successfully, the ip address, ; the origination port, the registration period, and the username of ; the UA will be set to database via realtime. ; If not present, defaults to 'yes'. Note: realtime peers will ; probably not function across reloads in the way that you expect, if ; you turn this option off. ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered? (yes|no|<seconds>) ; If set to yes, when the registration expires, the friend will ; vanish from the configuration until requested again. If set ; to an integer, friends expire within this number of seconds ; instead of the registration interval. ;ignoreregexpire=yes ; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the Asterisk database ; if you attempt to place a call to the peer, the existing information ; will be used in spite of it having expired ; ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether ; it has expired or not; if it expires while the realtime peer ; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------ ; Incoming INVITE and REFER messages can be matched against a list of 'allowed' ; domains, each of which can direct the call to a specific context if desired. ; By default, all domains are accepted and sent to the default context or the ; context associated with the user/peer placing the call. ; REGISTER to non-local domains will be automatically denied if a domain ; list is configured. ; ; Domains can be specified using: ; domain=<domain>[,<context>] ; Examples: ; domain=myasterisk.dom ; domain=customer.com,customer-context ; ; In addition, all the 'default' domains associated with a server should be ; added if incoming request filtering is desired. ; autodomain=yes ; ; To disallow requests for domains not serviced by this server: ; allowexternaldomains=no ;domain=mydomain.tld,mydomain-incoming ; Add domain and configure incoming context ; for external calls to this domain ;domain=126.96.36.199 ; Add IP address as local domain ; You can have several "domain" settings ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains ; Default is yes ;autodomain=yes ; Turn this on to have Asterisk add local host ; name and local IP to domain list. ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to ; non-peers, use your primary domain "identity" ; for From: headers instead of just your IP ; address. This is to be polite and ; it may be a mandatory requirement for some ; destinations which do not have a prior ; account relationship with your server. ;------------------------------ Advice of Charge CONFIGURATION -------------------------- ; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and ; AOC-E to snom endpoints. This option can be used both in the ; peer and global scope. The default for this option is off. ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; SIP channel. Defaults to "no". An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The SIP channel can accept jitter, ; thus a jitterbuffer on the receive SIP side will be used only ; if it is forced and enabled. ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP ; channel. Defaults to "no". ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is ; resynchronized. Useful to improve the quality of the voice, with ; big jumps in/broken timestamps, usually sent from exotic devices ; and programs. Defaults to 1000. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP ; channel. Two implementations are currently available - "fixed" ; (with size always equals to jbmaxsize) and "adaptive" (with ; variable size, actually the new jb of IAX2). Defaults to fixed. ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set. ; The option represents the number of milliseconds by which the new jitter buffer ; will pad its size. the default is 40, so without modification, the new ; jitter buffer will set its size to the jitter value plus 40 milliseconds. ; increasing this value may help if your network normally has low jitter, ; but occasionally has spikes. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ;----------------------------------------------------------------------------------- [authentication] ; Global credentials for outbound calls, i.e. when a proxy challenges your ; Asterisk server for authentication. These credentials override ; any credentials in peer/register definition if realm is matched. ; ; This way, Asterisk can authenticate for outbound calls to other ; realms. We match realm on the proxy challenge and pick an set of ; credentials from this list ; Syntax: ; auth = <user>:<secret>@<realm> ; auth = <user>#<md5secret>@<realm> ; Example: ;auth=mark:firstname.lastname@example.org ; ; 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 ; insecure ; trustrpid ; progressinband ; promiscredir ; useclientcode ; accountcode ; setvar ; callerid ; amaflags ; callcounter ; busylevel ; allowoverlap ; allowsubscribe ; allowtransfer ; ignoresdpversion ; subscribecontext ; template ; videosupport ; maxcallbitrate ; rfc2833compensate ; 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 ; registertrying ; timert1 ; timerb ; qualifyfreq ; t38pt_usertpsource ; contactpermit ; Limit what a host may register as (a neat trick ; contactdeny ; is to register at the same IP as a SIP provider, ; contactacl ; then call oneself, and get redirected to that ; ; same location). ; directmediapermit ; directmediadeny ; directmediaacl ; unsolicited_mailbox ; use_q850_reason ; maxforwards ; encryption ; description ; Used to provide a description of the peer in console output ; dtlsenable ; dtlsverify ; dtlsrekey ; dtlscertfile ; dtlsprivatekey ; dtlscipher ; dtlscafile ; dtlscapath ; dtlssetup ; ;------------------------------------------------------------------------------ ; DTLS-SRTP CONFIGURATION ; ; DTLS-SRTP support is available if the underlying RTP engine in use supports it. ; ; dtlsenable = yes ; Enable or disable DTLS-SRTP support ; dtlsverify = yes ; Verify that the provided peer certificate is valid ; dtlsrekey = 60 ; Interval at which to renegotiate the TLS session and rekey the SRTP session ; ; If this is not set or the value provided is 0 rekeying will be disabled ; dtlscertfile = file ; Path to certificate file to present ; dtlsprivatekey = file ; Path to private key for certificate file ; dtlscipher = <SSL cipher string> ; Cipher to use for TLS negotiation ; ; A list of valid SSL cipher strings can be found at: ; ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS ; dtlscafile = file ; Path to certificate authority certificate ; dtlscapath = path ; Path to a directory containing certificate authority certificates ; dtlssetup = actpass ; Whether we are willing to accept connections, connect to the other party, or both. ; ; Valid options are active (we want to connect to the other party), passive (we want to ; ; accept connections only), and actpass (we will do both). This value will be used in ; ; the outgoing SDP when offering and for incoming SDP offers when the remote party sends ; ; actpass ;[sip_proxy] ; For incoming calls only. Example: FWD (Free World Dialup) ; We match on IP address of the proxy for incoming calls ; since we can not match on username (caller id) ;type=peer ;context=from-fwd ;host=fwd.pulver.com ;[sip_proxy-out] ;type=peer ; we only want to call out, not be called ;remotesecret=guessit ; Our password to their service ;defaultuser=yourusername ; Authentication user for outbound proxies ;fromuser=yourusername ; Many SIP providers require this! ;fromdomain=provider.sip.domain ;host=box.provider.com ;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will ; ; accept both tcp and udp. The default transport type is only used for ; ; outbound messages until a Registration takes place. During the ; ; peer Registration the transport type may change to another supported ; ; type if the peer requests so. ;usereqphone=yes ; This provider requires ";user=phone" on URI ;callcounter=yes ; Enable call counter ;busylevel=2 ; Signal busy at 2 or more calls ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer ;port=80 ; The port number we want to connect to on the remote side ; Also used as "defaultport" in combination with "defaultip" settings ;--- sample definition for a provider ;[provider1] ;type=peer ;host=sip.provider1.com ;fromuser=4015552299 ; how your provider knows you ;remotesecret=youwillneverguessit ; The password we use to authenticate to them ;secret=gissadetdu ; The password they use to contact us ;callbackextension=123 ; Register with this server and require calls coming back to this extension ;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will ; ; accept both tcp and udp. Default is udp. The first transport ; ; listed will always be used for outgoing connections. ;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old ; ; message count will be stored in the configured virtual mailbox. It can be used ; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the ; ; mailbox. ; ; Because you might have a large number of similar sections, it is generally ; convenient to use templates for the common parameters, and add them ; the the various sections. Examples are below, and we can even leave ; the templates uncommented as they will not harm: [basic-options](!) ; a template dtmfmode=rfc2833 context=from-office type=friend [natted-phone](!,basic-options) ; another template inheriting basic-options directmedia=no host=dynamic [public-phone](!,basic-options) ; another template inheriting basic-options directmedia=yes [my-codecs](!) ; a template for my preferred codecs disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw ; Or, more simply: ;allow=!all,ilbc,g729,gsm,g723,ulaw [ulaw-phone](!) ; and another one for ulaw-only disallow=all allow=ulaw ; Again, more simply: ;allow=!all,ulaw ; and finally instantiate a few phones ; ; (natted-phone,my-codecs) ; secret = peekaboo ; (natted-phone,ulaw-phone) ; secret = not_very_secret ; (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 README.callingpres for more information ;[xlite1] ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed ;type=friend ;regexten=1234 ; When they register, create extension 1234 ;callerid="Jane Smith" <5678> ;host=dynamic ; This device needs to register ;directmedia=no ; Typically set to NO if behind NAT ;disallow=all ;allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes ;registertrying=yes ; Send a 100 Trying when the device registers. ;[snom] ;type=friend ; Friends place calls and receive calls ;context=from-sip ; Context for incoming calls from this user ;secret=blah ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions ;language=de ; Use German prompts for this user ;host=dynamic ; This peer register with us ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 ; IP used until peer registers ;mailbox=1234@context,2345 ; 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. ;[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.