81 lines
4.6 KiB
Plaintext
81 lines
4.6 KiB
Plaintext
[general]
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context=default ; Default context for incoming calls
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;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
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;realm=mydomain.tld ; Realm for digest authentication
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bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
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bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
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srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;domain=mydomain.tld ; Set default domain for this host
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;domain=mydomain.tld,mydomain-incoming
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;domain=1.2.3.4 ; Add IP address as local domain
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;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
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;autodomain=yes ; Turn this on to have Asterisk add local host
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;pedantic=yes ; Enable slow, pedantic checking for Pingtel
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;tos=184 ; Set IP QoS to either a keyword or numeric val
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tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
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maxexpiry=3600 ; Max length of incoming registration we allow
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defaultexpiry=120 ; Default length of incoming/outgoing registration
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;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
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;checkmwi=10 ; Default time between mailbox checks for peers
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;vmexten=voicemail ; dialplan extension to reach mailbox sets the
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;videosupport=yes ; Turn on support for SIP video
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;recordhistory=yes ; Record SIP history by default
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disallow=all ; First disallow all codecs
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allow=ulaw ; Allow codecs in order of preference
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allow=gsm ;
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musicclass=default ; Sets the default music on hold class for all SIP calls
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language=en ; Default language setting for all users/peers
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relaxdtmf=yes ; Relax dtmf handling
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rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
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;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
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trustrpid = no ; If Remote-Party-ID should be trusted
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sendrpid = yes ; If Remote-Party-ID should be sent
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progressinband=no ; If we should generate in-band ringing always
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;useragent=Asterisk PBX ; Allows you to change the user agent string
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promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
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;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
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dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
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;compactheaders = yes ; send compact sip headers.
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;sipdebug = yes ; Turn on SIP debugging by default, from
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;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
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;notifyringing = yes ; Notify subscriptions on RINGING state
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;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
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;regcontext=sipregistrations
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;registertimeout=20 ; retry registration calls every 20 seconds (default)
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;registerattempts=10 ; Number of registration attempts before we give up
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callevents=no ; generate manager events when sip ua performs events (e.g. hold)
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;externip = 192.168.1.1 ; Address that we're going to put in outbound SIP messages
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;externhost=foo.dyndns.net ; Alternatively you can specify an
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;externrefresh=10 ; How often to refresh externhost if
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localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
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localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
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localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
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localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
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nat=yes ; Global NAT settings (Affects all peers and users)
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canreinvite=no
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;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
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;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
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;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
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;ignoreregexpire=yes ; Enabling this setting has two functions:
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; domain=myasterisk.dom
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; domain=customer.com,customer-context
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; autodomain=yes
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; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
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#include sip-vicidial.conf
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; register SIP account on remote machine if using SIP trunks
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; register => testSIPtrunk:test@10.10.10.16:5060
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;
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; setup account for SIP trunking:
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; [SIPtrunk]
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; disallow=all
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; allow=ulaw
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; allow=alaw
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; type=friend
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; username=testSIPtrunk
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; secret=test
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; host=10.10.10.16
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; dtmfmode=inband
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; qualify=1000
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