[Asterisk-Users] FREE BSD

[Julio Tejera]

Yes !

Go to on the wiki ....

http://www.voip-info.org/wiki-Asterisk+FreeBSD

On 5.2 or higher there is also a "port"

-------
Ing. Julio Alvarez Tejera
Unix Trends
*BSD, Solaris & Linux
---------------
"extremely stable systems"
----- Original Message -----
From: "Alvaro Gonzalez" <alvaro.gonzalez at globaldatainternational.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Wednesday, December 15, 2004 7:35 AM
Subject: [Asterisk-Users] FREE BSD


> 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.
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>