Outgoing problem "Freeconet"

Hello,
I have a problem with outgoing connection :

sip.conf

[general]
context=default         
extensions.conf
bindport=5172           
srvlookup=yes            
defaultexpiry=60        
allowguest=no           
dtmfmode=rfc2833        
nat=yes                 
bindaddr=0.0.0.0
                                        
register => user:pass@sip.freeconet.pl/rafal



[freeconet-out]
type=peer                       
username=user              
secret=pass              
fromdomain=sip.freeconet.pl

context=freeconet             
host=sip.freeconet.pl
port=5060
outboundproxy=sip.freeconet.pl
outboundproxyport=5060
insecure=no                     

[freeconet-in1]                 
type=peer
fromdomain=sip.freeconet.pl
port=5060
context=freeconet
host=213.218.116.65
insecure=no

[freeconet-in2]
type=peer
fromdomain=sip.freeconet.pl
port=5060
context=freeconet
host=213.218.116.66
insecure=no

[freeconet-in3]
type=peer
fromdomain=sip.freeconet.pl
port=5060
context=freeconet
host=213.218.116.93
insecure=no

[freeconet-in4]
type=peer
fromdomain=sip.freeconet.pl
port=5060
context=freeconet
host=213.218.116.85
insecure=no

[freeconet-in5]
type=peer
fromdomain=sip.freeconet.pl
port=5060
context=freeconet
host=213.218.108.141
insecure=no

[freeconet-in6]
type=peer
fromdomain=sip.freeconet.pl
port=5060
context=freeconet
host=213.218.108.142
insecure=no

[100]                                   
context=freeconet                      
type=peer                               
username=100                            
secret=pass                          
callerid="Rafal" <100>                  
host=dynamic                                                                       
dtmfmode=rfc2833
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
insecure=no

and extensions.conf

[default]
exten => _.,1,Hangup

[freeconet]
exten => t,1,Hangup
exten => h,1,Hangup
exten => _X.,1,SIPAddHeader(X-Fid: ${SIPCALLID})
exten => _XXXXXXXXX,1,Dial(SIP/${EXTEN}@freeconet-out)
exten => rafal,1,Dial(SIP/100,45,Tt)
exten => _XXX,1,Dial(SIP/${EXTEN})
 -- SIP/freeconet-out-0000001b is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/100-0000001a' status is 'CONGESTION'
    -- Executing [h@freeconet:1] Hangup("SIP/100-0000001a", "") in new stack
  == Spawn extension (freeconet, h, 1) exited non-zero on 'SIP/100-0000001a'
       > doing dnsmgr_lookup for 'sip.freeconet.pl'
       > ast_get_srv: SRV lookup for '_sip._udp.sip.freeconet.pl' mapped to host server1.freeconet.pl, port 5060
Name/username              Host                                    Dyn Forcerport ACL Port     Status
100/100                    xx.xxx.xx.xx                            D   N             5099     Unmonitored
freeconet-in1              213.218.116.65                               N             5060     Unmonitored
freeconet-in2              213.218.116.66                               N             5060     Unmonitored
freeconet-in3              213.218.116.93                               N             5060     Unmonitored
freeconet-in4              213.218.116.85                               N             5060     Unmonitored
freeconet-in5              213.218.108.141                              N             5060     Unmonitored
freeconet-in6              213.218.108.142                              N             5060     Unmonitored
freeconet-out/rafalrk87    213.218.116.66                               N             5060     Unmonitored

no nat and firewall
asterisk 1.8 debian
any help for me
thanks

freeconet rejected the call with a status that Asterisk treats as busy. Increasing the logging/debugging level (and enabling debug logging) should reveal the status code.

This appears to be a support request, not a discussion topic!

Sorry.
I’m new and i need help

---
[Jul 11 00:54:29] NOTICE[14979]: chan_sip.c:20354 handle_response_invite: Failed to authenticate on INVITE to '"Rafal" <sip:100@sip.freeconet.pl>;tag=as67b060f8'
    -- SIP/freeconet-out-0000005c is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/100-0000005b' status is 'CONGESTION'

<--- Reliably Transmitting (NAT) to 5.226.xxx.xx:5099 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 5.226.xxx.xx:5099;branch=z9hG4bKPjh6touH-VMTuBIAgKmv3lyvOze-wS8jHU;received=5.226.xxx.xx;rport=5099
From: "100" <sip:100@92.xxx.xx.xx:5172>;tag=DpcwN4U.14Cp8ZUodQPKQvY7xQpBjYDz
To: sip:796595935@92.xxx.xx.xx:5172;tag=as31bd2b31
Call-ID: FPMv67rVSKDJOUI3a4r2GmTm.68eX5KO
CSeq: 11049 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<------------>
Really destroying SIP dialog '310e41574f1545ba13b7186a158f0287@sip.freeconet.pl' Method: INVITE
    -- Executing [h@freeconet:1] Hangup("SIP/100-0000005b", "") in new stack
  == Spawn extension (freeconet, h, 1) exited non-zero on 'SIP/100-0000005b'

<--- SIP read from UDP:5.226.xxx.xx:5099 --->
ACK sip:796595935@92.xxx.xx.xx:5172 SIP/2.0
Via: SIP/2.0/UDP 5.226.xxx.xx:5099;rport;branch=z9hG4bKPjh6touH-VMTuBIAgKmv3lyvOze-wS8jHU
Max-Forwards: 70
From: "100" <sip:100@92.xxx.xx.xx:5172>;tag=DpcwN4U.14Cp8ZUodQPKQvY7xQpBjYDz
To: sip:796595935@92.xxx.xx.xx:5172;tag=as31bd2b31
Call-ID: FPMv67rVSKDJOUI3a4r2GmTm.68eX5KO
CSeq: 11049 ACK
Route: <sip:92.xxx.xx.xx:5172;lr>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'FPMv67rVSKDJOUI3a4r2GmTm.68eX5KO' Method: ACK

<--- SIP read from UDP:5.226.xxx.xx:5099 --->

i need help

Your log starts after the important SIP messages. However, it does show that freeconet rejected your authentication data. The problem will be finding what they actually require. You can probably assume that username (now called defaultuser) is irrelevant, as it should be ignored once you are registered. This may be a system that needs fromuser, especially if it actually needs fromdomain.

Yea,
I add in sip.conf [freeconet-out] fromuser=MyUserName
And working now.
Thank you david55.