Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes.
Hi, I don't use a macro. I stay in the same dialplan (application) In the h exten I place a test (for example testThis is a test/test) If I look at the CLI and after I placed the example text in the variable CDR(Userfield), I see (with NoOp) that example text is placed in CDR(Userfield). But if I look in de Master.csv, I see that the example text is not the CDR(userfield) -- Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Tilghman Lesher Verzonden: 05-04-2011 00:08 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes. On Monday 04 April 2011 06:58:23 Arjan Kroon | Mobillion wrote: Hi, Does anybody have a solution to this problem? Because in this issue the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 The h extension should be in the context from which the Macro was called, not in the Macro context itself. -- Tilghman -- _ -- 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] Asterisk = Request a Code
Hi i want add a numeric password to a call in : User call to a number, Asterisk answer and request: please insert your pin code the user enter a numeric code of 4 number and # when asterisk have the code, he start a api. Anyone have a sample of extension.conf for this ? thanks Olivier -- _ -- 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] CDR fields not being written from h extension after Dial command completes.
Hi, If I try to call out with Queue mechanism and the call is answered then hangup, the CDR(userfield) in the h exten is placed in the CDR. So for now I see that this problem only occurs with a Dial in the dialplan. -- Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Arjan Kroon | Mobillion Verzonden: 05-04-2011 08:21 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes. Hi, I don't use a macro. I stay in the same dialplan (application) In the h exten I place a test (for example testThis is a test/test) If I look at the CLI and after I placed the example text in the variable CDR(Userfield), I see (with NoOp) that example text is placed in CDR(Userfield). But if I look in de Master.csv, I see that the example text is not the CDR(userfield) -- Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Tilghman Lesher Verzonden: 05-04-2011 00:08 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes. On Monday 04 April 2011 06:58:23 Arjan Kroon | Mobillion wrote: Hi, Does anybody have a solution to this problem? Because in this issue the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 The h extension should be in the context from which the Macro was called, not in the Macro context itself. -- Tilghman -- _ -- 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 -- _ -- 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 crashes on high IO load
Did you take a look at /var/log/syslog /var/log/asterisk/messages ? Using Debian? Take a look at iotop (apt-get install iotop). There you can see information about which process consumes high io load. Am 04.04.2011 17:23, schrieb Maximilian Grobecker: Hello Thorsten, the system has 4 GB RAM and about 2,5 GB free so swap space is not used or exhausted. Maybe the high load is not cause of this crashes but it's the only thing the crashes can be reproduced with. Thank you! Maximilian Grobecker Am 04.04.2011 16:03, schrieb Thorsten Göllner: Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- 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 -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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] CDR fields not being written from h extension after Dial command completes.
Hi, New update. When I use the option g in a dial then the CDR fields are not updated. When I perform a dial without the option g, for example rR then the CDR field will be written in the h exten. So therefore I lose the g option in the dial. -- Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Arjan Kroon | Mobillion Verzonden: 05-04-2011 09:32 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes. Hi, If I try to call out with Queue mechanism and the call is answered then hangup, the CDR(userfield) in the h exten is placed in the CDR. So for now I see that this problem only occurs with a Dial in the dialplan. -- Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Arjan Kroon | Mobillion Verzonden: 05-04-2011 08:21 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes. Hi, I don't use a macro. I stay in the same dialplan (application) In the h exten I place a test (for example testThis is a test/test) If I look at the CLI and after I placed the example text in the variable CDR(Userfield), I see (with NoOp) that example text is placed in CDR(Userfield). But if I look in de Master.csv, I see that the example text is not the CDR(userfield) -- Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Tilghman Lesher Verzonden: 05-04-2011 00:08 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes. On Monday 04 April 2011 06:58:23 Arjan Kroon | Mobillion wrote: Hi, Does anybody have a solution to this problem? Because in this issue the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 The h extension should be in the context from which the Macro was called, not in the Macro context itself. -- Tilghman -- _ -- 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 -- _ -- 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] Asterisk 1.8 and new the command: exten = _X., 4, Wait, 2
OK Dears; Is the exten = _X.,2,Wait,2 no more working with Asterisk 1.8? What is the equivalent? I installed Asterisk 1.8 and Star2Billing 1.9 but I am getting this error if someone can advise me: Executing [9615806234@a2billing:1] Answer(SIP/gwsshihabuddinkw-0014, ) in new stack [Apr 5 02:59:05] WARNING[2941]: pbx.c:4055 pbx_extension_helper: No application 'Wait,2' for extension (a2billing, 9615806234, 2) == Spawn extension (a2billing, 9615806234, 2) exited non-zero on 'SIP/gwsshihabuddinkw-0014' Now, my investigations: The extensions.conf: [a2billing] exten = _X.,1,Answer exten = _X.,2,Wait,2 exten = _X.,3,DeadAGI,a2billing.php exten = _X.,4,Wait,2 exten = _X.,5,Hangup ; From the other side: I did installations for Star2Billing version 1.9, I copied the a2billing.conf to the /etc/, also I enabled the manager.conf with port 5038. I copied a2billing.php and the lib to the agi-bin directory and I ran chmod +x for a2billing.php to make sure it is executable. And my php packages are: [root@Call-Bilal asterisk]# rpm -qa | grep php php-pgsql-5.2.9-2.fc10.i386 php-pear-1.7.2-2.fc10.noarch php-common-5.2.9-2.fc10.i386 php-pdo-5.2.9-2.fc10.i386 php-mbstring-5.2.9-2.fc10.i386 php-cli-5.2.9-2.fc10.i386 php-5.2.9-2.fc10.i386 php-mysql-5.2.9-2.fc10.i386 php-imap-5.2.9-2.fc10.i386 php-gd-5.2.9-2.fc10.i386 Note: I am able to place a normal call frome extension to extension but using a2billing, no success. Any help? Regards Bilal -- _ -- 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 1.8 and new the command: exten = _X., 4, Wait, 2
Change Wait,2 to wait(2) -Original Message- From: bilal ghayyad bilmar...@yahoo.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 5 Apr 2011 01:31:11 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.8 and new the command: exten = _X., 4, Wait, 2 OK Dears; Is the exten = _X.,2,Wait,2 no more working with Asterisk 1.8? What is the equivalent? I installed Asterisk 1.8 and Star2Billing 1.9 but I am getting this error if someone can advise me: Executing [9615806234@a2billing:1] Answer(SIP/gwsshihabuddinkw-0014, ) in new stack [Apr 5 02:59:05] WARNING[2941]: pbx.c:4055 pbx_extension_helper: No application 'Wait,2' for extension (a2billing, 9615806234, 2) == Spawn extension (a2billing, 9615806234, 2) exited non-zero on 'SIP/gwsshihabuddinkw-0014' Now, my investigations: The extensions.conf: [a2billing] exten = _X.,1,Answer exten = _X.,2,Wait,2 exten = _X.,3,DeadAGI,a2billing.php exten = _X.,4,Wait,2 exten = _X.,5,Hangup ; From the other side: I did installations for Star2Billing version 1.9, I copied the a2billing.conf to the /etc/, also I enabled the manager.conf with port 5038. I copied a2billing.php and the lib to the agi-bin directory and I ran chmod +x for a2billing.php to make sure it is executable. And my php packages are: [root@Call-Bilal asterisk]# rpm -qa | grep php php-pgsql-5.2.9-2.fc10.i386 php-pear-1.7.2-2.fc10.noarch php-common-5.2.9-2.fc10.i386 php-pdo-5.2.9-2.fc10.i386 php-mbstring-5.2.9-2.fc10.i386 php-cli-5.2.9-2.fc10.i386 php-5.2.9-2.fc10.i386 php-mysql-5.2.9-2.fc10.i386 php-imap-5.2.9-2.fc10.i386 php-gd-5.2.9-2.fc10.i386 Note: I am able to place a normal call frome extension to extension but using a2billing, no success. Any help? Regards Bilal -- _ -- 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] Asterisk = Request a Code
-Mensaje original- Olivier CALVANO Enviado el: martes, 05 de abril de 2011 9:16 Hi i want add a numeric password to a call in : User call to a number, Asterisk answer and request: please insert your pin code the user enter a numeric code of 4 number and # when asterisk have the code, he start a api. Anyone have a sample of extension.conf for this ? A couple of suggestions: * CLI core show application Authenticate * http://www.voip-info.org/wiki/view/Asterisk+user+authentication Best Regards, José Miguel Este correo electrónico y, en su caso, cualquier fichero anexo al mismo, contiene información de carácter confidencial exclusivamente dirigida a su destinatario o destinatarios. Si no es vd. el destinatario indicado, queda notificado que la lectura, utilización, divulgación y/o copia sin autorización está prohibida en virtud de la legislación vigente. En el caso de haber recibido este correo electrónico por error, se ruega notificar inmediatamente esta circunstancia mediante reenvío a la dirección electrónica del remitente. Evite imprimir este mensaje si no es estrictamente necesario. This email and any file attached to it (when applicable) contain(s) confidential information that is exclusively addressed to its recipient(s). If you are not the indicated recipient, you are informed that reading, using, disseminating and/or copying it without authorisation is forbidden in accordance with the legislation in effect. If you have received this email by mistake, please immediately notify the sender of the situation by resending it to their email address. Avoid printing this message if it is not absolutely necessary. -- _ -- 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 1.8 and new the command: exten = _X., 4, Wait, 2
Also change DeadAGI,a2billing.php to AGI(a2billing.php) -Original Message- From: bilal ghayyad bilmar...@yahoo.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 5 Apr 2011 01:31:11 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.8 and new the command: exten = _X., 4, Wait, 2 OK Dears; Is the exten = _X.,2,Wait,2 no more working with Asterisk 1.8? What is the equivalent? I installed Asterisk 1.8 and Star2Billing 1.9 but I am getting this error if someone can advise me: Executing [9615806234@a2billing:1] Answer(SIP/gwsshihabuddinkw-0014, ) in new stack [Apr 5 02:59:05] WARNING[2941]: pbx.c:4055 pbx_extension_helper: No application 'Wait,2' for extension (a2billing, 9615806234, 2) == Spawn extension (a2billing, 9615806234, 2) exited non-zero on 'SIP/gwsshihabuddinkw-0014' Now, my investigations: The extensions.conf: [a2billing] exten = _X.,1,Answer exten = _X.,2,Wait,2 exten = _X.,3,DeadAGI,a2billing.php exten = _X.,4,Wait,2 exten = _X.,5,Hangup ; From the other side: I did installations for Star2Billing version 1.9, I copied the a2billing.conf to the /etc/, also I enabled the manager.conf with port 5038. I copied a2billing.php and the lib to the agi-bin directory and I ran chmod +x for a2billing.php to make sure it is executable. And my php packages are: [root@Call-Bilal asterisk]# rpm -qa | grep php php-pgsql-5.2.9-2.fc10.i386 php-pear-1.7.2-2.fc10.noarch php-common-5.2.9-2.fc10.i386 php-pdo-5.2.9-2.fc10.i386 php-mbstring-5.2.9-2.fc10.i386 php-cli-5.2.9-2.fc10.i386 php-5.2.9-2.fc10.i386 php-mysql-5.2.9-2.fc10.i386 php-imap-5.2.9-2.fc10.i386 php-gd-5.2.9-2.fc10.i386 Note: I am able to place a normal call frome extension to extension but using a2billing, no success. Any help? Regards Bilal -- _ -- 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] call-limit bypass
I am using asterisk 1.4.2 and it usually does enforce the limit. yesterday and couple of times before was an exception. I am still trying to find the reason behind. Any more suggestions please? oh by the way * 1.8.1.1 does enforce the call limit, i tested it yesteday on sip channels. On Mon, Apr 4, 2011 at 11:48 PM, Bryant Zimmerman brya...@zktech.comwrote: From what I understand on the newer versions of asterisk call-limit does not limit calls anymore. You have to limit them from your code using call groups. From what I have seen on the 1.6x and 1.8 versions call-limit does not limit your call counts. We use code and the GROUP_COUNT to limit calls. If you use it right it is rock solid. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- *From*: Rizwan Hisham rizwanhas...@gmail.com *Sent*: Monday, April 04, 2011 12:30 PM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: [asterisk-users] call-limit bypass Hi everyone, one of our users last night bypassed asterisk call-limit limitation. I have no Idea how. Is it possible? Is there a bug in asterisk that can be manipulated for this purpose? The call-limit variable was to 2, and the user initiated 169 calls in 2 minutes each has duration at least 8 minutes. Please comment... Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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 -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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] dialplan is not finding my number asterisk 1.8.3
Jerry Geis wrote: I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a speaker attached. When asterisk first starts this works. In fact it works for some time. Then it just stops with this error on the CLI. [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_mediaport105' to extension '1105' rejected because extension not found in context 'smvoice-mediaport'. When doing the dialplan show it clearly in the context. [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1105' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] Its telling me it cannot find it. Its there - the dialplan shows its there. When I stop and start it works again for a little while. Matter of fact I just issued dialplan reload and calling into 1105 works again. Whats up? How do I get this to be consistent? Jerry I just looked in my extensions.conf and I do not have extenpatternmatchnew at all. My understanding is that it is off by default. my sip.conf has: register = mndemo_to_mediaport105:secret@mndemo ; Description: [mndemo_to_mediaport105] type=friend defaultname=mndemo_to_mediaport105 username=mndemo_to_mediaport105 secret=secret disallow=all allow=ulaw allow=alaw allow=gsm rtptimeout=60 host=192.168.1.58 context=smvoice-mediaport I was not aware I needed another context of : [mndemo_to_mediaport105] include = smvoice-mediaport The context is set above in sip.conf and that is what the CLI above is showing its using. Also my extensions.conf section is : -- [smvoice-mediaport-public-address] exten = s,1,System(/home/silentm/bin/smfunctions -stop) exten = s,n,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Hangup exten = h,1,System(/home/silentm/bin/smfunctions -start) [smvoice-mediaport] exten = public_address,1,Goto(smvoice-mediaport-public-address,s,1) #include /etc/asterisk/express.dnis.conf -- where express.dnis.conf has: ; Phone Caller ID DNIS Manager screen ; MMPCGA: VISUAL PC ROOM 105 - exten = 1105,1,Goto(smvoice-mediaport-public-address,s,1) --- Here is a call that works: == Using SIP RTP CoS mark 5 -- Executing [1105@smvoice-mediaport:1] Goto(SIP/mndemo_to_mediaport105-0003, smvoice-mediaport-public-address,s,1) in new stack -- Goto (smvoice-mediaport-public-address,s,1) -- Executing [s@smvoice-mediaport-public-address:1] System(SIP/mndemo_to_mediaport105-0003, /home/silentm/bin/smfunctions -stop) in new stack -- Executing [s@smvoice-mediaport-public-address:2] Playback(SIP/mndemo_to_mediaport105-0003, beep) in new stack -- SIP/mndemo_to_mediaport105-0003 Playing 'beep.gsm' (language 'en') -- Executing [s@smvoice-mediaport-public-address:3] Dial(SIP/mndemo_to_mediaport105-0003, Console/dsp) in new stack Call placed to 'dsp' on console Auto-answered -- Called dsp -- ALSA/dummy answered SIP/mndemo_to_mediaport105-0003 -- Executing [h@smvoice-mediaport-public-address:1] System(SIP/mndemo_to_mediaport105-0003, /home/silentm/bin/smfunctions -start) in new stack Hangup on console == Spawn extension (smvoice-mediaport-public-address, s, 3) exited non-zero on 'SIP/mndemo_to_mediaport105-0003' -- As I mentioned starting asterisk all this works. There is some random time later - perhaps days where it then stops finding the exten. Is there something I have wrong in the config above? Jerry -- _ -- 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 crashes on high IO load
Hi, the log files contained (sometimes) lines about refcount -1 in astobj.c. I also generated core dumps and analyzed them - but there were always errors in another module. Mabye I found the solution: Asterisk seems to crash when a required module cannot be loaded fast enough due to heavy disk usage. When I move the modules directory to another hard disk Asterisk runs fine. I'm using autoload=yes in modules.conf and have several noload lines in it. Is there a possibility to say asterisk to load all modules to RAM at start time and not on demand? Thanks and greetings Max Am 05.04.2011 09:45, schrieb Thorsten Göllner: Did you take a look at /var/log/syslog /var/log/asterisk/messages ? Using Debian? Take a look at iotop (apt-get install iotop). There you can see information about which process consumes high io load. Am 04.04.2011 17:23, schrieb Maximilian Grobecker: Hello Thorsten, the system has 4 GB RAM and about 2,5 GB free so swap space is not used or exhausted. Maybe the high load is not cause of this crashes but it's the only thing the crashes can be reproduced with. Thank you! Maximilian Grobecker Am 04.04.2011 16:03, schrieb Thorsten Göllner: Take a look with top at your system when high io load is seen. Maybe the machine is running out of ram and starts swapping? Am 04.04.2011 15:04, schrieb Maximilian Grobecker: Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly crashed without any further error messages. Are you experiencing the same problem? I'm really confused now why Asterisk crashes... Thank you! Maximilian Grobecker -- _ -- 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 -- _ -- 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] dialplan is not finding my number asterisk 1.8.3
Idea: If something is corrupting your dialplan, then this should reveal the extent of the corruption: You might, when the system is working properly, do a: asterisk -rx dialplan show somefile1 and then, when you are having problems, do a: asterisk -rx dialplan show somefile2 diff -u somefile1 somefile2 and see if this reveals anything juicy. murf On Tue, Apr 5, 2011 at 5:44 AM, Jerry Geis ge...@pagestation.com wrote: Jerry Geis wrote: I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a speaker attached. When asterisk first starts this works. In fact it works for some time. Then it just stops with this error on the CLI. [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_mediaport105' to extension '1105' rejected because extension not found in context 'smvoice-mediaport'. When doing the dialplan show it clearly in the context. [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1105' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] Its telling me it cannot find it. Its there - the dialplan shows its there. When I stop and start it works again for a little while. Matter of fact I just issued dialplan reload and calling into 1105 works again. Whats up? How do I get this to be consistent? Jerry I just looked in my extensions.conf and I do not have extenpatternmatchnew at all. My understanding is that it is off by default. my sip.conf has: register = mndemo_to_mediaport105:secret@mndemo ; Description: [mndemo_to_mediaport105] type=friend defaultname=mndemo_to_mediaport105 username=mndemo_to_mediaport105 secret=secret disallow=all allow=ulaw allow=alaw allow=gsm rtptimeout=60 host=192.168.1.58 context=smvoice-mediaport I was not aware I needed another context of : [mndemo_to_mediaport105] include = smvoice-mediaport The context is set above in sip.conf and that is what the CLI above is showing its using. Also my extensions.conf section is : -- [smvoice-mediaport-public-address] exten = s,1,System(/home/silentm/bin/smfunctions -stop) exten = s,n,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Hangup exten = h,1,System(/home/silentm/bin/smfunctions -start) [smvoice-mediaport] exten = public_address,1,Goto(smvoice-mediaport-public-address,s,1) #include /etc/asterisk/express.dnis.conf -- where express.dnis.conf has: ; Phone Caller ID DNIS Manager screen ; MMPCGA: VISUAL PC ROOM 105 - exten = 1105,1,Goto(smvoice-mediaport-public-address,s,1) --- Here is a call that works: == Using SIP RTP CoS mark 5 -- Executing [1105@smvoice-mediaport:1] Goto(SIP/mndemo_to_mediaport105-0003, smvoice-mediaport-public-address,s,1) in new stack -- Goto (smvoice-mediaport-public-address,s,1) -- Executing [s@smvoice-mediaport-public-address:1] System(SIP/mndemo_to_mediaport105-0003, /home/silentm/bin/smfunctions -stop) in new stack -- Executing [s@smvoice-mediaport-public-address:2] Playback(SIP/mndemo_to_mediaport105-0003, beep) in new stack -- SIP/mndemo_to_mediaport105-0003 Playing 'beep.gsm' (language 'en') -- Executing [s@smvoice-mediaport-public-address:3] Dial(SIP/mndemo_to_mediaport105-0003, Console/dsp) in new stack Call placed to 'dsp' on console Auto-answered -- Called dsp -- ALSA/dummy answered SIP/mndemo_to_mediaport105-0003 -- Executing [h@smvoice-mediaport-public-address:1] System(SIP/mndemo_to_mediaport105-0003, /home/silentm/bin/smfunctions -start) in new stack Hangup on console == Spawn extension (smvoice-mediaport-public-address, s, 3) exited non-zero on 'SIP/mndemo_to_mediaport105-0003' -- As I mentioned starting asterisk all this works. There is some random time later - perhaps days where it then stops finding the exten. Is there something I have wrong in the config above? Jerry -- _ -- 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 -- Steve Murphy ParseTree Corporation 57 Lane 17 Cody, WY 82414 ✉ m...@parsetree.com ☎ 307-899-5535 -- _ -- 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] dialplan is not finding my number asterisk 1.8.3
Steve Murphy wrote: Idea: If something is corrupting your dialplan, then this should reveal the extent of the corruption: You might, when the system is working properly, do a: asterisk -rx dialplan show somefile1 and then, when you are having problems, do a: asterisk -rx dialplan show somefile2 diff -u somefile1 somefile2 and see if this reveals anything juicy. murf Steve, That is a great idea. I did that the first time it happened. I dumped the dialplan, then I restarted and dumped again. it was the same. Being the first time I thought it was just a fluke but now it has happened a couple of times. I have not been able to narrow anything down. Thanks, jerry -- _ -- 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] dialplan is not finding my number asterisk 1.8.3
Jerry Geis wrote: Steve Murphy wrote: Idea: If something is corrupting your dialplan, then this should reveal the extent of the corruption: You might, when the system is working properly, do a: asterisk -rx dialplan show somefile1 and then, when you are having problems, do a: asterisk -rx dialplan show somefile2 diff -u somefile1 somefile2 and see if this reveals anything juicy. murf Steve, That is a great idea. I did that the first time it happened. I dumped the dialplan, then I restarted and dumped again. it was the same. Being the first time I thought it was just a fluke but now it has happened a couple of times. I have not been able to narrow anything down. Thanks, jerry Steve, perhaps I did something wrong the first time. As I just got the error again. I dumped the dialplan and my section: [ Context 'smvoice-mediaport' created by 'pbx_config' ] is empty. when I restart and dump again. [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1105' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] 'mediaport_direct' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] 'public_address' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] I have the correct data. The only thing I have in the dialplan for this box is: [smvoice-mediaport-public-address] exten = s,1,System(/home/silentm/bin/smfunctions -stop) exten = s,n,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Hangup exten = h,1,System(/home/silentm/bin/smfunctions -start) Can a system call be removing stuff from the dialplan? What next? Jerry -- _ -- 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] Iptables configuration to handle brute force registrations?
Hello I'm no expert of iptables, and it seems like it can handle banning IP's that are trying to register and fail too many times. I'd like to use this feature instead of having to install a second tool such as SSHGuard or BFS that parses the logs and reconfigure iptables on the fly. Is there a good iptables configuration that I could use as reference? FWIW, the kernel is uClinux 2.6.13.9, iptables is 1.3.6, ans it's a single-homed host so there's no need to handle the FORWARD chain. Thank you. -- _ -- 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] Vestec for Asterisk
Hi, I installed the Vestec module to one of my development Asterisk servers a few months ago but now I need to move the license to another host. Does anyone know how to do this? I've had a look on my Account page on the Digium website but it only shows the Language Pack, and I can't do anything with this either. Can anyone point me in the right direction please? Thanks Lee -- _ -- 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] Vestec for Asterisk
If you purchased from Diguim, call their tech support. If you purchased it from Vestec, you'll have to provide them with some paperwork or shell out some bucks. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer Sent: Tuesday, April 05, 2011 9:37 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Vestec for Asterisk Hi, I installed the Vestec module to one of my development Asterisk servers a few months ago but now I need to move the license to another host. Does anyone know how to do this? I've had a look on my Account page on the Digium website but it only shows the Language Pack, and I can't do anything with this either. Can anyone point me in the right direction please? Thanks Lee -- _ -- 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] Vestec for Asterisk
Talk to Vestec. As far as i know they are they ones that can re-issue the license code. Kashif Kahn (kahn at vestec.com) was very helpfull whenever i need it info for my project. Stelios On Tue, 2011-04-05 at 15:36 +0100, Lee Archer wrote: Hi, I installed the Vestec module to one of my development Asterisk servers a few months ago but now I need to move the license to another host. Does anyone know how to do this? I’ve had a look on my Account page on the Digium website but it only shows the Language Pack, and I can’t do anything with this either. Can anyone point me in the right direction please? Thanks Lee -- _ -- 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] Iptables configuration to handle brute force registrations?
On Tue, 5 Apr 2011, Gilles wrote: I'm no expert of iptables, and it seems like it can handle banning IP's that are trying to register and fail too many times. Is there a good iptables configuration that I could use as reference? Gordon Henderson posted a link to his script that handled failures above a threshold and some other cool stuff a few months back. Try searching the archives. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] allpage issu on asterisk 1.8.3.x
Hey Guys!
I have perl script for allpage which is working fine with asterisk 1.8.2.3
version but same script same dialplan wouldn't working on asterisk-1.8.3.2 is
there anything changes ?
If i run this script from command like it works but not from asterisk dialplan.
This script nothing but just connecting AMI interface and using Variable:
SIPADDHEADER=Alert-Info: Ring Answer variable to call all phones and putting
them in meetme conf room.
following is sample of script ( I am pasting half script )
# Now, we have an array (@tocall) with all valid SIP extensions.
while (my $sipxtn = shift @tocall) {
print VERBOSE \Doing $sipxtn\ 0\n;
# Open connection to AGI
my $tn = new Net::Telnet ( Port = $mgrport,
Prompt = '/.*[\$%#] $/',
Output_record_separator = '',
Input_Log= /tmp/input.log,
Output_Log= /tmp/output.log,
Errmode= 'return', );
$tn-open(127.0.0.1);
$tn-waitfor('/0\n$/');
$tn-print(Action: Login\n);
$tn-print(Username: $mgruser\n);
$tn-print(Secret: $mgrpass\n);
$tn-print(Events: off\n\n);
my ($pm, $m) = $tn-waitfor('/Authentication (.+)\n\n/');
if ($m =~ /Authentication failed/) {
print VERBOSE \Incorrect MGRUSER or MGRPASS - unable to connect to
manager interface\ 0\n;
exit;
}
$tn-print(Action: Originate\nChannel: SIP/$sipxtn\nContext:
all-page\nPriority: 1\n);
$tn-print(Variable: SIPADDHEADER=Alert-Info: Ring Answer\n);
$tn-print(Extension: s\n);
$tn-print(CallerID: System Page\n);
$tn-print(Action: Logoff\n\n);
$tn-close;
}
-S
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] dialplan is not finding my number asterisk 1.8.3
Oh, you are *not* going to like this, but you have a few different paths: 1. If the dialplan stuff is not really a memory corruption, but some sort of unplanned, but maybe accidentally programmed thing, either by you or something in the asterisk code, then: a. compile asterisk for debug. You can get in the menuselect stuff and make sure the debug compile options are turned on. Install it. b. Shut down asterisk c. fire it back up, under gdb: gdb full path to asterisk br ast_context_remove_extension_callerid2 comm 1 where c end run normal arguments to asterisk Then use asterisk as normal and wait for the problem to re-occur. Look to see if any calls to ast_context_remove_extension_callerid2 occurred (they will occur with dial reloads-- lots of them). You'll have to search carefully! The gdb messages could be buried in your console output. If the problem reoccurs, but you didn't see any calls to ast_context_remove_extension_callerid2, then it could be assumed that you are suffering some sort of memory corruption. Where it dies, or things start looking strange, may not indicate where or why the corruption is happening. In such a case, it really gets tricky to debug. Then we might resort to stuff like dmalloc, and others, to help spot where/when corruption occurs. Let's cross that bridge if we come to it. murf On Tue, Apr 5, 2011 at 7:30 AM, Jerry Geis ge...@pagestation.com wrote: Jerry Geis wrote: Steve Murphy wrote: Idea: If something is corrupting your dialplan, then this should reveal the extent of the corruption: You might, when the system is working properly, do a: asterisk -rx dialplan show somefile1 and then, when you are having problems, do a: asterisk -rx dialplan show somefile2 diff -u somefile1 somefile2 and see if this reveals anything juicy. murf Steve, That is a great idea. I did that the first time it happened. I dumped the dialplan, then I restarted and dumped again. it was the same. Being the first time I thought it was just a fluke but now it has happened a couple of times. I have not been able to narrow anything down. Thanks, jerry Steve, perhaps I did something wrong the first time. As I just got the error again. I dumped the dialplan and my section: [ Context 'smvoice-mediaport' created by 'pbx_config' ] is empty. when I restart and dump again. [ Context 'smvoice-mediaport' created by 'pbx_config' ] '1105' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] 'mediaport_direct' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] 'public_address' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config] I have the correct data. The only thing I have in the dialplan for this box is: [smvoice-mediaport-public-address] exten = s,1,System(/home/silentm/bin/smfunctions -stop) exten = s,n,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Hangup exten = h,1,System(/home/silentm/bin/smfunctions -start) Can a system call be removing stuff from the dialplan? What next? Oh, you are *not* going to like this, but you have a few different paths: 1. If the dialplan stuff is not really a memory corruption, but some sort of unplanned, but maybe accidentally programmed thing, either by you or something in the asterisk code, then: a. compile asterisk for debug. You can get in the menuselect stuff and make sure the debug compile options are turned on. Install it. b. Shut down asterisk c. fire it back up, under gdb: gdb full path to asterisk br ast_context_remove_extension_callerid2 comm 1 where c end run normal arguments to asterisk Then use asterisk as normal and wait for the problem to re-occur. Look to see if any calls to ast_context_remove_extension_callerid2 occurred (they will occur with dial reloads-- lots of them). You'll have to search carefully! The gdb messages could be buried in your console output. If the problem reoccurs, but you didn't see any calls to ast_context_remove_extension_callerid2, then it could be assumed that you are suffering some sort of memory corruption. Where it dies, or things start looking strange, may not indicate where or why the corruption is happening. In such a case, it really gets tricky to debug. Then we might resort to stuff like dmalloc, and others, to help spot where/when corruption occurs. Let's cross that bridge if we come to it. murf Jerry -- Steve Murphy ParseTree Corporation -- _ -- 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] allpage issu on asterisk 1.8.3.x
Nevermind,
I have solved it my self. this script wring some logs in /tmp and somehow
logfile was already there. so just deleted and it works!
-S
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 16:35:37 +
Subject: [asterisk-users] allpage issu on asterisk 1.8.3.x
Hey Guys!
I have perl script for allpage which is working fine with asterisk 1.8.2.3
version but same script same dialplan wouldn't working on asterisk-1.8.3.2 is
there anything changes ?
If i run this script from command like it works but not from asterisk dialplan.
This script nothing but just connecting AMI interface and using Variable:
SIPADDHEADER=Alert-Info: Ring Answer variable to call all phones and putting
them in meetme conf room.
following is sample of script ( I am pasting half script )
# Now, we have an array (@tocall) with all valid SIP extensions.
while (my $sipxtn = shift @tocall) {
print VERBOSE \Doing $sipxtn\ 0\n;
# Open connection to AGI
my $tn = new Net::Telnet ( Port = $mgrport,
Prompt = '/.*[\$%#] $/',
Output_record_separator = '',
Input_Log= /tmp/input.log,
Output_Log= /tmp/output.log,
Errmode= 'return', );
$tn-open(127.0.0.1);
$tn-waitfor('/0\n$/');
$tn-print(Action: Login\n);
$tn-print(Username: $mgruser\n);
$tn-print(Secret: $mgrpass\n);
$tn-print(Events: off\n\n);
my ($pm, $m) = $tn-waitfor('/Authentication (.+)\n\n/');
if ($m =~ /Authentication failed/) {
print VERBOSE \Incorrect MGRUSER or MGRPASS - unable to connect to
manager interface\ 0\n;
exit;
}
$tn-print(Action: Originate\nChannel: SIP/$sipxtn\nContext:
all-page\nPriority: 1\n);
$tn-print(Variable: SIPADDHEADER=Alert-Info: Ring Answer\n);
$tn-print(Extension: s\n);
$tn-print(CallerID: System Page\n);
$tn-print(Action: Logoff\n\n);
$tn-close;
}
-S
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[asterisk-users] agi voicemail callback
I'm wondering if there is a simply way to perform a voicemail callback feature using AGI. For instance, a caller leaves a voicemail, the voicemail will then call the owner of the voicemailbox determined by a database look up. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forwarding
Hi Rizwan
Thank you for your help i will test this solution and i will update you as
soon as i have any result.
Kind Regards
2011/4/4 Rizwan Hisham rizwanhas...@gmail.com
Do this:
exten= _0522XX,1,Goto(${CONTEXT},0520${EXTEN:4:},1)
you can also use the dial command for this as well
exten= _0522XX,1,Dial(Local/0520${EXTEN:4}@${SOMECONTEXT})
replace ${CoNTEXT} and ${SOMECONTEXT} with name of your contexts which
contains 0520 numbers.
I have not tested it, you can try it on your setup.
On Mon, Apr 4, 2011 at 7:00 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hello list,
i have one question related to call forwarding.
i have 2 number for the inbound and i want to configure asterisk like
that.
When the customer call the first number 0522XX the call will be
forwarding automatically to anther number 0520xx
Does anybody have a solution to this problem.
Thanks and Regards.
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Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
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[asterisk-users] agi create mailbox
Is it possible to create a voicemail box using AGI? How does asterisk know about mailboxes when using Asterisk with pure AGI? -- _ -- 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] agi voicemail callback
vip killa wrote: I'm wondering if there is a simply way to perform a voicemail callback feature using AGI I don't have an AGI, but I do have dial-plan code. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] agi voicemail callback
On Tue, 5 Apr 2011, vip killa wrote: I'm wondering if there is a simply way to perform a voicemail callback feature using AGI.For instance, a caller leaves a voicemail, the voicemail will then call the owner of the voicemailbox determined by a database look up. Use 'mailcmd' in voicemail.conf. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] agi create mailbox
On Tue, 5 Apr 2011, vip killa wrote: Is it possible to create a voicemail box using AGI? An AGI executes as a child process when a channel executes agi() via the dialplan. Are you intending to call into Asterisk and let the caller create mailboxes? All the AGI needs to do is add a line to the appropriate stanza in voicemail.conf. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] asterisk hints
Hey guys!
I am new in hints application. what is the use of this application ( i already
did google ) but still confused. If i want to use hint in my dialplan then
should i type each and every extension in hint dialplan or is there regex
available
something like following _XXX will watch all my extension. Because we have
more than 200 phones so its hard to write down each and every extension in hint
[hints]
exten = _XXX,hint,SIP/${EXTEN}
exten = 7527,hint,SIP/7527
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Re: [asterisk-users] asterisk hints
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011 12:54 PM
To: asterisk-users
Subject: [asterisk-users] asterisk hints
Hey guys!
I am new in hints application. what is the use of this application ( i
already did google ) but still confused. If i want to use hint in my
dialplan then should i type each and every extension in hint dialplan or is
there regex available
something like following _XXX will watch all my extension. Because we have
more than 200 phones so its hard to write down each and every extension in
hint
[hints]
exten = _XXX,hint,SIP/${EXTEN}
exten = 7527,hint,SIP/7527
The answer depends on the version you are using. Hints are (in my
experience) most useful for BLF and AMI applications.
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Re: [asterisk-users] asterisk hints
I am using asterisk-1.8.3.2
and we have polycom phones. how should i use hint ?
-S
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 12:56:58 -0500
Subject: Re: [asterisk-users] asterisk hints
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011
12:54 PM
To: asterisk-users
Subject: [asterisk-users] asterisk
hints
Hey guys!
I am new in hints application. what is the use of this application ( i already
did google ) but still confused. If i want to use hint in my dialplan then
should i type each and every extension in hint dialplan or is there regex
available
something like following _XXX will watch all my extension. Because we
have more than 200 phones so its hard to write down each and every extension in
hint
[hints]
exten = _XXX,hint,SIP/${EXTEN}
exten = 7527,hint,SIP/7527
The
answer depends on the version you are using. Hints are (in my experience) most
useful for BLF and AMI applications.
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Re: [asterisk-users] asterisk hints
On my Polycom 501's I use hints to populate a buddy list - I hit the
buddies softkey and can see if my buddy is on the line.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011 1:19 PM
To: asterisk-users
Subject: Re: [asterisk-users] asterisk hints
I am using asterisk-1.8.3.2
and we have polycom phones. how should i use hint ?
-S
_
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 12:56:58 -0500
Subject: Re: [asterisk-users] asterisk hints
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011 12:54 PM
To: asterisk-users
Subject: [asterisk-users] asterisk hints
Hey guys!
I am new in hints application. what is the use of this application ( i
already did google ) but still confused. If i want to use hint in my
dialplan then should i type each and every extension in hint dialplan or is
there regex available
something like following _XXX will watch all my extension. Because we have
more than 200 phones so its hard to write down each and every extension in
hint
[hints]
exten = _XXX,hint,SIP/${EXTEN}
exten = 7527,hint,SIP/7527
The answer depends on the version you are using. Hints are (in my
experience) most useful for BLF and AMI applications.
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Re: [asterisk-users] Iptables configuration to handle brute, force registrations?
fail2ban might be good for this. On 04/05/2011 01:00 PM, asterisk-users-requ...@lists.digium.com wrote: Date: Tue, 5 Apr 2011 08:44:41 -0700 (PDT) From: Steve Edwardsasterisk@sedwards.com Subject: Re: [asterisk-users] Iptables configuration to handle brute force registrations? On Tue, 5 Apr 2011, Gilles wrote: I'm no expert of iptables, and it seems like it can handle banning IP's that are trying to register and fail too many times. Is there a good iptables configuration that I could use as reference? Gordon Henderson posted a link to his script that handled failures above a threshold and some other cool stuff a few months back. smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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 hints
If i want to watch every phone status Idel or Inuse the how should i use hint
in my dialplan. I meant should i need to specify each and every extension ? or
is there any catch-all extensions ?
-Satish
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 13:20:45 -0500
Subject: Re: [asterisk-users] asterisk hints
On my Polycom 501’s I use hints to
populate a “buddy” list – I hit the buddies softkey and can
see if my “buddy” is on the line.
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011 1:19
PM
To: asterisk-users
Subject: Re: [asterisk-users]
asterisk hints
I am using asterisk-1.8.3.2
and we have polycom phones. how should i use hint ?
-S
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 12:56:58 -0500
Subject: Re: [asterisk-users] asterisk hints
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011
12:54 PM
To: asterisk-users
Subject: [asterisk-users] asterisk
hints
Hey guys!
I am new in hints application. what is the use of this application ( i already
did google ) but still confused. If i want to use hint in my dialplan then
should
i type each and every extension in hint dialplan or is there regex available
something like following _XXX will watch all my extension. Because we
have more than 200 phones so its hard to write down each and every extension in
hint
[hints]
exten = _XXX,hint,SIP/${EXTEN}
exten = 7527,hint,SIP/7527
The
answer depends on the version you are using. Hints are (in my experience)
most useful for BLF and AMI applications.
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Re: [asterisk-users] Iptables configuration to handle brute, force registrations?
On Tue, Apr 5, 2011 at 1:31 PM, Bill Michaelson b...@cosi.com wrote: fail2ban might be good for this. I think you missed the point, which is reducing the need for an external application that searches logs in order to determine whether or not to block an IP. Why run fail2ban and add overhead when you can just do the same thing with iptables itself? -- _ -- 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] dahdi and linux-2.6.38
Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] dahdi and linux-2.6.38
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 1:53 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi and linux-2.6.38 Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. You installed libpri ? -- _ -- 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] dahdi and linux-2.6.38
On 04/05/2011 01:52 PM, cov...@ccs.covici.com wrote: Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. After installing dahdi did you load your wctdm.ko driver? What is the output of 'cat /sys/module/dahdi/version'? What is the output from 'cat /proc/dahdi/1'? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] dahdi and linux-2.6.38
On 04/05/2011 01:54 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 1:53 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi and linux-2.6.38 Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. You installed libpri ? Hi Danny, Would it be possible to update your email client to add a prefix to quoted lines? It's hard for me to see where the message you're replying to ends and your reply begins. Thanks, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] dahdi and linux-2.6.38
On 04/05/2011 02:00 PM, Shaun Ruffell wrote: Would it be possible to update your email client to add a prefix to quoted lines? It's hard for me to see where the message you're replying to ends and your reply begins. Arghh. I meant that to be a private email. My apologies. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] dahdi and linux-2.6.38
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, April 05, 2011 2:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi and linux-2.6.38 On 04/05/2011 02:00 PM, Shaun Ruffell wrote: Would it be possible to update your email client to add a prefix to quoted lines? It's hard for me to see where the message you're replying to ends and your reply begins. Arghh. I meant that to be a private email. My apologies. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org Will see what I can do - I am using Outlook 2003 -- _ -- 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] Iptables configuration to handle brute, force registrations?
On Tue, 5 Apr 2011, Sherwood McGowan wrote: Why run fail2ban and add overhead when you can just do the same thing with iptables itself? Because it's not the same? The iptables approach is great because it is 'light-weight' and it should already 'be there.' Also, it can react quicker because it doesn't have to read log files to make a decision. The 'downside' of the iptables approach is that the blocks go away when iptables is reloaded -- like when the host is restarted. Probably not an issue with Gordon since his hosts stay up for years. I'm thinking the iptables approach supplemented with a script to periodically save the block list to disk would allow persistent blocks as well as letting you accumulating blocks between all your hosts. Which would still be much 'lighter' than fail2ban. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Iptables configuration to handle brute, force registrations?
On 4/5/2011 2:11 PM, Steve Edwards wrote: On Tue, 5 Apr 2011, Sherwood McGowan wrote: Why run fail2ban and add overhead when you can just do the same thing with iptables itself? Because it's not the same? The iptables approach is great because it is 'light-weight' and it should already 'be there.' Also, it can react quicker because it doesn't have to read log files to make a decision. The 'downside' of the iptables approach is that the blocks go away when iptables is reloaded -- like when the host is restarted. Probably not an issue with Gordon since his hosts stay up for years. I'm thinking the iptables approach supplemented with a script to periodically save the block list to disk would allow persistent blocks as well as letting you accumulating blocks between all your hosts. Which would still be much 'lighter' than fail2ban. Agreed on all points Steve. I've already implemented an auto save function, to workaround the drawback you mentioned. Are there possibly other drawbacks that I'm not seeing/remembering? I've been running an iptables based setup for some time, never really jumped into the fail2ban wagon -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- 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] dahdi and linux-2.6.38
Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 1:53 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi and linux-2.6.38 Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. You installed libpri ? I don't have any pri's. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] dahdi and linux-2.6.38
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi and linux-2.6.38 Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 1:53 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi and linux-2.6.38 Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. You installed libpri ? I don't have any pri's. I'd check the dmesg output - AFAIK you need libpri as a backbone for DAHDI (at least on some kernels). -- _ -- 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] dahdi and linux-2.6.38
Shaun Ruffell sruff...@digium.com wrote: On 04/05/2011 01:52 PM, cov...@ccs.covici.com wrote: Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. After installing dahdi did you load your wctdm.ko driver? What is the output of 'cat /sys/module/dahdi/version'? What is the output from 'cat /proc/dahdi/1'? I did load the driver, but I am not booted into that system, so I cannot give you the other version info. I did make and make install and I will check to make sure it got to the correct place. And it looks like it did -- the dahdi-version.h has a time stamp about 2 minutes before the timestamp of the modules in the kernel I was using. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] dahdi and linux-2.6.38
On Tue, Apr 05, 2011 at 02:03:20PM -0500, Danny Nicholas wrote: On 04/05/2011 02:00 PM, Shaun Ruffell wrote: Would it be possible to update your email client to add a prefix to quoted lines? It's hard for me to see where the message you're replying to ends and your reply begins. Will see what I can do - I am using Outlook 2003 In case you haven't seen it before (or for anyone else who is using Outlook): http://mailformat.dan.info/config/outlook.html -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Iptables configuration to handle brute, force registrations?
On Tue, 5 Apr 2011, Sherwood McGowan wrote: Why run fail2ban and add overhead when you can just do the same thing with iptables itself? On 4/5/2011 2:11 PM, Steve Edwards wrote: Because it's not the same? The iptables approach is great because it is 'light-weight' and it should already 'be there.' Also, it can react quicker because it doesn't have to read log files to make a decision. The 'downside' of the iptables approach is that the blocks go away when iptables is reloaded -- like when the host is restarted. Probably not an issue with Gordon since his hosts stay up for years. I'm thinking the iptables approach supplemented with a script to periodically save the block list to disk would allow persistent blocks as well as letting you accumulating blocks between all your hosts. Which would still be much 'lighter' than fail2ban. On Tue, 5 Apr 2011, Sherwood McGowan wrote: Agreed on all points Steve. I've already implemented an auto save function, to workaround the drawback you mentioned. Then you're already a couple of steps down the path further than me :) Are there possibly other drawbacks that I'm not seeing/remembering? I've been running an iptables based setup for some time, never really jumped into the fail2ban wagon I've never used fail2ban either. I don't think it's advantages are functional, but the more somewhat intangible: ) It's included with several of the all-in-one Asterisk distributions. ) It's documented. ) It's more flexible ) Somebody else gets to enhance and maintain the code. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] dahdi and linux-2.6.38
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, April 05, 2011 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi and linux-2.6.38 On Tue, Apr 05, 2011 at 02:03:20PM -0500, Danny Nicholas wrote: On 04/05/2011 02:00 PM, Shaun Ruffell wrote: Would it be possible to update your email client to add a prefix to quoted lines? It's hard for me to see where the message you're replying to ends and your reply begins. Will see what I can do - I am using Outlook 2003 In case you haven't seen it before (or for anyone else who is using Outlook): http://mailformat.dan.info/config/outlook.html -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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 [Danny Nicholas] Thanks for the tip maybe this will be better - though the outlook beast still seems to want to top-post! -- _ -- 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-users Digest, Vol 81, Issue 12
On 04/05/2011 03:06 PM, asterisk-users-requ...@lists.digium.com wrote: Message: 12 Date: Tue, 5 Apr 2011 13:36:21 -0500 From: Sherwood McGowansherwood.mcgo...@gmail.com Subject: Re: [asterisk-users] Iptables configuration to handle brute, force registrations? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Bill Michaelsonb...@cosi.com Message-ID:banlktimqrbfmqpoinrphr_rjekolbwp...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 On Tue, Apr 5, 2011 at 1:31 PM, Bill Michaelsonb...@cosi.com wrote: fail2ban might be good for this. I think you missed the point, which is reducing the need for an external application that searches logs in order to determine whether or not to block an IP. Why run fail2ban and add overhead when you can just do the same thing with iptables itself? I apologize for jumping into the middle without reading the beginning of the discussion in which this central requirement to avoid an external application was stated, as I now infer from Mr. McGowan. Sorry for missing the point. I'll have to read up on fail2ban also. I thought it monitored the tails of logs. I did not know that it searched them. My intent was to suggest using an established tool that would consolidate the IP blocking and unblocking function for all ports into a single application without imposing additional maintenance overhead of new code for this purpose. Obviously, I'm not seeing the big picture. Sorry for my myopic comments and for cluttering the list. I won't make the mistake of offering worthless contributions in the future. smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] dahdi and linux-2.6.38
On Tue, Apr 05, 2011 at 02:43:15PM -0500, Danny Nicholas wrote: On 04/05/2011 02:00 PM, Shaun Ruffell wrote: On Tue, Apr 05, 2011 at 02:03:20PM -0500, Danny Nicholas wrote: On 04/05/2011 02:00 PM, Shaun Ruffell wrote: Would it be possible to update your email client to add a prefix to quoted lines? It's hard for me to see where the message you're replying to ends and your reply begins. Will see what I can do - I am using Outlook 2003 In case you haven't seen it before (or for anyone else who is using Outlook): http://mailformat.dan.info/config/outlook.html Thanks for the tip maybe this will be better - though the outlook beast still seems to want to top-post! That works better for me. The only other thing I would recommend is triming out parts of the email your replying to that aren't pertinent to the discussion (i.e. the sigatures and footers). But regardless, thanks, I appreciate the efforts. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] dahdi and linux-2.6.38
Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi and linux-2.6.38 Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 1:53 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi and linux-2.6.38 Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. You installed libpri ? I don't have any pri's. I'd check the dmesg output - AFAIK you need libpri as a backbone for DAHDI (at least on some kernels). No dmesg output at all. Just when the modules were loaded, but not from the dahdi_cfg -vv -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] dahdi and linux-2.6.38
On 04/05/2011 02:26 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi and linux-2.6.38 Danny Nicholasda...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 1:53 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi and linux-2.6.38 Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. You installed libpri ? I don't have any pri's. I'd check the dmesg output - AFAIK you need libpri as a backbone for DAHDI (at least on some kernels). This is not true. libpri is used only in userspace, and has nothing to do with anything in the kernel. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] dahdi and linux-2.6.38
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, April 05, 2011 3:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] dahdi and linux-2.6.38 On 04/05/2011 02:26 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi and linux-2.6.38 Danny Nicholasda...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Tuesday, April 05, 2011 1:53 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi and linux-2.6.38 Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. You installed libpri ? I don't have any pri's. I'd check the dmesg output - AFAIK you need libpri as a backbone for DAHDI (at least on some kernels). This is not true. libpri is used only in userspace, and has nothing to do with anything in the kernel. -- Kevin P. Fleming [Danny Nicholas] Thanks for the correction - INKM - I Now Know More? -- _ -- 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] spa8000 t38 faxing
Ok thanks I found the problem The spa8000 has some bugs with t38 which are fixed in the spa2102 but not in the 8000 1. If the adapter starts with g711 It doesn't switch to t38 2. (This my problem) when it does go to t38 and the itsp asks for it to fallback to 9600 it doesn't fallback so they never end up speaking to each other These were fixed in the 2102 according to the release notes Well now I hope I could get someone at cisco to look at it because I have more than a dozen 8000's Thanks for your help -Original Message- From: Larry Moore lmo...@starwon.com.au Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 27 Mar 2011 11:26:35 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] spa8000 t38 faxing Perhaps this will help. I have a SPA8800 which has 4 x FXS 4 x FXO ports. It took me some time to produce a working configuration. In Asterisk I have the following where 904 is the extension of the fax-modem and itsp is you VoIP Service Provider. sip.conf [general] . . faxdetect=no t38pt_udptl=yes,redundancy,maxdatagram=400 . . [904] ; Cisco SPA8800 FXS Port 4 ; Analogue FAX Modem attached type=friend defaultuser=904 secret=secret call-limit=2 qualify=yes canreinvite=no directmedia=no directrtpsetup=no ignoresdpversion=yes transport=udp,tcp host=dynamic context=your_context faxdetect=no . . [itsp] . . faxdetect=yes ignoresdpversion=yes . . I am including information from my SPA8800 for one of the FXS ports I have a Fax Modem attached to, the key to getting it to work I believe is the FAX Tone Detect Mode. Audio Configuration Preferred Codec: G711a Second Preferred Codec: Unspecified Third Preferred Codec: UnspecifiedUse Pref Codec Only: no Silence Supp Enable: yes Silence Threshold: medium G729a Enable: no Echo Canc Enable: yes G723 Enable: no Echo Canc Adapt Enable: yes G726-16 Enable: no Echo Supp Enable: yes G726-24 Enable: no FAX CED Detect Enable: yes G726-32 Enable: no FAX CNG Detect Enable: yes G726-40 Enable: no FAX Passthru Codec: G711a DTMF Process INFO: yes FAX Codec Symmetric: yes DTMF Process AVT: yes FAX Passthru Method: ReINVITE DTMF Tx Method: AVT DTMF Tx Mode: Strict DTMF Tx Strict Hold Off Time: 40FAX Process NSE: no Hook Flash Tx Method: None FAX Disable ECAN: no Release Unused Codec: yes FAX Enable T38: yes FAX T38 Redundancy: 1 FAX Tone Detect Mode: callee only Symmetric RTP: yes Supplementary Service Settings CW Setting: noBlock CID Setting: no Block ANC Setting: noDND Setting: no CID Setting: yes CWCID Setting: yes Dist Ring Setting: yesSecure Call Setting: no Message Waiting: noAccept Media Loopback Request: automatic Media Loopback Mode: sourceMedia Loopback Type: media Larry. -- _ -- 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] minmessage / maxsilence in voicemail.conf
So I get this warning: app_voicemail.c:11084 load_config: maxsilence should be less than minmessage or you may get empty messages Can anyone explain that a little better? When would I end up getting empty messages if say minmessage was set to 3 seconds and maxsilence is set to 10? 10 seconds of silence would be more acceptable to me than setting maxsilence to 2 seconds and having an undue pause in the left vm causing a hangup, or setting maxsilence to 0 seconds and having a situation where the PRI or whatever gets wedged, and the voicemail just keeps recording for... a long time. hose -- _ -- 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] IAS trunk error AES encryption disabled. Install OpenSSL.
Hey Guys! I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error. -Satish == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-000d, orasebcamdial,7623) in new stack -- Executing [s@macro-orasebcamdial:1] Dial(SIP/7527-000d, iax2/orasebcam@orasebcam/7623) in new stack -- Called orasebcam@orasebcam/7623 [Apr 5 13:51:26] WARNING[9539]: /usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:145 __stub__ast_aes_set_encrypt_key: AES encryption disabled. Install OpenSSL. [Apr 5 13:51:26] WARNING[9539]: /usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:157 __stub__ast_aes_set_decrypt_key: AES encryption disabled. Install OpenSSL. [Apr 5 13:51:26] WARNING[9539]: /usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:157 __stub__ast_aes_set_decrypt_key: AES encryption disabled. Install OpenSSL. -- Call accepted by 172.30.245.208 (format gsm) -- Format for call is gsm -- IAX2/orasebcam-16782 is ringing -- IAX2/orasebcam-16782 is circuit-busy -- Hungup 'IAX2/orasebcam-16782' == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-orasebcamdial:2] Goto(SIP/7527-000d, s-CONGESTION,1) in new stack -- _ -- 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] Iptables configuration to handle brute, force registrations?
On Tue, Apr 5, 2011 at 2:40 PM, Steve Edwards asterisk@sedwards.comwrote: snip Are there possibly other drawbacks that I'm not seeing/remembering? I've been running an iptables based setup for some time, never really jumped into the fail2ban wagon I've never used fail2ban either. I don't think it's advantages are functional, but the more somewhat intangible: ) It's included with several of the all-in-one Asterisk distributions. ) It's documented. ) It's more flexible ) Somebody else gets to enhance and maintain the code. Fail2ban is easy. It's well documented and can be setup in just a few minutes. It's got an easy way to setup a whitelist that doesn't get banned (so you don't ban yourself or any of your trunks, etc), and you can use it for more than just asterisk blocking (I use it to monitor ssh and ftp as well). You can easily copy config files between systems, etc, plus all the things you mentioned Steve. That being said, it has several downsides too, i.e - whenever fail2ban is restarted, the fail2ban chains are flushed (this is occurs on system restarts as well). If you need to make changes to your iptables setup (i.e change an IP address of a service provider), you really want to unload fail2ban, make your changes directly to iptables, then save your new iptables setup, then restart fail2ban. Otherwise you'll end up saving your fail2ban chains in with your regular chains, and when you restart fail2ban, it'll try to add new f2b chains. And for some reason people seem to think that it requiring Python is a bad thing. But then again, I'm not running it on small systems - most of the systems I've put it on have plenty of excess cpu and memory, so that hasn't been an issue for me. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- 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] Iptables configuration to handle brute, force registrations?
First, this appears to be working for me though I'm not 100% sure of that and cannot guarantee it will for you in any way, shape or form. With the lawyering out of the way... I've seen fail2ban allow more than 500 failed SIP login attempts in under 30 seconds before adding an iptables rule to block the attacker. Likely I have it configured wrong but lately, I've been tinkering with iptables rules using the recent module as another layer of defense. Relevant lines from /etc/sysconfig/iptables on my CENTOS/Asterisk machine below... -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -m recent --set --name SIP -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -m recent --rcheck --name SIP --seconds 600 --hitcount 20 --rttl -j DROP -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -m recent --rcheck --name SIP --seconds 300 --hitcount 10 --rttl -j DROP -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -m recent --rcheck --name SIP --seconds 180 --hitcount 5 --rttl -j DROP -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -m recent --rcheck --name SIP --seconds 60 --hitcount 3 --rttl -j DROP -A RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT This blocks the attacker when too many new SIP connections happen in too short a period of time. I think fail2ban will now never sees enough failed logins to fire off a response. $0.02 On Tue, Apr 5, 2011 at 2:31 PM, Bill Michaelson b...@cosi.com wrote: fail2ban might be good for this. On 04/05/2011 01:00 PM, asterisk-users-requ...@lists.digium.com wrote: Date: Tue, 5 Apr 2011 08:44:41 -0700 (PDT) From: Steve Edwards asterisk@sedwards.com Subject: Re: [asterisk-users] Iptables configuration to handle brute force registrations? On Tue, 5 Apr 2011, Gilles wrote: I'm no expert of iptables, and it seems like it can handle banning IP's that are trying to register and fail too many times. Is there a good iptables configuration that I could use as reference? Gordon Henderson posted a link to his script that handled failures above a threshold and some other cool stuff a few months back. -- _ -- 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] Iptables configuration to handle brute, force registrations?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Paul Dugas Sent: Tuesday, April 05, 2011 4:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Iptables configuration to handle brute,force registrations? First, this appears to be working for me though I'm not 100% sure of that and cannot guarantee it will for you in any way, shape or form. With the lawyering out of the way... I've seen fail2ban allow more than 500 failed SIP login attempts in under 30 seconds before adding an iptables rule to block the attacker. Likely I have it configured wrong but lately, I've been tinkering with iptables rules using the recent module as another layer of defense. Relevant lines from /etc/sysconfig/iptables on my CENTOS/Asterisk machine below... snip [Danny Nicholas] I'm no expert, but as I see it, for fail2ban to work properly in a heavy attack environment, you MUST have your logs in realtime databases and preferably also roll them frequently. In normal Asterisk (as I use it), logs are written at the end of a call (not good for attack scenario unless attacks are quick and out) and in a heavy call environment, an attacker could make quite a bit of headway before the log could be processed. If you are realtime and rolling the logs hourly or so, fail2ban should work pretty well, but no guarantees. -- _ -- 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] Asterisk 1.8.3
I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. Thanks Bryant -- _ -- 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 crashes on high IO load
On 6/04/11 12:39 AM, Maximilian Grobecker wrote: Hi, the log files contained (sometimes) lines about refcount -1 in astobj.c. I also generated core dumps and analyzed them - but there were always errors in another module. Mabye I found the solution: Asterisk seems to crash when a required module cannot be loaded fast enough due to heavy disk usage. When I move the modules directory to another hard disk Asterisk runs fine. I'm using autoload=yes in modules.conf and have several noload lines in it. Is there a possibility to say asterisk to load all modules to RAM at start time and not on demand? You could compile Asterisk with embedded modules? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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 1.8.3
On Tuesday 05 April 2011 20:10:48 Bryant Zimmerman wrote: I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. This sounds like a deadlock of some kind. Asterisk has a debugging facility built-in for finding this type of problem, but you will need to compile in DONT_OPTIMIZE and DEBUG_THREADS. Also, it would be helpful, but not entirely necessary, to compile in BETTER_BACKTRACES. Once the problem occurs with the recompiled binary, issuing a core show locks should turn up an indication of where the problem lies. -- Tilghman -- _ -- 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
