[Asterisk-Users] PRI to SIP

[FaberK]

Hi Jens,
this is my sip.conf
---
[general]
context=default
fromdomain=192.168.1.188
port=5060
bindaddr=0.0.0.0
localnet = 192.168.1.0/255.255.255.0
srvlookup=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
language=it

register => 1:1 at sip.mydomain.com:5060/s

[1]
type=peer
username=1
fromuser=1
secret=1
Callerid=1
context=default
port=5060
dtmfmode=rfc2833
host=sip.mydomain.com
fromdomain=sip.mydomain.com
insecure=very
canreinvite = no
disallow=all
allow=alaw
allow=ulaw
---
and my extensions.conf
---
[general]
static=yes
writeprotect=no

[globals]
;TRUNK=Zap/g2
;TRUNKMSD=1
PRITRUNK1=Zap/g1
PRITRUNK2=Zap/g2
MYNUM=0123456

[default]
exten => _0123456XXX,1,Wait(0.75)
exten => _0123456XXX,2,SetCallerID(${CALLERID})
exten => _0123456XXX,3,Set(CALLERID=${CALLERID})
exten => _0123456XXX,4,Dial(SIP/${EXTEN:7}@sip.mydomain.com:5060,30,tr)
exten => _0123456XXX,103,Hangup

exten => t,1,Hangup

exten => i,1,Answer
exten => i,2,Playback(pbx-invalid)
exten => i,3,Hangup
---
Maybe I do not see my error...

Thanks

2005/11/14, Jens Kübler <cleanerx at au.hadiko.de>:
> Am Montag 14 November 2005 17:22 schrieb 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)
> Here we go
>
> You haven't disabled general authentication (if you wish to)
> or haven't set a proper default context in sip.conf
> or you aren't handling the default incoming sip context properly in
> extensions.conf
>
> Jens
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--
.:FaberK:.