| 00:03.14 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 00:04.53 | *** join/#asterisk theron (~theron@66.220.144.81) |
| 00:19.41 | *** part/#asterisk mjordan (~matt@nat/digium/x-yjfpkfrjjgcsideo) |
| 00:22.49 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-34-110.dynamic.qsc.de) |
| 00:26.26 | *** join/#asterisk Es0KrptiK (~~Es0Krpti@2605:f700:40:c00::6139:adb9) |
| 00:49.16 | *** join/#asterisk jsjc (~Adium@126.Red-79-153-28.dynamicIP.rima-tde.net) |
| 00:58.08 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.127) |
| 01:20.04 | *** join/#asterisk jpoz (~jpoz@184.169.152.1) |
| 01:29.04 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.133) |
| 01:34.10 | *** join/#asterisk MaliutaLap (~nobusines@59.167.17.219) |
| 01:45.09 | AceFrahm | Just in case anyone else is wondering, I fixed the Playback() problem by including |
| 01:45.15 | AceFrahm | nat=yes |
| 01:46.08 | AceFrahm | in the user's configuration. The problem was the media packets somehow weren't making it through 2 routers without it. |
| 01:48.10 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 01:48.19 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 01:56.45 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.188) |
| 02:01.30 | *** join/#asterisk ipengineer (~zconkle@108-206-120-218.lightspeed.mckntx.sbcglobal.net) |
| 02:03.01 | *** join/#asterisk smirker (~x@CPE-60-228-93-204.lns8.woo.bigpond.net.au) |
| 02:10.21 | *** join/#asterisk mjordan (~matt@75.76.55.191) |
| 02:10.21 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:11.21 | *** join/#asterisk D30 (~deo@203.177.9.66) |
| 02:16.20 | *** join/#asterisk D30_ (~deo@203.177.9.66) |
| 02:17.04 | *** join/#asterisk D30_ (~deo@222.127.13.226) |
| 02:33.01 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 02:34.18 | *** join/#asterisk s_enya (~enya@fw.technoface.co.jp) |
| 02:34.25 | s_enya | hello |
| 02:49.03 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 02:57.19 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.134) |
| 03:01.41 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 03:01.51 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 03:12.48 | *** join/#asterisk bitfury (~bitfury@unaffiliated/bitfury) |
| 03:58.03 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.134) |
| 04:04.53 | s_enya | I want to get files from under "asterisk/trunk/tests/channels" directory. but this directory is not contain svn checkout. why reason? |
| 04:56.23 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 04:56.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.133) |
| 05:24.19 | *** join/#asterisk theron (~theron@67.42.4.81) |
| 05:39.27 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.133) |
| 05:49.49 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 05:51.12 | *** part/#asterisk s_enya (~enya@fw.technoface.co.jp) |
| 05:53.05 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.133) |
| 06:04.16 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 06:06.04 | *** join/#asterisk kgunnIt (~kgunn@ec2-23-23-228-7.compute-1.amazonaws.com) |
| 06:12.23 | *** join/#asterisk hos7ein (~chatzilla@2.191.244.144) |
| 06:37.27 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 06:46.26 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 06:53.54 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 06:57.24 | *** part/#asterisk LiuYan (~LiuYan@211.154.128.133) |
| 07:00.33 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 07:00.54 | *** join/#asterisk MaliutaLap (~nobusines@eth637.qld.adsl.internode.on.net) |
| 07:02.03 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:09.21 | *** join/#asterisk D30 (~deo@203.177.9.66) |
| 07:09.43 | *** join/#asterisk andrewyager (~andrewyag@114.141.100.226) |
| 07:17.41 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 07:20.53 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 07:28.59 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 07:29.53 | *** join/#asterisk D30_ (~deo@203.177.9.66) |
| 07:31.56 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 07:35.54 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:36.21 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:49.48 | *** join/#asterisk bsdice (~bsdice@embinet.eu) |
| 07:51.25 | bsdice | morning |
| 07:52.35 | AceFrahm | no. |
| 07:56.46 | bsdice | yes |
| 07:59.08 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 08:15.00 | *** join/#asterisk andrewyager (~andrewyag@CPE-60-225-76-199.hhui1.cht.bigpond.net.au) |
| 08:18.58 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 08:19.14 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 08:34.36 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:40.54 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:48.55 | *** join/#asterisk D30_ (~deo@203.177.9.66) |
| 08:52.52 | *** join/#asterisk oatha (~oatha@201.86.221.98) |
| 08:55.54 | *** join/#asterisk D30_ (~deo@203.177.9.66) |
| 08:55.59 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 08:56.42 | *** part/#asterisk tarokinuu (~alexan@188.168.144.185) |
| 08:57.21 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 09:05.09 | *** join/#asterisk Rokfan (~Rokfan@D522448D.static.ziggozakelijk.nl) |
| 09:08.03 | *** join/#asterisk D30 (~deo@203.177.9.66) |
| 09:08.54 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 09:17.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.164) |
| 09:22.46 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 09:24.02 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 09:29.12 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 09:29.53 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.133) |
| 09:34.14 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 09:42.30 | *** join/#asterisk vlad_starkov (~vlad_star@194.67.31.130) |
| 09:47.05 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 09:50.31 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:50.32 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:57.42 | *** join/#asterisk moy_ (~moy@173.239.155.74) |
| 10:05.13 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 10:07.27 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 10:07.55 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 10:07.55 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 10:08.53 | *** join/#asterisk vlad_starkov (~vlad_star@194.67.31.130) |
| 10:25.53 | *** join/#asterisk vlad_starkov (~vlad_star@194.67.31.130) |
| 10:28.14 | *** join/#asterisk jacobkiers (~jacobkier@jacobkiers.net) |
| 10:30.29 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 10:30.32 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 10:39.19 | *** join/#asterisk Kraln (~kraln@69.169.90.240) |
| 10:49.42 | *** join/#asterisk vlad_starkov (~vlad_star@194.67.31.130) |
| 10:57.11 | *** join/#asterisk calum_ (~calum_@88-104-182-200.dynamic.dsl.as9105.com) |
| 10:57.29 | *** join/#asterisk vlad_starkov (~vlad_star@194.67.31.130) |
| 11:02.52 | *** join/#asterisk BarthezZ (~bart@monitoring.deheij-ict.nl) |
| 11:04.46 | jacobkiers | Hi all. What's wrong with this: https://gist.github.com/jacobkiers/9d5d9a2dfa0e9bdea703? I'm trying to conditionally set the account code, but the dial plan always skips to priority 2. |
| 11:05.10 | jacobkiers | I've redacted the console output to hide the phone number. It is actually 11 characters long. |
| 11:19.08 | wdoekes | [${LEN(${EXTEN})} > 17] <-- missing $ |
| 11:19.17 | wdoekes | $[...] |
| 11:19.57 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 11:20.10 | *** join/#asterisk makmak78 (~makmak78@195-67-63-194.customer.telia.com) |
| 11:20.15 | Penguin | You're awfully quick for it being so early in the morning. |
| 11:20.40 | wdoekes | $[...] evaluates to a number, 0 or not-0.. [...] (without $) simply evaluates to true, so you never end up in the else-case |
| 11:20.41 | makmak78 | Hello, anybody know if 183 session progress has an avent in Asterisk 1.4 AMI |
| 11:20.50 | makmak78 | event* |
| 11:20.59 | wdoekes | Penguin: 12:20 here |
| 11:21.01 | makmak78 | eg new state event |
| 11:21.03 | Penguin | Oh. Well... 15 minutes since asked. I guess you're not all THAT quick. |
| 11:21.33 | wdoekes | and not that early either ;) |
| 11:21.49 | Penguin | You answered within a minute of my reading his question. I thought he had just asked it. |
| 11:21.58 | Penguin | It's 0521 here, so I'm the one being slow. |
| 11:23.03 | *** join/#asterisk andrewyager (~andrewyag@CPE-60-225-76-199.hhui1.cht.bigpond.net.au) |
| 11:27.49 | Penguin | I should probably go back to bed. |
| 11:29.07 | *** join/#asterisk hos7ein (~chatzilla@2.191.244.144) |
| 11:45.44 | jacobkiers | wdoekes: Thanks, that worked :) |
| 11:51.14 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 12:02.04 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 12:02.48 | *** join/#asterisk CeBe (~CeBe@port-92-206-34-110.dynamic.qsc.de) |
| 12:09.26 | *** join/#asterisk vlad_starkov (~vlad_star@194.67.31.130) |
| 12:17.51 | *** join/#asterisk simNIX (~simNIX@92.254.60.156) |
| 12:18.37 | jacobkiers | Hmm. I would also like to strip the account code part for the CDRs. Using Set(CDR(dst)=${EXTEN:71}) does not work. Any ideas? |
| 12:18.48 | jacobkiers | I've also tried CDR(extension); didn't work either. |
| 12:21.30 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 12:23.08 | jacobkiers | Ah, my bad. |
| 12:23.12 | jacobkiers | It's a read-only variable. |
| 12:23.18 | mirela666 | :7:1 you wwanted? |
| 12:23.33 | mirela666 | aha |
| 12:23.58 | mirela666 | you can change CDR(dst) by sending it to another extension with Goto |
| 12:25.03 | mirela666 | for example you have 999 and 888 extensions in one context |
| 12:25.21 | mirela666 | and you dial 999 and your EXTEN will be 999 |
| 12:25.34 | mirela666 | then you do Got0(888,1) |
| 12:25.45 | mirela666 | your EXTEN will be 888 |
| 12:25.56 | mirela666 | and so CDR(dst) |
| 12:26.27 | jacobkiers | mirela666: I see. It was an optimisation anyway :) |
| 12:26.46 | jacobkiers | and yes, I actually meant :71; we have long account numbers... |
| 12:26.54 | mirela666 | oh |
| 12:27.12 | jacobkiers | had to patch asterisk for that, actually. |
| 12:27.27 | jacobkiers | apparently my C is better than my "Dialplan" ;) |
| 12:31.40 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 12:31.41 | mirela666 | hehehe good, EXTEN is limited to 30+ chars? |
| 12:31.59 | mirela666 | 30+- |
| 12:33.36 | *** join/#asterisk LiuYan (~LiuYan@222.125.137.217) |
| 12:52.14 | jacobkiers | mirela666: according to my patches by default the following |
| 12:52.20 | jacobkiers | Account code: 20 chars |
| 12:52.35 | jacobkiers | extension: 80 chars |
| 12:52.59 | jacobkiers | context: 80 |
| 12:53.13 | jacobkiers | but for some weird reason, the extension can also include the context :) |
| 12:53.31 | jacobkiers | so if exten+context > 80, it's bye-bye |
| 12:55.00 | jacobkiers | Here's my patch if you're interested: https://gist.github.com/jacobkiers/8004475 |
| 12:55.07 | jacobkiers | it's against Asterisk 11.5.1 |
| 13:00.35 | *** join/#asterisk rayzzz (rayzzz@gateway/shell/ircrelay.com/x-julrwuheehbevkld) |
| 13:06.22 | *** join/#asterisk Draecos (~Draecos@124-169-108-149.dyn.iinet.net.au) |
| 13:06.31 | *** join/#asterisk lnb (~lnb@CPE000347b24a71-CM602ad06bec2f.cpe.net.cable.rogers.com) |
| 13:07.12 | *** join/#asterisk makubi (~makubi@chello084113018217.7.12.vie.surfer.at) |
| 13:11.34 | mirela666 | jacobkiers: thx :0 |
| 13:12.18 | *** join/#asterisk Draecos (~Draecos@124-169-108-149.dyn.iinet.net.au) |
| 13:13.59 | *** join/#asterisk Draecos (~Draecos@124-169-108-149.dyn.iinet.net.au) |
| 13:15.53 | *** join/#asterisk vlad_starkov (~vlad_star@194.67.31.130) |
| 13:16.36 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 13:19.13 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 13:25.44 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:42.26 | *** part/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 13:43.15 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:43.18 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 13:58.08 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:58.08 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:59.11 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 14:00.28 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 14:03.26 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 14:03.30 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 14:05.55 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:13.11 | Katty | morning |
| 14:19.54 | *** join/#asterisk troyt (~troyt@2601:7:6d00:432:44dd:acff:fe85:9c8e) |
| 14:21.44 | Qwell | glomps Katty |
| 14:25.02 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:25.22 | Katty | hugs on Qwell |
| 14:25.47 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 14:30.41 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:32.19 | *** join/#asterisk theron (~theron@67.42.4.81) |
| 14:32.31 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 14:36.37 | resist0r | http://www.securityfocus.com/archive/1/530370 AST-2013-007: Asterisk Manager User Dialplan Permission Escalation |
| 14:40.29 | *** join/#asterisk human39_ (~human39@c-71-236-110-155.hsd1.pa.comcast.net) |
| 14:42.18 | [TK]D-Fender | resist0r: that patch solves very little |
| 14:42.26 | [TK]D-Fender | resist0r: AMI is still an open floodgate |
| 14:53.34 | *** join/#asterisk mjordan (~matt@nat/digium/x-ofllqvcywnklnxhq) |
| 14:53.34 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:54.47 | *** join/#asterisk gen1us2k (~gen1us2k@212.42.116.39) |
| 14:54.50 | gen1us2k | Hi all |
| 14:55.12 | gen1us2k | I have a problem around webrtc and asterisk with incoming calls |
| 14:56.06 | gen1us2k | how can I speed up RTP source probation? I have a approx. 10 seconds of one way audio |
| 14:57.31 | mjordan | gen1usk: change the probation config option in rtp.conf |
| 14:57.48 | gen1us2k | I set probation to 1 |
| 14:57.50 | gen1us2k | and to 2 |
| 14:57.53 | gen1us2k | and to 8 |
| 14:57.55 | mjordan | k, then probation is probably not your problem. |
| 14:57.58 | gen1us2k | and.. I have no changes |
| 14:58.28 | mjordan | probation of 1 means that after a single RTP packet it will lock onto that address and consider it the RTP source. |
| 15:02.11 | *** join/#asterisk BKhan (ca7d90e3@gateway/web/freenode/ip.202.125.144.227) |
| 15:02.20 | BKhan | Hi |
| 15:03.03 | BKhan | There is an issue as call connect to agents it got disconnected |
| 15:04.43 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 15:08.26 | [TK]D-Fender | BKhan: Show us the failed call with the appropriate channel debug enabled |
| 15:08.29 | [TK]D-Fender | ~pb |
| 15:08.29 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:08.31 | [TK]D-Fender | ^^^^ |
| 15:09.28 | BKhan | call not disconnected rtp packet not receving |
| 15:10.00 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 15:11.02 | *** join/#asterisk johnmilton (~johnmilto@ool-44c5f03d.dyn.optonline.net) |
| 15:11.55 | gen1us2k | if not probation then why I got this? |
| 15:15.13 | BKhan | D-Fender:We just analyes that rot was not send sending from our server. Cli is normal. |
| 15:15.32 | *** join/#asterisk mattcen (~mattcen@c110-22-201-130.sunsh4.vic.optusnet.com.au) |
| 15:17.47 | gen1us2k | I compiled asterisk-11-6-0 manuallyu |
| 15:18.21 | *** join/#asterisk johnmilton (~johnmilto@ool-44c5f03d.dyn.optonline.net) |
| 15:21.16 | *** join/#asterisk johnmilton (~johnmilto@ool-44c5f03d.dyn.optonline.net) |
| 15:22.50 | mjordan | gen1us2k: I don't think anyone can answer your question without seeing what is actually occurring. A debug trace showing the sip/rtp debug would be helpful (sip set debug on, rtp set debug on in the CLI). You could also try setting strictrtp to no (which completely disables strictrtp) - my guess is that won't help in your situation, but it's worth a try. |
| 15:24.26 | *** join/#asterisk rmill (~rmill@unaffiliated/rmill) |
| 15:24.27 | gen1us2k | I already tryed |
| 15:24.32 | gen1us2k | to set srtiprtp=no |
| 15:24.38 | gen1us2k | Not working at all |
| 15:24.52 | gen1us2k | I have one way packets |
| 15:24.59 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-ywmlyzbtsvldhysx) |
| 15:24.59 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:26.02 | gen1us2k | rtp packets going only from sip gateway and after 10 secs goes to two side |
| 15:26.09 | gen1us2k | from client and from gateway |
| 15:26.25 | gen1us2k | problem in routes? nat? or? |
| 15:26.40 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:26.40 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:27.24 | [TK]D-Fender | gen1us2k: We have no idea. You've shown precisely no debug for us to examine. |
| 15:27.46 | *** join/#asterisk hfp (~hfp@MTRLPQ0736W-LP130-01-2925269924.dsl.bell.ca) |
| 15:28.11 | hfp | Hi guys, is the RAN() function broken in 1.8? |
| 15:29.14 | mjordan | hfp: there is no function RAN |
| 15:29.44 | hfp | Ah my bad, a letter got erased...... |
| 15:30.51 | mjordan | No one has filed a bug against RAND, if that's what you meant |
| 15:31.22 | mjordan | why do you think it is broken? |
| 15:31.59 | [TK]D-Fender | I'm betting the "D" he missed was in his actual diaplan |
| 15:32.06 | [TK]D-Fender | so calling a non-existent function |
| 15:32.14 | [TK]D-Fender | And the actual one works fine |
| 15:32.20 | mjordan | heh |
| 15:32.34 | file | I RAN()... so far away |
| 15:32.43 | mjordan | file: was the destination pseudo-random? |
| 15:32.54 | file | maybe. |
| 15:33.05 | mjordan | well, if it was truly random... |
| 15:33.07 | [TK]D-Fender | big flock-up |
| 15:34.18 | *** join/#asterisk moy_ (~moy@173.239.155.74) |
| 15:34.37 | makubi | maybe he meant BRAN |
| 15:37.08 | makubi | ok |
| 15:37.23 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:38.22 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:47.49 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:50.17 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:50.29 | hfp | mjordan: Yes that is it, the "D" got erased somehow. I put it back and it's fine now |
| 15:51.34 | hfp | Another question: How would you go about sending a recorded mp3 as an attachment via email? |
| 15:52.29 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 15:52.31 | [TK]D-Fender | hfp: * can't record as MP#, so you'll have to do your own e-mail script that will convert first |
| 15:52.45 | [TK]D-Fender | hfp: read the voicemail.conf sample |
| 15:55.15 | hfp | [TK]D-Fender: Ok. I already have all the magic in place to convert to mp3 etc. I was using mutt but I was wondering if there is a better way |
| 15:55.25 | hfp | I'll check the voicemail.conf |
| 16:02.53 | *** join/#asterisk carrar (~tim@osburn.jp) |
| 16:04.11 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 16:08.36 | *** join/#asterisk navaismo (~navaismo@189.241.100.102) |
| 16:25.49 | *** join/#asterisk theron (~theron@66.220.144.81) |
| 16:31.52 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 16:37.17 | carrar | *Y*A*W*N* |
| 16:41.13 | hfp | When using System(), commands are run as root or as the user running the Asterisk daemon? |
| 16:41.38 | Greenlight | The user running asterisk |
| 16:42.43 | hfp | Hmmm. But this user has /sbin/false as his terminal in /etc/passwd, how can this possibly work? |
| 16:43.09 | Penguin | That's an interesting question. |
| 16:43.34 | Penguin | My asterisk shell is /bin/false, and I use System() all the time. |
| 16:45.50 | hfp | So do I... I am calling a bash script that eventually sends an email with mutt and I need to know which user I have to configure mutt for in order to get this working... |
| 16:48.09 | Penguin | I too use mutt to send email, but I don't configure mutt to do anything. |
| 16:49.27 | hfp | Penguin: Well as it stands, it doesn't send the email. Did you set a .muttrc? Where did you put it? |
| 16:50.24 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.98) |
| 16:50.31 | Penguin | System(/bin/echo "Stuff to say in the email."|/usr/bin/mutt -a ${FILE_TO_ATTACH} -s "A subject" -- ${EMAILADDR}) |
| 16:50.31 | Penguin | I just said I didn't configure mutt to do anything. |
| 16:52.20 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 16:55.22 | Penguin | Although, I do remember using the .muttrc for some testing when I was first trying to make things play nice. The file goes at ~asterisk/.muttrc |
| 16:57.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.98) |
| 17:00.42 | wdoekes | hfp: the terminal from passwd is not used, /bin/sh is invoked directly by asterisk. after all, you're trying to sh -c "some_stuff", not get a user-defined interactive terminal |
| 17:01.32 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 17:03.29 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 17:18.45 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 17:19.19 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 17:23.20 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 17:32.14 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.98) |
| 17:36.09 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 17:43.20 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.141) |
| 17:47.45 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:52.20 | *** join/#asterisk noko (pavel@gate6.zhovner.com) |
| 17:55.21 | noko | Hello guys. I'm desperate. I have Avaya 9611G Avaya deskphone. Documentation said that it support SIP, but actually very strange. |
| 17:55.43 | *** join/#asterisk rayzzz (rayzzz@gateway/shell/ircrelay.com/x-tfoddmbdhbzlgjcz) |
| 17:57.03 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:03.51 | newtonr | noko, What is the problem? |
| 18:04.54 | noko | It try to register with headers: |
| 18:04.58 | noko | Via: SIP/2.0/UDP 0.0.0.0:5060 |
| 18:05.10 | noko | Contact: <sip:100@0.0.0.0;transport=udp.... |
| 18:05.46 | newtonr | lol |
| 18:05.46 | [TK]D-Fender | ~pb |
| 18:05.46 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:05.48 | [TK]D-Fender | ^^^ |
| 18:05.52 | [TK]D-Fender | Show the actual full debug |
| 18:06.34 | *** join/#asterisk rayzzz (rayzzz@gateway/shell/ircrelay.com/x-hfztiwfwogfzaqsm) |
| 18:07.06 | newtonr | noko, yeah it would help to see the full SIP packets that Asterisk is receiving. |
| 18:08.19 | newtonr | noko, but otherwise, sounds like your phone is misbehaving or is misconfigured. I've not used that phone before. Also consider if there is any SIP ALG in the way that may be modifying things. |
| 18:08.55 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:10.20 | noko | here is the full debug http://pastebin.com/raw.php?i=SSAxbBmv |
| 18:11.00 | noko | in document's I found: SIP deskphones do not support Network Address Translation (NAT) |
| 18:13.58 | *** join/#asterisk gmalsack (~gmalsack@23.30.198.161) |
| 18:14.52 | gmalsack | ok, so who thinks they know their sip/udp stuff????? I've got a kicker of a packet trace for you... |
| 18:16.51 | *** join/#asterisk af_ (~af@93-43-29-101.ip90.fastwebnet.it) |
| 18:16.55 | Nugget | The REINVITE is coming from INSIDE THE HOUSE!!! |
| 18:17.08 | file | Nugget, lol |
| 18:17.14 | gmalsack | having problems getting a 7960 to register. |
| 18:17.44 | gmalsack | registration comes in from 172.24.3.254:51074 |
| 18:17.53 | Chainsaw | gmalsack: The 7960 is essentially SCCP hardware. It isn't that great at SIP. |
| 18:18.24 | gmalsack | asterisk responds with ack to 172.24.3.254:51074 |
| 18:18.38 | gmalsack | phone responds icmp port unreachable |
| 18:19.07 | gmalsack | inspecting the packet sip states source port is 5060, udp states source port is 51074.... thoughts? |
| 18:19.55 | *** join/#asterisk SuD (~alex@13-197-39-46.usuarios.innovasur.com) |
| 18:20.56 | Chainsaw | gmalsack: From what I remember, directmedia=no is vital. |
| 18:21.35 | Chainsaw | gmalsack: For the 7940 & 7960. I also needed dtmfmode=inband & progressinband=yes to avoid incorrect American tones. |
| 18:22.12 | newtonr | noko, uhh, yeah so your phone is either mis-configured or having an issue. Your best bet is to find someone experienced with that particular phone.. |
| 18:22.38 | gmalsack | ok thanks. however none of that has anything to do with the initial register request coming from the phone would it. |
| 18:22.39 | Chainsaw | gmalsack: (I suppose if you're in the States the default tones are great and you don't need the inband settings) |
| 18:23.41 | Chainsaw | gmalsack: If you're already sure what the issue is, I'm not sure I can add anything. |
| 18:26.54 | *** join/#asterisk bitfury (~bitfury@unaffiliated/bitfury) |
| 18:30.24 | [TK]D-Fender | gmalsack: Show us the actual debug |
| 18:40.05 | gmalsack | [TK]D-Fender: it's a wireshark file. doubt I can pastebin that.... |
| 18:40.45 | [TK]D-Fender | How about * CLI SIP debug? |
| 18:43.03 | gmalsack | here's the file: http://wikisend.com/download/257572/5001.pcap |
| 18:43.47 | gmalsack | sip debug: http://pastebin.com/6CYBCN7B |
| 18:45.52 | *** join/#asterisk mfu (~mfu@gw-office.mcosys.de) |
| 18:47.45 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:48.28 | [TK]D-Fender | gmalsack: I don't see anything wrong there... |
| 18:49.25 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.126) |
| 18:50.17 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 18:50.22 | gmalsack | [TK]D-Fender: the read from is coming from port 51108. however on the contact line states to contact 5060. when asterisk sends an ack to 51108, the phone responds with destination port unreachable |
| 18:51.12 | gmalsack | [TK]D-Fender: I'm assuming the destination port on the phone is unreachable because the phone is expecting traffic to come in on 5060, however asterisk is sending the ack to 51108 |
| 18:54.07 | gmalsack | I'm assuming the destination port on the phone is unreachable because the phone is expecting traffic to come in on 5060, however asterisk is sending the ack to 51108 |
| 18:54.41 | *** join/#asterisk crazed1 (~robertmos@unaffiliated/themrrobert) |
| 18:55.30 | crazed1 | Hey guys, I've got a tricky situation. I need to redirect two channels into a conference, but about 75% of the time, after i redirect the first channel, the bridged channel is hungup before the 2nd call to channelredirect. Sometimes it works perfectly, its purely random |
| 18:56.01 | crazed1 | how can i prevent that bridged channel from hanging up in the .05 seconds that its 'alone' |
| 18:57.00 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 18:57.36 | mjordan | crazed1: AMI or dialplan? |
| 18:57.44 | crazed1 | dialplan |
| 18:58.04 | mjordan | hm. |
| 18:58.11 | mjordan | AMI lets you redirect both channels in a bridge. |
| 18:58.36 | *** join/#asterisk apb1963 (~quassel@174.134.232.228) |
| 18:58.48 | mjordan | You could try using the 'g' flag in Dial, and ChannelRedirect the called party first |
| 18:59.01 | mjordan | the caller will then drop into the next priority after Dial, which could be a Wait() |
| 18:59.03 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:59.11 | crazed1 | has to be thru dialplan. a 3rd channel/extension dials an extension in the primary context, and this extension needs to move all 3 channels into a conference. tried both legs first, doesn't make a difference, still random drops |
| 18:59.18 | mjordan | that's worth a shot, at any rate. In general, when you're manipulating multiple channels at the same time, you're typically better served by AMI. |
| 18:59.27 | crazed1 | i'll try that mjordan |
| 18:59.47 | crazed1 | I can't use ami for my purpose tho, can i? |
| 19:00.15 | crazed1 | a 'manager' has to dial an extension which creates a conference with both legs of his agent's call |
| 19:00.43 | crazed1 | also mjordan, how do i make sure that normal channels that hangup don't get stuck on the wait() command |
| 19:01.27 | crazed1 | nvm on the last one |
| 19:04.18 | crazed1 | haha mjordan, you're a godsend, thanks! |
| 19:04.45 | crazed1 | that was brilliant, especially since i'm assuming you haven't had to do that yourself you just intuitively came to that idea |
| 19:05.21 | *** join/#asterisk TSM (~the_softw@46-65-201-69.zone16.bethere.co.uk) |
| 19:07.09 | *** join/#asterisk kresp0 (~kresp0@unaffiliated/kresp0) |
| 19:18.42 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 19:30.32 | *** join/#asterisk leedm777 (~dlee@nat/digium/x-tbvfeqzyyryncgoh) |
| 19:36.51 | *** join/#asterisk felipealmeida (~user@179.210.237.70) |
| 19:41.00 | mjordan | crazed1: np. You could do it through AMI by watching the channels and redirecting all of them into an extension that leads to a ConfBridge, but if the dialplan is working for you that's great |
| 19:46.05 | *** join/#asterisk gerhard7 (~gerhard7@77.172.47.159) |
| 19:59.48 | *** join/#asterisk gartral (~gartral@unaffiliated/gartral) |
| 20:02.10 | drmessano | Why my Akerisk no 64-bite windows? |
| 20:03.23 | drmessano | Are my enough gagabits? |
| 20:04.07 | dumby | Download 3 gagabits more RAM, it will work |
| 20:04.45 | drmessano | It say noo valid 32 bite appliction so I go two times |
| 20:04.47 | drmessano | :( |
| 20:05.48 | Penguin | You're supposed to get the 128 version and stop it at 50%. |
| 20:06.06 | Penguin | That bypasses the license validation. |
| 20:06.14 | eirirs | lol |
| 20:06.16 | dumby | ^ |
| 20:06.34 | dumby | Also make sure you downloaded RAM only from Microsoft, others is malware |
| 20:07.09 | drmessano | Why lisenses I think Akerisk sores open? |
| 20:07.25 | drmessano | I PAY????? |
| 20:07.30 | dumby | Get a season pass |
| 20:08.11 | drmessano | I think Akerisk free but not. I try Links? |
| 20:08.17 | drmessano | MS LINKS? |
| 20:08.46 | drmessano | Yeah I got nothin |
| 20:08.59 | navaismo | I knew it!!!! |
| 20:09.09 | navaismo | the troll was you! |
| 20:09.21 | drmessano | What gave me away? The nickname? |
| 20:09.32 | dumby | It was the smell |
| 20:09.38 | dumby | Stings the nostrils |
| 20:10.05 | drmessano | Well, I did spent 30 minutes wiping out a fire ants nest with wasp spray and contact cleaner earlier |
| 20:10.13 | drmessano | Maybe I smell like death and engineer |
| 20:10.23 | [TK]D-Fender | drmessano: Yes, to get 32 bite, you do 2 rounds on MS Links and the subtract your score from the 4 lowest holes. |
| 20:10.32 | drmessano | HAHAH |
| 20:11.00 | drmessano | I INSTATED MS LINKS AND NO IM ONLY GOLF??? WHY NO TALK? |
| 20:11.02 | [TK]D-Fender | drmessano: http://www.bizpacreview.com/2013/12/14/video-fire-ant-colony-casted-with-molten-aluminum-creates-controversy-88932 |
| 20:11.15 | drmessano | I saw that.. That was completely awesome |
| 20:11.40 | drmessano | Makes me want to carry molten aluminum to all my sites |
| 20:11.45 | drmessano | For the lulz |
| 20:11.48 | [TK]D-Fender | "T1000 Extermination, how may I direct your call?" |
| 20:11.53 | drmessano | haha |
| 20:12.27 | drmessano | Those poor ants waiting for John Connor |
| 20:13.07 | dumby | Those are the most metal ants alive, Dethklok could make a great song about them |
| 20:14.05 | drmessano | Rammstein made multiple albums to play while killing said ants |
| 20:14.44 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:15.36 | [TK]D-Fender | drmessano: Pink Panther.... |
| 20:15.42 | dumby | "Colony of Molten Metal" |
| 20:16.01 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 20:16.06 | [TK]D-Fender | drmessano: Dead ant! Dead ant! Dead ant! Dead ant! Dead ant! Dead ant! Dead ant!!!!!!!!!!!!!!!!!!!!! |
| 20:16.20 | drmessano | lol |
| 20:16.24 | ageis | The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? |
| 20:16.33 | ageis | this means GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|GatewayB) the | simply becomes a comma? |
| 20:17.55 | navaismo | yes |
| 20:19.23 | ageis | ty |
| 20:19.24 | *** join/#asterisk dxd828 (~dxd828@88-105-208-194.dynamic.dsl.as9105.com) |
| 20:20.55 | pigpen | I have a "sizing" topic for discussion. |
| 20:21.46 | pigpen | would you put ~100 sip phones with a PRI into production on a latest ATOM 64 bit box with SSD's ? |
| 20:22.19 | pigpen | I have done it with ~50 sip phones, 6 analog on digium cards with a rev older Atom 64 bit with no issues. |
| 20:22.20 | pigpen | thoughts? |
| 20:22.57 | pigpen | I guess largely it comes down to the transcoding. |
| 20:27.11 | navaismo | can anyone take a looik ont his PRI debug--->http://pastebin.com/cENGB4ay im dont know nothing about pri debug so im kind useless-actually i dont know nothing- but the calls arre dropping and i can only see the: [Dec 17 18:08:04] WARNING[27834]: sig_pri.c:5097 pri_dchannel: Span 2: SETUP on unconfigured channel 0/3 |
| 20:28.00 | navaismo | and beyond that where i can read about debugging and undersandig the pri debug? |
| 20:30.51 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 20:34.11 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:36.26 | pabelanger | pigpen, correct, transcoding is going to be killer |
| 20:36.45 | pigpen | pabelanger, but, with pri -> ulaw, there is little encoding right? |
| 20:37.25 | pigpen | navaismo, I would ensure you have the pri setup right. is this something new to an existing system, or is this a new turn up? |
| 20:38.10 | pigpen | pabelanger, yeah, unless I have my head up my ass, and it visits there often, there is no transcoding with PRI-SIP, SIP-SIP(as long as it is the same codec). |
| 20:38.20 | pigpen | again: my head may be visiting that special place. |
| 20:38.42 | pigpen | waits for TK to bitch slapped |
| 20:39.52 | [TK]D-Fender | pigpen: Your PRI is ULAW so no transcoding there. The load should be pretty low |
| 20:40.28 | pigpen | [TK]D-Fender, tks. that is what I was thinking, and I don't plan to use any other codec's. |
| 20:41.19 | pigpen | the most load would be an "all page". (school) and the Atom in place, the load is pretty low hitting ~51 devices. All ULAW. |
| 20:41.27 | pigpen | tks TK. |
| 20:41.32 | pigpen | again. |
| 20:41.47 | pabelanger | correct, if the codec is the same, you don't transcode |
| 20:42.31 | pigpen | yeah, I didn't think there would be, just wanted to make sure I didn't miss something. |
| 20:43.02 | [TK]D-Fender | pigpen: that page might be an issue. I'm not sure how it scales. |
| 20:43.33 | [TK]D-Fender | pigpen: You're basically jsut going from ULAW>SLIN across those which is about as low as it gets... but it is a lot of overall calls to span that for. |
| 20:43.36 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:43.45 | [TK]D-Fender | pigpen: might not have any issue... it's just the only funny thingg I see |
| 20:43.52 | pigpen | yeah. thinking about it, I'll find out when the next "announcement" happens and watch it closely. |
| 20:44.33 | pigpen | And thinking about it more, this is for a small Texas District. Likely the all-page will not happen to all 3 schools. |
| 20:45.07 | pigpen | never the less, if they need to do it, I want it to work. (ie: shooter in the building type event) |
| 20:45.12 | *** join/#asterisk human39 (~human39@71.236.110.155) |
| 20:45.14 | [TK]D-Fender | The worst part is the sync required for a 50-device page like that |
| 20:45.17 | [TK]D-Fender | call setups and all... |
| 20:45.25 | *** join/#asterisk darkdrgn2k3 (~darkdrgn2@209.90.253.66) |
| 20:45.26 | darkdrgn2k3 | hi all |
| 20:45.29 | pigpen | yeah, then take that to 100. |
| 20:45.39 | pigpen | [TK]D-Fender, more of a proc issue or ram? |
| 20:45.40 | darkdrgn2k3 | is there any way to re-construct an incomming fax if i have a PCAP of the t,38 transmission? |
| 20:45.45 | [TK]D-Fender | You start getting close to the poitn where "store & forward" becomes a better approach |
| 20:45.48 | pigpen | again, running SSD's so disk would be fast. |
| 20:45.56 | [TK]D-Fender | pigpen: it isn't a HDD issue |
| 20:46.12 | pigpen | yeah, didn't think so, but system runs a bit quicker with them. |
| 20:46.18 | pigpen | so Proc/Ram. |
| 20:48.05 | *** join/#asterisk Bkhan (ca7d90e3@gateway/web/freenode/ip.202.125.144.227) |
| 20:48.56 | Bkhan | Hi. There is an issue call got disconnected as it connected to agent |
| 20:49.52 | Bkhan | we can see on cli that call hanged up but on phone still bridging but no voice |
| 20:50.07 | [TK]D-Fender | Bkhan: And you were asked to show the complete debug of the failed call when you asked this 6 hours ago.... |
| 20:50.11 | [TK]D-Fender | Bkhan: And we never got it |
| 20:50.45 | Bkhan | [TK]D-Fender:oh sorry let me do it now. |
| 20:53.05 | Bkhan | [TK]D-Fender:http://pastebin.com/zQT3idLZ |
| 20:53.10 | Bkhan | please check |
| 20:54.30 | darkdrgn2k3 | so anyone have any idea how i can extract a fax from a pcap file using t.38 portooclo |
| 20:54.37 | [TK]D-Fender | [10:08][TK]D-FenderBKhan: Show us the failed call with the appropriate channel debug enabled |
| 20:54.48 | [TK]D-Fender | Bkhan: Llooking without SIP DEBUG there is a waste of time... |
| 20:55.18 | [TK]D-Fender | Bkhan: And that wasn't even a complete call with just basic verbose |
| 20:55.33 | [TK]D-Fender | Bkhan: And do not mask anything except passwords in there |
| 20:56.44 | Bkhan | [TK]D-Fender: this time around 180 calls on this server so its diffcult to get debugging logs this time |
| 20:58.34 | navaismo | we need to add somewhere the tools and the way to parse logs that ^ is a usual response |
| 21:00.15 | *** join/#asterisk Weezey (~ohno@i.am.weezey.com) |
| 21:01.00 | [TK]D-Fender | That isn't a response worth making really. Saying X is hard doesn't make it less necessary |
| 21:02.30 | *** join/#asterisk kgunnIt (~kgunn@ec2-23-23-228-7.compute-1.amazonaws.com) |
| 21:03.39 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-ewxrguqyfwpcbwwx) |
| 21:05.08 | navaismo | Bkhan: make a tcpdump tcapture, save in a pcap or cap file that output, open that with wireshark, go to telephony-->voip calls, locate your call, then make a "follow udp stream" export that to txt file and pastebin or copy and paste directly in the pastebin |
| 21:06.00 | carrar | heh |
| 21:06.09 | navaismo | loves searching in google and matching the same issue but got an unresolved thread-->http://lists.digium.com/pipermail/asterisk-users/2010-September/253825.html |
| 21:06.31 | [TK]D-Fender | Clear Buffer. Place call. Call Dies. Copy All to buffer. Total time : 10 SECONDS |
| 21:07.17 | [TK]D-Fender | Ceasing to whine like a little bitch when asked for debug : PRICELESS |
| 21:07.33 | Bkhan | navaismo: thanks let me check |
| 21:08.00 | [TK]D-Fender | Get a bigger buffer </brody> |
| 21:08.21 | [TK]D-Fender | Bkhan: No 3rd party. Just get it from * CLI with SIP DEBUG. |
| 21:10.23 | Bkhan | [TK]D-Fender: yes but there is 200 calls this time so i think tcpdump is better for the time being |
| 21:11.10 | [TK]D-Fender | Bkhan: How long will it take to fail? |
| 21:11.33 | [TK]D-Fender | You are cutting off what ASTERISK is saying about things thinking raw SIP explains everything |
| 21:11.43 | [TK]D-Fender | You are "thinking" wrong. |
| 21:11.57 | pigpen | Well, doing an all page Asterisk went from using .3% of the proc to 32% of the proc on an Intel x86_64 Atom CPU D525 @ 1.80 GHz |
| 21:12.13 | pigpen | Memory was not affected. |
| 21:12.20 | Bkhan | <[TK]D-Fender>: as call connected to agent |
| 21:12.31 | Bkhan | <[TK]D-Fender>:with in 2 seconds |
| 21:13.11 | [TK]D-Fender | pigpen: Sounds survivable |
| 21:14.34 | pigpen | yeah, an now there are much faster Atom procs. So 32% with ~50 devices |
| 21:15.00 | [TK]D-Fender | pigpen: I've got a 1U Atom S1260 server I'm about to deploy for my PRI terminated PBX |
| 21:15.21 | pigpen | D525 is 1.8 GHz, 2 core. |
| 21:15.34 | pigpen | the C2758 is 8 core at 2.4 GHz. |
| 21:15.46 | pigpen | [TK]D-Fender, Supermicro? |
| 21:16.51 | [TK]D-Fender | pigpen: yup - http://cpuboss.com/cpus/Intel-Atom-S1260-vs-Intel-Atom-D525#performance |
| 21:17.12 | [TK]D-Fender | Good boost there. |
| 21:17.19 | [TK]D-Fender | http://www.supermicro.com/products/system/1u/5017/sys-5017a-ef.cfm |
| 21:19.06 | pigpen | That was exactly what we were beginging to move to, we have been using the D525 for about 60 smaller deployments. |
| 21:20.18 | pigpen | and also using the D525 for our two BGP routers. |
| 21:20.25 | pigpen | at our datacenter. |
| 21:20.51 | pigpen | We have been very happy with them. We usually use SSD's in them with a Linux Raid. |
| 21:20.51 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 21:21.09 | [TK]D-Fender | The Supermicros with the D525 tended to all have Intels super-buggy NIC's in them which is why I went with this model |
| 21:21.31 | pigpen | heh, yeah. We started deploying them before the S1260 was released. |
| 21:21.48 | pigpen | but now they have some 1U C2750 & C2758's. |
| 21:21.53 | pigpen | I wonder how much... |
| 21:21.58 | pigpen | Oh, 4x nic. |
| 21:22.11 | pigpen | dedicated ipmi too. |
| 21:22.36 | pigpen | http://www.supermicro.com/products/system/1U/5018/SYS-5018A-FTN4.cfm |
| 21:23.47 | [TK]D-Fender | Looks really nice... |
| 21:24.05 | pigpen | yeah. Nice they have more nics. |
| 21:24.46 | pigpen | getting a build cost now. |
| 21:24.55 | [TK]D-Fender | Makes for a nifty routing box if you want a full PC doing it |
| 21:27.00 | [TK]D-Fender | Alrighty... checkout time... BBIAB |
| 21:27.20 | pigpen | right. We have used like Dell R710's in the past for routing boxes which work flawlessly. All on Gentoo. |
| 21:27.28 | pigpen | later. |
| 21:30.17 | drmessano | I like the R710s |
| 21:30.45 | pigpen | Yeah, they are ok. We had enough of Dell. Their SAN's were starting to really fail us. |
| 21:30.57 | *** part/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 21:31.03 | drmessano | At least it's not HP |
| 21:31.09 | pigpen | So we moved over to Supermicro. Custom building our own Servers, SAN's, phone systems, firewalls, etc... |
| 21:31.13 | drmessano | Hardly Passable |
| 21:31.15 | pigpen | HP, ugh. |
| 21:31.40 | drmessano | I built a PBX on a DL380G4 and I regret it every day |
| 21:31.47 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 21:31.53 | pigpen | We are doing some kickass SAN's here lately with some 36 bay chassis. |
| 21:31.54 | drmessano | I think I MAY have resolved the 2 year old ongoing issue |
| 21:32.11 | drmessano | Thats awesome |
| 21:32.12 | pigpen | heh. Sounds like HP. They have funky raid contollers. |
| 21:32.17 | drmessano | LOL |
| 21:32.27 | drmessano | How did you know the issue was RAID controller? |
| 21:32.55 | pigpen | my business partner is a kernel dev: he hates HP due to the raid contollers. |
| 21:33.08 | *** join/#asterisk LinoSP (~LinoSP@201.240.245.207) |
| 21:33.26 | pigpen | Yeah, we have done several 144 TB SAN's for bulk storage. Very sweet boxes. Dam heavy I might add. |
| 21:33.47 | pigpen | As of lately, we are not using any hardware Raid controllers. |
| 21:34.03 | drmessano | I suspected that was it, upgraded firmware about a year ago, but I think the upgrade failed. I never did check FW version after the upgrade. After it froze up on me 3 times in one day last week I ran the firmware update again, and it actually ran.. No mention of it being current. The FW is 3 years old |
| 21:34.05 | pigpen | None of the big boys (SAN wise) are using them, why should we. |
| 21:34.24 | pigpen | yeah, firmware bastards....they suck. |
| 21:34.35 | LinoSP | Hola a todos disculpen hay alguien q pueda ayudarme con un examen sobre asterisk? |
| 21:34.50 | drmessano | So I think I may NOW be on current firmware. If last years upgrade failed, I may have still been on 2006 era firmware |
| 21:34.54 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 21:34.56 | pigpen | drmessano, you can take that one. English only here. |
| 21:34.58 | drmessano | Current (2010) |
| 21:35.09 | pigpen | heh. yeah. |
| 21:35.22 | pigpen | We had a Dell SAN fail 5 drives in less than a half a second. |
| 21:35.30 | pigpen | we were not happy. |
| 21:35.39 | navaismo | LinoSP: uh? |
| 21:35.44 | drmessano | The box has been up for 10 days without dying. I'll be convinced if I break the old record of 30 |
| 21:35.48 | drmessano | Oh I bet |
| 21:36.11 | navaismo | LinoSP: preguntas especificas por favor |
| 21:36.14 | LinoSP | Hi everybody is there someone that can help me with an asterisk exam? |
| 21:36.23 | LinoSP | navaismo: |
| 21:36.24 | pigpen | Even worse. Engineering Firm. (civil) Bye by cad files. |
| 21:36.25 | drmessano | Mi Esposa es been tramplado por toros |
| 21:36.41 | drmessano | pigpen, ouch |
| 21:37.01 | LinoSP | navaismo: tengo un pdf |
| 21:37.10 | navaismo | LinoSP: ~ask |
| 21:37.15 | drmessano | I've luckily not had data loss yet. Just downtime and lots of "Why the FSCK did we put in that Asterisk crap???!!!!" |
| 21:37.20 | pigpen | yeah. December 14th, 2012 10:28 AM. I will never forget. |
| 21:38.04 | drmessano | Shitty RAID -> Shitty Server -> Shitty PBX -> Shitty idea <-- Process of blame, of course |
| 21:38.10 | pigpen | drmessano, heh, have them talk to me. Reliable as hell. Flexible as heaven. |
| 21:38.18 | navaismo | LinoSP: please dont PM |
| 21:38.57 | navaismo | LinoSP: we dont resolv school tests if you have a question ask |
| 21:39.20 | drmessano | I've been behind probably 60 Asterisk installs at this point. Most of them are side work or when I was with another employer. Figures the one box I have an issue with is at my current day job |
| 21:39.56 | drmessano | This is the only one that has ever cursed me like this |
| 21:39.57 | LinoSP | navaismo: ok ... I can't ask for a complete school question then? |
| 21:39.58 | pigpen | Ah, yeah. You know. |
| 21:40.19 | drmessano | But its the one I use every day and the one I get to hear about all the damn time |
| 21:40.21 | pigpen | drmessano, yeah, I have had a few of those. |
| 21:40.47 | pigpen | goofy hardware will "F" you quickly. |
| 21:40.53 | navaismo | LinoSP: You can ask an specific question, then wait patiently for the answer |
| 21:43.04 | pigpen | LinoSP, please remember, many of us learned Asterisk the hard way. We don't have a "Cert" in it. I would guess you will fall on deaf ears "getting help on a test". |
| 21:43.37 | pigpen | Also remember: Digium is in here too! They are watching. (kinda like the NSA) |
| 21:44.08 | pigpen | needs coffee. Grumpy. |
| 21:44.47 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 21:44.57 | drmessano | Goofy hardware indeed |
| 21:44.57 | *** join/#asterisk aarobc (~ac@asa-2.fbp.ore.fiber.net) |
| 21:45.55 | aarobc | So I'm following this guide here: http://highsecurity.blogspot.com/2013/02/setting-up-silk-codec-with-asterisk-1011.html and when I run “core show codecs” silk isn't showing up. it says that it loads it, but it's acting like it's not |
| 21:45.59 | WIMPy | Does that mean I shouldn't connect Digium hardware to the Internet? |
| 21:46.01 | drmessano | pigpen, I try not to forget that the DL380/DL385 was a COMPAQ product |
| 21:46.11 | aarobc | any ideas? |
| 21:46.31 | WIMPy | just got hold of some Asterisk appliance which contains some Digium cards. |
| 21:46.33 | drmessano | aarobc, SILK doesn't work for me either. I tried on several boxes |
| 21:46.47 | drmessano | aarobc, I felt like I may be missing something, but never could figure out what |
| 21:46.59 | pigpen | DL380...heh. Yeah. I remember. |
| 21:47.13 | LinoSP | pigpen: navaismo Gonna study hard for my tomorrow test thx anyway for remind me that I have to do it bymyself ;) |
| 21:47.20 | aarobc | drmessano: well that's frustrating. I also tried to get opus to work but even though it's listed there it doesn't seem to work either |
| 21:47.32 | drmessano | aarobc, I found no mention of dependencies, and no where else could I find someone having the same issue. Til now |
| 21:48.10 | drmessano | aarobc, is this a bare metal install or a VM? |
| 21:49.03 | aarobc | drmessano: vm. |
| 21:49.55 | drmessano | aarobc, same here. I could have sworn it loaded on a bare metal box, but I had too many other variables in play to blame it on a VM |
| 21:49.59 | aarobc | But I've done this sort of thing on freepbx before and it worked |
| 21:50.11 | aarobc | in a vm |
| 21:50.26 | drmessano | Asterisk 11? |
| 21:51.43 | pigpen | hasn't touched SILK. |
| 21:52.29 | aarobc | drmessano: indeed |
| 21:53.03 | drmessano | That question was kinda vague. The FreePBX install you had it working on was also 11? |
| 21:53.14 | aarobc | yes indeed |
| 21:53.18 | drmessano | Ok |
| 21:53.52 | drmessano | I think I have only ever had it loaded successfully on a bare metal Asterisk 10 box. Another variable I wanted to rule out |
| 21:53.58 | drmessano | So its not just 11 |
| 21:54.50 | aarobc | I'm going to spin up a new VM and see if I can get it working there. |
| 21:54.52 | *** join/#asterisk BlackDex (~BlackDex@ori.vyus.nl) |
| 22:00.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.126) |
| 22:01.33 | drmessano | aarobc, i'll be interested to see what happens. It's definitely fickle, but other than the failure in the full log ,there's little to go on |
| 22:02.44 | aarobc | drmessano: does the *43 echo test come stock on asterisk, or is that something else you have to configure? |
| 22:03.05 | aarobc | I'm too lazy to set up an incoming trunk |
| 22:03.35 | drmessano | No, but easy to build |
| 22:04.35 | drmessano | exten => 111,1,Echo |
| 22:05.20 | aarobc | drmessano: supid question: would you put that in sip.conf or where? |
| 22:06.30 | drmessano | extensions.conf |
| 22:06.57 | drmessano | You're going to need to know a lot more before building a box by hand, i'm afraid |
| 22:08.05 | aarobc | Indeed, I'm trying to not rely on freePBX anymore, so i'm forcing myself to learn asterisk manually. |
| 22:08.09 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 22:10.10 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 22:23.06 | pigpen | aarobc, very good idea. |
| 22:24.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:24.48 | navaismo | why is freepbx not good enough or...? |
| 22:25.31 | pigpen | freepbx is more about the bells and whistles. |
| 22:26.00 | pigpen | building an asterisk box up from scratch (in my case, compiled entire system from scratch) you have much, much more control. |
| 22:26.17 | pigpen | over the entire system. This makes it more of a "Enterprise" grade solution. |
| 22:26.24 | pigpen | very, very customizable. |
| 22:26.33 | *** join/#asterisk petris (~petris@192.184.93.7) |
| 22:26.46 | pigpen | and if something breaks, you can fix it, rather than depending on someone else to make it a priority. |
| 22:30.19 | navaismo | aha tell me more please |
| 22:30.46 | navaismo | not a joke ^ |
| 22:32.22 | dumby | well, it forces you to learn how to write all the configs and what they all do |
| 22:33.12 | drmessano | Matthew Jordan just sent me an email to tell me Asterisk 10 is dead. |
| 22:33.22 | drmessano | Hopefully he sent others the same email |
| 22:33.54 | Qwell | drmessano: nah, he figured you'd take care of it |
| 22:34.46 | drmessano | Qwell, I hope not. I have a horrible Klout score |
| 22:35.55 | drmessano | Right now it's 52. If they let me connect my PornTube account it would be in the high 90s though |
| 22:47.50 | mjordan | drmessano: nope, only to you |
| 22:48.21 | mjordan | you'll note Asterisk 10 is no longer in the #asterisk motd. That's the true sign that it has passed beyond the veil. |
| 22:49.27 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 22:50.50 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 22:51.18 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.114) |
| 22:52.23 | Chainsaw | mjordan: Then again, nobody ever updates that topic. Isn't DAHDI on 2.7.0.1 and the tools on 2.8.0? |
| 22:53.34 | mjordan | I update that topic every time I release Asterisk, so I wouldn't say no one |
| 22:53.42 | crazed1 | We get this about 2-3 times per month, out of nowhere the PBX 'crashes' and all the lines go dead. its unable to make new channels, but it seems the core is still running, i could open the cli, and core show channels was all blank |
| 22:54.06 | mjordan | but I can blame someone else for DAHDI |
| 22:54.12 | mjordan | pushes sruffell under the bus |
| 22:54.13 | crazed1 | a service asterisk restart fixes the problem but obviously its a huge productivity loss |
| 22:54.40 | sruffell | huh? ohhh…the topic. |
| 22:54.45 | Chainsaw | crazed1: Are you running out of FDs? |
| 22:55.38 | crazed1 | max files is 5 million, and asterisk restarts each night, so i wouldn't think so, its possible though. oh sorry i guess this should be freepbx |
| 22:55.46 | sruffell | Chainsaw: 2.8.0 all the way around (but now I'll have to lookup how to update the topic again) |
| 22:56.07 | Chainsaw | We broke sruffell :( |
| 22:56.53 | Chainsaw | reads up on JCL and connects to the Digium mainframe |
| 22:58.20 | crazed1 | Chainsaw: but it does feel like that sort of an issue. Its like the system just can't move forward, but the leftover channels finish their thing, except when they hit ChannelRedirect or similar, then they stop, until its all quiet, but i can CLI in |
| 22:58.55 | Chainsaw | crazed1: Definitely running out of a finite resource. Suggests a leak. |
| 23:00.13 | crazed1 | Chainsaw: anything I can do? i already restarted asterisk and not dont know what the resource usage was before that |
| 23:00.37 | Chainsaw | crazed1: You blew it up and didn't bring me the shrapnel. |
| 23:00.48 | Chainsaw | crazed1: Nothing that I can do for you now. |
| 23:00.55 | crazed1 | what can i do to collect in the future |
| 23:01.25 | Chainsaw | crazed1: If it's FreePBX, you really should ask how they want it. |
| 23:01.58 | crazed1 | okay, but i do a lot of pure asterisk work (its barely freepbx anymore), too, so how would you want it here |
| 23:02.05 | crazed1 | and i will ask them as well |
| 23:02.32 | Chainsaw | Remind me, didn't Fender have an article in the bot about backtraces and the like? |
| 23:02.58 | WIMPy | ~collectdebug |
| 23:02.58 | infobot | collectdebug is, like, a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 23:03.05 | WIMPy | That one? |
| 23:03.32 | Chainsaw | Thanks WIMPy. |
| 23:03.33 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 23:03.50 | Chainsaw | crazed1: All of that, when it's failed. |
| 23:06.11 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 23:06.25 | crazed1 | lol that'd be nice, but its a production machine, with 150k call per day, having all of that debug info on for possibly weeks until it happens again isn't practical, whats maybe the most important, debug? |
| 23:07.25 | Chainsaw | crazed1: I would like for you to fix my car. It has a rattle. I'm going to keep the garage locked though, because it's expensive. What's wrong with it? |
| 23:08.28 | *** join/#asterisk clopez_ (~tau@neutrino.es) |
| 23:08.31 | crazed1 | i have the core dump. and chainsaw: if you didnt't have the resources for surveilance, and monitoring all aspects of the car for weeks, then i might suggest monitoring 1 key component :) |
| 23:11.51 | navaismo | also if you compile asterisk with debug_threads run the core show locks command |
| 23:12.17 | WIMPy | That might kill you performance wise. |
| 23:15.15 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 23:15.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 23:15.35 | *** join/#asterisk thebmw (~thebmw@i.am.thebmw.net) |
| 23:17.12 | crazed1 | yea @ navaismo i've tried that the system can't run as production with debug_threads |
| 23:22.33 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 23:25.15 | *** join/#asterisk ehsjoar (~jonas@c-24-9-87-13.hsd1.co.comcast.net) |
| 23:31.08 | *** join/#asterisk troglobyte (~troglobyt@unaffiliated/troglobyte) |
| 23:32.55 | *** join/#asterisk dxd828 (~dxd828@88-105-208-194.dynamic.dsl.as9105.com) |
| 23:54.18 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.114) |
| 23:54.59 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |