diff options
Diffstat (limited to '1.2-netsec/configs/sip.conf.sample')
| -rw-r--r-- | 1.2-netsec/configs/sip.conf.sample | 441 |
1 files changed, 0 insertions, 441 deletions
diff --git a/1.2-netsec/configs/sip.conf.sample b/1.2-netsec/configs/sip.conf.sample deleted file mode 100644 index 3d9299c05..000000000 --- a/1.2-netsec/configs/sip.conf.sample +++ /dev/null @@ -1,441 +0,0 @@ -; -; SIP Configuration example for Asterisk -; -; Syntax for specifying a SIP device in extensions.conf is -; SIP/devicename where devicename is defined in a section below. -; -; You may also use -; SIP/username@domain to call any SIP user on the Internet -; (Don't forget to enable DNS SRV records if you want to use this) -; -; 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 -; -; Useful CLI commands to check peers/users: -; sip show peers Show all SIP peers (including friends) -; sip show users Show all SIP users (including friends) -; sip show registry Show status of hosts we register with -; -; sip debug Show all SIP messages -; -; reload chan_sip.so Reload configuration file -; Active SIP peers will not be reconfigured -; - -[general] -context=default ; Default context for incoming calls -;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp' - ; if asterisk was compiled with OSP support. -;realm=mydomain.tld ; Realm for digest authentication - ; defaults to "asterisk" - ; Realms MUST be globally unique according to RFC 3261 - ; Set this to your host name or domain name -bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) -bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) -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 - -;domain=mydomain.tld ; Set default domain for this host - ; If configured, Asterisk will only allow - ; INVITE and REFER to non-local domains - ; Use "sip show domains" to list local domains -;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 -;allowexternalinvites=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. -;pedantic=yes ; Enable slow, pedantic checking for Pingtel - ; and multiline formatted headers for strict - ; SIP compatibility (defaults to "no") -;tos=184 ; Set IP QoS to either a keyword or numeric val -;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none -;maxexpiry=3600 ; Max length of incoming registration we allow -;defaultexpiry=120 ; Default length of incoming/outoing registration -;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY -;checkmwi=10 ; Default time between mailbox checks for peers -;vmexten=voicemail ; dialplan extension to reach mailbox sets the - ; Message-Account in the MWI notify message - ; defaults to "asterisk" -;videosupport=yes ; Turn on support for SIP video -;recordhistory=yes ; Record SIP history by default - ; (see sip history / sip no history) - -;disallow=all ; First disallow all codecs -;allow=ulaw ; Allow codecs in order of preference -;allow=ilbc ; -;musicclass=default ; Sets the default music on hold class for all SIP calls - ; This may also be set for individual users/peers -;language=en ; Default language setting for all users/peers - ; This may also be set for individual users/peers -;relaxdtmf=yes ; Relax dtmf handling -;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity - ; when we're not on hold -;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity - ; when we're on hold (must be > rtptimeout) -;trustrpid = no ; If Remote-Party-ID should be trusted -;sendrpid = yes ; If Remote-Party-ID should be sent -;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 -;useragent=Asterisk PBX ; Allows you to change the user agent 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 SIP 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 - ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) - ; auto : Use rfc2833 if offered, inband otherwise - -;compactheaders = yes ; send compact sip headers. -;sipdebug = yes ; Turn on SIP debugging by default, from - ; the moment the channel loads this configuration -;subscribecontext = default ; Set a specific context for SUBSCRIBE requests - ; Useful to limit subscriptions to local extensions - ; Settable per peer/user also -;notifyringing = yes ; Notify subscriptions on RINGING state - -; -; 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. The actual extension is the 'regexten' parameter of the registering -; peer or its name if 'regexten' is not provided. More than one regexten may -; be supplied if they are separated by '&'. Patterns may be used in regexten. -; -;regcontext=sipregistrations -; -; Asterisk can register as a SIP user agent to a SIP proxy (provider) -; Format for the register statement is: -; register => user[:secret[:authuser]]@host[:port][/extension] -; -; 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). -; -; host is either a host name defined in DNS or the name of a section defined -; below. -; -; 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 type=peer and type=user sections for SIP providers -; (instead of type=friend) if you have calls in both directions - -;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 -;callevents=no ; generate manager events when sip ua performs events (e.g. hold) - -;----------------------------------------- NAT SUPPORT ------------------------ -; The externip, externhost and localnet settings are used if you use Asterisk -; behind a NAT device to communicate with services on the outside. - -;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages - ; if we're behind a NAT - - ; The externip and localnet is used - ; when registering and communicating with other proxies - ; that we're registered with -;externhost=foo.dyndns.net ; Alternatively you can specify an - ; external host, and Asterisk will - ; perform DNS queries periodically. Not - ; recommended for production - ; environments! Use externip instead -;externrefresh=10 ; How often to refresh externhost if - ; used - ; You may add multiple local networks. A reasonable set of defaults - ; are: -;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks -;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 nat= setting is used when Asterisk is on a public IP, communicating with -; devices hidden behind a NAT device (broadband router). If you have one-way -; audio problems, you usually have problems with your NAT configuration or your -; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP -; ports for incoming audio in rtp.conf -; -;nat=no ; Global NAT settings (Affects all peers and users) - ; yes = Always ignore info and assume NAT - ; no = Use NAT mode only according to RFC3581 - ; never = Never attempt NAT mode or RFC3581 support - ; route = Assume NAT, don't send rport - ; (work around more UNIDEN bugs) - -;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) - -;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'. - -;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 - -; 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. -; 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 - -; 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. - -[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 - -;------------------------------------------------------------------------------ -; Users and peers have different settings available. Friends have all settings, -; since a friend is both a peer and a user -; -; User config options: Peer configuration: -; -------------------- ------------------- -; context context -; permit permit -; deny deny -; secret secret -; md5secret md5secret -; dtmfmode dtmfmode -; canreinvite canreinvite -; nat nat -; callgroup callgroup -; pickupgroup pickupgroup -; language language -; allow allow -; disallow disallow -; insecure insecure -; trustrpid trustrpid -; progressinband progressinband -; promiscredir promiscredir -; useclientcode useclientcode -; accountcode accountcode -; setvar setvar -; callerid callerid -; amaflags amaflags -; call-limit call-limit -; restrictcid restrictcid -; subscribecontext subscribecontext -; mailbox -; username -; template -; fromdomain -; regexten -; fromuser -; host -; port -; qualify -; defaultip -; rtptimeout -; rtpholdtimeout -; sendrpid - -;[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 -;secret=guessit -;username=yourusername ; Authentication user for outbound proxies -;fromuser=yourusername ; Many SIP providers require this! -;fromdomain=provider.sip.domain -;host=box.provider.com -;usereqphone=yes ; This provider requires ";user=phone" on URI -;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer - -;------------------------------------------------------------------------------ -; Definitions of locally connected SIP phones -; -; type = user a device that authenticates to us by "from" field to place calls -; type = peer a device we place calls to or that calls us and we match by host -; type = friend two configurations (peer+user) in one -; -; For local phones, type=friend works most of the time -; -; If you have one-way audio, you propably 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 - -;[grandstream1] -;type=friend -;context=from-sip ; Where to start in the dialplan when this phone calls -;callerid=John Doe <1234> ; Full caller ID, to override the phones config -;host=192.168.0.23 ; we have a static but private IP address - ; No registration allowed -;nat=no ; there is not NAT between phone and Asterisk -;canreinvite=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 - ; (1 for the explicit peer, 1 for the explicit user, - ; remember that a friend equals 1 peer and 1 user in - ; memory) -;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 -;astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists - - -;[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 -;nat=yes ; X-Lite is behind a NAT router -;canreinvite=no ; Typically set to NO if behind NAT -;disallow=all -;allow=gsm ; GSM consumes far less bandwidth than ulaw -;allow=ulaw -;allow=alaw -;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes - - -;[snom] -;type=friend ; Friends place calls and receive calls -;context=from-sip ; Context for incoming calls from this user -;secret=blah -;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions -;language=de ; Use German prompts for this user -;host=dynamic ; This peer register with us -;dtmfmode=inband ; Choices are inband, rfc2833, or info -;defaultip=192.168.0.59 ; IP used until peer registers -;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator -;vmexten=voicemail ; dialplan extension to reach mailbox - ; sets the Message-Account in the MWI notify message - ; defaults to global vmexten which defaults to "asterisk" -;restrictcid=yes ; To have the callerid restriced -> sent as ANI -;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 -;username=polly ; Username to use in INVITE until peer registers - ; 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 -;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 -;defaultip=192.168.0.60 ; IP address to use if peer has not registred - -;[cisco1] -;type=friend -;secret=blah -;qualify=200 ; Qualify peer is no more than 200ms away -;nat=yes ; This phone may be natted - ; Send SIP and RTP to the IP address that packet is - ; received from instead of trusting SIP headers -;host=dynamic ; This device registers with us -;canreinvite=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 -;username=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 this device - |