106 lines
6.7 KiB
Plaintext
106 lines
6.7 KiB
Plaintext
[general]
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context=trunkinbound ; Default context for incoming calls
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allowguest=no ; Allow or reject guest calls (default is yes)
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allowoverlap=no ; Disable overlap dialing support. (Default is yes)
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;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
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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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;pedantic=yes ; Enable checking of tags in headers,
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;tos_sip=cs3 ; Sets TOS for SIP packets.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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;tos_video=af41 ; Sets TOS for RTP video packets.
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;maxexpiry=3600 ; Maximum allowed time of incoming registrations
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;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
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;defaultexpiry=120 ; Default length of incoming/outgoing registration
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;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
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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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;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
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;vmexten=voicemail ; dialplan extension to reach mailbox sets the
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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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mohinterpret=default
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mohsuggest=default
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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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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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videosupport=no ; Turn on support for SIP video. You need to turn this on
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;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
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callevents=yes ; generate manager events when sip ua
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;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
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;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
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;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
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;regcontext=sipregistrations
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rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
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;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
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;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
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;sipdebug = yes ; Turn on SIP debugging by default, from
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;recordhistory=yes ; Record SIP history by default
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;dumphistory=yes ; Dump SIP history at end of SIP dialogue
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;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
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;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
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notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
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notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
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limitonpeers = yes ; Apply call limits on peers only. This will improve
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;t38pt_udptl = yes ; Default false
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;register => 1234:password@mysipprovider.com
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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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;externip = 192.168.1.1 ; Address that we're going to put in outbound SIP
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;externhost=test.test.com ; Alternatively you can specify a domain
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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=force_rport,comedia ; Global NAT settings (Affects all peers and users)
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canreinvite=no ; Asterisk by default tries to redirect the
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;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
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;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
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;rtsavesysname=yes ; Save systemname in realtime database at registration
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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=mydomain.tld,mydomain-incoming
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;domain=1.2.3.4 ; Add IP address as local domain
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;allowexternaldomains=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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;fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
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jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
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jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
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jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
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jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
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jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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qualify=yes ; By default, qualify all peers at 2000ms
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limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers
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websocket_enabled=true ; Set to false to prevent chan_sip listening to websockets. Needed if using SIP & PJSIP on the same system
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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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