IRC log for #asterisk on 20150923

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09:17.39EncryptHEllo o/
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09:54.26EncryptI still have this question...
09:54.54*** part/#asterisk xpheres (~bodhi64@dslb-088-073-000-038.088.073.pools.vodafone-ip.de)
09:55.00EncryptIs it possible to connect to my remote asterisk server which is behind a NAT as well as I am behin a NAT?
09:55.16EncryptWith a kind of VPN technlogy
09:55.34EncryptI thought about doing an SSH tunnel
09:55.49EncryptBut since SSH is based on TCP, it is not possibe
09:55.52Encryptpossible*
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10:42.17catphishi seem to be experiencing a nasty crash, can anyone have a quick look and point me in the direction of what might be causing this? http://paste.codebasehq.com/pastes/hys0ngkntdml2lx1j0
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10:49.16catphishi've unregistered statis, i hope this helps
10:52.22catphishi think using module autoload was a mistake
10:52.55catphishmy realtime problems yesterday that i thought were caused by "sip reload", were actually caused by the whole app crashing :(
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13:11.16twanny796how can i determine a  .bin sound file format?
13:11.41WIMPyTry file. Maybe you're lucky.
13:12.11twanny796hello WIMPy ; file just gave me data />
13:12.24WIMPyBad luck then.
13:12.50WIMPyIf you have no other clues, you will have to experiment.
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13:13.46twanny796WIMPy: experiment with what?
13:14.12WIMPyverious formats until it sounds like something.
13:14.34WIMPyThat requires some luck, off course.
13:16.03twanny796WIMPy: with which software?
13:16.29WIMPyI'd try sox or maybe audacity.
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13:20.27cw1972open it in a hex editor, there may be a header to identify the format
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13:21.02twanny796cw1972: trying
13:21.33Reapsterheya, if I delete endpoint, aor and auth via pjsip dynamic config (/ari/asterisk/config/dynamic/res_pjsip/{endpoint,aor,auth}), the endpoint still shows up in /ari/endpoints, is there some other way to really delete an endpoint?
13:22.20filewhere are they stored?
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13:23.05Reapsterwhatever the default is, I haven't set up the sorcery stuff yet, so in memory?
13:23.22filethe only default is to pull from config file
13:23.27filewhich is read-only
13:23.34twanny796cw1972: excuse my ignorance which hex editor in linux?
13:25.17ReapsterI'll configure sorcery for it and check again then, restarting asterisk got rid of everything
13:25.20Reapsterthanks file
13:26.21fileif you have stuff configured properly so it has a real place to go and deleting it doesn't completely make it go again, then I'd file a bug - there may be a lifecycle problem with the internal Asterisk endpoint and the PJSIP endpoint where one disappears and the other doesn't
13:26.28fileit wasn't written with push in mind originally
13:26.44filethe end result being it may still appear in /ari/endpoints but the PJSIP endpoint would be gone
13:27.23twanny796cw1972: HARM_52 ???
13:29.12Reapsteroh ok, thanks for the heads up
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13:34.46Zogotis it right that you get 2 PeerStatus events in the AMI? Asterisk 13.3
13:35.36WIMPyYou probably get a lot. For what exactely?
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13:37.26gonzapatahello everybody
13:38.05ZogotWIMPy: for the same peer, i should of clarified further
13:38.16gonzapataI need some help with libss7 I have tried several asterisk version dahdi etc and asterisk crash
13:38.50WIMPyZogot: The question remains: For what exact occasion?
13:39.46Zogotwhen a phone unregisters/registers, i get the exact same event for peerStatus twice
13:39.58Zogothttps://gist.github.com/zogot/9aee2e2ecfacfe8a3375
13:41.19WIMPyI can't look at that, but if it's only one account registering, I'd expect only one event.
13:41.38ZogotWIMPy: same :p lol
13:41.45Zogoti get only 1 for the others
13:41.48ZogotDeviceStateChange and
13:41.52ZogotChallengeSent
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13:44.10catphishi seem to be experiencing a nasty crash, can anyone have a quick look and point me in the direction of what might be causing this? http://paste.codebasehq.com/pastes/hys0ngkntdml2lx1j0
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13:44.29WIMPyOh, is chan_pjsip that verbose? Anyway, file may be able to tell you more if he has the chance to spot it.
13:44.45catphishi've taken the step of removing my statis module, i hope this will prevent the problem for now
13:45.01WIMPycatphish: Strings with embennded newlines?
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13:45.41catphishWIMPy: i have no idea where the invalid strings are coming from, or what's trying to json encode them :(
13:46.19WIMPyAnd it's killing the whole process?
13:46.21fileZogot, https://issues.asterisk.org/jira/browse/ASTERISK-25116
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13:46.28catphishWIMPy: yes :(
13:46.43WIMPyThat's very evil.
13:47.11catphishmy verbosity was not very high, that paste is all the info i have
13:47.24Zogotfile: hmm thanks for the link. I am currently on chan_sip, so perhaps someone should comment it has issues in that also
13:47.46ZogotI could also make that comment :p
13:47.52fileyou could.
13:48.02catphishi've decided to stay on chan_sip and realtime for now, i realised the reason i was losing my peers was purely because asterisk was crashing :(
13:48.04seik0some says Hangup() in h is not very good (undefined, if strictly), but what is the best way just to stop working in h-extension ?
13:48.58seik0because i have "_." exten in same context and must define special extens to not be overwritten by "_."
13:49.11catphishthe ... in my paste isn't actually actually where lines have been removed, it literally restarts immediately after that JSON error :(
13:49.29WIMPyOh, chan_sip generates a ChallengeSent event? Never seen that.
13:50.39WIMPyseik0: The best way to use a _. is if you are able to Goto() another context where you don't have that pattern so the special extensions work.
13:51.28seik0WIMPy: even so first step with undefined "h" exten will make the same goto
13:51.58WIMPyYes, but you won't have anythign happening at that point, yet.
13:52.48seik0doesn't make any sense to me
13:52.58WIMPyYou won't get to the h extension for a call that hasn't been answered.
13:55.46seik0but if I really need "_." exten and then make Goto() to Some-Context, that  Some-Context also should  contain "_." exten and I will face the same problem
13:56.06[TK]D-Fenderit should NOT contain that pattern
13:56.17WIMPyThat depends on what you want to do.
13:56.18[TK]D-Fenderjump OUT of the context that has it
13:57.02seik0but it can only be achieved by excluding "_." from base context
13:57.41WIMPyIf you do a databse lookup for ${EXTEN} ot such, you could still Goto(SomeOtherContext,AA${EXTEN},1) and lookup ${EXTEN:2} in '_AA.' and would be safe,
13:57.46seik0so there should be one place where need for "_." should be resolved
13:58.02seik0and in that place I may problably face h-problem
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13:58.43WIMPyAnd y ou can still have a h extension in the same context, off course.
13:58.48seik0WIMPy: AA${EXTEN} looks a good workaround
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13:59.14seik0thanks
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14:31.16seik0surprisingly magical, but when Originating (say,  with call-file) to Local  channel , variables in call-file set with Set exist in both legs of Originate
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14:44.30[red]so in asterisk 13, are there any different settings for getting voicemail odbc storage working from 11? i'm pretty confident i have everything right, but asterisk keeps saving voicemails to /var/spool instead of the DB
14:44.39[red]no error messages generated by asterisk
14:45.11[TK]D-Fender* will only load one module to manager voicemail.  Apparently you re letting it load the statick file storage one.
14:45.43[red]hmmm, good point
14:45.48[red]i'll give that a try
14:45.54[TK]D-Fendermodules.conf <-
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14:51.17[red]any idea of what the module name is? i'm not finding it with CLI> module show
14:52.37dajaz27What does chan_sip.c:15307 -- Registration for sip provider timeout mean
14:52.40[TK]D-Fenderlook in your modules folder
14:52.54fileAsterisk doesn't have separate modules for voicemail by default, it's chosen at compile time in menuselect under the "Voicemail Build Options"
14:53.04[TK]D-Fenderdajaz27: Means * didn't get an answer.  You should be looking at what it requested and where it sent that request
14:53.25[TK]D-Fenderfile: Used to have multiple .so's , when did that change?
14:53.34filenever? it has always been like this
14:53.46filethere's outside patches to build separate ones, but afaik noone has ever submitted them
14:53.47[TK]D-Fenderrecalls seeing a piel before...
14:53.53[TK]D-Fenderpile*
14:54.56fileor they just do it as part of the build process
14:55.00dajaz27Thank you.
14:55.57[red]thanks guys, it was the compiler flag
14:56.05[red]feels like he should have known this
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15:01.28[red]a warning if you set the odbcstorage= config in voicemail.conf and have directory storage compiled would be super nice :)
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15:58.01SpeakerToMeatI'll add an extra question, what's a good brand and/or model of sip phone that is not too expensive (like big brands), but has good build quality and software in general?
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15:58.21[TK]D-FenderPolycom > All
15:58.43[TK]D-FenderAastra runs 2nd place, 3rd starts tying up with Yealink, Digium, Snom, etc
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15:59.37SpeakerToMeatNormally I wouldn't consider Polycom on the "not too expensive" realm. But maybe I'm mistaken?
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16:02.02[TK]D-FenderDepends where you are
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16:02.33[TK]D-FenderIf you're in North America then there is no reason to cheap out for anything else
16:02.45SpeakerToMeatNot in north america
16:03.05[TK]D-FenderIt then comes down to features as to whether Aastra has a partuculr model that is slightly better suited to your intended usage
16:03.25[TK]D-FenderCisco's SPA series are "good enough" as well.
16:03.45[TK]D-FenderAnd are generally a top value/$ elsewhere
16:06.35igcewieling1Has anyone ever solved an issue where Polycom phones seemingly randomly lose the contact directory or revert the contact directory to a previous version.   Seems to happen most often on firmware upgrades, but not always.
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16:17.20SpeakerToMeatI guess you guys deal with all sorts of users.... have you found good methods, or low resistance methods of allowing users to manage their own speed dial and/or manage the directory?
16:18.06WIMPyalways thought the phones do that.
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16:25.08[TK]D-FenderThey do.
16:25.20[TK]D-Fenderby default all phone directories are controlled by the phone itself
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16:28.26igcewieling1by default the phone uploads its contacts to MAC-directory.xml
16:34.04SpeakerToMeatOk, thanks
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17:23.21[red]igcewieling1: yes. you need to generate a directory listing for each polycom phone. the polycom phones only load the default diretory once if a phone specific directory doesn't exist
17:23.29[red]i made a script to help with this issue
17:23.35[red]at least for freepbx
17:24.09[red]https://git.talkroute.com/red/freepbx-polycom-directory/blob/master/updateContactDirectory.sh
17:27.21igcewieling1[red]: I am familiar with creating phone specific contact directories.  I'm not sure how that script would help with the contacts the person put in their contact directory.
17:27.33[red]oh gotcha
17:28.05[red]yeah, my experience with polycom is that either you can have contacts managed by a central server or manged by the end user
17:28.09[red]both doesn't work out well
17:28.22igcewieling1[red]: contacts are managed by the local user.
17:28.40[red]and it's still being overwritten/lost?
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17:29.23igcewieling1[red]: in VERY recent Polycom firmwares you can have a real global directories, see the release notes for the 5.x or 5.1.x firmwares.
17:29.47igcewieling1[red]: yes.  seemingly randomly, but might happen most often during firmware upgrades.
17:29.59[red]oh gotcha, sorry i haven't worked with the newer firmwares yet, don't think i may be of much help
17:30.30igcewieling1Had a customer complaining today that the 90 contacts they had put into their phone were all gone this morning.     Happens once a month or so.
17:31.17igcewieling1Once a month or so across the 1,000+ phones I manage.
17:31.23[red]ouch
17:32.23igcewieling1not all 1,000 phones, just one or two out of the 1,000.
17:33.13igcewieling1I suspect the "correct" solution is to set up an ldap server at each client.
17:33.53[red]that, or, maybe even a quick web script where folks can manage their contacts through a web interface
17:34.03[red]and then that generates the xml files for the phones
17:44.12igcewieling1[red]: heh, we did that about 5 years ago.  Doesn't work nearly as well as it sounds.
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18:31.57zopsiIs there any reason that explains upon reboot asterisk isn't recognizing my G.729 and DPMA licenses in /var/lib/asterisk/licenses? It is owned by asterisk:asterisk.
18:32.53TazzNZzopsi, what do you have to do to get it to work ?
18:33.41zopsiTazzNZ: what do you mean? They were being recognized fine before.
18:33.58zopsiTazzNZ: currently it does not work
18:34.16TazzNZright - so it was working, then your rebooted, and now it doesn't work at all ?
18:34.34TazzNZ(well DPMA and g729 :) )
18:34.35zopsiwell I can start asterisk and everything is fine except the licensed modules
18:34.45TazzNZright
18:34.47zopsiI'm on 13.5
18:35.03TazzNZif you open the asterisk cli
18:35.41zopsiit is open
18:36.13TazzNZand type module show like 729
18:36.17TazzNZwhat does it show ?
18:36.42zopsicodec_g729a.so                 Digium G.729 Annex A Codec (optimized fo 0          Running              core
18:37.23TazzNZand repeat that with "digi" instead of 729
18:38.46zopsihttps://pastebin.com/uhGwEsYA
18:39.14TazzNZso why do you think it's not working ?
18:39.26zopsiit doesn't detect the license and thus doesn't function
18:40.06zopsihttps://pastebin.com/40nNSrqT
18:40.26zopsiI've confirmed my two licenses are in /var/lib/asterisk/modules and accessible by asterisk user
18:40.28TazzNZunload and reload the module
18:40.30TazzNZmodule unload res_digium_phone.so
18:40.31TazzNZand
18:40.35TazzNZmodule load res_digium_phone.so
18:40.48TazzNZpastebin the load please
18:41.12zopsihm
18:41.37zopsihttps://pastebin.com/kVet9VgG
18:42.45TazzNZmodule reload res_digium_phone.so
18:42.50TazzNZthat might do it too
18:43.25zopsihttps://pastebin.com/nttNYM6g
18:44.38TazzNZwhat does "/var/lib/asterisk/licenses" show?
18:45.18zopsia ls or in cli?
18:45.28TazzNZwhoops
18:45.30TazzNZyes - ls
18:45.30TazzNZ:D
18:46.17zopsihttps://pastebin.com/DKDnG6q3
18:49.24TazzNZthe only digg's on mine is that I don't have group write
18:49.35TazzNZI am not sure that will change anything
18:49.41TazzNZbut let's try it
18:50.14zopsisure
18:50.37TazzNZchmod 644 -v /var/lib/asterisk/licenses/
18:50.51TazzNZchmod 644 -v /var/lib/asterisk/licenses/*
18:50.54TazzNZforgot the *
18:51.41zopsistill problem
18:51.50zopsihow can I tell if my hostid changed?
18:52.00TazzNZwhen you reload it still can't find it ?
18:52.16TazzNZwhy do you suspect your hostid has changed ?
18:52.17zopsiyeah still nothing
18:53.09zopsino reason I didn't change hardware or anything but I'm just trying to think of things
18:53.55TazzNZit won't "just change"
18:55.32zopsiwell apparently it did...
18:55.38zopsihow is the host-id calculated
18:56.00zopsimy g729 license that was working yesterday has a different host ID than the one that shows when reloading the module.
18:56.31TazzNZIn general, the host ID is not allowed to change on a given host.  Since the host ID is based on hardware fingerprinting, Digium acknowledges that sometimes hardware and systems fail, so it is necessary to move the keys to new hardware
18:56.36igcewieling1zopsi: based on the MAC addresses on the system.
18:56.42igcewieling1I don't know any more than that.
18:56.58zopsiliterally have no changed anything hardware on my server.
18:57.02zopsinot*
18:57.16TazzNZhttp://kb.digium.com/articles/FAQ/Introduction-to-Digium-Asterisk-Module-Licensing
19:01.53zopsiI have no idea how that happned, but I used my last registration on my g729 and DPMA and it fixed it
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19:32.16c|onemanwireshark playback of calls crash for anyone else?
19:32.31c|onemanclick on the waveform and *poof*
19:44.26SpeakerToMeatIs there a set out there of CC licensed basic default system recordings?
19:45.07SpeakerToMeatWell... actually, freepbx seems to come with a set, only, I can't hear them :/ I'll keep investigating
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23:42.52zopsion inbound calls I get a bunch of "res_pjsip/pjsip_distributor.c:221 find_dialog: Could not find matching INVITE transaction for CANCEL request" as well as 9 inbound calls for the same instance
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