[Asterisk-Users] FREE BSD

[Alvaro Gonzalez]

anynody knows if I Can install and run Asterisk under Free BSD?

thanks,

Alvaro

-----Mensaje original-----
De: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]En nombre de Rich
Adamson
Enviado el: martes, 14 de diciembre de 2004 20:14
Para: Asterisk Users Mailing List - Non-Commercial Discussion; William
Betts
Asunto: Re: [Asterisk-Users] 404 "Not Found" Sip Response


> The hardware I currently have is:
>
> TDM400P  with 3 FXO ports, and 1 FXS port
> 4 Cisco 7960 Phones (only 1 is currently configured for testing purposes)
> Asterisk on slack 10
>
> I can dial out just fine via the Cisco phone, but when I try to dail
> in I get the following output when I load asterisk up in debug mode.
>
>  -- Got SIP response 404 "Not Found" back from <ip_address_of_sip_phone>
>     -- SIP/20-e3a9 is circuit-busy
>
> I have looked several places for an answer to this and I haven't found
> one. Any input from the users on this would be a great help. Here is
> what is in my sip.conf and extensions.conf file.
>
> Thank You,
> William Betts
>
> [general]
> port=5060
> bindaddr=0.0.0.0
> tos=lowdelay
> disallow=all
> allow=ulaw
> context=local-access
>
> [20]
> type=friend
> username=w0
> secret=m3
> host=64.123.157.103
> canreinvite=no
> qualify=200
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> callerid=Daves Office <20>
>
>
>
> [extensions]
> exten => 20,1,Dial(SIP/20,20)
> exten => 20,2,Voicemail(u${EXTEN})
> exten => 20,3,Hangup
> exten => 20,102,Voicemail(b${EXTEN})
> exten => 20,103,Hangup
>
> [incoming]
>
> exten => s,1,Answer
> exten => s,2,DigitTimeout(10)
> exten => s,3,ResponseTimeout(20)
> exten => s,4,Dial(SIP/20,20)
> exten => t,1,Hangup
> include => extensions

Assuming that you have context=incoming on your fxo channels in zapata.conf,
then the above context=incoming should be okay for starters.

In your sip.conf file, the type=friend should not have host=64.123.157.103,
as 'friend' implies the phone is registering with asterisk and therefore
asterisk knows the IP from that registration.

In sip.conf, your start out with context=local-access and then define
extension 20 within "that" context. But, in extensions.conf you don't
have any definitions for [local-access]. It kind of looks like you
should change context=local-access to context=extensions in your
extensions.conf file.

If you can't make the phone operate without the host= statement, then
debug why the phone isn't registering correctly.


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