IRC log for #asterisk on 20180215

00:02.35*** join/#asterisk Amnesia (~Amnesia@unaffiliated/amnesia)
00:09.55*** join/#asterisk qxork (~qxork@unaffiliated/qxork)
00:14.30*** join/#asterisk talntid (~talntid@unaffiliated/talntid)
00:19.31*** join/#asterisk dar123 (~dar@107-203-254-117.lightspeed.sntcca.sbcglobal.net)
00:32.39*** join/#asterisk u0m3 (~u0m3@188.25.124.179)
00:54.09*** join/#asterisk DaRock (~Thunderbi@150.101.178.33)
01:19.57*** join/#asterisk infobot (ibot@rikers.org)
01:19.57*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.1 (2018/02/13), Standard: 15.2.1 (2018/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
01:32.41*** join/#asterisk nix8n82 (~AndChat62@2600:100e:b01c:a980:91c2:a7f9:7bff:638e)
01:34.16*** join/#asterisk AndChat-620489 (~AndChat62@2600:100e:b01c:a980:860:1ef8:dcdf:17ea)
02:10.22*** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net)
02:15.51*** join/#asterisk AndChat|620489 (~AndChat62@67-130-74-235.dia.static.qwest.net)
02:52.17madduckdoes anyone have any recommendations for how to set up the new CT target-based netfilter helpers to handle not only RTP 10000–20000, but also the rest of RTCP and RTSP?
02:52.35madduckor are we basically just talking about 1024–65535?
02:56.58*** join/#asterisk nix8n82 (~AndChat62@2600:100e:b01c:a980:a1a5:16a9:5c47:64e0)
02:58.30*** join/#asterisk AndChat-620489 (~AndChat62@67-130-74-235.dia.static.qwest.net)
03:11.00*** join/#asterisk Oooohboy (~Oooohboy@2605:6000:1711:c1ff:407d:634d:a14b:242e)
03:16.02*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
03:16.15*** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net)
03:16.54*** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi)
03:19.09*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
03:31.09*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
03:31.20*** join/#asterisk AndChat|620489 (~AndChat62@2600:100e:b01c:a980:db8:27d:cf29:631d)
03:33.18*** join/#asterisk Oooohboy (~Oooohboy@2605:6000:1711:c1ff:5de1:182:e755:ba7a)
03:33.58*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
03:35.43*** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net)
03:35.58*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
03:38.08*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
03:38.57*** join/#asterisk Oooohboy (~Oooohboy@2605:6000:1711:c1ff:5de1:182:e755:ba7a)
03:40.05*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
03:44.27*** join/#asterisk Oooohboy (~Oooohboy@2605:6000:1711:c1ff:5de1:182:e755:ba7a)
04:11.33*** join/#asterisk AndChat|620489 (~AndChat62@2600:100e:b01c:a980:7859:6158:ff3f:636e)
04:16.58*** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net)
04:29.49*** join/#asterisk AndChat|620489 (~AndChat62@2600:100e:b01c:a980:ad38:bed7:7a1f:7c59)
04:35.56*** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net)
04:43.12*** join/#asterisk AndChat|620489 (~AndChat62@2600:100e:b01c:a980:11de:9d9e:9593:7479)
04:44.19*** join/#asterisk K0HAX (~michael@28.139.154.104.bc.googleusercontent.com)
04:45.29*** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net)
04:59.58*** join/#asterisk NonSecwitter (~NonSecwit@unaffiliated/nonsecwitter)
05:23.51*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
05:25.55*** join/#asterisk AndChat|620489 (~AndChat62@2600:100e:b01c:a980:cd3b:5d1b:6592:d735)
05:28.27*** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net)
06:25.34*** join/#asterisk AndChat|620489 (~AndChat62@2600:100e:b01c:a980:7cbf:eb28:47ec:a37d)
06:26.29*** join/#asterisk AndChat-620489 (~AndChat62@67-130-74-235.dia.static.qwest.net)
06:38.56*** join/#asterisk ashutoshkrchaube (673399be@gateway/web/freenode/ip.103.51.153.190)
06:39.57ashutoshkrchaubeHi what is the maximum size in mb of voicemail that asterisk can play?
06:41.44*** join/#asterisk nix8n82 (~AndChat62@2600:100e:b01c:a980:bc98:f85c:4cd0:834e)
06:42.14*** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl)
06:42.35*** join/#asterisk AndChat|620489 (~AndChat62@67-130-74-235.dia.static.qwest.net)
07:19.37*** join/#asterisk jkroon (~jkroon@uls-154-73-35-201.wall.uls.co.za)
07:26.06*** join/#asterisk jamesaxl (~James_Axl@109.172.62.242)
07:27.06*** join/#asterisk nix8n82 (~AndChat62@67.130.74.235)
07:35.59*** join/#asterisk AndChat|620489 (~AndChat62@67-130-74-235.dia.static.qwest.net)
07:54.49*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
08:12.11*** join/#asterisk Oooohboy (~Oooohboy@72-48-251-4.dyn.grandenetworks.net)
08:21.21*** join/#asterisk Oooohboy (~Oooohboy@72-48-251-4.dyn.grandenetworks.net)
08:23.16*** join/#asterisk Oooohboy (~Oooohboy@72-48-251-4.dyn.grandenetworks.net)
08:32.25*** join/#asterisk hehol (~hehol@gatekeeper.loca.net)
08:34.35*** join/#asterisk visip (~visix@gateway/tor-sasl/visip)
08:51.04*** join/#asterisk liviuc (~liviuc@109.99.227.30)
08:57.02*** join/#asterisk DanB (~DanB@clt-195.192.204.39.ip-anschluss.net)
09:06.26*** part/#asterisk Amnesia (~Amnesia@unaffiliated/amnesia)
09:09.45DonkeyDongHi, i have a silly question. Is pjsip the new norm or are people in reality still using chan_sip?
09:34.39DanQuinneyDonkeyDong: no such thing as a silly question, you'll find ITSP's using both chan_sip and pjsip, we use pjsip and it has bitten us a couple of times - but in the grand scheme of things we prefer it. You'll find others really dislike pjsip mainly as it's not as tested in the wild as chan_sip is.
09:44.34DonkeyDongRight, thanks :-)
10:00.14SamotThe only ITSPs that have chan_sip and chan_pjsip have Asterisk. Those are not things in most other sip software.
10:03.05*** join/#asterisk [sr] (~kvirc@pal-213-228-163-73.netvisao.pt)
10:03.07[sr]howdy
10:05.57[sr]when i have external call's comming in, whethear they came from an ip trunk, or an ISDN trunk
10:06.17[sr]the field "dcontext" in cdr, is always "ext-group" ?
10:09.51[sr]from what i've analized yes, just want to confirm at 101% ;)
10:13.00SamotSo this is FreePBX?
10:19.22[sr]no, cdr!
10:19.24[sr]asterisk!!
10:20.02[sr]ok the context's names may be hammered by freepbx
10:29.43*** join/#asterisk Samael28 (~Samael28@176.104.56.91)
10:30.26SamotWell do these calls go to ring groups?
10:30.46SamotBecause ext-group is where all the Ring Group dialplan is held in FreePBX.
10:58.22*** join/#asterisk bounceman (~bounceman@185.32.9.250)
10:58.31[sr]oh ok, going to similate a call to an extension to see whats recordes there
10:59.55bouncemanHello, I have a very interesting issue. Have a look at this CLI trace http://codepad.org/8lYfFYzG What happens is that after an attended transfer is completed (10:12:05) Music on Hold starts on the transferee target. The channels was swapped at line 23 and Moh starts at line 25
11:00.19bouncemanI've seen people with similar issues on the asterisk jira, but no one really seems to have solved it.
11:00.49[sr]Samot: "from-did-direct",
11:01.04SamotThis is 100% FreePBX dialplan.
11:01.06bouncemanAlso at line 27 & 28 MoH is stopped and started for the incoming part (A)
11:01.21Samot[sr]: What's the issue?
11:03.02Samot[2018-02-15 10:11:24] VERBOSE[7902][C-00013ed2] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/foobar_sip_trunk-00022a25'
11:03.25Samot[2018-02-15 10:11:39] VERBOSE[8624][C-00013ed9] pbx_realtime.c: Executing [1023@from-sip:2] Dial("SIP/fredrik.bergqvist-00022a37", "SIP/margareta.brun,,tTxX")
11:03.46bouncemanSamot: yes, the A part was holded, that is why the MoH started at 10:11:24. Then B dialed C(Margareta)
11:04.22bouncemanBut why would MoH starts after the attended transfer was performed?
11:04.52SamotA called in.
11:04.54SamotB Answered.
11:04.57SamotB puts A on hold.
11:05.00SamotB calls C
11:05.03SamotB puts C on hold
11:05.09SamotB transfer A to C
11:05.17SamotB hangs up A, stops that music on hold
11:05.26bouncemanNo, C is never put on hold. The attended transfer was performed while B and C was talking
11:05.27[sr]Samot: none, i'll similate call's to see whats recorded there ;) i have my own frontend for cdr
11:05.29SamotB hangs up with, stops that music on hold.
11:05.58Samot[sr]: Every context you have mentioned for far is a FreePBX generated context.
11:06.06SamotFor years the same contexts.
11:06.14bounceman[2018-02-15 10:11:42] VERBOSE[8624][C-00013ed9] app_dial.c: SIP/margareta.brun-00022a38 answered SIP/fredrik.bergqvist-00022a37
11:06.20bouncemanThis is the part were C answers B
11:06.27SamotSo you either wrote you dialplan and just happened to use all the same context names, or this is a hack of FreePBX code.
11:06.30bouncemanThere is no hold between that time and where the transfer take part
11:06.56Samot2018-02-15 10:12:05] VERBOSE[8627][C-00013ed9] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/margareta.brun-00022a38'
11:07.08SamotMoH was triggered on C's channel.
11:07.12SamotSomeone put C on hold.
11:07.30SamotHow was the transfer made?
11:07.48SamotWhat method did B use to transfer the call?
11:08.00bouncemanSamot: Someone put C on hold
11:08.06bouncemanyes, someone or something
11:08.07SamotWhat method did B use to transfer the call?
11:08.26SamotDid they put A on hold and called C
11:08.33bouncemanBy method you mean attended transfer or featurecode?
11:08.50SamotAttended Transfer is a thing that can be done via feature code or not
11:08.53SamotDon't confuse it.
11:09.06SamotThis is an Attended Transfer despite how it was initiated.
11:09.18SamotSo did they press the "Transfer" key on the phone, did they use Feature Codes?
11:09.22bouncemanI will post the scenario here below
11:09.26SamotDid they put A on hold then call C
11:09.44Samot_How_ it was transferred will shed some light
11:10.27bouncemanA calls B
11:10.28bouncemanB answers A
11:10.28bouncemanB holds A
11:10.29bouncemanB calls C
11:10.31bouncemanC answers B
11:10.33bouncemanB use attended transfer to connect A with C
11:10.35bouncemanMusic on Hold starts
11:11.20bouncemanI spoke to other peers and they say it looks like an asterisk bug, I am just double checking if anyone here has any idea
11:11.28bouncemanWe're running 13.16.0 by the way
11:17.49SamotWait.
11:17.57SamotWhat do you mean "uses attended transfer"
11:18.02SamotHow do they actually transfer....
11:18.27SamotAttended Transfer is the process of holding one call and staying on the line until the other party is connected.
11:18.36SamotYOU make sure the second party is available and control the call
11:18.52SamotVs a BLIND transfer which just sends the call without you being involved
11:19.04SamotAgain, HOW is it being done?
11:19.09SamotAre they using a Feature Code?
11:19.20SamotAre they using the Transfer key on the phone?
11:19.31bouncemanYeah, so when C answers B and B can verify that C is available, there is a button in the softphone that handles that. I have not built the softphone myself so I cannot tell you how the code works in the backend, but I've requested that info and hope to receive it soon.
11:20.02SamotOK
11:20.06SamotSo in general...
11:20.26SamotThe "Transfer" buttons on phones will send a REFER to the system.
11:20.38bouncemanYes, that is what it does.
11:20.42SamotThe moment they hit the button, it puts the caller on hold.
11:20.43bouncemanAnd we get the Accepted and all that
11:20.53SamotAnd opens a new channel to dial.
11:21.04SamotYou dial that new destination, they either answer or don't.
11:21.25SamotUntil the "Transfer" button is hit again, the original call is on hold.
11:21.51*** join/#asterisk qxork (~qxork@unaffiliated/qxork)
11:22.30bouncemanOk, even if this were the case. Why would asterisk start MoH after C has been put into the bridge?
11:27.24bouncemanYou have to agree that does not make any sense, even if the scenario was as you described
11:33.13SamotAnd what do the parties hear?
11:33.19SamotWhat is the experience for them?
11:33.33SamotDoes C hear MoH after B hangs up?
11:33.37bouncemanC hears MoH
11:33.57bouncemanA hears MoH the entire time, until one of them eventually hangs up
11:34.15SamotThen something isn't right.
11:34.19SamotIn how this is being done.
11:34.25SamotHow often does this happen?
11:34.27fileyou need to provide the SIP traffic.
11:34.37SamotYeah a full sip debug would help
11:34.41SamotThat was going to be next
11:34.48SamotHow often does this happen?
11:34.49bouncemanIt depends, our customer that does 100 transfer each time can experience 5-30% failure per day.
11:34.52bouncemanIt is extremely random
11:35.06SamotSo it's the same person doing it?
11:35.41SamotOr is this an office and multiple users are having the issue?
11:35.49bouncemanNo, there are 20 agents, who all experience this. Both on internal and external transfers
11:36.12SamotAll using the same softphone?
11:37.15bouncemanYes
11:37.28bouncemanWe have experience this is other environents as well with the same setup
11:37.54SamotOther PBX systems?
11:38.01bouncemanNo, asterisk 11 and 13
11:38.09SamotWhat softphone?
11:38.25bouncemanFrom Xenialab
11:39.20SamotHas this been experienced on anyone using something that is not this softphone?
11:39.31bouncemanOk, so lets imagine that the softphone is to blame. That means that the Softphone will tell Asterisk to start moh? How is this done? Via AMI Events or can it be done in another way?
11:39.41fileSIP signaling.
11:39.51SamotWe need to see an actual SIP Debug
11:39.56bouncemanfile: how does a hold look like in SIP?
11:39.56SamotFull SIP messages.
11:40.24fileit's a reinvite with either sendonly or recvonly (I always get them reversed) or a connection address of 0.0.0.0
11:41.32bouncemanI have a pcap between the asterisk and the agent performing the transfer
11:42.03bouncemanhttps://uploadfiles.io/gr0g8
11:42.47bouncemanSo based on what file said the Hold was initiated at 65 seconds, and the attended transfer at 106 seconds.
11:45.39SamotPCAPs are before Asterisk touches/processes the call.
11:48.04bouncemanYeah hang on
11:50.31*** join/#asterisk lankanmon (~LKNnet@CPE64777dd7e053-CM64777dd7e050.cpe.net.cable.rogers.com)
11:51.11bouncemanhttps://uploadfiles.io/u5c2w so the asterisk is 35.157.118.140 in this pcap
11:51.44bouncemanPretty basic pcap really, since it is an internal transfer.
11:51.46bouncemanNot much fun going on
11:51.58SamotNo.
11:52.04SamotI mean, a Asterisk SIP Debug
11:52.22SamotWhere it shows how Asterisk sees, processes when in Asterisk
11:52.25bouncemanThought you said before asterisk touches the call
11:52.56SamotI was pointing out that a PCAP is before Asterisk processes the call
11:53.12SamotThat's at the system level..
11:53.28SamotAn Asterisk SIP Debug will show how Asterisk is processing the SIP message.
11:53.31bouncemanOk, I do not have that sip debug I am afraid.
11:54.11fileit also shows it in relation to what Asterisk is doing, when things occur
11:59.50SamotAnd this PCAP is one sided.
12:10.17bouncemanAny ideas how I can step up my game in regards to the debug data provided. I want to know WHY asterisk starts MoH and who told it to do so.
12:10.27fileI looked, there's already an issue open
12:11.43bouncemanYeah I submitted one, it was closed and moved to duplicate https://issues.asterisk.org/jira/browse/ASTERISK-27071
12:14.52SamotHave you tried the latest version?
12:15.20SamotOutside of the fact there were security flaws found in 13.17 <
12:18.01bouncemanI looked through the changelog, couldn't see anything that relate to this
12:18.32SamotSure.
12:18.46SamotBut that doesn't mean it wasn't fix just when something else was fixed.
12:18.52filethe person posted a change which resolved it for them, but never responded to review feedback
12:19.04fileit didn't use locking so it could crash randomly
12:19.15SamotI've got an office of lawyers that transfer calls all day
12:19.20SamotA lot of calls.
12:19.23bouncemanfile: you're talking about Jason's post?
12:19.34fileyes.
12:19.39fileit was put up for code review as well.
12:20.11bouncemanThere has not been any update since 27th of July except for the issue merge just now, which scares me. We might even put up a bounty on this bug.
12:21.12SamotTry updating.
12:21.36SamotAgain, anything less than 13.17.2 or 13.18 (can't remember which) is exposed to security flaws.
12:21.59SamotIt was enough for 11 to be updated 3 weeks before SFO ended.
12:22.33bouncemanSamot: do you have more info to share?
12:22.48SamotI would have to go back and look at the announcement releases..
12:22.57SamotBut this was back in Sep/Oct 2017
12:23.06SamotSince 11 died on Oct 25th 2017
12:23.24stefan27bounceman, can't you upgrade to latest and then manually merge in Jason's suggestion (at your own risk)?
12:23.36SamotOr even see if the merge is needed
12:23.59bouncemanWhat would be a potential issue of that fix?
12:24.06bouncemanfile mentioned crashing
12:24.21SamotFirst.
12:24.23SamotUpdate
12:24.35SamotTHEN see if you need to worry about a patch that doesn't using locking.
12:25.05Samot30% of failures is a high amount.
12:25.10bouncemanYeah, I just have to check the compability
12:25.20SamotIf this was still a bug in later versions, I would have had complaints.
12:25.24SamotCompability?
12:25.29SamotIt's 13.
12:25.42bouncemanWe had 80% of failures with internal transfers last friday
12:25.49SamotOK
12:25.51bouncemansomedays 0%
12:26.06SamotThis is either a bug in Asterisk
12:26.12SamotAn issue with the softphone
12:26.14SamotOr both
12:26.32filemy initial analysis could have also been incorrect and it could still be something separate, just be aware that chan_sip is community supported - so Digium doesn't touch the issues and it's up to others
12:26.35SamotYou can try a different softphone or a real IP phone to see if this issue exists...
12:26.44SamotYou can update Asterisk to see if this issue persists.
12:27.12SamotBut with the amount of transfers I have on this one system....
12:27.19SamotI would have seen this issue.
12:28.01bouncemanYeah, I will try an update. That is my first step
12:30.01bouncemanAnyone has experience with a bug boundy? Does it spark interest?
12:30.09bouncemanbounty
12:30.32SamotDon't know.
12:30.42SamotBecause that requires someone to get in and mess with the core code.
12:30.57SamotAgain, you're on a 10 month release.
12:30.59filesome have, some haven't, depends on what is needed and what is involved - chan_sip itself is messy and any change usually causes ripples, so that drives people away
12:31.18SamotSince that release we have gone from 13.16 to 13.19.1
12:31.27SamotThat's a lot of updates.
12:31.47SamotYou're also using a sip driver that isn't actively supported by Digium.
12:32.41SamotChan_SIP some saw nice updates in 14 but Digium didn't do them.
12:34.10SamotI'm going to guess over the next few years things are going to get wobbly for Asterisk users
12:35.07SamotMost ITSP/ISPs either 1) Benchmarked a pre-11 version of Asterisk and their configs are chan_sip only or 2) They never benchmarked, found generic how-to's for Asterisk and just shoved their details into them.
12:35.08bouncemanSo PJSIP?
12:35.52SamotAnd by and far, ITSP/ISPs don't know or care about Chan_SIP or Chan_PJSIP since they are both simply SIP.
12:36.24SamotSince 12 I've lost count of people claiming their ITSP "Doesn't support PJSIP"
12:37.18SamotPJSIP is the core driver.
12:37.22SamotSIP driver.
12:37.30SamotThat is where development is focused and driven.
12:37.46SamotAny updates for Chan_SIP are community driven and supplied.
12:40.13bouncemanGot you, would using WebRTC "remove" chan_sip from the equation?
12:40.20bouncemana webrtc softphone for an example
12:42.21fileunless you wrote your own signaling protocol and implemented it in Asterisk then you'd end up using either chan_sip or chan_pjsip for signaling
12:43.30SamotChan_SIP just went community support in recent history.
12:43.43SamotIt was still supported when WebRTC was introduced into Asterisk.
12:44.00bouncemanI dont mind using chan_pjsip
12:44.05bouncemanIt is chan_sip I want to flee
12:44.28SamotI would recommend a sandbox system then.
12:44.39SamotThe core concepts are still the same.
12:44.50SamotSetting names have been changed to support both types.
12:44.59SamotSo SIP/100 is PJSIP/100
12:45.23filehttps://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip
12:45.24bouncemanBut would it be a bad idea to try the transfer issue on pjsip instead?
12:45.30SamotDial() for PJSIP is different than Dail() for chan_SIP as PJSIP supports more than one contact.
12:45.37SamotNo.
12:45.44SamotWhat I'm saying is, just don't blanket move.
12:45.54SamotYou'll need to stand up a box side by side.
12:46.01SamotThe dialplan commands are different
12:46.10bouncemanI am aware, I've done pjsip configuration in the past. Just thought if it was worth a shot
12:46.12SamotThe logic/concept is still the same...
12:46.14SamotOK
12:46.16SamotIt is.
12:46.32SamotYou can eithe update and try chan_sip on 13.91.1
12:46.37SamotSee if the issue goes away
12:46.44SamotSince it probably might not be in a changelog
12:47.01SamotOr you can spin up a new box and test PJSIP
12:47.13bouncemanWell, I will try the asterisk update first
12:47.50bouncemanThe softphone should be OK either way? Would a standard SIP Stack work with PJSIP?
12:50.00fileSIP is SIP.
12:51.16SamotI just explained that
12:51.52fileheck, the PJSIP stack is used in tons of clients already
12:54.01fileSamot: I have the date on when we said chan_sip would not be core supported
12:54.07fileAugust 8th, 2014
12:54.17SamotOh
12:54.29SamotYeah, so a while
12:54.37SamotI though it was in 2015.
13:18.26*** join/#asterisk [TK]D-Fender (~joe@216.191.106.165)
13:50.30*** join/#asterisk freebs (~freebs@unaffiliated/freebs)
13:52.43bouncemanSamot: you can call me an idiot but I have failed to find those security things you mentioned. Would you care to guide me a bit further?
13:53.42filehttp://lists.digium.com/pipermail/asterisk-announce/2017-December/date.html http://lists.digium.com/pipermail/asterisk-announce/2017-November/date.html
13:54.28filehttps://www.asterisk.org/downloads/security-advisories
13:55.11bouncemanthank you file
14:01.05*** join/#asterisk qxork (~qxork@unaffiliated/qxork)
14:27.41*** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl)
14:32.13*** join/#asterisk brad_mssw (~brad@66.129.88.50)
14:33.45*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
14:38.17*** join/#asterisk ghoti (~paul@75.98.206.5)
14:45.52*** join/#asterisk NonSecwitter (~NonSecwit@unaffiliated/nonsecwitter)
14:58.04*** join/#asterisk dar123 (~dar@107-203-254-117.lightspeed.sntcca.sbcglobal.net)
14:59.17*** join/#asterisk kharwell (kharwell@nat/digium/x-fshymxsvkxclykkh)
14:59.17*** mode/#asterisk [+o kharwell] by ChanServ
15:17.49*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
15:22.52*** join/#asterisk bford (d8cff501@gateway/web/freenode/ip.216.207.245.1)
15:22.53*** mode/#asterisk [+o bford] by ChanServ
15:27.17*** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net)
15:39.27*** join/#asterisk dar123 (~dar@107-203-254-117.lightspeed.sntcca.sbcglobal.net)
15:41.43*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
15:44.01*** join/#asterisk rmudgett (rmudgett@nat/digium/x-ejrfueotmormcwfx)
15:44.02*** mode/#asterisk [+o rmudgett] by ChanServ
15:47.52*** join/#asterisk rpifan (~androirc@12.206.107.11)
15:47.57rpifanAnyone else at it expo
15:48.01rpifanAsterisk world east
15:54.54*** join/#asterisk troyt (~troyt@c-73-65-211-33.hsd1.ut.comcast.net)
15:55.19rpifanHi
16:03.39*** join/#asterisk miralin (~Thunderbi@194.8.128.80)
16:08.16*** join/#asterisk marlow2k (~Mario@rt-bb-d.Station-Berlin.Net)
16:08.32*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
16:10.09marlow2kHey guys! Today I have no real idea how to debug that. My asterisk was running for years, and after a 6 hour downtime of our internet connection, in the log I get the error chan_sip.c: sip_reg_timeout:    -- Registration for 'account@server' timed out, trying again
16:10.09marlow2kBut I can connect via telnet to that server and get answers. It looks like the connection was running for a long time and now after a restart something changed. Any ideas?
16:17.30*** join/#asterisk marlow2k (~Mario@rt-bb-d.Station-Berlin.Net)
16:17.51marlow2kSorry was disconnected. Any ideas?
16:17.52*** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net)
16:18.54*** join/#asterisk Worldexe (~Worldexe@95-107-33-134.dsl.orel.ru)
16:19.16*** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n)
16:19.17*** mode/#asterisk [+o cresl1n] by ChanServ
16:30.05*** join/#asterisk jkroon (~jkroon@165.16.204.162)
16:41.45*** join/#asterisk infobot (ibot@rikers.org)
16:41.45*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.1 (2018/02/13), Standard: 15.2.1 (2018/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
16:41.59marlow2kyes sip
16:42.41marlow2ksip show registry shows the state "Request Sent"
16:43.09marlow2kand i have the same for two independent sip servers
16:44.13*** join/#asterisk qxork (~qxork@unaffiliated/qxork)
16:46.59*** join/#asterisk startledmarmot (~startledm@2601:582:4301:a05c:ed92:14ec:1b39:9667)
17:00.39*** join/#asterisk rpifan (~androirc@12.206.107.11)
17:00.45rpifanHello
17:03.59*** join/#asterisk rwb (~Thunderbi@74.85.159.242)
17:06.13*** join/#asterisk viebig (c88fa9b5@gateway/web/freenode/ip.200.143.169.181)
17:06.47viebighi all! Is it possible to listen to SIP status using AMI or ARI ?
17:07.33viebigfor instance, I need to know all the sip status history of a call
17:20.03*** join/#asterisk dakudos (~dakudos@c-73-203-6-107.hsd1.co.comcast.net)
17:20.27*** join/#asterisk K0HAX (~michael@28.139.154.104.bc.googleusercontent.com)
17:26.51*** join/#asterisk [TK]D-Fender (~joe@216.191.106.165)
17:27.16[TK]D-Fenderviebig, What do you mean exactly?
17:33.09*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
17:35.04*** join/#asterisk jamesaxl (~James_Axl@109.172.62.242)
17:42.11*** join/#asterisk NotSecwitter (~NonSecwit@unaffiliated/nonsecwitter)
17:51.34*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
18:00.35viebig[TK]D-Fender: I need to reveive and record all SIP events
18:00.41*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
18:01.02viebig[TK]D-Fender: for instance, 183, 200 .. time between then etc
18:07.04[TK]D-FenderThere is no mechanism for that in Asterisk
18:39.03*** join/#asterisk ShaunR (~ShaunR@freenode/sponsor/NDChost.com)
18:41.01ShaunRAny recommendations on a good SIP Trunk provider in the US? Looking for a provider that allows a dynamic incomming concurrent connection count and mainly just charges for usage.
18:58.37[TK]D-Fender~itsplist-us
18:58.37infobotHere are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com
18:58.48[TK]D-Fendervoip.ms
19:03.16igcewielingI've used Vitelity since 2005, but for very low volume.
19:09.05*** join/#asterisk NonSecwitter (~NonSecwit@unaffiliated/nonsecwitter)
19:09.37*** join/#asterisk karl370 (~karl@cpe-142-129-161-190.socal.res.rr.com)
19:10.21drmessanoTwilio should be added to that list
19:11.23*** join/#asterisk startledmarmot (~startledm@2601:582:4301:a05c:7d1d:549f:5a09:17e6)
19:11.39*** join/#asterisk pruonckk (~pruonckk@135-143-11-177.raimax.com.br)
19:16.30*** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n)
19:16.30*** mode/#asterisk [+o cresl1n] by ChanServ
19:32.47*** join/#asterisk AndChat|620489 (~AndChat62@67-130-74-235.dia.static.qwest.net)
19:34.13*** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.213.148)
19:40.53*** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net)
19:51.50*** join/#asterisk Oooohboy (~Oooohboy@cpe-67-11-10-120.satx.res.rr.com)
19:51.53*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
20:03.22*** join/#asterisk nix8n82 (~AndChat62@2600:100e:b012:63f2:2102:b8df:1b2:4cc1)
20:05.22*** join/#asterisk AndChat-620489 (~AndChat62@67-130-74-235.dia.static.qwest.net)
20:05.47*** join/#asterisk Samael28 (~Samael28@176.104.56.91)
20:09.19karl370Hi everyone. I have a question in regards to a remote client having problems connecting to my hosted server. I'm wondering if it has something to do with an IPv6 address.
20:09.33karl370I'm hosting a server, straight Asterisk, v14, and an IPv4 address. The remote user had been able to connect to the server just fine, until today. The user's credentials are fine. I was able to connect to the server using the user's config file, downloaded to a phone at a different remote location. But the other user cannot connect. I used TeamViewer and did a factory reset on their phone, then set it up to download the config file fr
20:09.33karl370om my provisioning server (different than the asterisk server). The phone downloads the config file just fine, but I never see it attempt to register on my server (whereas the test phone did). Interestingly, on my provisioning server, I see that it is using an IPv4 address, but through a browser on their desktop, it shows both an IPv4 & IPv6 address. I can connect to the phone (through TeamViewer, using an IPv4 private address), but
20:09.34karl370am wondering if the phone is using the client's public address (potentially IPv6) and having issues. I'm using fail2ban, and have whitelisted the user's IPv4 address.
20:09.47karl370Having not done anything with IPv6, I was wondering if there is anything specific that needs to be done to allow IPv6 clients to connect? Additionally, does anybody have an idea as to what to try next to resolve this issue?
20:10.54*** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.213.148)
20:31.11*** join/#asterisk nix8n82 (~AndChat62@2600:100e:b012:63f2:496b:d3a2:c240:a2b4)
20:34.17*** join/#asterisk pruonckk (~pruonckk@177.11.143.135)
20:35.59*** join/#asterisk AndChat|620489 (~AndChat62@67-130-74-235.dia.static.qwest.net)
20:40.58*** join/#asterisk tzafrir (~tzafrir@62-90-199-247.barak.net.il)
20:49.24*** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl)
20:58.25*** part/#asterisk ShaunR (~ShaunR@freenode/sponsor/NDChost.com)
21:00.47*** join/#asterisk jhord (~jhord@c-73-181-14-245.hsd1.co.comcast.net)
21:00.52jhordwondering if someone could help with NAT hacking for Cisco phones
21:02.35Kobazhmmm
21:09.24*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
21:21.18*** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc)
21:24.16*** join/#asterisk thiagoc_ (~thiagoc@unaffiliated/thiagoc)
21:42.01*** join/#asterisk nix8n82 (~AndChat62@2600:100e:b012:63f2:7190:12e9:c2a6:56df)
21:42.23*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
21:43.09*** join/#asterisk AndChat-620489 (~AndChat62@67-130-74-235.dia.static.qwest.net)
21:45.34*** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net)
21:45.55MiccIs there a way to run some dialplan code whenever someone is placed on hold and picked up from hold?
21:46.11fileno
21:46.12MiccI'd like to avoid using AMI if at all possible.
21:46.25Miccok, good to know.
21:47.55*** join/#asterisk pruonckk (~pruonckk@135-143-11-177.raimax.com.br)
21:48.02MiccI guess I'll need to make an AMI proxy/aggregator since I may have multiple apps needing to get AMI events, no reason for asterisk to have to do all that work for multiple connections. Is there anything like that in open source project?
21:53.55KobazMicc: i have a patch that adds hold unhold events
21:54.05Kobazyou would need ami
21:54.10Kobazor stasis or whatever
21:54.20Kobazyou can't run dialplan nicely because it takes over the channel... ie, no audio bridging
21:54.30Kobazwhile your dialplan is running, no one can talk
21:54.44Kobaz(even if you *could* run dialplan on hold/unhold)
21:58.25*** join/#asterisk nix8n82 (~AndChat62@2600:100e:b012:63f2:75e4:7436:4526:1f0b)
21:59.05*** join/#asterisk AndChat|620489 (~AndChat62@67-130-74-235.dia.static.qwest.net)
21:59.42*** join/#asterisk jeffspeff (~Jeff@12.49.160.131)
22:00.06kai[El]hey jeff
22:01.18*** join/#asterisk SSlater (~simon@mail.favour.com.au)
22:14.57*** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net)
22:18.40*** join/#asterisk pruonckk (~pruonckk@135-143-11-177.raimax.com.br)
22:26.02*** join/#asterisk AndChat|620489 (~AndChat62@2600:100e:b010:8dc6:b94b:9626:fdff:aa5e)
22:33.24*** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net)
22:42.49*** join/#asterisk kharwell (kharwell@nat/digium/x-iovxeshrqaplebme)
22:42.49*** mode/#asterisk [+o kharwell] by ChanServ
22:47.14*** join/#asterisk JohnWigley (~JohnWigle@johnwigley.plus.com)
22:51.17*** join/#asterisk rwb (~Thunderbi@65.183.151.121)
23:05.32*** join/#asterisk Alex_Bkash (84cde551@gateway/web/freenode/ip.132.205.229.81)
23:13.21*** join/#asterisk AndChat|620489 (~AndChat62@2600:100e:b010:8dc6:2c2b:d985:fa15:6f42)
23:22.20*** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com)
23:25.29*** join/#asterisk nix8n82 (~AndChat62@67-130-74-235.dia.static.qwest.net)
23:33.46*** join/#asterisk paulgrmn_ (~paulgrmn@184.75.212.68)
23:35.00*** join/#asterisk AndChat|620489 (~AndChat62@67-130-74-235.dia.static.qwest.net)
23:39.06*** join/#asterisk pruonckk (~pruonckk@135-143-11-177.raimax.com.br)
23:42.00*** join/#asterisk freebs (freebs@gateway/vpn/privateinternetaccess/freebs)

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.