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" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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 > [EMAIL PROTECTED] > 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 > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
