[asterisk-users] sip to sip ?

[Rob Schall]

If you're getting a 404, I would assume it is reacting like any other
non-connection would (http, etc). Do you know if the packets are
reaching the phone, or if the phone is registering its correct IP
Address? If it is registering, but no packets are reaching it, could it
be a routing issue?

Rob

Chris Hills wrote:
> Dennis Kavadas wrote:
>
>> hi all
>>
>> i've just setup an * box and want to test voip calling, initially from
>> sip user to sip user...
>>
>> local sip users can call each other, no issues.
>>
>> problem arises when i try and call a remote sip account, my * box
>> always returns "SIP/2.0 404 Not Found"
>>
>> any ideas ?
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> Dennis
>
> I use the following as my default context:-
>
> [default]
> exten => _X.,1,NoOp(Incoming Call from ${CALLERID} for
> ${EXTEN}@${SIPDOMAIN})
> exten => _X.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
> exten => _X.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
> exten => _X.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
> exten => _X.,5,GotoIf($[${SIPDOMAIN} = ${MYIP}]?10)
> exten => _X.,6,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
> exten => _X.,7,NoOp(@${SIPDOMAIN} is remote - forwarding...)
> exten => _X.,8,Macro(uridial,${EXTEN}@${SIPDOMAIN})
> exten => _X.,9,HangUp()
> exten => _X.,10,Goto(default-noturi,${EXTEN},1)
> exten => h,1,HangUp()
> exten => s-BUSY,1,Congestion
> exten => s-CHANUNAVAIL,1,Congestion
> exten => s-CONGESTION,1,Congestion
>
> [macro-uridial]
> exten => s,1,NoOp(Outbound SIP URI call ${ARG1})
> exten => s,2,SetCIDNum(0123456789)
> exten => s,3,Dial(SIP/${ARG1})
> exten => s,4,Congestion()
>
>
> HTH
>