IRC log for #asterisk on 20160830

01:18.30*** join/#asterisk ipengineer (~zconkle@108-206-120-218.lightspeed.mckntx.sbcglobal.net)
01:29.09*** join/#asterisk acidfu_ (~acidfoo@24-212-247-227.cable.teksavvy.com)
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02:57.19*** join/#asterisk hiyo (~hiyo@unaffiliated/hiyo)
03:09.32hiyoHello I am having a problem with calling over OpenVPN, I get silent calls. However when in the LAN of the server, it works fine
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03:50.41*** part/#asterisk hiyo (~hiyo@unaffiliated/hiyo)
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05:19.44*** join/#asterisk ruben23 (~OpenDIAL@112.201.10.245)
05:20.38ruben23sterisk*CLI> dahdi show channels  -----> No such command ‘dahdi show status’ (type ‘core show help dahdi show’ for other possible commands)
05:21.12ruben23* asterisk*CLI> dahdi show channels
05:21.16ruben23any ide aguys
05:27.05*** join/#asterisk miralin (~Thunderbi@194.8.128.48)
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07:37.41ChannelZruben23, yeah.. DAHDI isn't loaded
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08:02.01*** join/#asterisk jkroon (~jkroon@uls-154-73-35-201.wall.uls.co.za)
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08:15.08teaspoonIs it possible to dynamically create peers and add them to a hunt group?
08:16.08teaspoonThe example I have in mind right now is for an asterisk+iaxmodem+hylafax monstrosity, where I want to be able to spin up a couple of extra iaxmodem instances without having to update iax.conf
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09:00.03*** join/#asterisk bounceman (~bounceman@185.32.9.250)
09:00.20bouncemanHi, if I enable tls in manager.conf. Will that only make it possible to do requests over https?
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09:25.57bouncemanEarth is calling
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10:04.22*** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh)
10:04.25wasanzyHi
10:04.51wasanzyis there a way to determine if an audio has not finished playing?
10:22.42*** join/#asterisk sekil (~sekil@nat-73.net011.net)
10:35.15jkroonhi all, using htop I'm seeing that at least one asterisk thread is starting to approach 100% cpu.  what can I do to figure out what that thread is.
10:36.27jkroonI'm afraid that it's sporadically hitting 100% and causing quality problems, so I'd like to confirm which thread this is so that I can determine what I need to do to fix the problem>
10:37.19SamotWhat do the logs say is happening?
10:37.29SamotWhat are the actual resources being used?
10:38.41jkroonsamot, overall CPU >= 100% (meaning more than one core is being consumed).
10:38.52jkroonlogs themselves are all the normal logs that I expect to see.
10:39.21jkroonon RTCP stats I'm seeing some jitter, but I only started logging that *after* I first saw the problem (which is varying).
10:39.38SamotHow many calls?
10:39.54jkroonCPU is the only resource that I'm concerned about based on iotop, iostat, top, htop and iftop.
10:39.54SamotWhat activity is happening on the box right now?
10:40.02jkroon112 active calls - 24238 calls processed
10:40.07jkrooncombination of IAX and SIP channels.
10:40.28SamotWhat's the resources on the box?
10:40.39SamotHow much RAM and CPU?
10:40.44jkrooni'm aware that SIP is running in a single thread, but that's for port 5060 only, and shouldn't disturb RTP - which is what it seems is being disturbed.
10:41.39jkroonbox has 10G RAM, of whch free -m is saying 8.2G is being used, with 486M free, and 4M shared, buff/cache @ 1.2G and 1.2G available.
10:42.05jkroonaccording to top asterisk is using a mere 240MB of RAM.
10:42.22SamotCPU?
10:42.38jkroonIntel(R) Xeon(R) CPU E5420  @ 2.50GHz - 8 threads.
10:42.50SamotSingle core?
10:42.55SamotDual core?
10:42.59jkroonwell, 4 cores with 2 threads IIRC ... let me check google.
10:44.07wasanzyhow do I determine if a user terminate a call in asterisk?
10:44.10Samot~pb your top
10:44.16Samot~pb
10:44.17infobotit has been said that 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
10:44.58Samotwasanzy: You mean you want to know if the user (extension) ended the call or the other party?
10:45.07*** join/#asterisk Iamnach0 (~Iamnacho@ip24-252-4-195.om.om.cox.net)
10:45.19jkroon4 cores as per http://ark.intel.com/products/33927/Intel-Xeon-Processor-E5420-12M-Cache-2_50-GHz-1333-MHz-FSB
10:45.20wasanzySamot: yes
10:45.36jkroontop
10:45.44jkroonsorry, wrong window
10:45.54wasanzyeg, an audio could be playing and in the middle, a user can decide to hang the call. I want to know if that happens
10:46.39jkroonthe task id of the thread consuming the most CPU is the same as the main pid.
10:46.59SamotShow it.
10:47.09*** join/#asterisk zapata (~zapata@2a02:b18:581:10:f072:ea29:7a0e:e3d1)
10:47.30Samotwasanzy: It's in the logs. You're going to have to catch things on the hangup..
10:47.34Samotin the h extension.
10:47.51jkroonand seemingly i can't c&p htop ... https://pastebin.co.za/5706163895140352 - normal top (atm, that asterisk CPU spikes up to 180% at times)
10:48.07wasanzyso I can define ext => h, blalabla?
10:50.00jkroonwasanzy, that's generally the idea yes ... although, determining from which side the HANGUP came is something I've not yet managed
10:50.18jkroonSamot, will a screenshot from htop be ok?  I'm unable to copy&paste the text
10:50.20wasanzyok
10:50.24SamotSo, what is mysql doing?
10:50.36*** join/#asterisk Tiffon (~name@unaffiliated/tiff0n)
10:50.38SamotSince according to this it's taking up more resources than Asterisk.
10:51.03jkroonmostly i've got a process that hooks the AMI and logs RTCP reports to MySQL.
10:51.03SamotWell more memory and big chunk of CPU.
10:51.17jkroonit's using more RAM yes.  and most likely more disk IO too.
10:51.33jkroonnot even one core, overall 81.9% CPU idle of the 8 cores.
10:51.36SamotAnd almost 50% CPU.
10:51.48jkrooni'm worried about single threads taking a whole core on it's own (ie, 100%)
10:52.34*** join/#asterisk kline (~kline@freenode/staff/euncs.kline)
10:53.48jkroonoh, and I've got astdb on a ramdisk ... because that's known to cause MAJOR problems.
10:53.50SamotWhat happens when you stop the AMI process that's writing logs?
10:53.56jkroonwell, with handling of SIP.
10:54.00*** join/#asterisk kline (~kline@freenode/staff/enucs.kline)
10:54.04jkroonI wrote that specifically to try and pinpoint the problem.
10:54.13jkroonso problem started before I hooked into the AMI there.
10:54.34SamotIs it constant?
10:54.40SamotDoes it go away if the call volume drops?
10:54.47jkroonit's variable.
10:54.56jkroonI haven't yet graphed call volume vs problems.
10:55.26jkroonLet me quickly see if I can create a call volume histogram from the CDR records to see if that correlates with the problems.
10:56.24jkroonwe have an idea of when the problems occured.
10:56.32SamotWhen?
10:56.51jkroon8:15 SAST and then again around 9:55 SAST etc ...
10:56.56jkroonso we've got approximate time slots.
11:01.31jkroonthis is super strange.  asterisk is using less CPU now with more calls than it did a moment back.
11:04.47SamotActive calls take less resources vs setup/tear down
11:05.26jkroonfair enough.  but setup also shouldn't affect threads shuffling rtp should they?
11:05.41jkroonand it seems rtp is affected in some way or another somewhow.
11:05.48SamotWell it's real time
11:05.55SamotSo if the server is being lagged.
11:06.11SamotThen it's going to impact performance.
11:06.32SamotSo yes, poor audio quality is a result of a taxed server.
11:07.01jkroonat 100 concurrent calls?
11:07.20jkroonwe've done 130+ before ... same version of asterisk.  i'm trying to figure out what changed ... but cannot pinpoint anything.
11:07.25SamotWhat else is being done on the server?
11:07.43jkroonasterisk + mysql ... and a handful of helper processes.
11:08.07SamotWhat are the users doing?
11:08.08jkroonalso started tcpdump in an effort to confirm the udp flows for rtp + sip (and also capturing 5060 tcp).
11:08.11SamotAre their queues?
11:08.21SamotHow many Ring Groups, IVRs?
11:08.25SamotConference calls?
11:08.34jkroonno ring groups, no IVRs, no conference calls.
11:08.39jkroonno queues.
11:09.00jkroonthis box purely switches between incoming channel and outgoing channel, and then sporadically some voicemail.
11:09.15voipmonknot a true statement.
11:09.35jkroonok, ... truth to the best of my understanding ... why do you disagree?
11:09.41voipmonkoffload mysql to another box , offload tcpdump to another box and use a mirror port on a beefy switch that can handle it then you might be able to say "purely"
11:09.43jkroonudp      768      0 0.0.0.0:29590           0.0.0.0:*                           11057/asterisk
11:10.00voipmonkand tell me more about astdb
11:10.14jkroonhttp://jkroon.blogs.uls.co.za/it/voip/asterisk-massively-speeding-up-those-register-requests
11:10.29jkroonwe've mounted a ramdisk on /var/lib/asterisk/astdb and told asterisk to put astdb in there.
11:10.44jkroonthe reason for that is the fsync() that sqlite executes causes the sip thread to block.
11:10.49voipmonkits horrible that you are only doing 120+ calls on this box -
11:11.12voipmonkhttp://jkroon.blogs.uls.co.za/it/voip/asterisk-massively-speeding-up-those-register-requests
11:11.14voipmonkis this you?
11:11.23jkroonvoipmonk, and thus why i'm here looking for help.  previous estimates and calculations showed we should be doing a lot more.
11:11.23jkroonyes.
11:11.43voipmonkthis shouldn't happen until you get into the 1200's at least
11:12.01voipmonkwhere'd you dig up this proc?
11:12.07voipmonkcat /proc/cpuinfo
11:12.17jkroonyes, from /proc/cpuinfo.
11:12.48voipmonkwhat switch are you using?
11:13.22voipmonkdid I read that you are suggesting that calls are dropping?
11:13.49voipmonkheh - can you explain what the customer experience is?
11:14.05SamotHow many endpoints on registering to the system?
11:14.11voipmonkyes
11:14.12voipmonkgood one
11:14.37jkroon1595 sip peers [Monitored: 380 online, 401 offline Unmonitored: 12 online, 802 offline]
11:14.46voipmonkyeah
11:14.50jkroon376 iax2 peers [142 online, 16 offline, 218 unmonitored]
11:15.36jkroonswitch is an HP 1910-24G, neither it's error counters nor that on the server is reporting any problems.
11:15.37SamotWhat is the avg Registration Expires set for the endpoints?
11:15.47jkroon60 minutes IIRC.
11:16.02jkroonlet me confirm.
11:16.19jkroonrx_no_buffer_count: 27128059
11:16.25jkroonI lied.  error counters on eth0.
11:16.35SamotSo when you are doing your tcpdumps or sip debugs do you see register attempts from endpoints that are not yours?
11:16.49jkroonlogs aren't reporting failed registration attempts.
11:17.11SamotI didn't ask about logs.
11:17.18SamotI asked about live data.
11:17.32jkroondidn't check tcpdump, will do a quick check.
11:17.52jkrooni have increased the rx and tx ring buffers on the ethernet interfaces now, that may (or may not) help.
11:19.24SamotAnd while you're at it, see how many call attempts are being made as well.
11:19.34SamotThey don't need to register to try and send calls through you.
11:19.37jkroonINVITE?
11:19.46SamotYes
11:20.22SamotIs this box directly on the Internet or on a LAN?
11:20.53jkroonserver => switch => firewall => switch => 100Mbps link => open internet.
11:21.04jkroonthat 100Mbps is a peering link.
11:21.11SamotSo it's behind a LAN
11:21.19jkrooncurrently running at around 30-60Mbps variable during the day.
11:21.20SamotThat doesn't tell me if it's on a LAN
11:21.32jkroonwell, strictly speaking it's always on some kind of LAN.
11:21.43jkroonyou're asking whether port 5060 is internet accessible and the answer is yes.
11:22.02SamotAre the IPs on the box private LAN IPs?
11:22.09jkroonno, public IPs, no NAT.
11:22.11SamotOr does it listen directly on a public IP?
11:22.31SamotSo the answer is yes, the box is directly on the Internet.
11:23.30jkroonok, yes, the box is directly on the internet, and the firewall doesn't filter a lot of traffic for this machine, all SIP is allowed in, along with the RTP ranges, and IAX/2 traffic.  Logs on the firewall doesn't show any traffic being dropped by the firewall, and hardware counters seems sane too.
11:23.45jkroonerror counters ...
11:24.26jkroonso nothing dropped by interface/switch itself that I can find.  the switch did report 30 input packet errors overall on the switch over something like 3 billion packtes, so I think we can leave that out of the equation.
11:28.23SamotSo it's on the Internet and the firewall let's pretty much everything through..
11:28.43SamotNeed to see how many people are actually banging at the door on your box.
11:32.38*** join/#asterisk troyt (~troyt@2601:681:4600:7641:44dd:acff:fe85:9c8e)
11:36.32SamotYou have things like allowguest=no and allow_sip_anon=no?
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11:43.39*** part/#asterisk AviiNL (~AviiNL@185.21.52.255)
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11:49.55jkroonSamot, allowguest = no
11:51.24jkroonI cannot find that SIP has a allow_sip_anon setting.
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12:03.57SamotI think that goes in extensions.conf
12:04.20*** join/#asterisk SunGod (~SunGod@92.55.116.125)
12:07.00jkroonSamot, that seems to be a freepbx setting?
12:07.01SamotWell allow_sip_anon is more a global variable for dialplan actions.
12:07.11jkroonwhich we're not using.
12:07.15SamotIt's not a SIP setting.
12:07.35jkroonlooks like a dialplan global variable, specific to freepbx based on what I can find.  we're not using freepbx.
12:07.36SamotIt's a global variable...yes..I was looking in the wrong SSH tab on my boxes.
12:10.17jkroonno problem.
12:10.35jkroonwe've been running in approximately this config for ~8 years now, and definitely no changes since Feb.
12:10.55jkroonwell, other than adding more endpoints ... so I'm thinking it's load related somehow.
12:20.14*** join/#asterisk sekil (~sekil@jabber.net011.net)
12:23.38jkrooni picked up that the hardware rx buffers was at the default 256.  I increased those to 4096.  time will tell if this makes a difference.
12:24.14WIMPyHardware?
12:24.26jkroonethtool -g eth0 ... those settings.
12:24.52WIMPyAh
12:25.20jkroonethtool -S eth0 showed rx_no_buffer_count: 27128059
12:25.20jkroonthat's 27 *million* packets that won't show up in other error counters which I was previously looking at and monitoring.
12:25.39jkroonfrom a switch perspective the tx is successful, and from the rx_errors the packet was successfully received but due to no space in the buffers couldn't be passed to the OS.
12:26.41jkroonmostly in past experience for most drivers/hardware implementations (I'm not actually too clued up on where/how those counters are implemented and it seems to be driver specific) no buffer accounted to rx_errors and more specifically rx_missed_errors which doesn't seem to be the case here.
12:26.45WIMPyThat sounds pretty bad once again.
12:27.02jkroonwhich is WHY my existing monitoring didn't pick up on it.
12:28.16jkroon27128059/56564770852 = 0.05% over 570 days, 17:08.
12:28.43jkroonso most likely the problem just became noticable now it was probably there for a longer time.
12:30.04WIMPyThat's the point where I expect someone to say it wouldn't have happened on BSD...
12:32.03jkroonwell, with ping to another destination on another network I'm still seeing 0.008 % loss (ping -f).  not sure if it's worth chasing that down.
12:32.56jkroonand if it wouldn't (by logical reason, not just because someone said so) have happened on BSD I'd be willing to put that to a test.  If I can rig one.
12:50.50*** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net)
13:01.06*** join/#asterisk guest9103 (~bounceman@185.32.9.250)
13:01.37guest9103Hi, in the manager.conf. How should I specify the permit option if I wish to allow multiple IP-addresses? I find no docs explaining this.
13:07.50[TK]D-FenderI found a ton just googling it right now
13:07.52[TK]D-Fenderhttp://www.voip-info.org/wiki/view/Asterisk+config+manager.conf
13:07.57[TK]D-Fendershows multiple used directly
13:08.08[TK]D-Fenderthe sample config itself shows the ACL option
13:08.17[TK]D-Fenderhttps://wiki.asterisk.org/wiki/display/AST/Named+ACLs
13:08.32[TK]D-FenderWhich is pretty clearly documented on the official wiki.
13:11.02guest9103What the heck is "Named ACL"?
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13:12.57[TK]D-FenderACL.
13:13.07[TK]D-Fendera csettting right there in the sample config
13:13.58[TK]D-Fender[TK]D-Fender> https://wiki.asterisk.org/wiki/display/AST/Named+ACLs <- I think the very first paragraph answers that already
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13:29.50guest9103Great!
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13:37.23guest9103[TK]D-Fender: do you happen to know if the pem file needs to be generated using the as_tls script or is it okay to generate it using openssl req?
13:38.01[TK]D-FenderI don't
13:38.19guest9103[TK]D-Fender: You are the man of the day anyway.
13:39.00guest9103Accept
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14:18.17punk_hello everyone :)
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15:22.42jmetroCan anyone remind me the name of the default hold music? :)
15:24.07[TK]D-Fender[default]
15:24.33jmetroThe Mp3 <.< i think its "Cold Morning Coffee" ?
15:24.48[TK]D-FenderIIRC * includes 4 or so
15:26.26jmetrohmm...
15:26.46*** join/#asterisk Chotaire (chotaire@oahu.chotaire.net)
15:44.54ghotiSo ... I have a slew of what appear to be "zombie" channels "core show channels verbose" output is here: [root@pbx ~]# asterisk -rx 'core show channels verbose'
15:44.57ghotiChannel              Context              Extension        Prio State   Application  Data                      CallerID        Duration Accountcode PeerAccount BridgedTo
15:45.01ghotiSIP/222-00000532     from-internal        STARTMEETME         4 Up      ConfBridge   822,,,                    222             4122:55:                         (None)
15:45.05ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    3024:46:                         (None)
15:45.09ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    1027:33:                         (None)
15:45.13ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    193:05:1                         (None)
15:45.17ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    1145:44:                         (None)
15:45.22ghotiBridge/0x7f256800ba0 default              s                   1 Up      (None)       (None)                                    4122:54:                         (None)
15:45.25ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    118:28:4                         (None)
15:45.29ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    3165:56:                         (None)
15:45.33ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    3024:57:                         (None)
15:45.37ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    3166:07:                         (None)
15:45.42ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    1196:13:                         (None)
15:45.46ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    433:42:2                         (None)
15:45.50ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    1196:41:                         (None)
15:45.54ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    4122:54:                         (None)
15:45.58ghotiBridge/0x7f256800ba0 default              s                   1 Up      (None)       (None)                                    4122:54:                         (None)
15:46.02ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    3165:57:                         (None)
15:46.06ghotiConfBridgeRecorder/c default              s                   1 Up      (None)       (None)                                    118:31:2                         (None)
15:46.10ghoti17 active channels
15:46.13ghoti1 active call
15:46.15ghoti16876 calls processed
15:46.17ghotiCRAP, sorry about that, meant to paste the patebin.ca link. :-(
15:46.46ghotihttp://pastebin.ca/3707994, fwiw. :-P
15:47.36ghotiThese appear to have been valid calls that never got removed from .. some internal table. And they're affecting my stat gathering, though I don't think they're actually affecting performance or trunk usage.
15:48.07ghotiAny idea what I can do to clear them? I imagine that rebooting the box would do the trick, but I'd prefer something less .. drastic.
15:58.37*** join/#asterisk Chotaire (chotaire@oahu.chotaire.net)
16:16.29*** join/#asterisk matrix1233 (~matrix123@197.0.177.30)
16:26.36rwbHi, does anyone know why the extensions don't ring?  I have analog phones with FXS extentions.  If I dial an extention, it doesn't ring, but when I pick it up, it connects!
16:28.20WIMPyWrong voltage, wrong frequency, too little power.
16:30.48rwbHmm, they are old analog phones, but there should be enough power from the card (the supply is in fine)  Not sure how to test that.
16:31.49SamotIs the ringer turned on?
16:32.44SamotOr turned up?
16:33.08SamotI don't know how many times that was the cause of "service not working right" when I was doing residential services.
16:33.19*** join/#asterisk miralin (~Thunderbi@194.8.128.48)
16:35.10*** join/#asterisk lgaetz (~lgaetz@66.185.28.100)
16:35.21rwbyea, are, I was hoping for maybe some setting somewhere as I am quite new to this.  I'll have to figure out a way to test the phones themselves.
16:36.31SamotDo you have an ATA or even a POTS line?
16:38.10rwbI don't, so far no way to test.  I will be getting some "ring detectors" shipped today though, so I'll try those directly.
16:39.21rwbI'm just really hoping is not the pci board or modules!
16:40.30SamotWhat does 'dahdi show cadences' have?
16:40.49Samotor 'dahdi show status'
16:43.27*** part/#asterisk lgaetz (~lgaetz@66.185.28.100)
16:49.04rwbI see numbers for r1: - r4 but not sure what those mean.
16:50.40*** join/#asterisk boris_rh (d0414927@gateway/web/freenode/ip.208.65.73.39)
16:53.42boris_rhHello, I was looking to use ARI snoop feature: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Channels+REST+API#Asterisk13ChannelsRESTAPI-snoopChannel
16:54.13boris_rhbut then I came across this limitation: http://lists.digium.com/pipermail/asterisk-dev/2015-November/075157.html
16:54.46boris_rhdoes not it make whole whisper feature useless or I am missing something?
16:56.07fileuseless only if you have no media going to the channel
16:58.45boris_rhthanks @file can you please elaborate ?
16:59.03boris_rhif channel is in conference
16:59.12fileif the channel has no media being sent to it, then the whisper can't mix itself in
16:59.27filea conference generates media always, so it would work
16:59.41filethe same limitation applies to ChanSpy
17:00.11boris_rhso why guy at http://lists.digium.com/pipermail/asterisk-dev/2015-November/075157.html not happy?
17:00.15boris_rhis not it his case?
17:00.32boris_rhwhen he has conf bridge
17:00.42boris_rhand another bridge for spying
17:00.48filein his case the channel was NOT in a bridge
17:01.03fileand ARI does not generate a constant media stream to the channel, unless you call /silence
17:01.42boris_rhThanks @file, so if I have some bridge with two channels
17:01.48filethen it's fine
17:01.57boris_rhI invoke snoop on one of the channel
17:02.12boris_rhand add snoop channel to another bridge
17:02.27boris_rhwhere I also adding whisperring person
17:02.37boris_rhis it right way to do?
17:02.40fileyes
17:03.13boris_rhgreat, but it sounds to me the same case as at http://lists.digium.com/pipermail/asterisk-dev/2015-November/075157.html
17:03.23boris_rhmaybe I missing something
17:03.36filein that case the channel being snooped on was doing nothing
17:03.40fileit was waiting for ARI operations
17:03.47filewhen a channel is waiting in ARI *NO AUDIO* is sent to it
17:03.56filesince there's no audio the whispering can't mix the whispered audio in
17:04.19fileif the channel is in a bridge with another channel then media IS being sent to it so the whispering will work
17:04.59boris_rhOK, cool, thanks a lot @file, it is really encouraging , I will give it a try
17:06.14boris_rhcan I create more then one snooping channel for the same channel?
17:06.23fileyes.
17:07.44boris_rh@file: great, thanks
17:09.22*** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-bxfbilmunrrhfjob)
17:10.35*** join/#asterisk Euler (~Euler@pool-173-48-218-40.bstnma.fios.verizon.net)
17:10.41Eulerhi all
17:11.22Eulerupgrading to v13 from like v1.4, and asterisk doesnt seem to handle ENV vars same way;
17:11.47Eulerthat is, the ENV my AGI script has access to
17:12.29[TK]D-FenderThat's nearly a decade-long jump
17:12.47Eulertried passing in via AGI() command (e.g., AGI(/path/to/script, ${ENV(VARNAME)}), but it comes through as empty string
17:12.52[TK]D-FenderShow the actual failure & code BTW
17:14.56Eulerbasically, i have a linux ENV var that i want to pass in and it doesnt seem to be available
17:15.55[TK]D-FenderRight from the start that'd have nothing to do with AGI, and rather that dialplan function.
17:16.35Euleri think the big difference is that it is running now as "asterisk" user, and i'm not sure that user has the same env as other users
17:17.08Euleryea, i don't think it's related to AGI() specifically; that's just where i noticed it
17:17.53Euleraccording to extensions.conf doc, "Unix/Linux environmental variables can be reached with the ENV dialplan function: ${ENV(VARIABLE)}"
17:18.06Eulerbut that seems to be empty for my env var
17:18.20Eulerso i'm guessing the asterisk user doesnt have it for some reason
17:19.27WIMPyDo you start Asterisk the same way you did before?
17:20.12Eulerit's a newer OS, so it uses "service" calls instead of older init script
17:27.01Eulerwell, looks like i can set specific env vars in the systemd script
17:27.28Eulerso, i'm past that hump
17:27.37Eulerthanks for listening!
17:28.53Euler(ffr, adding the var to /usr/lib/systemd/system/asterisk.service , in format like: Environment=MYVAR=someval )
17:31.23punk_is there any way I can manipulate an incomming call from the console? so that it gets redirected to another extension?
17:32.05punk_I guess I could create hooks/checks in the extensions.conf file which looked at some file or database to see if the call should go somewhere else, but is there a better way? :-)
17:34.25[TK]D-Fenderpunk_, depends when and how you actually want these things to happen
17:34.36[TK]D-Fenderpunk_, because you just described 2 completely different things
17:34.44[TK]D-Fenderpunk_, So clarify your actual goal
17:38.00punk_[TK]D-Fender: Customer calls the queue (choose 1 for sales... etc), at the same time I have an IRC bot that tells me who is calling/number, I want to be able to type !mine or !transfer <phonenum> and it goes directly to the correct person :-)
17:38.45[TK]D-FenderThat would be an AMI Redirect you're looking for.
17:39.24[TK]D-FenderAnd nothing to do with a database lookup, etc.  You are deciding live yourself external to the PBX.
17:39.39punk_Thank you for the push in the right direction, I will investigate the AMI :)
17:42.18*** join/#asterisk infinity_ (~brendon@web2.artsopolis.com)
18:01.38punk_[TK]D-Fender: Quick question, probably a noob one, which channel should I use for the API when its a call coming over a SIP Trunk?
18:02.22[TK]D-FenderThe channel you want to hijack clearly
18:02.35punk_[TK]D-Fender: But I only know the call id?
18:03.19*** join/#asterisk matrix1233 (~matrix123@197.0.177.30)
18:03.31[TK]D-FenderIs that ... a question?
18:03.44punk_is that ... the same? the channel and the call id?
18:04.07[TK]D-FenderNo.
18:04.19[TK]D-Fendercall-id = SIP thing
18:04.43[TK]D-Fenderchannel = asterisk thing that encapsulates the call.  * channel can be of ANY type
18:05.44punk_I just noticed there is something called "channel redirect" in the asterisk console, I guess I could use that instead of the API?
18:05.58punk_since my bot can do asterisk -rx
18:06.10[TK]D-Fenderthat could work
18:08.55punk_[TK]D-Fender: I figured it out, great, thank you so much for your help and the push in the right direction. Have a nice day! :)
18:09.20[TK]D-FenderYou're welcome
18:12.24rwbAnyone know if 24G phone wire thick enough to ring older dial phones with bells?
18:14.54Nuggettakes a moment to say thanks for the fact that he'll never have to know that.
18:25.32gruetzkopfover which distance?
18:26.03gruetzkopf24 AWG is 0.5mm diameter, that should be plenty.
18:26.24gruetzkopfsource: run stuff like that (although not on a FX* pci card)
18:37.26rwbmaybe 200 ft max each chan.  I have a patch panel connected with a short telco cable to a 24 line FXS pci card.  I'm thinking of driving some old dial phones with the 24g.  I think it should be fine as well.
18:38.36rwbNo "big" bells or lights without a seperate relay...
18:46.40gruetzkopfi run pretty big bells over like 1000ft of .6mm wire, you should be fine
18:50.02rwbgood to know. Thanks!
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19:47.04rwbIs there a good way to trouble shoot why some FXS ports are dead?  I'm pretty new on this.  In my "Full Report" screen I see "State:Unavailable" for the ones that seem dead (no dial tone)?
20:00.01rwbOr another way to ask; Does it mean I have a bad board and/or FXS modules if I don't hear dial tone from ALL of the ports?
20:00.52[TK]D-Fenderdo you have a proper indications.conf?  Set your zones right?
20:00.53rwbOr, could this be configuration error (me)
20:01.04[TK]D-FenderTested with other phoens to validate the current draw as being sane?
20:01.50rwbSo far I'm using the web admin only.  Yes, I use the same phone on all the ports, and about half of them are silent
20:02.08rwbsame phone, and same patch cable.
20:02.40*** join/#asterisk jeffspeff (~Jeff@12.49.160.131)
20:03.02rwbI see "State:Unavailable" in the reports area for the ones that seem dead
20:03.37rwbI'm wondering if I just need to "enable" them somewhere
20:03.45[TK]D-Fenderhave you tried a DIFFERNT PHONE?
20:03.54[TK]D-Fenderand a DIFFERENT CABLE
20:04.13rwbyes, same results with 3 different phones, 3 cables, on all ports
20:04.38rwball old really old phones though...
20:04.46[TK]D-FenderThe back to the first 2 thing I told you to look for.
20:05.21rwbahh, I lost track. about when was that?
20:06.05rwboh, sones and indications... I will look at those
20:06.46jeffspeffi've done a fresh install of * on a google compute instance. i'm able to run * using 'asterisk -cv' but i can't get the service to act right. here's a paste of the error and some other details http://pastebin.com/UTLmsh3a any ideas? thanks
20:21.02*** join/#asterisk kubrrick (~Mutter@78.80.9.109.rev.sfr.net)
20:24.29kubrrickHello guys, i m looking for srtp on my asterisk server, can someone tell me how to set cipher suite for srtp ??
20:57.52*** join/#asterisk Qwell (~north@asterisk/developer/Qwell)
20:57.52*** mode/#asterisk [+o Qwell] by ChanServ
21:08.19*** join/#asterisk Pegasus_RPG (~Icedove@47.142.203.153)
21:09.20Pegasus_RPGHello. I'm trying to make a generic "pause queue member" application using  exten => *30*,1,PauseQueueMember(,${SIPUSERAGENT})  on * 11 but it keeps saying the interface parameter is missing
21:09.45Pegasus_RPGIf I manually specify e.g. SIP/User1   that works
21:10.09Pegasus_RPGSo I guess my question is: how can I strip off the transaction ID from the User agent string?
21:10.38voipmonkCUT it out
21:11.14Pegasus_RPGtries PauseQueueMember(,SubString(${SIPUSERAGENT},0,9))
21:11.42Pegasus_RPGXD  Attempt to pause interface SubString(,0,9), not found
21:12.03Pegasus_RPGreads about cut
21:17.25Pegasus_RPGtries a few things with it
21:20.55Pegasus_RPGRight now I have exten => *30*,1,Set(Member=${CUT(SIPUSERAGENT,,1)})   but it's not splitting on the -   (SIPUSERAGENT contains  SIP/User1-000003b  )
21:21.26Pegasus_RPGAnd I want just the SIP/User1  part
21:21.30Pegasus_RPGvoipmonk: ^
21:22.14voipmonkthat doesn't look right -
21:22.43voipmonkhold on - bee's like my vape - and this form doesn't do well with stings - moving
21:23.29*** join/#asterisk clopez (~tau@neutrino.es)
21:23.43Pegasus_RPGwait I think i found a trick with ${foo:0:-8}
21:25.05Pegasus_RPGDAMMIT  exten => *30*,1,Set(Member=${SIPUSERAGENT:0:-9})  isn't working either
21:25.15Pegasus_RPG(Shows "Member="  )
21:28.55Pegasus_RPGAUGH ${SIPUSERAGENT} * - SIP user agent (deprecated)
21:29.01Pegasus_RPGSo much old info on voip-wiki
21:31.34*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
21:43.11*** join/#asterisk jkroon (~jkroon@2c0f:f720:1:0:fcb3:e235:8b0a:1f6b)
21:47.54Pegasus_RPGgot it! ${CHANNEL:0:-9}
21:48.16Pegasus_RPGThanks to the real wiki https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables
21:48.31Pegasus_RPGvoipmonk: ^
22:32.24kunwon1Pegasus_RPG: i'm not sure if the length of a channel name is a constant, that may have weird corner cases
22:33.37Pegasus_RPGkunwon1: I'm sure. I'm open to better ideas. :)
22:38.30kunwon1Pegasus_RPG: is ${BLINDTRANSFER} populated?
22:43.38kunwon1looks likke not
22:47.49Pegasus_RPGThe question is: is the 8-digit transaction ID (or whatever it's called) always 8 digits?
22:48.24Pegasus_RPG(the -9 strips off the last 9 characters so should work fine for arbitrary length strings so long as that ID is constant length)
22:48.27WIMPyThe only thing we know about channel names is that they start with a channeltype name and a slash.
22:48.53kunwon1Pegasus_RPG: i figured it out, here you go  same => n,Set(cutVar=${CUT(CHANNEL,,-3)})
22:49.10kunwon1that puts the device name in ${cutVar}
22:50.00kunwon1maybe this would have corner cases too? unsure, i'm fairly certain it would work with my pure-SIP environment
22:50.38*** join/#asterisk acidfu_ (~acidfoo@24-212-247-227.cable.teksavvy.com)
22:50.39WIMPyFor the "corner case" of using chan_sip it will work.
22:50.41kunwon1i wonder if pausequeuemember needs SIP/ in front? the line i just posted includes that, but adding another CUT should be able to remove it
22:50.47Pegasus_RPGWIMPy: I guess * needs an INTERFACE variable then :)
22:51.01Pegasus_RPGkunwon1: it does
22:51.05WIMPyWhat would that be?
22:51.18kunwon1good times
22:51.36kunwon1WIMPy: is chan_sip in the minority these days?
22:51.37Pegasus_RPGWIMPy: e.g. SIP/User1
22:51.49Pegasus_RPGWIMPy: ZAP/foo
22:51.51Pegasus_RPGetc
22:52.03WIMPyNope. Don't think so. But still there are many other channeltypes/
22:52.07kunwon1true
22:52.08WIMPy.
22:52.45Pegasus_RPGkunwon1: how does your CUT line differ from my substring one?
22:52.58kunwon1my CUT line doesn't care how long the part after the final hyphen is
22:53.03Pegasus_RPGah okay
22:53.24kunwon1i believe the part after the final hyphen is of variable length, but i have no supporting evidence
22:54.04Pegasus_RPGDoesn't work for me though
22:54.25kunwon1:/
22:54.37WIMPyThe only thing we know about channel names is that they start with a channeltype name and a slash. -- That means everything else is unknown!
22:54.38Pegasus_RPGsec, trying stuffing it into an intemediate varable
22:55.03WIMPyThat also means that thee's no generic way to identify devices.
22:55.36Pegasus_RPGkunwon1: yep, I needed to put the cut result into a variable, then call that from PauseQueueMember()
22:56.15kunwon1ah, good times
22:57.25Pegasus_RPGoh wait, no it still doesn't work :/
22:57.44kunwon1try changing -2 to -1, i have more hyphens in my device names than normal, i think
22:58.32WIMPyIt might be easier to just use the CANNEL function.
22:58.34Pegasus_RPGMine are like  SIP/User1-00000059
22:58.53WIMPyNot generic, either, though.
22:59.21*** part/#asterisk kharwell (kharwell@nat/digium/x-wmzjtfmjleisfxqs)
23:01.28kunwon1hmm, ${CHANNEL(peername)}
23:01.30kunwon1TIL
23:10.48*** join/#asterisk newtonr (~newtonr@173-21-146-94.client.mchsi.com)
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