;
; 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,