;
; SIP Configuration example for Asterisk

[general]
context=pri_from_als		; Default context for incoming calls
;allowguest=no			; Allow or reject guest calls (default is yes, this can also be set to 'osp'
				; if asterisk was compiled with OSP support.
allowoverlap=yes                 ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
realm=asterisk			; Realm for digest authentication
				; defaults to "asterisk"
				; Realms MUST be globally unique according to RFC 3261
				; Set this to your host name or domain name
bindport=5060			; UDP Port to bind to (SIP standard port is 5060)

bindaddr=172.16.4.4		; 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
				
disallow=all			; First disallow all codecs
allow=g729
allow=ulaw			; Allow codecs in order of preference
;allow=g726
;
regcontext=mt_thorium
;
;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
; no reason for Asterisk to stay in the media path, the media will be redirected.
; This does not really work with in the case where Asterisk is outside and have
; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
;
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,
;------------------------------------------------------------------------------
; Definitions of locally connected SIP phones
[garrish]
type=friend
username=garrish                  ; Polycom 600
secret=******
callerid=garrish <1234567890>
context=mt_thorium
subscribecontext=mt_thorium	; Only allow SUBSCRIBE for local extensions
call-limit=10
;nat=yes                        ; This phone may be natted
host=dynamic
canreinvite=yes                  ; Asterisk by default tries to redirect the
videosupport=no
qualify=200                     ; Qualify peer is no more than 200ms away
qualifysmoothing=yes
mailbox=0452@mt_thorium         ; Set MWI

[polycom4]
type=friend
username=polycom4                  ; Polycom 600
secret=*****
callerid=polycom4 <2345678901>
context=mt_thorium
subscribecontext=mt_thorium	; Only allow SUBSCRIBE for local extensions
call-limit=10
;nat=yes                        ; This phone may be natted
host=dynamic
dtmfmode=rfc2833		; Choices are inband, rfc2833, or info
canreinvite=yes                  ; Asterisk by default tries to redirect the
videosupport=no
qualify=200                     ; Qualify peer is no more than 200ms away
qualifysmoothing=yes
mailbox=0453@mt_thorium  	; Set MWI
