[Asterisk-Users] PRI to SIP

[Jens Kübler]

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