Outbound SIP not working

Hello…

Here is the CLI from asterisk whenever I try to make an inbound call:

................ ]
Asterisk Ready.
*CLI> [Oct 20 11:06:08] NOTICE[28467]: chan_sip.c:12336 handle_response_peerpoke: Peer 'varphonex' is now Reachable. (73ms / 2000ms)
[Oct 20 11:06:08] NOTICE[28467]: chan_sip.c:12336 handle_response_peerpoke: Peer 'luke' is now Reachable. (221ms / 2000ms)
[Oct 20 11:06:08] NOTICE[28467]: chan_sip.c:12336 handle_response_peerpoke: Peer 'sipura1' is now Reachable. (341ms / 2000ms)
[Oct 20 11:06:20] NOTICE[28525]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/varphonex-081e1df0' not posted

All I end up getting is a “fast busy”

Here is my sip.conf

[general]
context=default			; Default context for incoming calls
allowoverlap=no			; Disable overlap dialing support. (Default is yes)
bindport=5060			; UDP Port to bind to (SIP standard port is 5060)
				; bindport is the local UDP port that Asterisk will listen on
bindaddr=0.0.0.0			; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes			; Enable DNS SRV lookups on outbound calls
canreinvite=no

[sipura1]
type=friend
secret=******
qualify=yes 	
nat=yes	
host=dynamic 
canreinvite=no 
context=phone1 

[luke]
type=friend
secret=******
qualify=yes 
nat=yes 
host=dynamic 
canreinvite=no 
context=phone1 

trustrpid = yes
sendrpid = yes

register => 4213002:dci123@sip.varphonex.com:5060

[varphonex]
type=friend
username:1234567
realm=sip.varphonex.com
canreinvite=yes
secret=*****
insecure=very
host=sip.varphonex.com
disallow=all
allow=ilbc
allow=ulaw
allow=alaw
qualify=yes
nat=no
outboundproxy=sip.varphonex.com
#regexten=yes
auth=1234567:******@sip.varphonex.com:5060[/code]

and here is my extensions.conf:
[code][general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp				; Console interface for demo
IAXINFO=guest					; IAXtel username/password
TRUNKMSD=1					; MSD digits to strip (usually 1 or 0)

[dundi-e164-canonical]

[dundi-e164-customers]

[dundi-e164-via-pstn]

[dundi-e164-local]
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

[dundi-e164-switch]
switch => DUNDi/e164

[dundi-e164-lookup]
include => dundi-e164-local
include => dundi-e164-switch

[macro-dundi-e164]
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)

[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext

[trunkint]
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunkld]
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunklocal]
exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[trunktollfree]
exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

[international]
ignorepat => 9
include => longdistance
include => trunkint

[longdistance]
ignorepat => 9
include => local
include => trunkld

[local]
ignorepat => 9
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider

include => parkedcalls

[macro-trunkdial]
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp

[macro-stdexten];
exten => s,1,Dial(${ARG2},20)			; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${ARG1},u)	; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)		; If they press #, return to start

exten => s-BUSY,1,Voicemail(${ARG1},b)		; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)		; If they press #, return to start

exten => _s-.,1,Goto(s-NOANSWER,1)		; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1})		; If they press *, send the user into VoicemailMain

[macro-stdPrivacyexten];
exten => s,1,Dial(${ARG2},20|p)			; Ring the interface, 20 seconds maximum, call screening 
						; option (or use P for databased call screening)
exten => s,2,Goto(s-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${ARG1},u)	; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)		; If they press #, return to start

exten => s-BUSY,1,Voicemail(${ARG1},b)		; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)		; If they press #, return to start

exten => s-DONTCALL,1,Goto(${ARG3},s,1)		; Callee chose to send this call to a polite "Don't call again" script.

exten => s-TORTURE,1,Goto(${ARG4},s,1)		; Callee chose to send this call to a telemarketer torture script.

exten => _s-.,1,Goto(s-NOANSWER,1)		; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1})		; If they press *, send the user into VoicemailMain

[macro-page];
exten => s,1,ChanIsAvail(${ARG1}|js)			; j is for Jump and s is for ANY call
exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA")			; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)	; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp()					; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1}||)
exten => s,n(fail),Hangup


[demo]
exten => s,1,Wait(1)			; Wait a second, just for fun
exten => s,n,Answer			; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5)	; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10)	; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats)	; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct)	; Play some instructions
exten => s,n,WaitExten			; Wait for an extension to be dialed.

exten => 2,1,BackGround(demo-moreinfo)	; Give some more information.
exten => 2,n,Goto(s,instruct)

exten => 3,1,Set(LANGUAGE()=fr)		; Set language to french
exten => 3,n,Goto(s,restart)		; Start with the congratulations

exten => 1000,1,Goto(default,s,1)
exten => 1234,1,Playback(transfer,skip)		; "Please hold while..." 
					; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})

exten => 1235,1,Voicemail(1234,u)		; Right to voicemail

exten => 1236,1,Dial(Console/dsp)		; Ring forever
exten => 1236,n,Voicemail(1234,b)		; Unless busy

exten => #,1,Playback(demo-thanks)		; "Thanks for trying the demo"
exten => #,n,Hangup			; Hang them up.

exten => t,1,Goto(#,1)			; If they take too long, give up
exten => i,1,Playback(invalid)		; "That's not valid, try again"

exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default)	; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo)	; Couldn't connect to the demo site
exten => 500,n,Goto(s,6)		; Return to the start over message.


exten => 600,1,Playback(demo-echotest)	; Let them know what's going on
exten => 600,n,Echo			; Do the echo test
exten => 600,n,Playback(demo-echodone)	; Let them know it's over
exten => 600,n,Goto(s,6)		; Start over

exten => 76245,1,Macro(page,SIP/Grandstream1)
; or if your peernames are the same as extensions
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
;
;
; System Wide Page at extension 7999
;
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n|d)

; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})


[default]
include => demo

[phone1]
exten => 100,1,Dial(SIP/luke)
exten => 101,1,Dial(SIP/sipura1)
exten => 600,1,Playback(demo-echotest)	; Let them know what's going on
exten => 600,n,Echo			; Do the echo test
exten => 600,n,Playback(demo-echodone)	; Let them know it's over
exten => 600,n,Goto(s,6)		; Start over
exten => _X.,1,Answer
exten => _X.,2,Wait(2)
exten => _X.,3,Dial(SIP/varphonex)
#exten => _X.,4,Hangup
#exten => _9.,1,Dial(SIP/${EXTEN:1}@my_provider,30,r)[/code]

And here is the Debug from my outbound sip channel:

[code]SIP Debugging Enabled for IP: 208.239.76.163:5060
*CLI> Audio is at 72.167.14.79 port 15110
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 208.239.76.163:5060:
INVITE sip:sip.varphonex.com SIP/2.0
Via: SIP/2.0/UDP 72.167.14.79:5060;branch=z9hG4bK7acfc37c;rport
From: "luke" <sip:luke@72.167.14.79>;tag=as47f1eb1f
To: <sip:sip.varphonex.com>
Contact: <sip:luke@72.167.14.79>
Call-ID: 5a5a31493f49c69a08d898b41dcbde4d@72.167.14.79
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 20 Oct 2007 18:30:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 28432 28432 IN IP4 72.167.14.79
s=session
c=IN IP4 72.167.14.79
t=0 0
m=audio 15110 RTP/AVP 0 97 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from 208.239.76.163:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 72.167.14.79:5060;received=72.167.14.79;branch=z9hG4bK7acfc37c;                                             rport=5060
From: "luke" <sip:luke@72.167.14.79>;tag=as47f1eb1f
To: <sip:sip.varphonex.com>
Call-ID: 5a5a31493f49c69a08d898b41dcbde4d@72.167.14.79
CSeq: 102 INVITE


<------------->
--- (6 headers 0 lines) ---

<--- SIP read from 208.239.76.163:5060 --->
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 72.167.14.79:5060;received=72.167.14.79;branch=z9hG4bK7acfc37c;                                             rport=5060
From: "luke" <sip:luke@72.167.14.79>;tag=as47f1eb1f
To: <sip:sip.varphonex.com>;tag=SDec0m199-14ec6e58
Call-ID: 5a5a31493f49c69a08d898b41dcbde4d@72.167.14.79
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Transmitting (no NAT) to 208.239.76.163:5060:
ACK sip:sip.varphonex.com SIP/2.0
Via: SIP/2.0/UDP 72.167.14.79:5060;branch=z9hG4bK7acfc37c;rport
From: "luke" <sip:luke@72.167.14.79>;tag=as47f1eb1f
To: <sip:sip.varphonex.com>;tag=SDec0m199-14ec6e58
Contact: <sip:luke@72.167.14.79>
Call-ID: 5a5a31493f49c69a08d898b41dcbde4d@72.167.14.79
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Oct 20 11:30:27] NOTICE[16335]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/var                                             phonex-081e1df0' not posted
Really destroying SIP dialog '5a5a31493f49c69a08d898b41dcbde4d@72.167.14.79' Met                                             hod: INVITE
Reliably Transmitting (no NAT) to 208.239.76.163:5060:
OPTIONS sip:sip.varphonex.com SIP/2.0
Via: SIP/2.0/UDP 72.167.14.79:5060;branch=z9hG4bK24cbba20;rport
From: "asterisk" <sip:asterisk@72.167.14.79>;tag=as103d6411
To: <sip:sip.varphonex.com>
Contact: <sip:asterisk@72.167.14.79>
Call-ID: 72a4e86f0a449d04317d54963d63739f@72.167.14.79
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 20 Oct 2007 18:31:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---

<--- SIP read from 208.239.76.163:5060 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 72.167.14.79:5060;received=72.167.14.79;branch=z9hG4bK24cbba20;rport=5060
From: "asterisk" <sip:asterisk@72.167.14.79>;tag=as103d6411
To: <sip:sip.varphonex.com>;tag=aprqngfrt-fsuhvb20000c6
Call-ID: 72a4e86f0a449d04317d54963d63739f@72.167.14.79
CSeq: 102 OPTIONS
Allow: INVITE,ACK,BYE,REGISTER,CANCEL,PRACK,INFO,NOTIFY,REFER

I know its alot to read through. If anyone can see what I am doing, let me wrong… let me know. thanks.

this one is too large to read…
create 2 users and and your own context for playing a file and try…
copy and paste that context only and the error code in the cli, if it is still not working…may be i can help u…

This:

should be:

exten => _X.,3,Dial(SIP/varphonex/${EXTEN})

or

exten => _X.,3,Dial(SIP/varphonex/${EXTEN:1})

if using a prefix.

Regards.

Marco Bruni