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#723023
;
; SIP Configuration example for Asterisk
[general]
#include additional_a2billing_sip.conf
#include additional_a2billing_iax.conf
#include sip_registrations.conf
#include sip_additional.conf
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
; bindport is the local UDP port that Asterisk will listen on
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
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
; and subscriptions (seconds)
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
defaultexpiry=300 ; Default length of incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
; Defaults to 100 ms
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
; fully. Enable this option to not get error messages
; when sending MWI to phones with this bug.
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to "asterisk"
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=alaw
;allow=ilbc ; see doc/rtp-packetization for framing options
;
language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
seragent=PBX ; Allows you to change the user agent string
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.
;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
rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
; on the audio channel
; when we're not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
; more database transactions if you are using realtime.
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
;canreinvite=yes ; Asterisk by default tries to redirect the
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
;canreinvite=nonat ; An additional option is to allow media path redirection
;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,