ViaTalk

Never mind, I figured it out. For thos of you who need it though, here it is.....

register=[YOUR PHONE NUMBER]:[YOUR PASSWORD]@neptune.vtnoc.net/[YOUR PHONE NUMBER]

[viatalk1]
username=[YOUR PHONE NUMBER]
type=peer
secret=[YOUR PASSWORD]
qualify=yes
nat=no
insecure=very
host=neptune.vtnoc.net
fromuser=[YOUR PHONE NUMBER]
fromdomain=neptune.vtnoc.net
dtmfmode=inband
dtmf=inband
context=from-pstn
canreinvite=yes
authuser=[YOUR PHONE NUMBER]
allow=ulaw

Hope this helps....

Grymlock........
:-D

This got me MU&CH further down the road. Thanks...

However, while my default extension does finally ring it hangs up after the 1st ring every time.

I suspected the softphone I was using but I can make calls back and forth between it and another softphone on another computer.

When I don't have the softphone active and there is not default extension registered. The call goes to voice mail perfectly. This leads me to believe I have the Trunk configured correctly.

Any ideas to even trouble shoot this will be appreciated.

Thanks :-?

One or two more points.
One, that is all 11 digits (same as your login on their management).
Two, You also will need to have host=neptune.vtnoc.net, on your
inbound settings.

allow=ulaw
disallow=all
dtmf=inband
dtmfmode=inband
type=friend
host=netptune.vtnoc.net

I had to add the inband things to get callback decoding DTMF.
(AND had to get them to do the same at VT)
There are two other ways to get calls if you don't have the host=
in your inbound.
One, turn on allow anon sip in general settings. (more on that later)
Two, Put your full DID in as an inbound route.
There are tons of stuff you can do with anonymous sip and such
services as ENUM and direct dialing between boxes, look for a thread on this soon.

Sincerely,
Jeff Kephart
(look for me on irc #freepbx and #voipguys)
PS:I endorse Viatalk to anyone who asks, as E-Bay would say, Good-Voipers... :-)

I'm up and running perfectly now. Or at least so it seems.

The problem was actually with ViaTalk. I had to get down to sniffing packets with Ethereal to figure this one out.

Anyway, I let them know and they figured out it for some odd reason had to do with the setting of the SERVICE DISTRUPED phone number I had set.

This is the number all calls are to be forwarded to durring internet service interuption to my location. I had it set to my Cell number.

Anyway the deleted it and now nothing but joy.

Grymlock or anyone else that may know,

I too have ViaTalk, but I am unable to make the incoming lines work. And 'priority support' has shoved me off because I use TrixBox. Im not sure where these settings need to be placed in the gui, any suggestions?

The following incoming settings will work - I got them to work on my setup:

allow=ulaw
context=global
disallow=all
dtmf=inband
dtmfmode=inband
host=mercury.vtnoc.net
secret=YOURPASSWORD
type=peer
username=YOURPHONENUMBER

For User Context enter your phone number. Hope this helps. I am running 1.2.3 with the latest FreePBX 2.2.0 RC1 that was just released.

Ok, I've gotten outbound working great! However, Inbound calling ends up with the "The number you have dialed is not in service...". I have tried both suggestions by Johnk and Grymlock. Viatalk's Inbound Call log shows the calls being answered as well. My only incoming call route (Any #/Any CID) is set to send it to my test extension.

2 Hours later: I've discovered, that for some reason my Viatalk trunk is not registered. I found this through : Asterisk CLI: sip show subscriptions.

I'm assuming this is why incoming calls come in and asterisk thinks it is an unknown sip caller...

Any ideas? THANKS in advance!

Edit:8/3/07 I missed in the earlier instructions where "allow anonymous sip calls" was required to be on. Works like a champ now!

but on dslreports there seem to be a fair number of folks who have problems more often than not.