[asterisk-users] Cannot make softphone outbound calls, "user not found?"

[BSumrall]

I can call my asterisk server from a land line, but I cannot make an
outgoing call from a softphone to a land line. 
The softphone says, "user not found". 
Teliax has tripped the switch to allow authentication to be in the body of
the pasket. 
Still doesn't work. 

Here is my extensions.conf 

[general] 
static=yes 
writeprotect=no 
clearglobalvars=no 

[globals] 
CONSOLE=Console/dsp ; Console interface for demo 
IAXINFO=guest ; IAXtel username/password 
TRUNK=Zap/g2 ; Trunk interface 
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) 
PSTN=Zap/g2 



[default] 
exten => 8005181896,1,Answer() 
exten => 8005181896,2,Playback(dir-intro) 
exten => 8005181896,3,Queue(service|t||8005181896|45) 

[outbound] 
exten => 8008629121,1,Answer() 
exten => 8008629121,2,Playback(demo-congrats) 
exten => 8008629121,3,AgentLogin() 

exten => h,1,DeadAGI(postqueue.agi) 

[8008629121] 
;exten => 8008629121,1,Answer() 
;exten => 8008629121,1,DIAL(SIP/user,20) 

[204] 
exten => _1XXXXXXXXXX,2,DIAL,(IAX2/xxxx at teliax/${EXTEN},30,tr) 
exten => 204,3,Answer 
exten => 204,4,Hangup 

here is my sip.conf 

register => xxxx:xxxxxxxxxxx at voip-co3.teliax.com 
[authentication] 
auth = xxxxx:xxxxxxxxxxxx at voip-co3.teliax.com 

[teliax] 
context=default 
type=friend 
username=xxxxx 
user=xxxxx 
host=voip-co3.teliax.com 
secret=xxxxxxxxxxxxx 
insecure=very 
canreinvite=no 
disallow=all 
allow=ulaw 
allow=alaw 
allow=gsm 

[204] 
user=204 
context=internal 
type=friend 
secret=xxxxxxxxx 
insecure=very 
canreinvite=no 
context=home 
host=dynamic 
disallow=all 
allow=ulaw 
allow=alaw 
allow=ilbc ; preference 
allow=gsm 
nat=no 

Here is the debug 


SIP/2.0 404 Not Found 
Via: SIP/2.0/UDP
66.176.193.46:5072;branch=z9hG4bKf8452f47-cdf8-1810-869e-0013d3ee21fe;receiv
ed=66.176.193.46;rport=5072 
From: "Brad S"
<sip:204 at xx.xx.xx.xx>;tag=74272f47-cdf8-1810-869a-0013d3ee21fe 
To: <sip:19544790554 at xx.xx.xx.xx>;tag=as7878bf48 
Call-ID: 59212f47-cdf8-1810-869a-0013d3ee21fe at usmc 
CSeq: 2 INVITE 
User-Agent: Asterisk PBX 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY 
Supported: replaces 
Content-Length: 0 


<------------> 
Scheduling destruction of SIP dialog
'59212f47-cdf8-1810-869a-0013d3ee21fe at usmc' in 32000 ms (Method: INVITE) 

<--- SIP read from 66.176.193.46:5072 ---> 
ACK sip:19544790554 at xx.xx.xx.xx SIP/2.0 
CSeq: 2 ACK 
Via: SIP/2.0/UDP
66.176.193.46:5072;branch=z9hG4bKf8452f47-cdf8-1810-869e-0013d3ee21fe;rport 
From: "Brad S"
<sip:204 at xx.xx.xx.xx>;tag=74272f47-cdf8-1810-869a-0013d3ee21fe 
Call-ID: 59212f47-cdf8-1810-869a-0013d3ee21fe at usmc 
To: <sip:19544790554 at xx.xx.xx.xx>;tag=as7878bf48 
Proxy-Authorization: Digest username="204", realm="asterisk",
nonce="59963977", uri="sip:19544790554 at 66.109.17.92", algorithm=md5,
response="08af1ad02b83d5a1c8a4fb442588d9ea" 
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE 
Content-Length: 0 
Max-Forwards: 70 

I can see the it is putting my extension @ my_sip_server, but it should be
looking at the body of the message right now. 

How do I make it put the user account in the header instead of the
extension?