[asterisk-users] IAX and rsa
Hi I am tyring to connect two * boxes over IAX with rsa, but I am having a slight problem. It just doesn't work. My configuration looks like this: iax.conf on box 1 [asterisk2] type=friend context=main auth=rsa inkey=asterisk2.mydomain.com outkey=asterisk1.mydomain.com host=asterisk2.mydomain.com extensions.conf looks like this: exten = _XX.,1,Dial(IAX2/asterisk2/${EXTEN}) iax on box 2 [asterisk1] type=friend context=main auth=rsa inkey=asterisk1.mydomain.com outkey=asterisk2.mydomain.com host=asterisk1.mydomain.com extensions.conf looks like this exten = _XX.,1,Dial(IAX2/asterisk1/${EXTEN}) I generated the key with astgenkey -n asterisk1.mydoamin.com on box 1 and astgenkey -n asterisk2.mydomain.com on box 2. I have also exchanged the .pub files between the servers. When I try to call, I can see on a console that the call is not authenticated. I know I did something wrong (but what?). Is it possible to have rsa authentication with type=friend? Any help would be appreciated. Cheers Andrutto -- Zobacz samochody przyszlosci! http://link.interia.pl/f199d ___ --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] H323
Hi What is the best solution for H323 in asterisk -- h323 in source, -- oh323 or -- ooh323c? which is most robust and reliable? Which supports gatekeeper functionality? Best wishes Andrutto -- Najnowsze fakty!!! http://link.interia.pl/f1996 ___ --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] NAT problems
Hi, Does anyone know how to solve this issue. I have Asterisk box on public IP and three clients connected to it. Unfortunately they are behind NAT (simple one-to-one). Those three clients can make outgoing calls hassle free, but when I try to make a call between them something is not right. I am using Linksys PAP-2 (two clients are connected to it) and one phone connected to planet VIP-156. When I try to make call between the phones connected to Linksys I am getting 488 Not Acceptable Here and when I try to reach the phone connected to planet I am getting silence after answer, but the phone can ring so I think that it is a RTP issue. I know that it is caused by the NAT, does anyone know how can I configure this to work appropriately. Cheers Andrutto -- Zostan Dziewczyna Lata! http://link.interia.pl/f1997 ___ --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] Re: NAT problems
Strange?!? These three phones are using g726 (this codec is configured in sip.conf and in SIP ATA as well). -- Zostan Dziewczyna Lata! http://link.interia.pl/f1997 ___ --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] extensions.conf
Hi Does anyone know how big extensions.conf can be? I am trying to set up Asterisk which will have about 45 000 lines in extensions.conf. Is there any limitation about the amount of lines in that file? Cheers Andrutto -- Zobacz nowosci salonu moto w Madrycie http://link.interia.pl/f1961 ___ --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] RE: extensions.conf
Hi Write a perl script that generates a mock 45,000 extensions.conf file, with 45,000 incrementing extensions, throw in a couple of contexts. Start Asterisk and see what happens. I will do that first thing in the morning :) Actually i've done 50,000+ line dialplans using my Asterisk::LCR dialplan generator, and asterisk has been just fine with it. Sound of relief :) Thanks for help I will let you know about the results. Cheers -- Zobacz nowosci salonu moto w Madrycie http://link.interia.pl/f1961 ___ --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] Problem after upgrade to 1.2.7.1
Hi Last friday I have upgraded to Asterisk 1.2.7.1 (bristuff-0.3.0-PRE-1p.tar.gz). Since that I have a problem with my Asterisk box. I am receiving these messages: May 24 09:30:11 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2). May 24 09:30:12 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2). May 24 09:31:03 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 1). May 24 09:31:13 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 4, stat = 0xff, card = 2). May 24 09:31:14 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2). Before the update this problem didn't exist. My configuration looks like this: card 0 ISDN NT mode card 1 ISDN TE mode card 2 ISDN TE mode card 3 TDM400P I haven't got any interrupts issues CPU0 CPU1 0: 336828 0 XT-PIC timer 1: 2 0 XT-PIC keyboard 2: 0 0 XT-PIC cascade 5: 26801377 0 XT-PIC zaphfc 7: 26801425 0 XT-PIC zaphfc 10:3350188 0 XT-PIC wctdm 11: 18702 0 XT-PIC eth0 12: 26801422 0 XT-PIC zaphfc 14: 8297 0 XT-PIC ide0 NMI: 0 0 LOC: 336781 336779 ERR: 2 MIS: 0 I tried to use google but unfortunately I haven't found anything helpful. Does anyone can tell what is wrong? Is it because Asterisk 1.2.7.1 or because the bristuff. Or maybe I did something wrong??? I can make and receiving calls with no problems. But the messages are very annoying. Please help!!! Cheers Andrutto -- Tysiace zdjec swietnych samochodow! http://link.interia.pl/f1944 ___ --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] Alarmreceiver
Hi, I just want to ask if anyone has some experience with Alarmreceiver application in * 1.2? Is this application reliable (according to wiki it isn't)? I managed to communicate Asterisk (Alarmreceiver) with a burglar alarm, but it behaves very strange. Sometimes alarmreceiver is able to get some events but sometimes not. Maybe there are some other non commercial applications which work under linux? Andrutto Cheers --- Fotoerotica! http://link.interia.pl/f1904 ___ --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] RE: Fax, txfax -bizarre thing
Hi, andrutto wrote: I was wondering why asterisk - great telecommunication program - has such a weak fax support. Because it's a PBX and not a fax server. Yeah, but to traditional PBX central you can plug fax machine hassle free. I don%u2019t want to use such a big thing like fax server (another machine and another cost ). I just want to connect to Asterisk one or two fax devices like a normal phone and use them in traditional way I would say paper way. Is this going to work with Asterisk. Or fax plus Asterisk is just something that will not work. Best regards Andrutto -- Kliknij po wiecej! http://link.interia.pl/f18ed ___ --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] RE: Fax, txfax -bizarre thing
Hi, I having similar problem. Unfortunately each thread is archive leads to nowhere. I read a post in which similar problem was solved by changing rxgain and txgain to 15. Maybe this would help. Does anyone have common problems? I was wondering why asterisk - great telecommunication program - has such a weak fax support. I am talking about mail to fax and fax to mail. Or maybe I am the only one who has the problems with it. If someone has some experience please help. Best wishes Andrutto -- Kliknij po wiecej! http://link.interia.pl/f18ed ___ --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] Postgres
Hi, I am new to Asterisk problems. Could anyone tell me how to install asterisk with postgres cdr feature. Because I install asterisk 1.2 from newest Bristuff and I do not have it Thanks in advance Cheers Andrutto Szukasz pracy? Szukamy pracownikow: dziennikarzy, webmasterow, specjalisty ds. badan, promotion managera, administratorow baz danych, programistow Windows i wielu innych! http://link.interia.pl/f18e6 ___ --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] Weird behaviour
Hi, I noticed this weird behavior - in my office I use mixed phone technology. I use Sip and Zap phones, analog and ISDN. I also defined a pickup feature and everything works prima to the time when I want to pickup call with ISDN phone. The console says (when I press my pickup extension *6) no such extension. Why? Other phones do not need special definition in extension.conf. When I added extra line in my dialplan - exten = *6,1,Pickup(group) - the ISDN phone was able to pickup all channels instead of Sip. To be certain the Sip channel was picked up, but the phone did not stop to ring. All phones are in the same pickup and call group. Do I need something special in my zapata.conf to normally pickup all channels or do I have to add some application to asterisk? Or I just did something wrong? Regards Andrutto -- Prawie 40.000 samochodow na sprzedaz! http://link.interia.pl/f18b2 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] VIP-050
Hi, I want to extend my asterisk stuff and buy some Planet devices, to be certain I'm going to buy PLANET VIP-050 with FXO and FXS modules. Has anyone heard about it. Is it compatible with Asterisk, or it would cause a lot of problems. Dose anyone have some experience with it?? All the best Andrutto -- Oferty sprzedazy samochodow... http://link.interia.pl/f18b1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Re: Re: Nokia 32 Terminal
AbdelRahman Tarzi ha scritto: If you wish to connect it to an FXS you will need a special cable which Nokia sells.. you don't really need a special cable for FXS, the cable is a standard phone cable with a j11 4/6 pin plug. Just read the tech manual from the nokia website for the pinout. Connecting to an FXO (which expects a line) is the default. Check the normal stuff (like dialstring) before you suspect the device.. They're really maintenance-free !! I have a problem with the external antenna. No signal gain with it connected to the nokia 32 terminal. You can play with the AT command via serial port to see the signal quality level, and I advice you to disable the gsm call waiting service. Sergio Hi, I tried everything with no success. I even restore the factory defaults but without positive effect (call waiting service was disable). The terminal could not be broken because I receive calls. I checked the monitor and the signal was very strong. Dose anyone have some other tips. Regards Andrutto -- Startuj z INTERIA.PL! http://link.interia.pl/f186c ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Nokia 32 Terminal
Hi, Does anyone have some experience with Nokia 32 Terminal (it is an analog GSM Gateway)? After a configuration I can make only incoming calls, I'm not able to do any outgoing. Nokia signalize an error (4 short tones), when I try to phone someone. I tried postpaid simcards as well as prepaid simcards with the same result. Does anyone try to connect this gateway to Asterisk PBX if so what were the results? All the best Andrutto -- Jedyny taki czat... http://link.interia.pl/f18b0 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sipura SPA-3000 strange behaviour
Hi, Few days ago I bought a Sipura SPA-3000 Gateway. Outgoing calls works fine but incomming calls behave very strange. When I dial my Sipura from outside and cancel befor picking up, the phone still rings for about one minut. What is wrong - the Sipura Gateway or I did something wrong with configuration. Does anyone know what I should do with this problem :). All the best Andrutto - Najwiekszy MOTO-serwis w Polsce http://link.interia.pl/f18af ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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