; ; 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] canreinvite=no 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=papinicomputer.it ; 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=62.48.35.22 ; 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=papinicomputer.it ; 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/outgoing 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=it ; 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 ; Valid values: yes, no, never Default: never ;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 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected, ; for any reason, always reject with '401 Unauthorized' ; instead of letting the requester know whether there was ; a matching user or peer for their request ; behind a NAT device to communicate with services on the outside. externip = 62.48.35.22 ; 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=yes ; 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|) ; 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 [authentication] [000001] type=friend callerid="PapiniComputer ASS 1 <0000001>" secret=******* host=dynamic nat=yes canreinvite=no dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw dtmfmode = rfc2833 context=papinicomputer subscribecontext=blf call-limit=1 limitonpeers=yes qualify=yes