Re: [Asterisk-Users] pridialplan/TON question
Hi Peter! FYI: Yesterday i put Asterisk between a Hicom 350E and a Telekom Austria (TA) PRI. Both use TON=unknown for called number, but Hicom always uses TON=international for calling number whereas TA uses a dynamic TON for calling number. Thus, for incoming calls (PSTN-PBX) the presented caller number will be incorrect = CALLINGTON is needed to fix this. Peter, please send me the patch. regards, Klaus Peter Svensson wrote: On Tue, 26 Apr 2005, Marc Storck wrote: I have a Digium E100P card, with an EuroISDN PRI E1. On incoming calls the CALLINGTON variable is empty. I have the latest stable version of asterisk. Do I have to use another variable or is the TON only support in CVS? CALLINGTON was not populated in -stable. Tha patch was only added to -head. It is not that hard to add, I can send you our old patch if you want it. Peter ___ Asterisk-Users mailing list Asterisk-Users@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@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
Re: [Asterisk-Users] pridialplan/TON question
Klaus Darilion wrote: Hi! I'm trying to understand how asterisk handles the TON (using the pridialplan=... directive). Setting the TON for outgoing calls using pridialplan and prilocaldialplan works fine. But how can I query and process the TON for incoming calls? e.g. in the follwing scenario: PBX--- asterisk PSTN 1. The PBX sends SETUP messages with the appropriate TON. I want to rewrite the called number into a common format to make an ENUM lookup. Thus, I need to query the TON sent by the PBX and add the correct prefixes. 2. Further, in case of unsuccessful ENUM lookups, I want to forward the SETUP message to the PSTN, again using the appropriate TON. CVS version allows the setting of pridialplan=dynamic. But I want to use stable as this is for a stable machine. Can I implement this with stable asterisk? I always thought that if you set pridialplan=unknown the telco would not munge the digits. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] pridialplan/TON question
Eric Wieling aka ManxPower wrote: Klaus Darilion wrote: ... e.g. in the follwing scenario: PBX--- asterisk PSTN 1. The PBX sends SETUP messages with the appropriate TON. I want to rewrite the called number into a common format to make an ENUM lookup. Thus, I need to query the TON sent by the PBX and add the correct prefixes. 2. Further, in case of unsuccessful ENUM lookups, I want to forward the SETUP message to the PSTN, again using the appropriate TON. CVS version allows the setting of pridialplan=dynamic. But I want to use stable as this is for a stable machine. Can I implement this with stable asterisk? I always thought that if you set pridialplan=unknown the telco would not munge the digits. Anyway, if I set TON to unknown, I have to send the number according to the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the PBX does not use UNKNOWN, I have to translate the numbers out of their original TON to ton=unknown. Therefore, I need to process the incoming TON. How do I handle this? regards, klaus ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] pridialplan/TON question
On Tue, 26 Apr 2005, Klaus Darilion wrote: Anyway, if I set TON to unknown, I have to send the number according to the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the PBX does not use UNKNOWN, I have to translate the numbers out of their original TON to ton=unknown. Therefore, I need to process the incoming TON. How do I handle this? You have two options: 1) Use the CALLINGTON variable in the dialplan. This is only for the calling party number, not the called party number. 2) Use the internationalprefix, nationalprefix, localprefix etc settings in the zapata.conf file. I _think_ this will affect both the interpretation of calling and called party and possibly also the TON of the called number for outgoing links. I am nut sure under which circumstances these variables are applied. Isdn handling in Asterisk tends to be these kinds of hacks. Peter ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] pridialplan/TON question
Anyway, if I set TON to unknown, I have to send the number according to the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the PBX does not use UNKNOWN, I have to translate the numbers out of their original TON to ton=unknown. Therefore, I need to process the incoming TON. How do I handle this? You have two options: 1) Use the CALLINGTON variable in the dialplan. This is only for the calling party number, not the called party number. 2) Use the internationalprefix, nationalprefix, localprefix etc settings in the zapata.conf file. I _think_ this will affect both the interpretation of calling and called party and possibly also the TON of the called number for outgoing links. I am nut sure under which circumstances these variables are applied. Isdn handling in Asterisk tends to be these kinds of hacks. I have a Digium E100P card, with an EuroISDN PRI E1. On incoming calls the CALLINGTON variable is empty. I have the latest stable version of asterisk. Do I have to use another variable or is the TON only support in CVS? Marc -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] pridialplan/TON question
Hi Peter! Peter Svensson wrote: On Tue, 26 Apr 2005, Klaus Darilion wrote: Anyway, if I set TON to unknown, I have to send the number according to the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the PBX does not use UNKNOWN, I have to translate the numbers out of their original TON to ton=unknown. Therefore, I need to process the incoming TON. How do I handle this? You have two options: 1) Use the CALLINGTON variable in the dialplan. This is only for the calling party number, not the called party number. Bad thing. I guess this is an important feature when interacting with existing PBXs. How are other people deal with this (processing the TON of the called number)? 2) Use the internationalprefix, nationalprefix, localprefix etc settings in the zapata.conf file. I _think_ this will affect both the interpretation of calling and called party and possibly also the TON of the called number for outgoing links. I am nut sure under which circumstances these variables are applied. Nothing of this is included in stable version. I'm sure I'm not the first person putting an asterisk box between a PBX and the telco line. Is everboy using asterisk CVS out there? regards, klaus ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] pridialplan/TON question
Marc Storck wrote: Anyway, if I set TON to unknown, I have to send the number according to the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the PBX does not use UNKNOWN, I have to translate the numbers out of their original TON to ton=unknown. Therefore, I need to process the incoming TON. How do I handle this? You have two options: 1) Use the CALLINGTON variable in the dialplan. This is only for the calling party number, not the called party number. 2) Use the internationalprefix, nationalprefix, localprefix etc settingsin the zapata.conf file. I _think_ this will affect both theinterpretation of calling and called party and possibly also theTON of the called number for outgoing links. I am nut sure underwhich circumstances these variables are applied. Isdn handling in Asterisk tends to be these kinds of hacks. I have a Digium E100P card, with an EuroISDN PRI E1. On incoming calls the CALLINGTON variable is empty. I have the latest stable version of asterisk. Do I have to use another variable or is the TON only support in CVS? CALLINGTON is only in CVS. internationalprefix, ... is only in CVS or using stable patched with bristuff. regards, klaus ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] pridialplan/TON question
On Tue, 26 Apr 2005, Klaus Darilion wrote: You have two options: 1) Use the CALLINGTON variable in the dialplan. This is only for the calling party number, not the called party number. Bad thing. I guess this is an important feature when interacting with existing PBXs. How are other people deal with this (processing the TON of the called number)? It would not be very hard to create a CALLEDTON variable. The information is sent from libpri to chan_zap. A few more fields in chan_zap and a little bit of code in pbx.c. I really _ really_ wish asterisk would stop using the pseudo-variables and simply store stuff in the dialplan variables (like PRI_CAUSE etc already do). These pseudo-variables are stupid since in most cases reading and writing them is not time critical. 2) Use the internationalprefix, nationalprefix, localprefix etc settings in the zapata.conf file. I _think_ this will affect both the interpretation of calling and called party and possibly also the TON of the called number for outgoing links. I am nut sure under which circumstances these variables are applied. Nothing of this is included in stable version. I'm sure I'm not the first person putting an asterisk box between a PBX and the telco line. Is everboy using asterisk CVS out there? We use cvs from an old date (predating these functions) but with quite a few additional patches of our own. there is currenctly a showstopper bug where the dtmf-detecting dsp is disabled on outbound call legs. Bad if you need #-transfers. Peter ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] pridialplan/TON question
On Tue, 26 Apr 2005, Marc Storck wrote: I have a Digium E100P card, with an EuroISDN PRI E1. On incoming calls the CALLINGTON variable is empty. I have the latest stable version of asterisk. Do I have to use another variable or is the TON only support in CVS? CALLINGTON was not populated in -stable. Tha patch was only added to -head. It is not that hard to add, I can send you our old patch if you want it. Peter ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] pridialplan/TON question
Peter Svensson wrote: Nothing of this is included in stable version. I'm sure I'm not the first person putting an asterisk box between a PBX and the telco line. Is everboy using asterisk CVS out there? We use cvs from an old date (predating these functions) but with quite a few additional patches of our own. there is currenctly a showstopper bug where the dtmf-detecting dsp is disabled on outbound call legs. Bad if you need #-transfers. Looks like most of those CVS features are also in bristuff (patch to stable-1.0.6) regards, klaus ___ Asterisk-Users mailing list Asterisk-Users@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
