[ehodge22]

I can call between the 2 phones I have, but when I try to call out I get a fast busy. From the Asterisk console I see the follow error;
senna*CLI>
<--- SIP read from UDP://<myip>:5060 --->
ACK sip:12123182000@<myip>:5060 SIP/2.0
Via: SIP/2.0/UDP <myip>:5060;branch=z9hG4bKb082f925D2D66254
From: "user" <sip:1002@myip>;tag=E1FCE27F-6E81431E
To: <sip:12123182000@<myip>;user=phone>;tag=as389d252a
CSeq: 2 ACK
Call-ID: 5c19d92-55f39c79-3f9e7388@<myip>
Contact: <sip:1002@<myip>:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
Authorization: Digest username="1002", realm="asterisk", nonce="4f0b0d91", uri="sip:12123182000@<myip>:5060;user=phone", response="c8e813601f663689b7c93749309bc613", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
----------------------------------------------------
my sip.conf entries;

[general]
port=5060
bindaddrr=0.0.0.0
disallow-all
allow=ulaw
context=default
allowoverlap=no

[inphonex]
disallow=all
allow=gsm
allow=g729
allow=g723
username="username"
type=peer
secret="password"
host=sip.inphonex.com
fromuser="username"
fromdomain=sip.inphonex.com
context=from-inphonex
canreinvite=no

[sip.inphonex.com]
username="username"
user="username"
type=friend
insecure=very
host=sip.inphonex.com
fromdomain=sip.inphonex.com
context=from-pstn

-----------------------
extensions.conf entries;
[general]
static=yes
writeprotect=yes
autofallthrough=yes
extenpatternmatchnew=true
clearglobalvars=no
priorityjumping=no

[default]
exten => s,1,Answer
exten => s,2,Dial(SIP/1000....) use the dial from my other post
exten => s,3,Voicemail(1000@default)
exten => s,4,playback(vm-goodbye)
exten => s,5,hangup

[incoming_calls]
exten => s,1,Wait(1)
exten => s,n,Answer()
exten => s,1,Goto(default|1001)
;exten => s,n(menu),Playback(vm-brief-menu)
;exten => s,n(exten),Background(vm-enter-num-to-call)
;exten => s,n,WaitExten(5)
exten => s,n(goodbye),Playback(vm-goodbye)
exten => s,n(end),Hangup();

include => extens
exten => i,1,Playback(vm-invalid)
exten => i,n,Goto(s,exten) ; optionally, transfer to operator
exten => t,1,Goto(s,goodbye)

[extens]
include => stdexten
exten => 1001,1,Gosub(stdexten(1001,SIP/user1))
exten => 1001,n,Goto(s,exten;
exten => 1002,1,Gosub(stdexten(1002,SIP/user2))
exten => 1002,n,Goto(s,end0

*********Help***************

[RusTus]

I am not seeing any outbound rule in your extensions.conf

nor do I see an outboundproxy= in your sip.conf

and I am assuming you are calling out via SIP here, theres also no register => line in your sip.conf as well. Right now Asterisk has no idea what to do when you dial an outside.