[general]
context=trunkinbound            ; Default context for incoming calls
allowguest=no                  ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
;realm=mydomain.tld             ; Realm for digest authentication
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
;domain=mydomain.tld            ; Set default domain for this host
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4                 ; Add IP address as local domain
;allowexternalinvites=no        ; Disable INVITE and REFER to non-local domains
;autodomain=yes                 ; Turn this on to have Asterisk add local host
;pedantic=yes                   ; Enable slow, pedantic checking for Pingtel
;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
;videosupport=yes               ; Turn on support for SIP video
;recordhistory=yes              ; Record SIP history by default
disallow=all                    ; First disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
allow=gsm                       ;
musicclass=default              ; Sets the default music on hold class for all SIP calls
language=en                     ; Default language setting for all users/peers
relaxdtmf=yes                   ; Relax dtmf handling
rtptimeout=60                   ; Terminate call if 60 seconds of no RTP activity
;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP activity
trustrpid = no                  ; If Remote-Party-ID should be trusted
sendrpid = yes                  ; If Remote-Party-ID should be sent
progressinband=no               ; If we should generate in-band ringing always
;useragent=Asterisk PBX         ; Allows you to change the user agent string
promiscredir = no       ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no               ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833              ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes           ; send compact sip headers.
;sipdebug = yes                 ; Turn on SIP debugging by default, from
;subscribecontext = default     ; Set a specific context for SUBSCRIBE requests
;notifyringing = yes            ; Notify subscriptions on RINGING state
;alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to be rejected,
;regcontext=sipregistrations
;registertimeout=20             ; retry registration calls every 20 seconds (default)
;registerattempts=10            ; Number of registration attempts before we give up
callevents=no                   ; generate manager events when sip ua performs events (e.g. hold)
;externip = 192.168.1.1     ; Address that we're going to put in outbound SIP messages
;externhost=foo.dyndns.net      ; Alternatively you can specify an
;externrefresh=10               ; How often to refresh externhost if
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
nat=yes                         ; Global NAT settings  (Affects all peers and users)
canreinvite=no
;rtcachefriends=yes             ; Cache realtime friends by adding them to the internal list
;rtupdate=yes                   ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes                ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes            ; Enabling this setting has two functions:
; domain=myasterisk.dom
; domain=customer.com,customer-context
; autodomain=yes
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to

#include sip-vicidial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@10.10.10.16:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000