Hi everyone, newbie help requested for Viatalk to PIAF/FreePBX configuration. I did search and couldn’t find this so please, oh please, be kind. My setup is a Walmart special, Orgasmatron PIAF, Linksys Router with correct ports forwarded, ATT dsl (ISP), dyndns.org (for my changing IP), ordered Vitelity DID (still waiting - 14 days and counting – backordered, ugh). I currently have a Viatalk account (2 +years with a PAP2 adapter) and want to configure a trunk and an inbound and outbound route in my new PBX while I await for Vitelity. My Viatalk account has an Asterisk configuration section that gives me the recommended settings, however, I am not sure what goes where in Free PBX gui. I know enough that I am not supposed to edit the sip.conf and extensions.conf files manually (Freepbx manages these). Here is what Viatalk recommends:
SIP.CONF
[general]
context=default
bindport=5060
port=5060
bindaddr=0.0.0.0
recordhistory=yes
disallow=all (where does all of this go in FreePBX)?
allow=ulaw
allow=gsm
trustrpid=yes
sendrpid=yes
dtmfmode=inband
relaxdtmf=yes
realm=asterisk
; REGISTRATION
register => 1815XXXXXXX:[email protected]/1815XXXXXXX (X=my phone # and Y= my password)
; TRUNK CONFIGURATION
[viatalk]
type=friend
authuser=1815XXXXXXX (X=my phone #)
username=1815XXXXXXX (X=my phone #)
fromuser=1815XXXXXXX (X=my phone #)
fromdomain=chicago-1a.vtnoc.net
host=chicago-1a.vtnoc.net
secret=YYYYYYYY (Y=my password)
insecure=very
qualify=3600
nat=no ; switch to yes if behind nat (try to avoid it if at all possible)
; PEER CONFIGURATION
[1000]
type=peer
nat=yes ; allows you to use a softphone/adapter behind nat
host=dynamic
canreinvite=yes
username=1000
secret=password ; this can be anything you want
Extensions.conf
[general]
static=yes
writeprotect=yes
[globals]
CONSOLE=Console/dsp
NPX=815
PEER=1000 ; The peer you setup in sip.conf for your softphone/adapter
TRUNK=viatalk ; The name of the trunk you defined
[default]
include=incoming
include=outgoing
[incoming]
exten => 1815XXXXXXX,1,Dial(SIP/${PEER},60,r) (X=my phone #)
exten => 1815XXXXXXX,2,Hangup (X=my phone #)
[outgoing]
exten => 911,1,Dial(SIP/911@${TRUNK},60,r)
exten => 411,1,Dial(SIP/411@${TRUNK},60,r)
exten => *123,1,Dial(SIP/*123@${TRUNK},60,r)
exten => _NXXXXXX,1,Goto(1${NPX}${EXTEN},1) ; if dialing 7 digits, prepend 1 + Area Code
exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) ; if dialing 10 digits, prepend 1
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@${TRUNK},60,r)
exten => _1NXXNXXXXXX,2,Playtones(480+620/250,0/250) ; play a fast busy (reorder) tone
exten => _1NXXNXXXXXX,3,Congestion
; For International dialing [Optional]
exten => _011X.,1,Dial(SIP/${EXTEN}@${TRUNK},60,r)
exten => _011X.,2,Playtones(480+620/250,0/250) ; play a fast busy (reorder) tone
exten => _011X.,3,Congestion
; in the case of an invalid number or a time-out hangup
exten => i,1,Hangup
exten => t,1,Hangup
Am I going to use all of this? I basically want to set up a single trunk and have all incoming and outgoing calls go through this trunk.
Thanks a bunch