Re: [asterisk-users] My Asterisk Box was hacked
Hello! First of all, you should disable unused VoIP protocols. Than remove all guest accounts from used protocols, disable guest unauth access. Always use strong passwords for accounts, for users on your system. Passwords shouldn't be eq username. Move port binds on LAN network for all active services as much as you can (i.e. SHH should be on WAN too I think). Use iptables for blocking password bruteforce. Try to install fail2ban with jails for asterisk, ssh, HTTP and other public services. Then you can try to install PSAD (port scan autodetect) to prevent attacks. And never use default context in asterisk for word calls directions. And you should always keep your software up to date. There much more security issues than you think. Good Luck! On 21.07.2011 09:29, Malvin Rito wrote: Hi List, My asterisk box was hacked! Can anyone help on how do I secure my asterisk box, currently my box is installed with 2 NIC. 1st NIC is for LAN access and 2nd NIC has a public IP which is registered to our VoIP Provider. As I remember I already tried putting our Box on NAT but unfortunately due to some issue like call is dropped after 30 seconds and sometimes voice are not heard. Then we disable again the NAT. Your advise will be much appreciated. Thanks in advance. Regards, Malvin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My Asterisk Box was hacked
Yeap, drop out box is normal idea. But it's strongly wired what type of hack was. If it was only traffic leak without any footsteps in your system (shell history, files modification time, logs) I don't think that box couldn't be used any longer. Try to use port knocking ( http://www.portknocking.org/ ) for opening SSH ports for more secure access. And if you have enough time, box could be reinstalled. Malvin Rito is right. Attacker could place rootkit on your system that couldn't easily detected. On 21.07.2011 10:31, Steve Edwards wrote: On 21.07.2011 09:29, Malvin Rito wrote: My asterisk box was hacked! On Thu, 21 Jul 2011, Захаров Антон wrote: First of all, you should disable unused VoIP protocols. Once a box has been hacked you cannot trust anything. Disconnect the box from the network, save whatever DATA ONLY you cannot live without, DBAN the disk and start over. Before you re-install the OS, read up on what you should have done the first time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pridialplan/ prilocaldialplan
I don't think so. I think it's a predefined pairs of TON and NPI, that could not be set separately. On 24.05.2011 04:35, Rafael dos Santos Saraiva wrote: did not work!! Bug in Asterisk?? :( Rafael 2011/5/20 Захаров Антон ins...@mail.ru mailto:ins...@mail.ru Yeap, I couldn't set Private TON too. Try to set all _prefix variables in chan_dahdi.conf and use dynamic prilocaldialplan. On 19.05.2011 21:30, Rafael dos Santos Saraiva wrote: Hi I change the chan_dahdi.conf and restart dahdi: prilocaldialplan=private pridialplan=private But, in debug i see the following informations: 1 Calling Number (len= 8) [ Ext: 0 TON: *National Number (2) * NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation permitted, user number not screened (0) '1570' ] 1 [70 0a 80 30 38 31 37 34 37 39 35 36] 1 Called Number (len=12) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '81747956' ] I set Private TON, but display National TON. Thank's Att, Rafael Saraiva 2011/5/19 Захаров Антон ins...@mail.ru mailto:ins...@mail.ru Hello. To apply this settings you should restart dahdi (dahdi restart in CLI). About influence you could read here: http://markmail.org/message/rpd2aewiu2soostz On 19.05.2011 06:05, Rafael dos Santos Saraiva wrote: Hi I'm beginner in list. I have doubts about the options pridialplan and prilocaldiaplan in chan_dahdi.conf. I interconnect the Asterisk with a Siemens PBX, but i saw that the changes in the file do not take effect in debug of the span or calling/called number. How to use this options? In that cases to use? Ps.: sorry for the english, i'm brazilian. Thanks -- Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pridialplan/ prilocaldialplan
Yeap, I couldn't set Private TON too. Try to set all _prefix variables in chan_dahdi.conf and use dynamic prilocaldialplan. On 19.05.2011 21:30, Rafael dos Santos Saraiva wrote: Hi I change the chan_dahdi.conf and restart dahdi: prilocaldialplan=private pridialplan=private But, in debug i see the following informations: 1 Calling Number (len= 8) [ Ext: 0 TON: *National Number (2) * NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation permitted, user number not screened (0) '1570' ] 1 [70 0a 80 30 38 31 37 34 37 39 35 36] 1 Called Number (len=12) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '81747956' ] I set Private TON, but display National TON. Thank's Att, Rafael Saraiva 2011/5/19 Захаров Антон ins...@mail.ru mailto:ins...@mail.ru Hello. To apply this settings you should restart dahdi (dahdi restart in CLI). About influence you could read here: http://markmail.org/message/rpd2aewiu2soostz On 19.05.2011 06:05, Rafael dos Santos Saraiva wrote: Hi I'm beginner in list. I have doubts about the options pridialplan and prilocaldiaplan in chan_dahdi.conf. I interconnect the Asterisk with a Siemens PBX, but i saw that the changes in the file do not take effect in debug of the span or calling/called number. How to use this options? In that cases to use? Ps.: sorry for the english, i'm brazilian. Thanks -- Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pridialplan/ prilocaldialplan
Hello. To apply this settings you should restart dahdi (dahdi restart in CLI). About influence you could read here: http://markmail.org/message/rpd2aewiu2soostz On 19.05.2011 06:05, Rafael dos Santos Saraiva wrote: Hi I'm beginner in list. I have doubts about the options pridialplan and prilocaldiaplan in chan_dahdi.conf. I interconnect the Asterisk with a Siemens PBX, but i saw that the changes in the file do not take effect in debug of the span or calling/called number. How to use this options? In that cases to use? Ps.: sorry for the english, i'm brazilian. Thanks -- Att, Rafael Saraiva -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Type of number in outgoing SETUP frame
Hello, all. Could you tell me how to set the type of number in the outgoing SETUP message sent over PRI trunk? I need to have: Called Number (len=18) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '318989263037666' ] But always have: Called Number (len=18) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '318989263037666' ] NPI should be ISDN/Telephony Numbering Plan (E.164/E.163) and TON: Unknown Number Type (0). My current chan_dahdi settings: switchtype=euroisdn pridialplan=dynamic prilocaldialplan=unknown unknownprefix = 3189 internationalprefix = +8 nationalprefix = +7 ;localprefix = +7495 ;privateprefix = +7499 Seems that 'unknownprefix' do not affect. I don't know how to influence on TON correctly and couldn't find this information in Internet. Thanks for your attention! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP bad request
Try to look in 'sip set debug peer user'. On 29.04.2011 18:10, Mike wrote: Hi, I have been getting reports phones ringing only a tiny moment and then going to voicemail. CLI output shows: -- SIP/user-0006fcdd is ringing -- Got SIP response 400 Bad Request back from 23.23.23.23 -- SIP/user-0006fcdd is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Which does explain it. How can I find the root cause of “bad request”? Call-limit is very high for this sip user, so I`m not reaching that limit for sure. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: No Internet, no asterisk
I have enabled DNS manager in /etc/asterisk/dnsmgr.conf. It helps me. On 19.04.2011 14:05, Niccolò Belli wrote: Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto: Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. srvlookup = no didn't help. What about putting my provider's name in /etc/hosts? Should it solve the problem? A caching nameserver is not a viable solution because I want it working even after a month without internet access. Cheers, Darkbasic P.S. Why nobody ever fixed this annoying bug? Is there a special reason behind? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Internet, no asterisk
Could 'dnsmgr' help? On 18.04.2011 14:30, A J Stiles wrote: On Monday 18 Apr 2011, Niccolò Belli wrote: As soon as the Internet connection goes down, phones stop working. I want to be able to use pstn, isdn and the gsm gateway even if the Internet connection goes down, how can I achieve it? You most probably are using a nameserver somewhere else on the Internet; and when Asterisk can't see the nameserver, everything stops working. The solution is to run your own nameserver. Install bind (probably easier from your distro's package repository than from source, since the former will include some sane config) on the Asterisk box, and reconfigure networking so as only to use 127.0.0.1 as a nameserver. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Asterisk / API / Perl
Hello!
Try to use ${CHANNEL} instead of agi_type.
It will be like this:
$typ = $AGI-get_variable('CHANNEL');
@tmp_array=split(/\//, $typ);
$typ = $tmp_array[0];
and
$src=$AGI-get_variable('cdr(src)');
On 05.03.2011 10:25, Olivier CALVANO wrote:
Hi
i want use the API on my asterisk 1.6, but i have a small problems :
In extension, i start it :
exten = _X.,3,AGI(My-Script.agi)
The perl agi file are started without problems
but i want get into this script a lot of variable:
Type (SIP or IAX)
src (from cdr)
but that's don't work:
use Asterisk::AGI;
use lib /var/lib/asterisk/agi-bin;
$AGI = new Asterisk::AGI;
$typ = $AGI-get_variable('agi_type');
$typ don't have SIP or IAX, same test without succes:
$typ = $AGI-get_variable('type');
anyone know this problems ?
thanks
Olivier
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Re: [asterisk-users] Asterisk 1.8.3
On 11.02.2011 12:37, Ishfaq Malik wrote: Hi Does anyone have any rough idea how far away 1.8.3 is? We can't deploy 1.8 yet because of this issue https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 Have you tried issue18403.patch ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Timing cable usage necessity
On 26.11.2010 17:29, David Backeberg wrote: 2010/11/25 Захаров Антонins...@mail.ru: Hello everyone. I have a timing slips errors and I can't understand what source of the problem is. My installation has 2 digium cards: TE420 and TE220 cards in one server. There are 3 spans (E1) to PSTN and 3 spans to internal PBS stations - normal installation for transit communication. Span configuration is: span=1,1,0,ccs,hdb3 #TE420 - first port. To PSTN. span=2,0,0,ccs,hdb3 #TE420 - second port. To PBX. span=3,2,0,ccs,hdb3 #TE420 - third port. To PSTN. span=4,0,0,ccs,hdb3 #TE420 - fourth port. To PBX. span=5,3,0,ccs,hdb3 #TE220 - first port. To PSTN. span=6,0,0,ccs,hdb3 #TE220 - second port. To PBX. I should to say, that PBXs are interconnected through router (doesn't know anything about it). So all schema looks like this: http://yfrog.com/jjschemaj Spans 1-5 works fine, but on span 6 (marked bold) I have rising timing slips counter. I think it's appearing because I'm getting a primary timing source on span 1 - first port on TE420. But TE220 doesn't use it's span 5 for timing source, because it has priority 3, so it could be a sync problem. Am I wrong? I'm started to think about timing cable for syncing timing on first card and second. Should I use it? It's a problem to bought cable in our city (Russia,Moscow). All resellers sell only cards. Could I use floppy or IDE cable to interconnect cards? As I see in picture of cable, it's a direct 16 pin cable. Does anybody know something about timing cable for different cards? How I can solve my problem? I can't answer your problem about how you find one in Moscow, but I can tell you that when I've installed two cards in a single server I have used the timing ribbon cable. I have no idea whether it made a difference, and I now build my servers differently. You could ask Digium directly, as they probably know their resellers. Something that may help you is that you can make sure you set the priority on the cards differently from each other. There is a dial with a pointer that tells you which card is prioritized. Whenever I had two in the same server I made sure that one card was set higher than the other. As I think, i forgot to set priority on TE420 and TE220. Could this be the cause of an slip error? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Timing cable usage necessity
Здравствуйте. Спасибо за ответ. Меня какраз интересуют проблемы, которые решает этот кабель. Достать его теоретически мы сможем. Другой вопрос, в чем же его уникальность? И почему, например, нельзя использовать floppy кабель? Когда я поставил floppy кабель вместо официального и модулю указал опцию timingcable=1, у меня либо вообще система падать стала (так, что коннект по ssh терялся), либо в asterisk я получал сообщение Bad HDLC и поток не поднимался. Hello. Thank you for your reply. I'm interested in problems that solves this cable. We can get it theoretically. Another question, what is its uniqueness? And why, for example, we can not use the floppy cable? When I put the floppy cable instead of an official and set the option module timingcable = 1, I either do a system has to fall (so that the connection was lost over ssh) or I get the message Bad HDLC in asterisk and the E1 flow was not raised. On 29.11.2010 10:16, Grigoriy Puzankin wrote: Здравствуйте, Антон. Мы заказывали кабель у Мототелекома (если не ошибаюсь). В наличии у них его нет, но под заказ с очередной поставкой они могут его достать. Либо купите через e-bay или вражеский интернет-магазин. У нас он соединяет две 4-портовые карточки. Не помню, что именно было до того, как его поставили, но какие-то проблемки были. 25.11.2010 19:23, Захаров Антон пишет: Hello everyone. I have a timing slips errors and I can't understand what source of the problem is. My installation has 2 digium cards: TE420 and TE220 cards in one server. There are 3 spans (E1) to PSTN and 3 spans to internal PBS stations - normal installation for transit communication. Span configuration is: span=1,1,0,ccs,hdb3 #TE420 - first port. To PSTN. span=2,0,0,ccs,hdb3 #TE420 - second port. To PBX. span=3,2,0,ccs,hdb3 #TE420 - third port. To PSTN. span=4,0,0,ccs,hdb3 #TE420 - fourth port. To PBX. span=5,3,0,ccs,hdb3 #TE220 - first port. To PSTN. span=6,0,0,ccs,hdb3 #TE220 - second port. To PBX. I should to say, that PBXs are interconnected through router (doesn't know anything about it). So all schema looks like this: http://yfrog.com/jjschemaj Spans 1-5 works fine, but on span 6 (marked bold) I have rising timing slips counter. I think it's appearing because I'm getting a primary timing source on span 1 - first port on TE420. But TE220 doesn't use it's span 5 for timing source, because it has priority 3, so it could be a sync problem. Am I wrong? I'm started to think about timing cable for syncing timing on first card and second. Should I use it? It's a problem to bought cable in our city (Russia,Moscow). All resellers sell only cards. Could I use floppy or IDE cable to interconnect cards? As I see in picture of cable, it's a direct 16 pin cable. Does anybody know something about timing cable for different cards? How I can solve my problem? Thanks for attention -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Timing cable usage necessity
Hello everyone. I have a timing slips errors and I can't understand what source of the problem is. My installation has 2 digium cards: TE420 and TE220 cards in one server. There are 3 spans (E1) to PSTN and 3 spans to internal PBS stations - normal installation for transit communication. Span configuration is: span=1,1,0,ccs,hdb3 #TE420 - first port. To PSTN. span=2,0,0,ccs,hdb3 #TE420 - second port. To PBX. span=3,2,0,ccs,hdb3 #TE420 - third port. To PSTN. span=4,0,0,ccs,hdb3 #TE420 - fourth port. To PBX. span=5,3,0,ccs,hdb3 #TE220 - first port. To PSTN. span=6,0,0,ccs,hdb3 #TE220 - second port. To PBX. I should to say, that PBXs are interconnected through router (doesn't know anything about it). So all schema looks like this: http://yfrog.com/jjschemaj Spans 1-5 works fine, but on span 6 (marked bold) I have rising timing slips counter. I think it's appearing because I'm getting a primary timing source on span 1 - first port on TE420. But TE220 doesn't use it's span 5 for timing source, because it has priority 3, so it could be a sync problem. Am I wrong? I'm started to think about timing cable for syncing timing on first card and second. Should I use it? It's a problem to bought cable in our city (Russia,Moscow). All resellers sell only cards. Could I use floppy or IDE cable to interconnect cards? As I see in picture of cable, it's a direct 16 pin cable. Does anybody know something about timing cable for different cards? How I can solve my problem? Thanks for attention -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Realtime and MySQL
[ivr_holiday] switch = Realtime/ivr_holid...@extensions where 'ivr_holidays' is context and 'extensions' is table On 01.10.2010 12:52, Phibee Network Operation Center wrote: Hi i am not a expert on Asterisk and search a lot of small information : I use Asterisk 1.6.1.4 with MySQL. That's work and in my extension.conf, i have: [as5300-incoming] switch = Realtime and in extconfig.conf extensions = mysql,general,VOIP_Extensions A lot of Extension are into the table VOIP_Extensions. I am search to know if i can add a : [beta-incoming] switch = Realtime but not use the table VOIP_Extensions but VOIP_Extensions_Beta Anyone know if it's possible ? (use two table for extension) Thanks Jerome SCHEVINGT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI
Hello everyone. I have server with 2E1 PCI card, asterisk 1.4.35, dahdi 2.4.0, libpri 1.4.12-beta2. One PRI trunk looks to PSTN and take a clocksource from telco. Another trunk looks to PBX with DECT system. Some outgoing calls from asterisk to PSTN drops. The last message that exists before hanging up process is: DEBUG[28467] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI/... This frame come when call already established. So, when it come, the call drops. FRAME_CONTROL (8) means 'Congestion' according to 'frame.h'. I have already set debug 6, verbose 6 and enabled EXTENSIVE debugging on span, but couldn't find incoming frame from telco with information of 'Congestion' on this channel. I want to debug this message. I want to know where the root of my problem. And I'm sure that it's only my problem. That's why I didn't create issue ticket on bug tracker. So default methods of debug didn't show me control frames. I have a call log: http://pastebin.mozilla-russia.org/107089 Part of the full log file at the moment when this FRAME have been got: http://pastebin.mozilla-russia.org/107090 Part of the full log file from start of the call to drop: http://pastebin.com/MphaCkiV I have from 3 to 5 call drops in hour so it's reproduce periodically : http://pastebin.com/rdnYR8dU http://pastebin.com/KUJDPd3C I'm sorry, but I can't remember what E1 card placed in server. But it could be Digium or OpenVox. Here is output of some commands: lspci: 02:00.0 Communication controller: Digium, Inc. Wildcard TE210P dual-span T1/E1/J1 card 3.3V (rev 02) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- INTx- Latency: 32 Interrupt: pin A routed to IRQ 16 Region 0: Memory at d010 (32-bit, non-prefetchable) [size=128] Kernel driver in use: wct4xxp Kernel modules: wct4xxp dmesg: wct4xxp :02:00.0: PCI INT A - GSI 16 (level, low) - IRQ 16 wct4xxp :02:00.0: Found TE2XXP at base address d010, remapped to f90d4000 wct4xxp :02:00.0: DMA memory base of size 2048 at f6829000. Read: f6829400 and Write f6829000 wct4xxp :02:00.0: Firmware Version: c01a wct4xxp :02:00.0: Burst Mode: On wct4xxp :02:00.0: FALC Framer Version: 2.1 or earlier wct4xxp :02:00.0: Board ID: 00 wct4xxp :02:00.0: Reg 0: 0x36829400 wct4xxp :02:00.0: Reg 1: 0x36829000 wct4xxp :02:00.0: Reg 2: 0xd018 wct4xxp :02:00.0: Reg 3: 0x wct4xxp :02:00.0: Reg 4: 0x wct4xxp :02:00.0: Reg 5: 0xd0100014 wct4xxp :02:00.0: Reg 6: 0xc01a wct4xxp :02:00.0: Reg 7: 0x1f00 wct4xxp :02:00.0: Reg 8: 0x wct4xxp :02:00.0: Reg 9: 0x wct4xxp :02:00.0: Reg 10: 0xd0100028 IRQ 16/wct2xxp: IRQF_DISABLED is not guaranteed on shared IRQs wct4xxp :02:00.0: Found a Wildcard: Wildcard TE210P dahdi_hardware: pci::02:00.0 wct4xxp+ d161:0210 Wildcard TE210P As I think, it's really Digium TE210P. But I don't think, it's a pci card problem because call drops exist only on outgoing calls and 98% DIDs is mobile phone numbers. Any suggestions how to see this frame and who was the sender? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
