[Asterisk-Users] PRI to SIP

[FaberK]

Hi guys,
this is the scenario:
PRI <->Asterisk<->SER
If I call from a Sip(SER) user everything is good, I can call
anywhere, but if I try to call from outside(PRI) everything is
wrong!!!
This is the CLI for an incoming call:
------------------
ast*CLI>
    -- Executing SetCallerID("Zap/14-1", "outside") in new stack
    -- Executing Set("Zap/14-1", "CALLERID=outside") in new stack
    -- Executing Dial("Zap/14-1",
"SIP/201 at sip.mydomain.com:5060|30|r") in new stack
    -- Accepting call from 'outside' to 'mynumber201' on channel 0/14, span 1
    -- Called 201 at sip.mydomain.com:5060
Nov 14 17:15:47 NOTICE[8459]: chan_sip.c:9492 handle_response_invite:
Failed to authenticate on INVITE to '"Unknown"
<sip:Unknown at 192.168.1.188>;tag=as6261e060'
    -- SIP/sip.mydomain.com:5060-5eda is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Congestion("Zap/14-1", "") in new stack
    -- Channel 0/14, span 1 got hangup request
  == Spawn extension (default, 0666620201, 4) exited non-zero on 'Zap/14-1'
    -- Hungup 'Zap/14-1'
ast*CLI>
------------------
my extensions:
------------------
[general]
static=yes
writeprotect=no

[globals]
;TRUNK=Zap/g2
;TRUNKMSD=1
PRITRUNK1=Zap/g1
PRITRUNK2=Zap/g2
MYNUM=1234567

[default]

exten => _1234567XXX,1,SetCallerID(${CALLERID})
exten => _1234567XXX,2,Set(CALLERID=${CALLERID})
exten => _1234567XXX,3,Dial(SIP/${EXTEN:7}@sip.mydomain.com:5060,30,r)
exten => _1234567XXX,103,Hangup
------------------
Where I'm wrong?
What's missing?
Thanks!
--
.:FaberK:.