Re: [asterisk-users] Timing cable usage necessity
On 26.11.2010 17:29, David Backeberg wrote: 2010/11/25 Захаров Антонins...@mail.ru: Hello everyone. I have a timing slips errors and I can't understand what source of the problem is. My installation has 2 digium cards: TE420 and TE220 cards in one server. There are 3 spans (E1) to PSTN and 3 spans to internal PBS stations - normal installation for transit communication. Span configuration is: span=1,1,0,ccs,hdb3 #TE420 - first port. To PSTN. span=2,0,0,ccs,hdb3 #TE420 - second port. To PBX. span=3,2,0,ccs,hdb3 #TE420 - third port. To PSTN. span=4,0,0,ccs,hdb3 #TE420 - fourth port. To PBX. span=5,3,0,ccs,hdb3 #TE220 - first port. To PSTN. span=6,0,0,ccs,hdb3 #TE220 - second port. To PBX. I should to say, that PBXs are interconnected through router (doesn't know anything about it). So all schema looks like this: http://yfrog.com/jjschemaj Spans 1-5 works fine, but on span 6 (marked bold) I have rising timing slips counter. I think it's appearing because I'm getting a primary timing source on span 1 - first port on TE420. But TE220 doesn't use it's span 5 for timing source, because it has priority 3, so it could be a sync problem. Am I wrong? I'm started to think about timing cable for syncing timing on first card and second. Should I use it? It's a problem to bought cable in our city (Russia,Moscow). All resellers sell only cards. Could I use floppy or IDE cable to interconnect cards? As I see in picture of cable, it's a direct 16 pin cable. Does anybody know something about timing cable for different cards? How I can solve my problem? I can't answer your problem about how you find one in Moscow, but I can tell you that when I've installed two cards in a single server I have used the timing ribbon cable. I have no idea whether it made a difference, and I now build my servers differently. You could ask Digium directly, as they probably know their resellers. Something that may help you is that you can make sure you set the priority on the cards differently from each other. There is a dial with a pointer that tells you which card is prioritized. Whenever I had two in the same server I made sure that one card was set higher than the other. As I think, i forgot to set priority on TE420 and TE220. Could this be the cause of an slip error? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] change date
Am 29.11.2010 08:20, schrieb Tilghman Lesher: On Saturday 27 November 2010 04:52:31 Klaus Schwarzkopf wrote: Hi, why have many files on http://downloads.asterisk.org/pub/telephony/asterisk/releases/ the change date 18 aug 2009? See: asterisk-1.2.24-patch.gz07-Aug-2007 17:103.2K asterisk-1.2.24-patch.gz.asc07-Aug-2007 17:101.1K asterisk-1.2.24-patch.gz.sha107-Aug-2007 17:10 67 asterisk-1.2.24.tar.gz18-Aug-2009 16:33 28M asterisk-1.2.24.tar.gz.asc18-Aug-2009 16:331.0K asterisk-1.2.24.tar.gz.sha118-Aug-2009 16:33 65 asterisk-1.2.25-patch.gz29-Nov-2007 15:591.5K asterisk-1.2.25-patch.gz.asc29-Nov-2007 15:59567 I try to repair the openembedded recipes an the recipe have also an different checksum. NOTE: fetch http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1 .2.24.tar.gz NOTE: The checksums for '/home/klaus/development/oe/downloads/asterisk-1.2.24.tar.gz' did not match. Expected MD5: '63dc8b7be4cd10375c5fbda893c780bc' and Got: 'db7bcaaa494804af361157a37c224dfa' Expected SHA256: '9debaf410636fa477e1e1f09fe0b16a1c2814afaf7195f34f29e4ce5b8debbbd' and Got: 'eed3493b1409d7100e0f983af0486bd7f8965e9e47b7a6d5ab8539b2dd3609aa' NOTE: Your checksums: SRC_URI[md5sum] = db7bcaaa494804af361157a37c224dfa SRC_URI[sha256sum] = eed3493b1409d7100e0f983af0486bd7f8965e9e47b7a6d5ab8539b2dd3609aa Due to a licensing issue with some of the files we distributed with previous tarballs, we removed those files from archived tarballs in order to avoid continuing to distribute those files in any form. So yes, the checksums will have changed, although the checksums we distribute with the tarballs were also updated at the same time. Given that most of the changes since 1.2.24 have been security fixes, I would strongly encourage you to update your packages. There is no excuse for distributing vulnerable packages beyond the date that the vulnerability is disclosed, plus a brief period necessary for releasing updated packages. Additionally, the 1.2 branch has been EOLed, which means if any additional security issues are found, we will not be releasing updated packages to deal with those issues. For this reason, you would be better off putting forth the work to release packages based upon 1.4 or 1.8. Thanks for the detailed information. There are recipes with the new version. I recommend to delete the old one. Greetings, Klaus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Timing cable usage necessity
Здравствуйте. Спасибо за ответ. Меня какраз интересуют проблемы, которые решает этот кабель. Достать его теоретически мы сможем. Другой вопрос, в чем же его уникальность? И почему, например, нельзя использовать floppy кабель? Когда я поставил floppy кабель вместо официального и модулю указал опцию timingcable=1, у меня либо вообще система падать стала (так, что коннект по ssh терялся), либо в asterisk я получал сообщение Bad HDLC и поток не поднимался. Hello. Thank you for your reply. I'm interested in problems that solves this cable. We can get it theoretically. Another question, what is its uniqueness? And why, for example, we can not use the floppy cable? When I put the floppy cable instead of an official and set the option module timingcable = 1, I either do a system has to fall (so that the connection was lost over ssh) or I get the message Bad HDLC in asterisk and the E1 flow was not raised. On 29.11.2010 10:16, Grigoriy Puzankin wrote: Здравствуйте, Антон. Мы заказывали кабель у Мототелекома (если не ошибаюсь). В наличии у них его нет, но под заказ с очередной поставкой они могут его достать. Либо купите через e-bay или вражеский интернет-магазин. У нас он соединяет две 4-портовые карточки. Не помню, что именно было до того, как его поставили, но какие-то проблемки были. 25.11.2010 19:23, Захаров Антон пишет: Hello everyone. I have a timing slips errors and I can't understand what source of the problem is. My installation has 2 digium cards: TE420 and TE220 cards in one server. There are 3 spans (E1) to PSTN and 3 spans to internal PBS stations - normal installation for transit communication. Span configuration is: span=1,1,0,ccs,hdb3 #TE420 - first port. To PSTN. span=2,0,0,ccs,hdb3 #TE420 - second port. To PBX. span=3,2,0,ccs,hdb3 #TE420 - third port. To PSTN. span=4,0,0,ccs,hdb3 #TE420 - fourth port. To PBX. span=5,3,0,ccs,hdb3 #TE220 - first port. To PSTN. span=6,0,0,ccs,hdb3 #TE220 - second port. To PBX. I should to say, that PBXs are interconnected through router (doesn't know anything about it). So all schema looks like this: http://yfrog.com/jjschemaj Spans 1-5 works fine, but on span 6 (marked bold) I have rising timing slips counter. I think it's appearing because I'm getting a primary timing source on span 1 - first port on TE420. But TE220 doesn't use it's span 5 for timing source, because it has priority 3, so it could be a sync problem. Am I wrong? I'm started to think about timing cable for syncing timing on first card and second. Should I use it? It's a problem to bought cable in our city (Russia,Moscow). All resellers sell only cards. Could I use floppy or IDE cable to interconnect cards? As I see in picture of cable, it's a direct 16 pin cable. Does anybody know something about timing cable for different cards? How I can solve my problem? Thanks for attention -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk on smartphone?
Hello Some SOHO prospects only have a cellphone and I was wondering if someone had investigate running Asterisk on a smartphone, to perform tasks such as IVR, CID rewriting, voice-mail, notifications through e-mails, etc.? 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
Re: [asterisk-users] New implementation asterisk
This sounds a bit suspect. What sound files are you talking about? Voicemail? Prompts? Responses? Dictation? Phone call recordings, outgoing and incoming to and from the call center. That explanation sounds bogus. Where are you seeing segmentation errors? What processes are faulting? Do you see the same thing if you restart Asterisk or MySQL? In a production environment, 'seg-faulting' is unacceptable and needs to be resolved. Indeed, that's why I'm opting for a fresh install and this time setting it up in the right way. ) What does the system do? It is supposed to record the calls of the call agents there is about 10 - 15 call agents. The reason for storing them is that a third party should be able to retrieve them through our corporate website. I don't understand why it is necessary to store these files into a database? To me this is not needed at all and would GREATLY improve performance if we just used a path to the files stored into the database. ) What version of Asterisk? 1.8 ) What channel technologies are used? I will have to get back to you on this, I'm not exactly sure. ) Average simultaneous calls? 8 * 5 mins ) Peak simultaneous calls? 15 * 5 mins Thanks for the quick reply -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: vrijdag 26 november 2010 21:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] New implementation asterisk You have provided no details to base suggestions on, so take any suggestions with a grain of 'salt.' On Fri, 26 Nov 2010, Edwin Blommaerts wrote: -The people that setup our server made asterisk write the sound files onto the hard drive, and then somewhat later store these files into my-sql. Is this the proper way to do it? Or would it make more sense to just have my-sql store the path to the file? Or is it possible to store the data directly into my-sql from asterisk? So in general what is the best way to make this happen? Has anyone implemented something similar? This sounds a bit suspect. What sound files are you talking about? Voicemail? Prompts? Responses? Dictation? Our current server now shows segmentation errors at boot and they claim that this is due to queries on the database. That explanation sounds bogus. Where are you seeing segmentation errors? What processes are faulting? Do you see the same thing if you restart Asterisk or MySQL? In a production environment, 'seg-faulting' is unacceptable and needs to be resolved. I’ve also checked the query it returns the data to us in less then 3 seconds, so I’m doubting this can be the trouble. In some environments, 3 seconds might as well be never. I'd aim for sub-second response for anything that affects call processing. -Sometimes the phone lines hang waiting, people can’t call out because asterisk seems to be holding on to these lines for a certain amount of time. What could be the cause of this? Does this happen with every call? What does the console log (with debug and verbose cranked up) look like when this happens? -What is the best hardware architecture for asterisk implementation? My-sql and asterisk on separate servers, or doesn’t it really matter? With no details, this comes down to personal preference. For 'big systems,' I like to 'front-end' a pool of Asterisk servers with a couple of boxes running OpenSIPS and I like to run databases on boxes by themselves. OpenSIPS lets you load balance and take boxes out of production easily. I think of database servers and 'telephony servers' as having different characteristics and needs. I really hope to get some advice, because, I’ve began to doubt/question the people who set the server up. I’m happy with any information, good references that you can give me. If you would like more concrete suggestions, please reply with more details. Basic details like: ) What does the system do? ) What version of Asterisk? ) What channel technologies are used? ) Average simultaneous calls? ) Peak simultaneous calls? may shed light on an appropriate solution. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 This email is scanned with Mcafee Groupshield. --- This email and any attached files are confidential and may be legally privileged. If you are not the intended recipient, any disclosure, reproduction, copying, distribution, or other dissemination or use of this communication is strictly prohibited. If you have received this transmission in error please notify the sender immediately
[asterisk-users] resending cause codes
hello,
i'm testing sending ISDN cause codes to customer pbx (test scenario for
unallocated number)
topology:
PSTN-E1-AsteriskA-AsteriskB-SOMEPBX
INVITE from SOMEPBX to PSTN
AsteriskA sends to AsteriskB
Status-Line: SIP/2.0 503 Service Unavailable
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
how can i resend HangupCauseCode from AsteriskB to SOMEPBX?
i'm tried this on AsteriskB
exten = _X.,1,Dial(SIP/AsteriskA,${EXTEN})
exten = _X.,n,Hangup(${SIP_HEADER(X-Asterisk-HangupCauseCode)})
thanks
--
---
Marek Cervenka
===
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Re: [asterisk-users] Asterisk on smartphone?
Do you mean, using the smart phone as an Asterisk server, or as a device (i.e., an extension)? I think running Asterisk in server mode would run up against blocking of SIP traffic on most voice networks. Also, you would probably run into issues with battery life, and with availability (what if you are flying and have to turn off your smart phone - suddenly the phone number wouldn't go to the IVR or voice mail any more). I would also be concerned about constantly changing IP addresses, firewalling within the carrier network, ... As an extension, I believe there are some softphone implementations available for Android (they might still run up against blocked SIP traffic, though). Alternatively, you can probably also use follow me or plain old forwarding to get calls to the cell phone, and DNDI for dialing outgoing calls. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Monday, November 29, 2010 2:09 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk on smartphone? Hello Some SOHO prospects only have a cellphone and I was wondering if someone had investigate running Asterisk on a smartphone, to perform tasks such as IVR, CID rewriting, voice-mail, notifications through e-mails, etc.? 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 -- _ -- 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] IAX2 and INVAL packets
On 11/18/2010 08:01 PM, Sebastian wrote: Is anybody here familiar with the meaning of INVAL packets for IAX2? Every few days I get a dropped outgoing call in the middle of the conversation (the outgoing call has been connected for few minutes) when an incoming call comes in. The log reads the following when this happens: Just answering myself here. After spending few months troubleshooting this completely erratic fault - it turns out that it was the ADSL router. I still don't know for certain the exact mechanism of the fault - but it looks like my IAX provider would receive some packets which were invalid from my direction - and hang-up calls randomly. There was no rhyme or reason for it. Sometime it would happen few times a day, sometime once every few days. Sometime 10 seconds into the call, sometime when another call would come in. But it always seemed to be connected with the received INVAL message in the logs. I replaced the router (a TP-Link ADSL router) with a Belkin router (that's what I had to hand) and everything is fine now for more then a week. The other router did not give any signs of trouble otherwise - all other Internet, email, openvpn and any other traffic going through it was fine. But somehow it would generate this problem with Asterisk. I just thought it might help somebody some day. Sebastian [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Immediately destroying 2963, having received INVAL [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Destroying call 2963 [Nov 17 15:25:04] DEBUG[11242] chan_iax2.c: We're hanging up IAX2/ihs_trunk_out-2963 now... [Nov 17 15:25:04] VERBOSE[11242] chan_iax2.c: -- Hungup 'IAX2/ihs_trunk_out-2963' And more setup details, for those who still have the will to live :-) Asterisk version: 1.6.2.13 Internal externsions: everything on SIP - 3 Grandstream GXP-2000, 2 analog phones on a pci OpenVox card and 2 Linphone softphones Trunks: IAX2 Trunks provider: Gradwell Asterisk machine: 800Mhz Intel Pentium, 512MB of RAM Internet connection: Tiscali business ADSL I am happy to post here any config files and logs you might think would be relevant. This is not consistent - and I've managed to have 4 concurrent calls which held 30 minutes (before I hung them up) when I tried. So not easy to replicate. Sebastian -- _ -- 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] How to hangup all channels
Thank you, i want to follow your idea, how i can send and receive data from/to
Command Line in PHP Script?Thank you in advance
Date: Sat, 27 Nov 2010 08:45:47 -0800
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to hangup all channels
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
1) sudo /etc/init.d/asterisk restart
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x 'channel request hangup ${CHANNEL_NAME} for
each channel.
3) Write a script to do #2 using AMI.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Preserve CallerID on transfers
2010/11/27 Fabiano Carlos Heringer b...@grupoheringer.com.br Hi, it´s possible to mantain the original CallerId when making transfers? (atx or blind) Example: A calls to B, A transfer to C, C see the CallerID of B, and not A... It´s possible? yes Thanks1 -- _ -- 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] How to hangup all channels
2 ways:
Read http://www.voip-info.org/wiki/view/Asterisk+AGI
or in PHP - system (asterisk -rx 'core restart now' /dev/null);
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe D'alessio
Sent: 29 November 2010 14:47
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to hangup all channels
Thank you, i want to follow your idea, how i can send and receive data from/to
Command Line in PHP Script?
Thank you in advance
Date: Sat, 27 Nov 2010 08:45:47 -0800
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to hangup all channels
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
1) sudo /etc/init.d/asterisk restart
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x 'channel request hangup ${CHANNEL_NAME} for
each channel.
3) Write a script to do #2 using AMI.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] How to hangup all channels
Un-top-posting...
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
On Sat, 27 Nov 2010, Steve Edwards wrote:
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x 'channel request hangup ${CHANNEL_NAME} for
each channel.
3) Write a script to do #2 using AMI.
On Mon, 29 Nov 2010, Giuseppe D'alessio wrote:
Thank you, i want to follow your idea, how i can send and receive data
from/to Command Line in PHP Script?Thank you in advance
1) Use 'popen().'
2) Use 'proc_open().'
3) Google for 'asterisk php ami.'
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] How to hangup all channels
Un-top-posting... From: Giuseppe D'alessio Thank you, i want to follow your idea, how i can send and receive data from/to Command Line in PHP Script? On Mon, 29 Nov 2010, Andrew Thomas wrote: Read http://www.voip-info.org/wiki/view/Asterisk+AGI An AGI is executed in the context of a channel. Are you suggesting the OP write an AGI so he can call into his system to ask it to hang up all channels? -- 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] Firewalling and Asterisk
On Sun, 28 Nov 2010, Jeremy Kister wrote: On 11/28/2010 12:03 PM, Silver Thorne wrote: So, I am wondering if anyone has a firewall/IP tables statement that keep out unauthorised users? No one seems to get in as we use really http://jeremy.kister.net/code/iptables/ if you already have an iptables configuration, the throttle section is important. if not, the iptables.init script can likely drop in place. if you only need north-american ip addresses to talk to your asterisk box, i suggest you also run the make-non-na.pl from cron every week. +1 Jeremy - these scripts, for NA PBXes, are perfect (and even without the heavy handed blocking of the rest of the world, the iptables stuff is invaluable). If I am digesting it correctly, this set of iptables rules does exactly what fail2ban would do, minus the logging, and without the overhead of a scripting language, correct? Love it! j -- _ -- 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] How to hangup all channels
Re-top-posting... I was merely pointing out that AGI exists (teach a man to fish...)! Sorry for not being as perfect as you... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: 29 November 2010 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to hangup all channels Un-top-posting... From: Giuseppe D'alessio Thank you, i want to follow your idea, how i can send and receive data from/to Command Line in PHP Script? On Mon, 29 Nov 2010, Andrew Thomas wrote: Read http://www.voip-info.org/wiki/view/Asterisk+AGI An AGI is executed in the context of a channel. Are you suggesting the OP write an AGI so he can call into his system to ask it to hang up all channels? -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] Firewalling and Asterisk
On 11/29/2010 11:03 AM, Jeff LaCoursiere wrote: If I am digesting it correctly, this set of iptables rules does exactly what fail2ban would do, minus the logging, and without the overhead of a scripting language, correct? Very similar to fail2ban, but not quite the same: * this'll block hosts based on X authentication attempts (good OR bad) (fail2ban only counts bad attempts) * this cannot detect encrypted attempts (SIPS), fail2ban can -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] How to hangup all channels
Un-un-top posting... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: 29 November 2010 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to hangup all channels Un-top-posting... From: Giuseppe D'alessio Thank you, i want to follow your idea, how i can send and receive data from/to Command Line in PHP Script? On Mon, 29 Nov 2010, Andrew Thomas wrote: Read http://www.voip-info.org/wiki/view/Asterisk+AGI An AGI is executed in the context of a channel. Are you suggesting the OP write an AGI so he can call into his system to ask it to hang up all channels? On Mon, 2010-11-29 at 16:11 +, Andrew Thomas wrote: Re-top-posting... I was merely pointing out that AGI exists (teach a man to fish...)! Sorry for not being as perfect as you... I think you are missing the point entirely. The OP wants to hangup all channels, and appears to want to do so from a web interface, which makes sense. Writing an AGI does not make sense, as you would have to call into the PBX to execute it, and basically hangup on yourself. Don't get me wrong, it would work, but it seems like a bad design. Teach a man to fish, but teach him to do it with a fishing rod, not a crowbar. j -- _ -- 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 on smartphone?
On Mon, 29 Nov 2010 02:26:35 -0800, Kevin Keane wrote: Do you mean, using the smart phone as an Asterisk server, or as a device (i.e., an extension)? I think running Asterisk in server mode would run up against blocking of SIP traffic on most voice networks. Also, you would probably run into issues with battery life, and with availability (what if you are flying and have to turn off your smart phone - suddenly the phone number wouldn't go to the IVR or voice mail any more). I would also be concerned about constantly changing IP addresses, firewalling within the carrier network, ... As an extension, I believe there are some softphone implementations available for Android (they might still run up against blocked SIP traffic, though). Alternatively, you can probably also use follow me or plain old forwarding to get calls to the cell phone, and DNDI for dialing outgoing calls. Actually there is a project called Serval that runs Asterisk on Android as a means to create a mobile mesh. It's related to the Village Telco project. As such David Rowe has some observations about it on bis blog. http://www.rowetel.com/blog/?p=168 Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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] Firewalling and Asterisk
On Sun, Nov 28, 2010 at 12:24 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 28 Nov 2010, Silver Thorne wrote: I have noticed lately that there have been several attempts to hack our Asterisk server. So, I am wondering if anyone has a firewall/IP tables statement that keep out unauthorised users? 0) Read the list archives, this comes up weekly. 1) Determine who (in terms of external IP addresses) should be allowed to connect to your server. 2) Create a list of iptables commands to allow those IP addresses. 3) Deny everybody else. 4) Use 'fail2ban' or something similar to detect abusive addresses and block them, if only for an [hour|day|week] or so. Even if you have 'mobile' users who 'need to connect from everywhere' you can probably define 'everywhere' a bit better like 'not from North Korea' or 'not from Africa' -- with suitable apologies to readers from North Korea or Africa. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 I agree with Steve, this is the safest way to tackle it. For the road warriors that demand an extension, I use SNOM 370VPN if they want to carry around a real phone or openvpn x-lite on their laptops. Thanks, Steve T -- _ -- 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] ID'ing failed auth IPs
So when someone's brute forcing your server is there a way to identify the originating IPs without using a tcpdump? When I get a failed auth on the console it shows 'acco...@asteriskserver' then tag=as25ca5023 (or some random string, though it's a bit odd as alwaysauthreject = yes is on in sip.conf). Anyway, the logs don't show anything more useful either. Is there something obvious I'm missing? Cranking up verbosity on the console doesn't seem to do anything. 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] How to initiate a two-party call from within Asterisk
The desired result is that user A's phone rings; when he picks it up, user B is dialled, and user A's channel is connected to that. (This is to be a back-end for a web-based address book.) -- _ -- 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] ID'ing failed auth IPs
On Mon, Nov 29, 2010 at 2:01 PM, Hose hose+aster...@bluemaggottowel.com wrote: So when someone's brute forcing your server is there a way to identify the originating IPs without using a tcpdump? When I get a failed auth on the console it shows 'acco...@asteriskserver' then tag=as25ca5023 (or some random string, though it's a bit odd as alwaysauthreject = yes is on in sip.conf). Anyway, the logs don't show anything more useful either. Is there something obvious I'm missing? Cranking up verbosity on the console doesn't seem to do anything. hose You can use IPTABLES to log all traffic on a port for you. Instead of ACCEPT or DROP use LOG. -- _ -- 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] Trouble with TE122 on HP DL120G6 - can't disable USB
I have recently built a single-T1 Asterisk box using an HP DL120G6 with a Digium TE122 card. I was finding that I was getting missed interrupts on the TE122, causing the driver to report that it was increasing latency. It kept doing this until the T1 did not work reliably. I tried my usual procedure of disabling the USB subsystem in the BIOS (or so I thought), but I found that the USB drivers still got loaded and the missed interrupts still occurred. So I added nousb as a kernel boot option, which successfully prevented the USB drivers being loaded, and got rid of the missed interrupts. Unfortunately, this also stopped the PS/2 keyboard port working. It turns out that the PS/2 ports on the DL120G6 are not REAL PS/2 ports on an 8042 controller, but just PS/2 to USB converters going to the motherboard's USB subsystem. And turning off USB in the BIOS merely disconnects the system's external USB ports. That seriously sucks, IMHO! So, does anyone know how to get this system working reliably, so that the USB drivers do not cause the TE122 driver to miss interrupts? Using Zaptel 1.4.12.1, with Asterisk 1.2. Please don't tell me try DAHDI unless you KNOW that this issue has been fixed in DAHDI. Because my application is based on Asterisk 1.2, DAHDI would be majorly painful to try, and will only be a very last resort. However, if it has been fixed in DAHDI, and someone can point me at the specific fix, I'd be happy to try back-porting it. Thanks in advance, Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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] Stability..
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub dotnet...@gmail.com wrote: Sorry, what I meant was: server*CLI remove extension (hit tab) segfault.. 1.4.22 It could be an extension name Where is the error trapping if this is the case.. Who writes this shit? If you remove an extension that is being used, control could flow into the now non-loaded extension, and THAT is what caused your core dump. Don't do that. If you want to NEVER load extensions and also not crash asterisk, you're better off taking a look at modules.conf, particular making entries that begin with noload = You can also take a look at 1.6 and make menuconfig, and just not build modules that you don't want. Then they'll really never load at startup. -- _ -- 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] Trouble with TE122 on HP DL120G6 - can't disable USB
On 11/29/2010 11:11 AM, Tony Mountifield wrote: I have recently built a single-T1 Asterisk box using an HP DL120G6 with a Digium TE122 card. I was finding that I was getting missed interrupts on the TE122, causing the driver to report that it was increasing latency. It kept doing this until the T1 did not work reliably. I tried my usual procedure of disabling the USB subsystem in the BIOS (or so I thought), but I found that the USB drivers still got loaded and the missed interrupts still occurred. So I added nousb as a kernel boot option, which successfully prevented the USB drivers being loaded, and got rid of the missed interrupts. Unfortunately, this also stopped the PS/2 keyboard port working. It turns out that the PS/2 ports on the DL120G6 are not REAL PS/2 ports on an 8042 controller, but just PS/2 to USB converters going to the motherboard's USB subsystem. And turning off USB in the BIOS merely disconnects the system's external USB ports. That seriously sucks, IMHO! So, does anyone know how to get this system working reliably, so that the USB drivers do not cause the TE122 driver to miss interrupts? DAHDI does add idle buffers which can allow the max latency to be caped at something low. This change went in revision 7517 [1]. You would still have data problems in the channel but you wouldn't have to worry about the framer getting confused. Some other things you might try: 1) Is there an option for legacy keyboard emulation in your BIOS that you could disable? It could be that there is a long running System Management Interrupt running to see if it should make the USB keyboard look like a PS/2 keyboard for DOS, etc.. 2) Do you have the latest BIOS for the DL120G6? 3) Update your kernel to the 2.6.32 stable series in case the problem really is in the USB stack. 4) Use /proc/irq/IRQ num/smp_affinity to force the USB interrupts onto CPU0 and the TE122 interrupts onto CPU1 (assuming the DL120G6 is dual core). Cheers, Shaun [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=7517 -- 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] Asterisk on smartphone?
On Mon, Nov 29, 2010 at 11:08:36AM +0100, Gilles wrote: Hello Some SOHO prospects only have a cellphone and I was wondering if someone had investigate running Asterisk on a smartphone, to perform tasks such as IVR, CID rewriting, voice-mail, notifications through e-mails, etc.? I believe someone has already built Asterisk on Maemo (for N900) - http://my-maemo.com/software/applications.php?name=AsteriskfldAuto=1081faq=37 Sadly, no support for DAHDI ;-( -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] HA8 + B400M not configured with genconf_parameters
On 11/26/2010 05:05 AM, Olivier wrote: Hi, On a Lenny system, with dahdi 2.4.0, libpri 1.4.11.5 and asterisk 1.6.1.18, I inserted a new Digium HA8 + B400M card. My usual installation fails. I can see it listed : # lspci -n | grep d161 01:0b.0 0200: d161:8007 (rev 11) # lspci -vn 01:0b.0 0200: d161:8007 (rev 11) Subsystem: d161:8007 Flags: medium devsel, IRQ 22 I/O ports at d400 [size=256] Memory at fe5fe800 (32-bit, non-prefetchable) [size=1K] Expansion ROM at 2002 [disabled] [size=128K] Capabilities: [c0] Power Management version 2 Kernel modules: wctdm24xxp # lsdahdi # modprobe wctdm24xxp # dahdi_genconf -v modules Default parameters from /etc/dahdi/genconf_parameters Generating /etc/dahdi/modules # cat modules # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules) on Fri Nov 26 12:02:11 2010 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. wctdm24xxp # dahdi_genconf -v system Default parameters from /etc/dahdi/genconf_parameters Empty configuration -- no spans Generating /etc/dahdi/system.conf Is there something I could do to change this ? Does the modprobe wctdm24xxp command complete successfully? Is there anything in dmesg or in the output of dahdi_scan after loading the driver? Cheers, 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 2.4.0 produces HDLC errors with echo canceler
On 11/27/2010 11:03 AM, James Lamanna wrote: Hi, After upgrading to DAHDI 2.4.0 from Zaptel, I noticed a lot of HDLC errors on my console: [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 These errors prevented calls from being made and received on my PRI spans. This seems similar to bug 15498: https://issues.asterisk.org/view.php?id=15498 Which says this was fixed in 2.2...so maybe it got back into 2.4? I can get rid of the errors by disabling the mg2 echo canceler in /etc/dadhi/system.conf. Do you have a hardware echocan module installed on your card? If so, it's strange indeed that the error goes away when you disable mg2. What is the complete output of your /etc/dahdi/system.conf? -- 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] Stability..
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub dotnet...@gmail.com wrote: Sorry, what I meant was: server*CLI remove extension (hit tab) segfault.. 1.4.22 It could be an extension name Where is the error trapping if this is the case.. Who writes this shit? If you get hurt do you blame your Mama for having some fun in the past? I sure do now. On 28 November 2010 22:21, dotnetdub dotnet...@gmail.com wrote: Beautiful.. Asterisk 1.4.22 remove extension and hit tab from the CLI.. -- _ -- 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] How to initiate a two-party call from within Asterisk
On Mon, Nov 29, 2010 at 11:07 AM, Roger Burton West ro...@firedrake.orgwrote: The desired result is that user A's phone rings; when he picks it up, user B is dialled, and user A's channel is connected to that. (This is to be a back-end for a web-based address book.) This is click-to-call. It can be done with the Asterisk Manager Interface (AMI). See this site: http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html -- Chris -- _ -- 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 on smartphone?
On Mon, 29 Nov 2010, Gilles wrote: Hello Some SOHO prospects only have a cellphone and I was wondering if someone had investigate running Asterisk on a smartphone, to perform tasks such as IVR, CID rewriting, voice-mail, notifications through e-mails, etc.? While I can run Asterisk on my Nokia N900, I have to say it's purely for my own benefit and to be a bit geeky... Personally people like this are cheakskates and not worth your time and effort in trying to implment a solution for them unless they're going to spend 1000's with you. Gordon -- _ -- 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] Stability..
On 29 November 2010 18:52, C F shma...@gmail.com wrote: On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub dotnet...@gmail.com wrote: Sorry, what I meant was: server*CLI remove extension (hit tab) segfault.. 1.4.22 It could be an extension name Where is the error trapping if this is the case.. Who writes this shit? If you get hurt do you blame your Mama for having some fun in the past? I sure do now. Apologies very long and bad day yesterday stuck on a customer site for 14 hours doing something that should take 1 hour 50 max but I'm not even going to go into it... Sorry for being an asshole. Regards On 28 November 2010 22:21, dotnetdub dotnet...@gmail.com wrote: Beautiful.. Asterisk 1.4.22 remove extension and hit tab from the CLI.. -- _ -- 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] kernel: dahdi: Detected time shift.
On 11/25/10 7:12 AM, Jonas Kellens wrote: @ Shaun Ruffell : What do you mean by Wall time ? This server is indeed also time server (ntpd is running) Basically that the time on the server matches up with the time actual time you would see on a wall clock. Based on your response to Willaim Stillwell below, it sounds like it is keeping accurate wall time. @ Mark Deneen : No, no monitor attached. This is a Xen VPS. I do have a VPS interface, but this is also frozen when the server hangs... @ William Stillwell : I run Asterisk 1.6.2.10 with Dahdi 2.4.0 (as timing source). Time on my server seems very consistent. When doing /usr/sbin/ntpdate (once a month) there is a very very small offset. How often does this happen? Daily, Weekly, etc? Also, when you say the server freezes is it completely locked or can you make the caps lock key LED go on and off on your keyboard? Also, you said that this started occurring lately. Was there some other system change that was made about the time you noticed the problems start? -- 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
[asterisk-users] OT: for those wondering on the stability
r...@pbx:~# uptime 23:10:15 up 606 days, 9:38, 1 user, load average: 0.31, 0.08, 0.02 Customer called they are having a scheduled power outage for most of the day because of construction if I can shut down the machine gracefully. So I decided to run uptime first. Enjoy -- _ -- 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
