diff options
| author | Matthew Jordan <mjordan@digium.com> | 2014-07-17 21:17:28 +0000 |
|---|---|---|
| committer | Matthew Jordan <mjordan@digium.com> | 2014-07-17 21:17:28 +0000 |
| commit | fc0fecb4768d696db3324bcf6dd03325bb4cd513 (patch) | |
| tree | 12615f96e88382b2824d4901f6949571e41ea2e4 /configs/samples/sip.conf.sample | |
| parent | 1ce23d4534994fdd8bfb8ad3b9ca1884194097be (diff) | |
configs: Move sample config files into a subdirectory of configs
This moves all samples configs from configs/ to configs/samples. This allows
for additional sets of sample configuration files to be added in the future.
Review: https://reviewboard.asterisk.org/r/3804/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'configs/samples/sip.conf.sample')
| -rw-r--r-- | configs/samples/sip.conf.sample | 1571 |
1 files changed, 1571 insertions, 0 deletions
diff --git a/configs/samples/sip.conf.sample b/configs/samples/sip.conf.sample new file mode 100644 index 000000000..0d81a6b63 --- /dev/null +++ b/configs/samples/sip.conf.sample @@ -0,0 +1,1571 @@ +; +; SIP Configuration example for Asterisk +; +; Note: Please read the security documentation for Asterisk in order to +; understand the risks of installing Asterisk with the sample +; configuration. If your Asterisk is installed on a public +; IP address connected to the Internet, you will want to learn +; about the various security settings BEFORE you start +; Asterisk. +; +; Especially note the following settings: +; - allowguest (default enabled) +; - permit/deny/acl - IP address filters +; - contactpermit/contactdeny/contactacl - IP address filters for registrations +; - context - Which set of services you offer various users +; +; SIP dial strings +;----------------------------------------------------------- +; In the dialplan (extensions.conf) you can use several +; syntaxes for dialing SIP devices. +; SIP/devicename +; SIP/username@domain (SIP uri) +; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] +; SIP/devicename/extension +; SIP/devicename/extension/IPorHost +; SIP/username@domain//IPorHost +; +; +; Devicename +; devicename is defined as a peer in a section below. +; +; username@domain +; Call any SIP user on the Internet +; (Don't forget to enable DNS SRV records if you want to use this) +; +; devicename/extension +; If you define a SIP proxy as a peer below, you may call +; SIP/proxyhostname/user or SIP/user@proxyhostname +; where the proxyhostname is defined in a section below +; This syntax also works with ATA's with FXO ports +; +; SIP/username[:password[:md5secret[:authname]]]@host[:port] +; This form allows you to specify password or md5secret and authname +; without altering any authentication data in config. +; Examples: +; +; SIP/*98@mysipproxy +; SIP/sales:topsecret::account02@domain.com:5062 +; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1 +; +; IPorHost +; The next server for this call regardless of domain/peer +; +; All of these dial strings specify the SIP request URI. +; In addition, you can specify a specific To: header by adding an +; exclamation mark after the dial string, like +; +; SIP/sales@mysipproxy!sales@edvina.net +; +; A new feature for 1.8 allows one to specify a host or IP address to use +; when routing the call. This is typically used in tandem with func_srv if +; multiple methods of reaching the same domain exist. The host or IP address +; is specified after the third slash in the dialstring. Examples: +; +; SIP/devicename/extension/IPorHost +; SIP/username@domain//IPorHost +; +; CLI Commands +; ------------------------------------------------------------- +; Useful CLI commands to check peers/users: +; sip show peers Show all SIP peers (including friends) +; sip show registry Show status of hosts we register with +; +; sip set debug on Show all SIP messages +; +; sip reload Reload configuration file +; sip show settings Show the current channel configuration +; +;------- Naming devices ------------------------------------------------------ +; +; When naming devices, make sure you understand how Asterisk matches calls +; that come in. +; 1. Asterisk checks the SIP From: address username and matches against +; names of devices with type=user +; The name is the text between square brackets [name] +; 2. Asterisk checks the From: addres and matches the list of devices +; with a type=peer +; 3. Asterisk checks the IP address (and port number) that the INVITE +; was sent from and matches against any devices with type=peer +; +; Don't mix extensions with the names of the devices. Devices need a unique +; name. The device name is *not* used as phone numbers. Phone numbers are +; anything you declare as an extension in the dialplan (extensions.conf). +; +; When setting up trunks, make sure there's no risk that any From: username +; (caller ID) will match any of your device names, because then Asterisk +; might match the wrong device. +; +; Note: The parameter "username" is not the username and in most cases is +; not needed at all. Check below. In later releases, it's renamed +; to "defaultuser" which is a better name, since it is used in +; combination with the "defaultip" setting. +;----------------------------------------------------------------------------- + +; ** Old configuration options ** +; The "call-limit" configuation option is considered old is replaced +; by new functionality. To enable callcounters, you use the new +; "callcounter" setting (for extension states in queue and subscriptions) +; You are encouraged to use the dialplan groupcount functionality +; to enforce call limits instead of using this channel-specific method. +; You can still set limits per device in sip.conf or in a database by using +; "setvar" to set variables that can be used in the dialplan for various limits. + +[general] +context=public ; Default context for incoming calls. Defaults to 'default' +;allowguest=no ; Allow or reject guest calls (default is yes) + ; If your Asterisk is connected to the Internet + ; and you have allowguest=yes + ; you want to check which services you offer everyone + ; out there, by enabling them in the default context (see below). +;match_auth_username=yes ; if available, match user entry using the + ; 'username' field from the authentication line + ; instead of the From: field. +allowoverlap=no ; Disable overlap dialing support. (Default is yes) +;allowoverlap=yes ; Enable RFC3578 overlap dialing support. + ; Can use the Incomplete application to collect the + ; needed digits from an ambiguous dialplan match. +;allowoverlap=dtmf ; Enable overlap dialing support using DTMF delivery + ; methods (inband, RFC2833, SIP INFO) in the early + ; media phase. Uses the Incomplete application to + ; collect the needed digits. +;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) + ; Default is enabled. The Dial() options 't' and 'T' are not + ; related as to whether SIP transfers are allowed or not. +;realm=mydomain.tld ; Realm for digest authentication + ; defaults to "asterisk". If you set a system name in + ; asterisk.conf, it defaults to that system name + ; Realms MUST be globally unique according to RFC 3261 + ; Set this to your host name or domain name +;domainsasrealm=no ; Use domains list as realms + ; You can serve multiple Realms specifying several + ; 'domain=...' directives (see below). + ; In this case Realm will be based on request 'From'/'To' header + ; and should match one of domain names. + ; Otherwise default 'realm=...' will be used. +;recordonfeature=automixmon ; Default feature to use when receiving 'Record: on' header + ; from an INFO message. Defaults to 'automon'. Works with + ; dynamic features. Feature must be usable on requesting + ; channel for it to work. Setting this value to a blank + ; will disable it. +;recordofffeature=automixmon ; Default feature to use when receiving 'Record: off' header + ; from an INFO message. Defaults to 'automon'. Works with + ; dynamic features. Feature must be usable on requesting + ; channel for it to work. Setting this value to a blank + ; will disable it. + +; With the current situation, you can do one of four things: +; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1 +; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1 +; c) Listen on the IPv4 wildcard. Example: bindaddr=0.0.0.0 +; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=:: +; (You can choose independently for UDP, TCP, and TLS, by specifying different values for +; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".) +; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat. +; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.) +; +; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061 +; for TLS). +; IPv4 example: bindaddr=0.0.0.0:5062 +; IPv6 example: bindaddr=[::]:5062 +; +; The address family of the bound UDP address is used to determine how Asterisk performs +; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only +; AAAA records are considered. In case d), both A and AAAA records are considered. Note, +; however, that Asterisk ignores all records except the first one. In case d), when both A +; and AAAA records are available, either an A or AAAA record will be first, and which one +; depends on the operating system. On systems using glibc, AAAA records are given +; priority. + +udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) + ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) + +; When a dialog is started with another SIP endpoint, the other endpoint +; should include an Allow header telling us what SIP methods the endpoint +; implements. However, some endpoints either do not include an Allow header +; or lie about what methods they implement. In the former case, Asterisk +; makes the assumption that the endpoint supports all known SIP methods. +; If you know that your SIP endpoint does not provide support for a specific +; method, then you may provide a comma-separated list of methods that your +; endpoint does not implement in the disallowed_methods option. Note that +; if your endpoint is truthful with its Allow header, then there is no need +; to set this option. This option may be set in the general section or may +; be set per endpoint. If this option is set both in the general section and +; in a peer section, then the peer setting completely overrides the general +; setting (i.e. the result is *not* the union of the two options). +; +; Note also that while Asterisk currently will parse an Allow header to learn +; what methods an endpoint supports, the only actual use for this currently +; is for determining if Asterisk may send connected line UPDATE requests and +; MESSAGE requests. Its use may be expanded in the future. +; +; disallowed_methods = UPDATE + +; +; Note that the TCP and TLS support for chan_sip is currently considered +; experimental. Since it is new, all of the related configuration options are +; subject to change in any release. If they are changed, the changes will +; be reflected in this sample configuration file, as well as in the UPGRADE.txt file. +; +tcpenable=no ; Enable server for incoming TCP connections (default is no) +tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) + ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) + +;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no) +;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces) + ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061) + ; Remember that the IP address must match the common name (hostname) in the + ; certificate, so you don't want to bind a TLS socket to multiple IP addresses. + ; For details how to construct a certificate for SIP see + ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs + +;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number + ; of seconds a client has to authenticate. If + ; the client does not authenticate beofre this + ; timeout expires, the client will be + ; disconnected. (default: 30 seconds) + +;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of + ; unauthenticated sessions that will be allowed + ; to connect at any given time. (default: 100) + +;websocket_write_timeout = 100 ; Default write timeout to set on websocket transports. + ; This value may need to be adjusted for connections where + ; Asterisk must write a substantial amount of data and the + ; receiving clients are slow to process the received information. + ; Value is in milliseconds; default is 100 ms. + +transport=udp ; Set the default transports. The order determines the primary default transport. + ; If tcpenable=no and the transport set is tcp, we will fallback to UDP. + +srvlookup=yes ; Enable DNS SRV lookups on outbound calls + ; Note: Asterisk only uses the first host + ; in SRV records + ; Disabling DNS SRV lookups disables the + ; ability to place SIP calls based on domain + ; names to some other SIP users on the Internet + ; Specifying a port in a SIP peer definition or + ; when dialing outbound calls will supress SRV + ; lookups for that peer or call. + +;pedantic=yes ; Enable checking of tags in headers, + ; international character conversions in URIs + ; and multiline formatted headers for strict + ; SIP compatibility (defaults to "yes") + +; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters. +;tos_sip=cs3 ; Sets TOS for SIP packets. +;tos_audio=ef ; Sets TOS for RTP audio packets. +;tos_video=af41 ; Sets TOS for RTP video packets. +;tos_text=af41 ; Sets TOS for RTP text packets. + +;cos_sip=3 ; Sets 802.1p priority for SIP packets. +;cos_audio=5 ; Sets 802.1p priority for RTP audio packets. +;cos_video=4 ; Sets 802.1p priority for RTP video packets. +;cos_text=3 ; Sets 802.1p priority for RTP text packets. + +;maxexpiry=3600 ; Maximum allowed time of incoming registrations (seconds) +;minexpiry=60 ; Minimum length of registrations (default 60) +;defaultexpiry=120 ; Default length of incoming/outgoing registration +;submaxexpiry=3600 ; Maximum allowed time of incoming subscriptions (seconds), default: maxexpiry +;subminexpiry=60 ; Minimum length of subscriptions, default: minexpiry +;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions +;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention) + ; Default value is 70 +;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds + ; and reported in milliseconds with sip show settings. + ; Set to low value if you use low timeout for NAT of UDP sessions + ; Default: 60 +;qualifygap=100 ; Number of milliseconds between each group of peers being qualified + ; Default: 100 +;qualifypeers=1 ; Number of peers in a group to be qualified at the same time + ; Default: 1 +;keepalive=60 ; Interval at which keepalive packets should be sent to a peer + ; Valid options are yes (60 seconds), no, or the number of seconds. + ; Default: 0 +;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY +;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC + ; fully. Enable this option to not get error messages + ; when sending MWI to phones with this bug. +;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in + ; the From: header as the "name" portion. Also fill the + ; "user" portion of the URI in the From: header with this + ; value if no fromuser is set + ; Default: empty +;vmexten=voicemail ; dialplan extension to reach mailbox sets the + ; Message-Account in the MWI notify message + ; defaults to "asterisk" + +; Codec negotiation +; +; When Asterisk is receiving a call, the codec will initially be set to the +; first codec in the allowed codecs defined for the user receiving the call +; that the caller also indicates that it supports. But, after the caller +; starts sending RTP, Asterisk will switch to using whatever codec the caller +; is sending. +; +; When Asterisk is placing a call, the codec used will be the first codec in +; the allowed codecs that the callee indicates that it supports. Asterisk will +; *not* switch to whatever codec the callee is sending. +; +;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec + ; rather than advertising all joint codec capabilities. This + ; limits the other side's codec choice to exactly what we prefer. + +;disallow=all ; First disallow all codecs +;allow=ulaw ; Allow codecs in order of preference +;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization + ; for framing options +;autoframing=yes ; Set packetization based on the remote endpoint's (ptime) + ; preferences. Defaults to no. +; +; This option specifies a preference for which music on hold class this channel +; should listen to when put on hold if the music class has not been set on the +; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer +; channel putting this one on hold did not suggest a music class. +; +; This option may be specified globally, or on a per-user or per-peer basis. +; +;mohinterpret=default +; +; This option specifies which music on hold class to suggest to the peer channel +; when this channel places the peer on hold. It may be specified globally or on +; a per-user or per-peer basis. +; +;mohsuggest=default +; +;parkinglot=plaza ; Sets the default parking lot for call parking + ; This may also be set for individual users/peers + ; Parkinglots are configured in features.conf +;language=en ; Default language setting for all users/peers + ; This may also be set for individual users/peers +;tonezone=se ; Default tonezone for all users/peers + ; This may also be set for individual users/peers + +;relaxdtmf=yes ; Relax dtmf handling +;trustrpid = no ; If Remote-Party-ID should be trusted +;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no) +;sendrpid = rpid ; Use the "Remote-Party-ID" header + ; to send the identity of the remote party + ; This is identical to sendrpid=yes +;sendrpid = pai ; Use the "P-Asserted-Identity" header + ; to send the identity of the remote party +;rpid_update = no ; In certain cases, the only method by which a connected line + ; change may be immediately transmitted is with a SIP UPDATE request. + ; If communicating with another Asterisk server, and you wish to be able + ; transmit such UPDATE messages to it, then you must enable this option. + ; Otherwise, we will have to wait until we can send a reinvite to + ; transmit the information. +;trust_id_outbound = no ; Controls whether or not we trust this peer with private identity + ; information (when the remote party has callingpres=prohib or equivalent). + ; no - RPID/PAI headers will not be included for private peer information + ; yes - RPID/PAI headers will include the private peer information. Privacy + ; requirements will be indicated in a Privacy header for sendrpid=pai + ; legacy - RPID/PAI will be included for private peer information. In the + ; case of sendrpid=pai, private data that would be included in them + ; will be anonymized. For sendrpid=rpid, private data may be included + ; but the remote party's domain will be anonymized. The way legacy + ; behaves may violate RFC-3325, but it follows historic behavior. + ; This option is set to 'legacy' by default +;prematuremedia=no ; Some ISDN links send empty media frames before + ; the call is in ringing or progress state. The SIP + ; channel will then send 183 indicating early media + ; which will be empty - thus users get no ring signal. + ; Setting this to "yes" will stop any media before we have + ; call progress (meaning the SIP channel will not send 183 Session + ; Progress for early media). Default is "yes". Also make sure that + ; the SIP peer is configured with progressinband=never. + ; + ; In order for "noanswer" applications to work, you need to run + ; the progress() application in the priority before the app. + +;progressinband=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] + +;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=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections + ; The certificates must be sorted starting with the subject's certificate + ; and followed by intermediate CA certificates if applicable. + ; Default is to look for "asterisk.pem" in current directory + +;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections. + ; If no tlsprivatekey is specified, tlscertfile is searched for + ; for both public and private key. + +;tlscafile=</path/to/certificate> +; If the server your connecting to uses a self signed certificate +; you should have their certificate installed here so the code can +; verify the authenticity of their certificate. + +;tlscapath=</path/to/ca/dir> +; A directory full of CA certificates. The files must be named with +; the CA subject name hash value. +; (see man SSL_CTX_load_verify_locations for more info) + +;tlsdontverifyserver=[yes|no] +; If set to yes, don't verify the servers certificate when acting as +; a client. If you don't have the server's CA certificate you can +; set this and it will connect without requiring tlscafile to be set. +; Default is no. + +;tlscipher=<SSL cipher string> +; A string specifying which SSL ciphers to use or not use +; A list of valid SSL cipher strings can be found at: +; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS +; +;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2. + ; Specify protocol for outbound client connections. + ; If left unspecified, the default is sslv2. +; +;--------------------------- SIP timers ---------------------------------------------------- +; These timers are used primarily in INVITE transactions. +; The default for Timer T1 is 500 ms or the measured run-trip time between +; Asterisk and the device if you have qualify=yes for the device. +; +;t1min=100 ; Minimum roundtrip time for messages to monitored hosts + ; Defaults to 100 ms +;timert1=500 ; Default T1 timer + ; Defaults to 500 ms or the measured round-trip + ; time to a peer (qualify=yes). +;timerb=32000 ; Call setup timer. If a provisional response is not received + ; in this amount of time, the call will autocongest + ; Defaults to 64*timert1 + +;--------------------------- RTP timers ---------------------------------------------------- +; These timers are currently used for both audio and video streams. The RTP timeouts +; are only applied to the audio channel. +; The settings are settable in the global section as well as per device +; +;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity + ; on the audio channel + ; when we're not on hold. This is to be able to hangup + ; a call in the case of a phone disappearing from the net, + ; like a powerloss or grandma tripping over a cable. +;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity + ; on the audio channel + ; when we're on hold (must be > rtptimeout) +;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open + ; (default is off - zero) + +;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------ +; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions. +; This mechanism can detect and reclaim SIP channels that do not terminate through normal +; signaling procedures. Session-Timers can be configured globally or at a user/peer level. +; The operation of Session-Timers is driven by the following configuration parameters: +; +; * session-timers - Session-Timers feature operates in the following three modes: +; originate : Request and run session-timers always +; accept : Run session-timers only when requested by other UA +; refuse : Do not run session timers in any case +; The default mode of operation is 'accept'. +; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs. +; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs. +; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'. +; uac - Default to the caller initially refreshing when possible +; uas - Default to the callee initially refreshing when possible +; +; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other +; endpoint's preference for who will handle refreshes. Asterisk will never override the +; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint +; fighting over who sends the refreshes. This holds true for the initiation of session +; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or +; whether Asterisk is currently the refresher or not. +; +;session-timers=originate +;session-expires=600 +;session-minse=90 +;session-refresher=uac +; +;--------------------------- SIP DEBUGGING --------------------------------------------------- +;sipdebug = yes ; Turn on SIP debugging by default, from + ; the moment the channel loads this configuration +;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:password@mysipprovider.com +; +; This will pass incoming calls to the 's' extension +; +; +;register => 2345:password@sip_proxy/1234 +; +; Register 2345 at sip provider 'sip_proxy'. Calls from this provider +; connect to local extension 1234 in extensions.conf, default context, +; unless you configure a [sip_proxy] section below, and configure a +; context. +; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] +; Tip 2: Use separate inbound and outbound sections for SIP providers +; (instead of type=friend) if you have calls in both directions +; +;register => 3456@mydomain:5082::@mysipprovider.com +; +; Note that in this example, the optional authuser and secret portions have +; been left blank because we have specified a port in the user section +; +;register => tls://username:xxxxxx@sip-tls-proxy.example.org +; +; The 'transport' part defaults to 'udp' but may also be 'tcp', 'tls', 'ws', or 'wss'. +; Using 'udp://' explicitly is also useful in case the username part +; contains a '/' ('user/name'). + +;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. +; +; 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 peers should be able to pass directmedia to each other + ; (There is no default setting, this is just an example) + ; Use this if some of your phones are on IP addresses that + ; can not reach each other directly. This way you can force + ; RTP to always flow through asterisk in such cases. +;directmediaacl=acl_example ; Use named ACLs defined in acl.conf + +;ignoresdpversion=yes ; By default, Asterisk will honor the session version + ; number in SDP packets and will only modify the SDP + ; session if the version number changes. This option will + ; force asterisk to ignore the SDP session version number + ; and treat all SDP data as new data. This is required + ; for devices that send us non standard SDP packets + ; (observed with Microsoft OCS). By default this option is + ; off. + +;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=) + ; Like the useragent parameter, the default user agent string + ; also contains the Asterisk version. +;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=) + ; This field MUST NOT contain spaces +;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media) + ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if + ; the peer does not support SRTP. Defaults to no. +;encryption_taglen=80 ; Set the auth tag length offered in the INVITE either 32/80 default 80 +; +;avpf=yes ; Enable inter-operability with media streams using the AVPF RTP profile. + ; This will cause all offers and answers to use AVPF (or SAVPF). This + ; option may be specified at the global or peer scope. +;force_avp=yes ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for + ; media streams when appropriate, even if a DTLS stream is present. +;----------------------------------------- REALTIME SUPPORT ------------------------ +; For additional information on ARA, the Asterisk Realtime Architecture, +; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration +; +;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list + ; just like friends added from the config file only on a + ; as-needed basis? (yes|no) + +;rtsavesysname=yes ; Save systemname in realtime database at registration + ; Default= no + +;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) + ; If set to yes, when a SIP UA registers successfully, the ip address, + ; the origination port, the registration period, and the username of + ; the UA will be set to database via realtime. + ; If not present, defaults to 'yes'. Note: realtime peers will + ; probably not function across reloads in the way that you expect, if + ; you turn this option off. +;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule + ; as if it had just registered? (yes|no|<seconds>) + ; If set to yes, when the registration expires, the friend will + ; vanish from the configuration until requested again. If set + ; to an integer, friends expire within this number of seconds + ; instead of the registration interval. + +;ignoreregexpire=yes ; Enabling this setting has two functions: + ; + ; For non-realtime peers, when their registration expires, the + ; information will _not_ be removed from memory or the Asterisk database + ; if you attempt to place a call to the peer, the existing information + ; will be used in spite of it having expired + ; + ; For realtime peers, when the peer is retrieved from realtime storage, + ; the registration information will be used regardless of whether + ; it has expired or not; if it expires while the realtime peer + ; is still in memory (due to caching or other reasons), the + ; information will not be removed from realtime storage + +;----------------------------------------- SIP DOMAIN SUPPORT ------------------------ +; Incoming INVITE and REFER messages can be matched against a list of 'allowed' +; domains, each of which can direct the call to a specific context if desired. +; By default, all domains are accepted and sent to the default context or the +; context associated with the user/peer placing the call. +; REGISTER to non-local domains will be automatically denied if a domain +; list is configured. +; +; Domains can be specified using: +; domain=<domain>[,<context>] +; Examples: +; domain=myasterisk.dom +; domain=customer.com,customer-context +; +; In addition, all the 'default' domains associated with a server should be +; added if incoming request filtering is desired. +; autodomain=yes +; +; To disallow requests for domains not serviced by this server: +; allowexternaldomains=no + +;domain=mydomain.tld,mydomain-incoming + ; Add domain and configure incoming context + ; for external calls to this domain +;domain=1.2.3.4 ; Add IP address as local domain + ; You can have several "domain" settings +;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains + ; Default is yes +;autodomain=yes ; Turn this on to have Asterisk add local host + ; name and local IP to domain list. + +; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to + ; non-peers, use your primary domain "identity" + ; for From: headers instead of just your IP + ; address. This is to be polite and + ; it may be a mandatory requirement for some + ; destinations which do not have a prior + ; account relationship with your server. + +;------------------------------ Advice of Charge CONFIGURATION -------------------------- +; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and + ; AOC-E to snom endpoints. This option can be used both in the + ; peer and global scope. The default for this option is off. + + +;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- +; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a + ; SIP channel. Defaults to "no". An enabled jitterbuffer will + ; be used only if the sending side can create and the receiving + ; side can not accept jitter. The SIP channel can accept jitter, + ; thus a jitterbuffer on the receive SIP side will be used only + ; if it is forced and enabled. + +; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP + ; channel. Defaults to "no". + +; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. + +; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is + ; resynchronized. Useful to improve the quality of the voice, with + ; big jumps in/broken timestamps, usually sent from exotic devices + ; and programs. Defaults to 1000. + +; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP + ; channel. Two implementations are currently available - "fixed" + ; (with size always equals to jbmaxsize) and "adaptive" (with + ; variable size, actually the new jb of IAX2). Defaults to fixed. + +; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set. + ; The option represents the number of milliseconds by which the new jitter buffer + ; will pad its size. the default is 40, so without modification, the new + ; jitter buffer will set its size to the jitter value plus 40 milliseconds. + ; increasing this value may help if your network normally has low jitter, + ; but occasionally has spikes. + +; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". + +;----------------------------------------------------------------------------------- + +[authentication] +; Global credentials for outbound calls, i.e. when a proxy challenges your +; Asterisk server for authentication. These credentials override +; any credentials in peer/register definition if realm is matched. +; +; This way, Asterisk can authenticate for outbound calls to other +; realms. We match realm on the proxy challenge and pick an set of +; credentials from this list +; Syntax: +; auth = <user>:<secret>@<realm> +; auth = <user>#<md5secret>@<realm> +; Example: +;auth=mark:topsecret@digium.com +; +; You may also add auth= statements to [peer] definitions +; Peer auth= override all other authentication settings if we match on realm + +;------------------------------------------------------------------------------ +; DEVICE CONFIGURATION +; +; SIP entities have a 'type' which determines their roles within Asterisk. +; * For entities with 'type=peer': +; Peers handle both inbound and outbound calls and are matched by ip/port, so for +; The case of incoming calls from the peer, the IP address must match in order for +; The invitation to work. This means calls made from either direction won't work if +; The peer is unregistered while host=dynamic or if the host is otherise not set to +; the correct IP of the sender. +; * For entities with 'type=user': +; Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't +; call them) and are matched by their authorization information (authname and secret). +; Asterisk doesn't rely on their IP and will accept calls regardless of the host setting +; as long as the incoming SIP invite authorizes successfully. +; * For entities with 'type=friend': +; Asterisk will create the entity as both a friend and a peer. Asterisk will accept +; calls from friends like it would for users, requiring only that the authorization +; matches rather than the IP address. Since it is also a peer, a friend entity can +; be called as long as its IP is known to Asterisk. In the case of host=dynamic, +; this means it is necessary for the entity to register before Asterisk can call it. +; +; Use remotesecret for outbound authentication, and secret for authenticating +; inbound requests. For historical reasons, if no remotesecret is supplied for an +; outbound registration or call, the secret will be used. +; +; For device names, we recommend using only a-z, numerics (0-9) and underscore +; +; For local phones, type=friend works most of the time +; +; If you have one-way audio, you probably have NAT problems. +; If Asterisk is on a public IP, and the phone is inside of a NAT device +; you will need to configure nat option for those phones. +; Also, turn on qualify=yes to keep the nat session open +; +; Configuration options available +; -------------------- +; context +; callingpres +; permit +; deny +; secret +; md5secret +; remotesecret +; transport +; dtmfmode +; directmedia +; nat +; callgroup +; pickupgroup +; language +; allow +; disallow +; autoframing +; insecure +; trustrpid +; trust_id_outbound +; progressinband +; promiscredir +; useclientcode +; accountcode +; setvar +; callerid +; amaflags +; callcounter +; busylevel +; allowoverlap +; allowsubscribe +; allowtransfer +; ignoresdpversion +; subscribecontext +; template +; videosupport +; maxcallbitrate +; rfc2833compensate +; Note: app_voicemail mailboxes must be in the form of mailbox@context. +; mailbox +; session-timers +; session-expires +; session-minse +; session-refresher +; t38pt_usertpsource +; regexten +; fromdomain +; fromuser +; host +; port +; qualify +; keepalive +; defaultip +; defaultuser +; rtptimeout +; rtpholdtimeout +; sendrpid +; outboundproxy +; rfc2833compensate +; callbackextension +; timert1 +; timerb +; qualifyfreq +; t38pt_usertpsource +; contactpermit ; Limit what a host may register as (a neat trick +; contactdeny ; is to register at the same IP as a SIP provider, +; contactacl ; then call oneself, and get redirected to that +; ; same location). +; directmediapermit +; directmediadeny +; directmediaacl +; unsolicited_mailbox +; use_q850_reason +; maxforwards +; encryption +; description ; Used to provide a description of the peer in console output +; dtlsenable +; dtlsverify +; dtlsrekey +; dtlscertfile +; dtlsprivatekey +; dtlscipher +; dtlscafile +; dtlscapath +; dtlssetup +; dtlsfingerprint +; ignore_requested_pref ; Ignore the requested codec and determine the preferred codec +; ; from the peer's configuration. +; + +;------------------------------------------------------------------------------ +; DTLS-SRTP CONFIGURATION +; +; DTLS-SRTP support is available if the underlying RTP engine in use supports it. +; +; dtlsenable = yes ; Enable or disable DTLS-SRTP support +; dtlsverify = yes ; Verify that provided peer certificate and fingerprint are valid +; ; A value of 'yes' will perform both certificate and fingerprint verification +; ; A value of 'no' will perform no certificate or fingerprint verification +; ; A value of 'fingerprint' will perform ONLY fingerprint verification +; ; A value of 'certificate' will perform ONLY certficiate verification +; dtlsrekey = 60 ; Interval at which to renegotiate the TLS session and rekey the SRTP session +; ; If this is not set or the value provided is 0 rekeying will be disabled +; dtlscertfile = file ; Path to certificate file to present +; dtlsprivatekey = file ; Path to private key for certificate file +; dtlscipher = <SSL cipher string> ; Cipher to use for TLS negotiation +; ; A list of valid SSL cipher strings can be found at: +; ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS +; dtlscafile = file ; Path to certificate authority certificate +; dtlscapath = path ; Path to a directory containing certificate authority certificates +; dtlssetup = actpass ; Whether we are willing to accept connections, connect to the other party, or both. +; ; Valid options are active (we want to connect to the other party), passive (we want to +; ; accept connections only), and actpass (we will do both). This value will be used in +; ; the outgoing SDP when offering and for incoming SDP offers when the remote party sends +; ; actpass +; dtlsfingerprint = sha-1 ; The hash to use for the fingerprint in SDP (valid options are sha-1 and sha-256) + +;[sip_proxy] +; For incoming calls only. Example: FWD (Free World Dialup) +; We match on IP address of the proxy for incoming calls +; since we can not match on username (caller id) +;type=peer +;context=from-fwd +;host=fwd.pulver.com + +;[sip_proxy-out] +;type=peer ; we only want to call out, not be called +;remotesecret=guessit ; Our password to their service +;defaultuser=yourusername ; Authentication user for outbound proxies +;fromuser=yourusername ; Many SIP providers require this! +;fromdomain=provider.sip.domain +;host=box.provider.com +;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will +; ; accept both tcp and udp. The default transport type is only used for +; ; outbound messages until a Registration takes place. During the +; ; peer Registration the transport type may change to another supported +; ; type if the peer requests so. + +;usereqphone=yes ; This provider requires ";user=phone" on URI +;callcounter=yes ; Enable call counter +;busylevel=2 ; Signal busy at 2 or more calls +;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer +;port=80 ; The port number we want to connect to on the remote side + ; Also used as "defaultport" in combination with "defaultip" settings + +;--- sample definition for a provider +;[provider1] +;type=peer +;host=sip.provider1.com +;fromuser=4015552299 ; how your provider knows you +;remotesecret=youwillneverguessit ; The password we use to authenticate to them +;secret=gissadetdu ; The password they use to contact us +;callbackextension=123 ; Register with this server and require calls coming back to this extension +;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will +; ; accept both tcp and udp. Default is udp. The first transport +; ; listed will always be used for outgoing connections. +;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old +; ; message count will be stored in the configured virtual mailbox. It can be used +; ; by any device supporting MWI by specifying <configured value>@SIP_Remote as the +; ; mailbox. + +; +; Because you might have a large number of similar sections, it is generally +; convenient to use templates for the common parameters, and add them +; the the various sections. Examples are below, and we can even leave +; the templates uncommented as they will not harm: + +[basic-options](!) ; a template + dtmfmode=rfc2833 + context=from-office + type=friend + +[natted-phone](!,basic-options) ; another template inheriting basic-options + directmedia=no + host=dynamic + +[public-phone](!,basic-options) ; another template inheriting basic-options + directmedia=yes + +[my-codecs](!) ; a template for my preferred codecs + disallow=all + allow=ilbc + allow=g729 + allow=gsm + allow=g723 + allow=ulaw + ; Or, more simply: + ;allow=!all,ilbc,g729,gsm,g723,ulaw + +[ulaw-phone](!) ; and another one for ulaw-only + disallow=all + allow=ulaw + ; Again, more simply: + ;allow=!all,ulaw + +; and finally instantiate a few phones +; +; [2133](natted-phone,my-codecs) +; secret = peekaboo +; [2134](natted-phone,ulaw-phone) +; secret = not_very_secret +; [2136](public-phone,ulaw-phone) +; secret = not_very_secret_either +; ... +; + +; Standard configurations not using templates look like this: +; +;[grandstream1] +;type=friend +;context=from-sip ; Where to start in the dialplan when this phone calls +;recordonfeature=dynamicfeature1 ; Feature to use when INFO with Record: on is received. +;recordofffeature=dynamicfeature2 ; Feature to use when INFO with Record: off is received. +;callerid=John Doe <1234> ; Full caller ID, to override the phones config + ; on incoming calls to Asterisk +;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'. +;host=192.168.0.23 ; we have a static but private IP address + ; No registration allowed +;directmedia=yes ; allow RTP voice traffic to bypass Asterisk +;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone +;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time + ; from the phone to asterisk (deprecated) + ; 1 for the explicit peer, 1 for the explicit user, + ; remember that a friend equals 1 peer and 1 user in + ; memory + ; There is no combined call counter for a "friend" + ; so there's currently no way in sip.conf to limit + ; to one inbound or outbound call per phone. Use + ; the group counters in the dial plan for that. + ; +;mailbox=1234@default ; mailbox 1234 in voicemail context "default" +;disallow=all ; need to disallow=all before we can use allow= +;allow=ulaw ; Note: In user sections the order of codecs + ; listed with allow= does NOT matter! +;allow=alaw +;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! +;allow=g729 ; Pass-thru only unless g729 license obtained +;callingpres=allowed_passed_screen ; Set caller ID presentation + ; See function CALLERPRES documentation for possible + ; values. + +;[xlite1] +; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! +; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed +;type=friend +;regexten=1234 ; When they register, create extension 1234 +;callerid="Jane Smith" <5678> +;host=dynamic ; This device needs to register +;directmedia=no ; Typically set to NO if behind NAT +;disallow=all +;allow=gsm ; GSM consumes far less bandwidth than ulaw +;allow=ulaw +;allow=alaw +;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes +;registertrying=yes ; Send a 100 Trying when the device registers. + +;[snom] +;type=friend ; Friends place calls and receive calls +;context=from-sip ; Context for incoming calls from this user +;secret=blah +;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions +;language=de ; Use German prompts for this user +;host=dynamic ; This peer register with us +;dtmfmode=inband ; Choices are inband, rfc2833, or info +;defaultip=192.168.0.59 ; IP used until peer registers +;mailbox=1234@context,2345@context ; Mailbox(-es) for message waiting indicator +;subscribemwi=yes ; Only send notifications if this phone + ; subscribes for mailbox notification +;vmexten=voicemail ; dialplan extension to reach mailbox + ; sets the Message-Account in the MWI notify message + ; defaults to global vmexten which defaults to "asterisk" +;disallow=all +;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! + + +;[polycom] +;type=friend ; Friends place calls and receive calls +;context=from-sip ; Context for incoming calls from this user +;secret=blahpoly +;host=dynamic ; This peer register with us +;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info +;defaultuser=polly ; Username to use in INVITE until peer registers +;defaultip=192.168.40.123 + ; Normally you do NOT need to set this parameter +;disallow=all +;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! +;progressinband=no ; Polycom phones don't work properly with "never" + + +;[pingtel] +;type=friend +;secret=blah +;host=dynamic +;insecure=port ; Allow matching of peer by IP address without + ; matching port number +;insecure=invite ; Do not require authentication of incoming INVITEs +;insecure=port,invite ; (both) +;qualify=1000 ; Consider it down if it's 1 second to reply + ; Helps with NAT session + ; qualify=yes uses default value +;qualifyfreq=60 ; Qualification: How often to check for the + ; host to be up in seconds + ; Set to low value if you use low timeout for + ; NAT of UDP sessions +; +; Call group and Pickup group should be in the range from 0 to 63 +; +;callgroup=1,3-4 ; We are in caller groups 1,3,4 +;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 +;namedcallgroup=engineering,sales,netgroup,protgroup ; We are in named call groups engineering,sales,netgroup,protgroup +;namedpickupgroup=sales ; We can do call pick-p for named call group sales +;defaultip=192.168.0.60 ; IP address to use if peer has not registered +;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address +;permit=192.168.0.60/255.255.255.0 +;permit=192.168.0.60/24 ; we can also use CIDR notation for subnet masks +;permit=2001:db8::/32 ; IPv6 ACLs can be specified if desired. IPv6 ACLs + ; apply only to IPv6 addresses, and IPv4 ACLs apply + ; only to IPv4 addresses. +;acl=named_acl_example ; Use named ACLs defined in acl.conf + +;[cisco1] +;type=friend +;secret=blah +;qualify=200 ; Qualify peer is no more than 200ms away +;host=dynamic ; This device registers with us +;directmedia=no ; Asterisk by default tries to redirect the + ; RTP media stream (audio) to go directly from + ; the caller to the callee. Some devices do not + ; support this (especially if one of them is + ; behind a NAT). +;defaultip=192.168.0.4 ; IP address to use until registration +;defaultuser=goran ; Username to use when calling this device before registration + ; Normally you do NOT need to set this parameter +;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device +;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will + ; cause the given audio file to + ; be played upon completion of + ; an attended transfer to the + ; target of the transfer. + +;[pre14-asterisk] +;type=friend +;secret=digium +;host=dynamic +;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine. + ; You must have this turned on or DTMF reception will work improperly. +;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets + ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the + ; external IP address of the remote device. If port forwarding is done at the client side + ; then UDPTL will flow to the remote device. |