once more :-) hi,
I see. No, that will not work with this box and the original firmware. :-( You could send me the pages and descriptions you found on manipulated firmwares for use with asterisk off this list. Then I can take a look at them and tell you, if it will work or what it will do. :-) Nice weekend to everyone! Martin ----- Original Message ----- From: "Manuel Dominguez" <[EMAIL PROTECTED]> To: <[email protected]> Sent: Saturday, July 29, 2006 10:57 AM Subject: [asterisk-users] Re: Fritz!Box Fon ATA Hi Martin, No exactly. The Fritz!Box is connected to Asterisk using SIP. Not a direct connection between FXS ports and Asterisk. I would like to use this box like a Sipura 3000. This Sipura has 1 FXO port and 1 FXS port. You can use and register these ports in Asterisk independently. You register de FXS port like a normal extension in SIP.conf and you can use the FXO port for outbound calls from any extension (SIP or analog phones using FXS ports). With Fritz!Box to redirect all the calls from ISDN to Asterisk the only possibility we found is in the Rufumleitung menu. But in this menu you can't select the FXO port to redirect to Asterisk. You must select the FXS port (FON 1 or 2). This is ok but you can't use these ports to add other extensions. I find much information people making new firmware, changing settings inside Linux, using in asterisk... but always in German. I try to translate with Google but it is really complicated and my English is also terrible. Thanks, Manuel Message: 3 Date: Fri, 28 Jul 2006 23:08:00 +0200 From: "Martin Schrott - Thinking-Systems" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Re: Fritz!Box Fon ATA To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Hi Manuel, :-) If I understood you correctly, Your Fritz!Box and Asterisk are also connected via the fxs Ports? Then you should also be able to send incoming calls to this ports. Search for settings of Nebenstellen, eingehende Anrufe or ankommende Gespräche... But I do not see, where the sence would be, when you also can send directly to a Sip extension?! When you connect Asterisk via the fxs Ports, then you could directly dial out, without a Direktruf/Calltrough and pin. But Fritz!box is not really very userfriendly and not at least flexible. You can hardly do special configurations. :-( I am happy, that the things work as i supposed them to do. Best greetings from Austria Martin ----- Original Message ----- From: "Manuel Dominguez" <[EMAIL PROTECTED]> To: <[email protected]> Sent: Friday, July 28, 2006 9:39 PM Subject: [asterisk-users] Re: Fritz!Box Fon ATA Hi Martin, you say only a bit of work? ;-) 1. Incoming Yes, works like you suggest me!! The problem is that using this method, it's not possible to use the FXS ports in the Fritz!Box like normal extensions from Asterisk. We only use it to forward calls to a SIP extension. 2. Outbound I don't understand exactly your comments but I think is working. I go to the Rufumleitung -> Durchwahl (Call Through) aktiv -> definierte Durchwahl. In the combo box "Durchwahl für Anrufe auf der Rufnummer" I select my connection to Asterisk. I write a PIN and in the combo box "Anrufe weiterverbinden über die Rufnummer" I select the Festnetz. >From a SIP phone, I make a call to the extension selected in "Durchwahl für Anrufe auf der Rufnummer". In that moment another tone appears, I enter the PIN and I can make an external call from the SIP phone. Thanks for you help & greeting from Spain Manuel -----Mensaje original----- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: viernes, 28 de julio de 2006 13:50 Message: 16 Date: Fri, 28 Jul 2006 13:53:50 +0200 From: "Martin Schrott - Thinking-Systems" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Re: Fritz!Box Fon ATA To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Hi again, should both be possible. With a bit of work ;-) 1. incoming. You will have to set Rufumleitung to your choosen sip destination. telefonie> Rufumleitung> set the fone, that is set up to be ringing to be forwarded to your sip extension. As named in your extensions.conf local context. All incoming calls should then be forwarded to your asterisk. 2. Outbound Not as easy. Maybe you can realize that as follows: Telefonie> Rufumleitung> Callthrough (Direktdurchwahl) You may be able to set internet calls from a given did to be presented a callthrough option. Set internetcalls e.g. from 12345 to callthrough via Festnetz (psd) Then it should be possible to dial through when calling from Asterisk to your Fritz!Box if your callerid is 12345. (Never tested this. But with a bit of luck and time you can do it :-) ) all the best hth Martin ----- Original Message ----- From: "Manuel Dominguez" <[EMAIL PROTECTED]> To: <[email protected]> Sent: Friday, July 28, 2006 12:13 PM Subject: [asterisk-users] Re: Fritz!Box Fon ATA Hi Martin, Thank you for your comments. I made more or less these settings and in this moment I can make call from de FXS port to asterisk and from asterisk to FXS ports. My problem it's the FXO part of this ATA. I want to redirect all the incoming ISDN calls to a SIP phone or to an autoatendant and to make outgoing calls from sip phones (asterisk). I'm not sure if it s possible make this work using this ATA and the necessary settings. Manuel ------------------------------ Message: 8 Date: Fri, 28 Jul 2006 09:59:07 +0200 From: "Martin Schrott - Thinking-Systems" <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Fritz!Box Fon ATA To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Hi Manuel, I do not know the ATA, but we have a Fritz!Box fon, wich maybe is equal in setting it up. If you have any problems understanding the german setup, you can contact me, so I can help you in translating the needed Words :-) Normally you only have to do this on the Webinterface: Telefonie> Internettelefonie> Internetrufnummern> Neue Internetrufnummer> Internetrufnummer: Your VOIP number, or if using with isdn, then the msn. Do not use Internetrufnummer zum Anmelden verwenden! Registrar: the ip or host of your provider or Asterisk. If you have a own Asterisk use yur ip adress. There is a bug using hostnames. Benutzername: Username Passwort / Kennwort : password Do only fill out this fields, then it should work. If you put in any proxy or Stun Servers it may not work. (our experience) hth, Martin ----- Original Message ----- From: "Manuel Dominguez" <[EMAIL PROTECTED]> To: <[email protected]> Sent: Friday, July 28, 2006 9:25 AM Subject: [asterisk-users] Fritz!Box Fon ATA Hi, I have bought a Fritz!Box Fon ATA in eBay. I'm trying to find information about configuration this box in Asterisk. Its possible use this box like a normal ATA (sipura 3000.) receiving and making ISDN calls from Asterisk? Somebody has information in English about this box? Some example settings? Another problem is that firmware is in German. I have tried to change it but was not possible to use a difference language. Some ideas? Any help would be greatly appreciated Manuel _______________________________________________ --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 _______________________________________________ --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 _______________________________________________ --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 _______________________________________________ --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
