[general]
#include sip_general_additional.conf
bindport = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)

canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=h264
allow=h261
allow=h263
allow=h263p
videosupport=yes
maxcallbitrate=512

; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; Reported as required for Asterisk 1.4
notifyringing=yes
notifyhold=yes
limitonpeers=yes

; enable and force the sip jitterbuffer. If these settings are desired
; they should be set in the sip_general_custom.conf file as this file
; will get overwritten during reloads and upgrades.
;
; jbenable=yes
; jbforce=yes

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.

[100]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=100@device
host=dynamic
dtmfmode=rfc2833
disallow=all
dial=SIP/100
context=from-internal
canreinvite=no
callgroup=
callerid=device <100>
allow=g729,g723,ulaw,h264,h263,h263p
accountcode=
call-limit=50

[101]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=101@device
host=dynamic
dtmfmode=rfc2833
disallow=all
dial=SIP/101
context=from-internal
canreinvite=no
callgroup=
callerid=device <101>
allow=g729,g723,ulaw,h264,h263,h263p
accountcode=
call-limit=50

[102]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=102@device
host=dynamic
dtmfmode=rfc2833
disallow=
dial=SIP/102
context=from-internal
canreinvite=no
callgroup=
callerid=device <102>
allow=all
accountcode=
call-limit=50