Viatalk and FreePBX configuration troubles

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/[email protected]${TRUNK},60,r)
exten => 411,1,Dial(SIP/[email protected]${TRUNK},60,r)
exten => *123,1,Dial(SIP/*[email protected]${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

Ok, This part:

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

should already be in your sip.conf file. If it is not and you are running PIAF, you will need to add this at the command line with vi or nano.

Now open FreePBX and create a new trunk. Put viatalk as the trunk name and the rest of this goes into Peer Details

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)
context=from-trunk

Notice I added the line at the bottom. This tells asterisk what to do with inbound calls

Erase everything in the User section of the page

At the bottom enter this in the registration string

1815XXXXXXX:[email protected]/1815XXXXXXX

HTH,

I’ve copied the information over from the AussieVoIP site, hopefully it will still work for you:

http://www.freepbx.org/support/documentation/howtos/howto-setting-up-voip-provider-trunks/viatalk

Thanks for pointing me in the right direction. The trunk is working and the incoming and outgoing routes work as expected. However, the information in the [general] field is not in my sip.conf file. That file doesn’t have much except a warning not to edit , as Free PBX will overwrite on reload. I placed the [general] information in the sip.custom.conf file, is that right? The reason I ask is that there is no DTMF happiness. Thanks again I know I am almost there!!!

The DTMF issue is specifically addressed…
http://www.freepbx.org/support/documentation/howtos/howto-setting-up-voip-provider-trunks/viatalk