[general]
context=default                 ; Default context for incoming calls
;allowguest=no                  ; Allow or reject guest calls (default is yes)
;match_auth_username=yes        ; if available, match user entry using the
                                ; 'username' field from the authentication line
                                ; instead of the From: field.
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
                                ; Default is enabled. The Dial() options 't' and 'T' are not
                                ; related as to whether SIP transfers are allowed or not.
realm=tacoms.fin.miss4             ; Realm for digest authentication
                                ; defaults to "asterisk". If you set a system name in
                                ; asterisk.conf, it defaults to that system name
                                ; Realms MUST be globally unique according to RFC 3261
                                ; Set this to your host name or domain name
udpbindaddr=10.10.3.12             ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
                                ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)

tcpenable=yes                    ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=10.10.3.12             ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
                                ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)

;tlsenable=no                   ; Enable server for incoming TLS (secure) connections (default is no)
;tlsbindaddr=10.10.0.12            ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
                                ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
                                ; Remember that the IP address must match the common name (hostname) in the
                                ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
                                ; For details how to construct a certificate for SIP see 
                                ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs

;tlscertfile=asterisk.pem       ; Certificate file (*.pem only) to use for TLS connections 
                                ; default is to look for "asterisk.pem" in current directory
srvlookup=no                    ; Enable DNS SRV lookups on outbound calls

; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
;tos_sip=cs3                    ; Sets TOS for SIP packets.
;tos_audio=ef                   ; Sets TOS for RTP audio packets.
;tos_video=af41                 ; Sets TOS for RTP video packets.
;tos_text=af41                  ; Sets TOS for RTP text packets.

;cos_sip=3                      ; Sets 802.1p priority for SIP packets.
;cos_audio=5                    ; Sets 802.1p priority for RTP audio packets.
;cos_video=4                    ; Sets 802.1p priority for RTP video packets.
;cos_text=3                     ; Sets 802.1p priority for RTP text packets.

disallow=all                   ; First disallow all codecs
allow=alaw                     ; Allow codecs in order of preference
allow=g729
allow=h263
allow=h263p
allow=h261
;relaxdtmf=yes                  ; Relax dtmf handling
;trustrpid = no                 ; If Remote-Party-ID should be trusted
;sendrpid = yes                 ; If Remote-Party-ID should be sent
dtmfmode = rfc2833             ; Set default dtmfmode for sending DTMF. Default: rfc2833
;dtmfmode=inband
videosupport=yes               ; Turn on support for SIP video. You need to turn this

;--------------------------- SIP DEBUGGING ---------------------------------------------------
sipdebug = yes                 ; Turn on SIP debugging by default, from
                                ; the moment the channel loads this configuration
;recordhistory=yes              ; Record SIP history by default 
                                ; (see sip history / sip no history)
;dumphistory=yes                ; Dump SIP history at end of SIP dialogue
                                ; SIP history is output to the DEBUG logging channel
;
; Specify a few phones
; v
;
[9876541234567]
context=default
username=9876541234567
type=friend
secret=yourpassword
qualify=yes
nat=no
host=dynamic
careinvite=no
port=5060

[9876541234568]
context=default
username=9876541234568
type=friend
secret=yourpassword
qualify=yes
nat=no
host=dynamic
careinvite=no
port=5060