Re: [asterisk-users] Is there a smarter way to ban expensive calls indial plan?
Hi, try to list the blocked numbers first! Then you should be able to use wildcards without a problem. :-) That was the solution for the same problem at our dialplan. hth Martin - Original Message - From: Chris Blunt To: asterisk-users@lists.digium.com Sent: Tuesday, August 01, 2006 4:16 PM Subject: [asterisk-users] Is there a smarter way to ban expensive calls indial plan? Hi List, I need a bit of advice please. I want to ban calls to expensive destinations such as cell phones. This is fairly simple here in the UK because all cell phone numbers begin with a 7 where as all geographic numbers begin 1 and 2 Elsewhere this is different, take Andorra for example all numbers begin 376, cell phone numbers are 3763, 3764 and 3765 So if I try the following dial plan my pattern always matches the first wild card Exten = _00376.,1,Dial(my iax terminiator) Exten = _003763.,1,Congestion Exten = _003764.,1,Congestion Exten = _003765.,1,Congestion I seem to have been able to fix this with adding an x after the 6 in the first extension to make the patterns all the same length and thus making a better match with the blocked numbers. Example: Exten = _00376x.,1,Dial(my iax terminiator) Exten = _003763.,1,Congestion Exten = _003764.,1,Congestion Exten = _003765.,1,Congestion This is just so long winded, and you can imagine doing this for a huge list of destinations. If any one can suggest an improved or more efficient way of doing this, I would be greatly appreciated! Best regards Chris -- Chris Blunt Entropy IT Ltd ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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
Re: [asterisk-users] Re: Fritz!Box Fon ATA
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: asterisk-users@lists.digium.com 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 asterisk-users@lists.digium.com 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: asterisk-users@lists.digium.com 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 asterisk-users@lists.digium.com 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
[asterisk-users] agentcallbacklogin Asterisk V1.210 and v1.4
Hello :-) I just read, that the agentcallbacklogin will be marked as depreciated in v1.4 and we should use dynamic members. What do you think of this? Is it possible to use the dynamic members instead with all the features? 1. Maybe somebody can give me a hint, how to set up the following with dynamic members: Now we use agentcallbacklogin for our agents to logon and tell the number, where to call them. When a call is waiting, they get dialed and have to accept the waiting call with #. (is this also possible with dynamic members? We cannot run makros out of a queue, so how can we request the Buttonpress of a #? Any ideas? Also the login was very easy with agentcallbacklogin and I think we would have to write our own for dynamic members, or is there an equal function? Maybe anybody of you can help me or has a example configuration. 2. Our Queues do ignore the leavewhenempty=yes I read, that there is a bug on that?! Does it work any way? How could we set that up? and By the way someting different: 3. I just added an applicationmap and made a featurekey for saying the callerid. When I press the defined Button *1 the Callee or caller gets the announcement, but after that goes on in the extensionplan and is not getting back to the other partie. The second one is hung up. Is there anything I can configure to prevent that? Thank you all and have a nice day/evening depending where you are ;-) Martin ___ --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
Re: [asterisk-users] Fritz!Box Fon ATA
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: asterisk-users@lists.digium.com 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
Re: [asterisk-users] Re: Fritz!Box Fon ATA
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: asterisk-users@lists.digium.com 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 asterisk-users@lists.digium.com 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: asterisk-users@lists.digium.com 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 -- Message: 9 Date: Fri, 28 Jul 2006 09:04:26 +0100 From: Steve Davies [EMAIL PROTECTED] Subject: Re: [asterisk-users] SNOM 360 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 7/28/06, Koopmann, Jan-Peter [EMAIL PROTECTED] wrote: On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote: Does anyone know how to set up QoS on the SNOM 360 ? Thanks. What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a Snom 360 that will manage things for you. AFAIK all you can do is tell the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the task of the equipment managing the bottleneck (firewall, router whatever) to use this information and manage your traffic accordingly. As I understand it, you can set a QoS priority if the phone is in a VLAN. When you configure the (Tagged) VLAN, you can specify the priority of the packets in the VLAN. Otherwise, newer firmware allows the setting of TOS values IIRC. Regards, Steve -- Message: 10 Date: Fri, 28 Jul 2006 10:11:35 +0200 From: Olivier MONNET [EMAIL PROTECTED] Subject: [asterisk-users] PAP2T always busy on incoming calls with zaptel To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message
Re: [asterisk-users] Re: Fritz!Box Fon ATA
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: asterisk-users@lists.digium.com 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 asterisk-users@lists.digium.com 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: asterisk-users@lists.digium.com 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 asterisk-users@lists.digium.com 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
Re: [asterisk-users] Transfers - No ringback or moh
Hi Mike, Hi all, really works. ;-) But that can not be the solution for the future? :-) Can it? I think there should be an ANSWER() implimented in the Transfer function to prevent this problem ... Or does anybody have other ideas? greetings, Martin - Original Message - From: Mike Dawson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 26, 2006 4:32 PM Subject: Re: [asterisk-users] Transfers - No ringback or moh I get round this bug by replacing: exten = X,1,Dial(sip/blah) with: exten = X,1,Answer exten = X,n,Dial(sip/blah) It means the call is in an answered state before it starts ringing but it doesn't seem to cause any major problems. Mike Martin Schrott - Thinking-Systems wrote: Hi all, I cannot exactly reproduce your problems, but I can tell you, what problem we have on this topic: a calles b. b takes the call and can speak to a. b sets up a attendend transfer (via the softkey configured in asterisk) to c and hears ringing. a hears music on hold. b hears ringing if c answeres and b hanges up, everything is fine. now the problem: if b hangs up, before c has answered (during ringing) a will loose the connection and also be hanged up. I think this should not happen! The transfer should automatically be changed to blind and a should get the ringing played back instead of b. Hope, you can understand my problem and may have any ideas or thoughts. Greetings and Thanks, Martin ___ --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
Re: [asterisk-users] Transfers - No ringback or moh
Hi all, I cannot exactly reproduce your problems, but I can tell you, what problem we have on this topic: a calles b. b takes the call and can speak to a. b sets up a attendend transfer (via the softkey configured in asterisk) to c and hears ringing. a hears music on hold. b hears ringing if c answeres and b hanges up, everything is fine. now the problem: if b hangs up, before c has answered (during ringing) a will loose the connection and also be hanged up. I think this should not happen! The transfer should automatically be changed to blind and a should get the ringing played back instead of b. Hope, you can understand my problem and may have any ideas or thoughts. Greetings and Thanks, Martin ___ --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
[asterisk-users] playback / stream file
Hi all, I wonder if there is a simple solution for my needs: I have a archive, that includes different soundfiles in mp3 format. Some 11khz some 44.1khz and so on. Now I want to play them back to a caller, but asterisk only can play them in mono 8khz. So is there a possibility to use any realtime function as we can do at the musiconhold with -flag to set 8khz mono? Or do we have to convert to wave and then to gsm? Hope there is a simple way to play back our files without converting them. Thanks a lot for your help. Martin www.thinking-systems.eu [EMAIL PROTECTED] we think a step beyond -- JEtzt gesünder Rauchen - In jedem Rauchverbot www.dasneuerauchen.com ___ --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
