IRC log for #asterisk on 20170517

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00:22.59*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); 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
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01:20.10salviadudI'm trying to debug a cisco phone I converted to sip
01:20.37salviadudI can't seem to register it, I am following pretty good instructions because the phone does try to register, but I get a bunch of 401 errors.
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10:44.37RomanDcozHello All, I am just streaming audio file using "$agi->stream_file("1495017496-1574156409", '#');" but file not playing and direct hangup the call.
10:44.37RomanDcozHere is the logs : https://pastebin.com/qe39D9fD
10:44.47RomanDcozCan any one please guide me ? I am stuck here.
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11:08.00Dirk23Hi. is it possible (for monitoring) to log in to asterisk cli, send sip show peers (parse the output) and exit cli?
11:10.50Dirk23with a script
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11:19.27dadrcasterisk -rx 'sip show peers'
11:20.36dadrcDirk23: ↑
11:20.53Dirk23jap, found that one too. Thnx
11:21.47Martin`ah that is easy :D
11:21.58Dirk23indeed
11:24.51RomanDcoz<PROTECTED>
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11:27.18Martin`Hm don't know, never did that before with an agi
11:31.48RomanDcozThanks. If any one facing same kind of issue then please suggest me.
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12:45.16paule32hello
12:45.25paule32i would like to implement an echo test
12:45.29paule32i get this:
12:45.35paule32[May 17 14:41:56] WARNING[24075][C-00000000]: file.c:701 ast_openstream_full: File /var/lib/asterisk/sounds/de/demo-echotest.gsm does not exist in any format
12:45.35paule32[May 17 14:41:56] WARNING[24075][C-00000000
12:45.54paule32but the file exists
12:46.30file<PROTECTED>
12:46.52paule32i add gsm
12:46.59paule32after filename
12:47.03filedon't
12:47.13fileAsterisk will automatically find the best file it can
12:47.16paule32exten => 81,3,Playback(/var/lib/asterisk/sounds/de/demo-echotest.gsm)
12:47.55paule32when i use:
12:47.55paule32exten => 81,3,Playback(demo-echotest)
12:48.00paule32the same error
12:48.06paule32file not found
12:48.19fileremove the ".gsm" from your first one, do a dialplan reload, and try again
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12:49.12paule32http://codepad.org/FNrI276v
12:49.23paule32this was the original
12:51.03paule32when reload in cli
12:51.06paule32[May 17 14:50:30] NOTICE[24057][C-00000001]: chan_sip.c:25450 handle_request_invite: Call from '' (77.64.233.141:5061) to extension '81' rejected because extension not found in context 'default'.
12:52.52fileis the extension '81' reachable from the context 'default'?
12:55.21*** join/#asterisk drathir (~kamiljk8@unaffiliated/drathir)
12:56.01[TK]D-FenderClearly not...
12:56.40[TK]D-Fenderpaule32> exten => 81,3,Playback(/var/lib/asterisk/sounds/de/demo-echotest.gsm) <- here you specified the full path, but included the extension which you cannot do.
12:57.04[TK]D-Fender<paule32> exten => 81,3,Playback(demo-echotest) <- Here you removed the extension ... AND the full path
12:57.27[TK]D-FenderDon't pollute your tests by changing *2* things at a time.
12:58.45paule32http://codepad.org/nFYA0hK2
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12:59.40[TK]D-Fender"core set verbose 10" <- we aren't seeing your dialplan execution
13:00.06[TK]D-Fenderand your earlier sample seemed to imply it's in a DE folder but we have no proof what the current language is
13:00.20paule32german
13:00.24[TK]D-FenderAnd we don't see the files themselves
13:00.33[TK]D-FenderI don't see PROOF that the language is set <-
13:00.52paule32http://codepad.org/kahl7pb0
13:01.04[TK]D-FenderShow a new call with the folder dump
13:01.25[TK]D-Fenderexten => _3X,n,Dial,SIP/${EXTEN}|55|Ttr <- no good
13:01.32paule32http://codepad.org/AVIUZG4H
13:01.53[TK]D-Fender| = invalid, and your app data should be in (), not separated by a comma
13:02.29*** join/#asterisk drathir (~kamiljk8@unaffiliated/drathir)
13:02.35[TK]D-FenderShow the new call with proper verbose and the sound folder "ls -la" dump
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13:03.34paule32http://codepad.org/LfQB0elM
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13:04.55[TK]D-Fenderwaits for the other half
13:07.50paule32http://codepad.org/q7P8chGF
13:08.43[TK]D-FenderFix your ownerships <-
13:09.14[TK]D-Fender75% = ROOT
13:09.19paule32the Dial plan,  instead |  a comma?
13:09.45[TK]D-Fender[TK]D-Fender> exten => _3X,n,Dial,SIP/${EXTEN}|55|Ttr <- no good
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13:10.05[TK]D-Fenderexten => _3X,n,Dial(SIP/${EXTEN},55,Ttr) <- good
13:10.33paule32thank you
13:10.54paule32and context ankommend, same?
13:11.15paule32exten => 621,1,Dial,SIP/30|30|r
13:11.23[TK]D-FenderSame sort of thing
13:11.44paule32is that old convention (the parts from me) ?
13:12.04[TK]D-Fenderyes
13:12.07[TK]D-FenderVERY old
13:12.20[TK]D-Fender10 years...
13:12.24paule32uh
13:12.35paule32and _3X ..
13:13.04[TK]D-FenderWhat about it?
13:13.22paule32other convention, or correct
13:13.45[TK]D-Fender?
13:13.57paule32what means _
13:14.12[TK]D-Fender~book
13:14.12infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
13:14.21[TK]D-FenderRead up on your dialplan basics
13:14.30paule32thank you
13:14.35[TK]D-Fenderpattern <-
13:14.53paule32i have no isdn, a dsl phone line
13:15.30paule32can i do my self outgoing calls to other pbx - in other region
13:15.44paule32the good old telephony
13:16.04paule32i have a 56k modem card, is it enough if possible?
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13:16.45[TK]D-Fenderno
13:17.38[TK]D-FenderDAHDI supports a very specific set of chipsets for analog interfaces.  Generic modems are no good
13:17.51paule32i have an very old book here, that describes that is possible to connect with other pbx
13:18.06*** join/#asterisk brad_mssw (~brad@66.129.88.50)
13:18.45paule32and now, i read here, isdn
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13:19.12[TK]D-FenderNot with a cheap modem
13:20.09[TK]D-FenderThere are several choices for analog, BRI, PRI, CAS, etc
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13:28.03paule32http://codepad.org/a5tmMyHG
13:28.10paule32something wrong
13:28.30paule32with wich number, i have to log in?
13:28.36paule32i get only
13:28.47paule32QouteCom
13:29.04paule32woops i mean, i get online at home server
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13:29.38paule32and get the messages above
13:30.01[TK]D-Fender?
13:30.05paule32http://codepad.org/a5tmMyHG
13:30.11paule32i call 81
13:30.20paule32and get the message in cli
13:30.21paule32http://codepad.org/a5tmMyHG
13:30.26[TK]D-Fender[May 17 15:26:39] NOTICE[24363][C-00000003]: chan_sip.c:25450 handle_request_invite: Call from '621' (77.64.233.141:5061) to extension '81' rejected because extension not found in context 'default'.
13:30.30[TK]D-FenderMeans what it says
13:30.38[TK]D-FenderThere is no match for 81 in [default]
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13:30.44[TK]D-Fenderthere is not hing to guess about
13:30.55[TK]D-Fenderit is telling exactly what was requested, and where it is looking for it
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13:31.39paule32[default]
13:31.39paule32include => lokal
13:31.39paule32include => echotest
13:31.46[TK]D-Fenderhttp://codepad.org/kahl7pb0
13:31.49[TK]D-FenderTHIS showed 81
13:32.07[TK]D-Fenderhttp://codepad.org/a5tmMyHG <- Here is see ***600***
13:32.12[TK]D-FenderThese are NOT the same at all
13:32.50paule32so far i read is 600 default by standard installation
13:32.54paule32that i have
13:33.14paule32the standard number of echotest
13:33.14[TK]D-FenderYou showed 1 part of code, and the dialplan execution does NOT match
13:33.21paule32woops
13:33.22paule32sorry
13:33.55paule32http://codepad.org/PGaH0EAa
13:35.17[TK]D-Fender"dialplan show" <-
13:36.07[TK]D-FenderAnd your CLI output from before STILL shows it executing a 600... which is NOT in your dialplan bits you've been showing us
13:36.18[TK]D-FenderCode doesn't come out of thing air...
13:37.15paule32http://codepad.org/KOpLxTVW
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13:38.19[TK]D-Fender[ Context 'default' created by 'pbx_lua' ]
13:38.19[TK]D-Fender<PROTECTED>
13:38.19[TK]D-Fender<PROTECTED>
13:38.30[TK]D-FenderGet rid of LUA & AEL
13:38.59paule32that not my config
13:39.03paule32that is default
13:39.05[TK]D-Fenderit clearly blew away your default context
13:39.09[TK]D-FenderYou left that there
13:39.11[TK]D-Fenderit is screwing you
13:39.13[TK]D-Fenderremove it
13:39.16paule32ok
13:39.22paule32moment please
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14:24.56paule32jens@rechner ~ $ sudo asterisk -r
14:24.56paule32Privilege escalation protection disabled!
14:24.56paule32See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
14:24.56paule32Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
14:25.07paule32i delete the .ctl file
14:25.14paule32but asterisk will not start
14:25.33[TK]D-Fenderit is complaining that it does nto exist...
14:25.38[TK]D-Fenderand you tell us you DELETED IT
14:25.42[TK]D-FenderDo you see a problem with this?
14:25.48paule32hehe
14:25.49paule32ook
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14:26.52paule32rechner jens # touch /var/run/asterisk/asterisk.ctl
14:26.53paule32rechner jens # asterisk -r
14:26.57[TK]D-FenderSo We've gone though involdali dialplan.  broken permissions.  Other configs overriding your evolving dialplan, and NONE of these things have been shown to have been fixed.  And now you're breaking new things
14:27.12[TK]D-FenderYou can't just TOUCH the file
14:27.21[TK]D-Fenderit is supopsed to hold the PID for the running * process <-
14:27.49paule32yeah
14:27.55paule32how to start asterisk?
14:28.05paule32service asterisk start ?
14:28.24[TK]D-Fenderprobably
14:28.32[TK]D-Fenderdepending on your install method & distro
14:28.39paule32linux mint
14:28.42paule32same problem
14:28.52[TK]D-Fenderof course that assumes it isn't already running
14:29.18[TK]D-FenderAnd by douing "touch" like that... guess who the OWNER is?
14:29.24[TK]D-FenderYou keep screwing things with permissions....
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14:58.04igcewielingI miss the days of interesting questions.
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15:03.29SamotWell I just got one.
15:03.52SamotSomeone just emailed me about helping them with their 3rd party Node.js SIP client..
15:04.20SamotThat uses SUBSCRIBE/NOTIFY for CallerID. Which is odd, to say the least.
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16:25.44DanQuinneyha Samot, I hope you told them where to go?
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16:36.43*** join/#asterisk carlosdienstmann (~t7DS@50.144.20.201.rev.defferrari.com.br)
16:40.31carlosdienstmannhello guys, i'm suffering with an issue for some days, trying to write the CEL events on mysql by odbc. In fact i had success to write the events.. but they going withou columns expec.. like        > [INSERT INTO cel () VALUES ()]
16:42.22carlosdienstmannanyone knows where I had to define the values of the insert ? on cel.conf i defined events=APP_START,CHAN_START,CHAN_END,ANSWER,HANGUP,BRIDGE_START,BRIDGE_END,LINKEDID_END
16:42.41carlosdienstmannfor master.csv it worked nice
16:46.49paule32hello
16:47.16paule32how can i remove the items, so only my own rules are loaded:
16:47.17paule32http://codepad.org/EafFDaNT
16:47.18paule32?
16:47.57[TK]D-FenderStop loading those modules or trash their configs
16:48.04[TK]D-Fender[pbx_ael
16:49.24[TK]D-FenderAnd I no longer see your extensions.conf in there at all
16:58.34paule32[TK]D-Fender: i have download the latest version
16:58.42paule32compile and install it
16:58.49[TK]D-Fender...
16:58.50paule32no more old stuff
16:59.39[TK]D-FenderThen you've wasted out time in the earlier debugging.
16:59.57[TK]D-FenderYou were probably ONE step away from getting things working and the you started to BREAk everything and then trashed the whole thing
17:00.02[TK]D-FenderAnd are making this a guessing game
17:09.10paule32i have a connection now, and echotest work
17:09.46paule32[ Context 'parkedcalls' created by 'res_parking/default' ]
17:09.46paule32<PROTECTED>
17:09.53paule32in which file is that ?
17:10.58igcewielingpaule32: that is one of the contexts which are more trouble than it is worth.
17:11.55igcewielingpaule32: features.conf
17:12.23[TK]D-FenderI fail to see a PROBLEM having this
17:12.45[TK]D-Fender<igcewieling> paule32: features.conf <- not in modern *
17:12.59[TK]D-Fender#preoccupiedwith1985
17:13.14igcewieling[TK]D-Fender: interesting.  What config file it is it then?
17:15.40igcewielingActually, I'm stuck in 2014-10-23, the day before Asterisk 13 was released.
17:15.59igcewielingI suppose that would not make as interesting a song title.
17:16.21igcewielingLong live Asterisk 11!
17:17.29SamotWell at least until Oct 25th.
17:18.24paule32thanks
17:18.29paule32you are super heros
17:19.03paule32is it possible to call mobile phones?
17:20.24paule32also i can't see contexts 700 in the conf
17:20.29paule32features.conf
17:20.36paule32it is a file from the compilation
17:20.43paule32by "make samples"
17:20.53paule32all text is commented
17:21.21Samotfeatures.conf only plays a part of Park if you want a feature code to do the transfer for you to a slot
17:21.33SamotOtherwise you transfer the call to the lot or a specific slot.
17:21.57paule32yes, but it is default sample text
17:22.07paule32so i think i can delete the file
17:22.14SamotWhy?
17:22.27SamotAre you saying you don't want feature codes?
17:22.33SamotOr that you don't want Parking?
17:22.52paule32i would like learn the procedures of asterisk
17:23.06paule32so, the basics are echotest
17:23.06SamotSo then deleting them would be a waste.
17:23.17SamotBecause you can't learn about them if they aren't there.
17:25.25paule32yeah, i delete the files from older version, and then, i could not start asterisk anymore , therefore i was a little time idle here, because i download and compile the latest version, and i saw that the sip and extension,conf from older version work on the new version - with the changes that TK wrote here few hour's ago
17:26.40SamotSettings really haven't changed much over the releases.
17:26.51SamotNew features/settings have been added..
17:27.06SamotBut even deprecated settings still function.
17:27.17paule32yeah
17:28.21paule32now, i have the source code, and i am not have to deal with old repos
17:28.35paule32i work under linux, but lazy sysadmin
17:28.37paule32:-)
17:29.13Samotfile: So 11.25.1 is probably going to be the final release of 11.x unless something requires a huge security fix before Oct 25th, I'm guessing.
17:29.25fileyes.
17:29.36SamotOK.
17:32.38igcewielingAsterisk 11 will not suddenly break when support expires.
17:32.57fileyes, and no
17:33.10fileas newer compiler releases occur older versions can sometimes no longer build
17:35.04SamotI'm not sure SFO counts as "support"
17:35.22SamotBut yeah, 11 isn't just going to go belly up.
17:36.19rmudgettThat's why people still run DOS 3.3
17:36.31SamotOr Asterisk 1.8
17:36.33SamotOr 1.4
17:36.40fileor why rmudgett runs antiquated versions of Ubuntu
17:36.45file:P
17:36.47paule32[ Context 'parkedcalls' created by 'res_parking/default' ]
17:37.05paule32is that context automatically created?
17:37.25*** join/#asterisk matrix1233 (~matrix123@41.230.44.234)
17:37.30paule32i wipe data, reload asterisk, and can't see progress
17:38.23igcewielingfile: that is an edge case and I doubt it will happen for a while, especially since we are not upgrading the OS so only minor updates to things like compilers.
17:38.30paule32file: you don't like ubuntu?
17:38.41filepaule32: I use Ubuntu
17:38.55paule32o, sorry misread
17:38.56filermudgett just holds on to the version he is currently on for far too long
17:39.11igcewielingpaule32: yes, it is automatically created
17:39.11paule32have read the debian info
17:39.19*** join/#asterisk Bud82 (5ed33bfe@gateway/web/freenode/ip.94.211.59.254)
17:39.24igcewielingWhy are you still wasting your time on this?
17:39.43paule32with wipe data?
17:39.52igcewielingwith trying to remove that context.
17:40.17paule32ah, ok, i had thought that is defined in a config
17:40.32SamotThe wiki and the sample files tell you
17:40.41SamotThe the partext is created automatically.
17:41.00igcewielingres_parking is tied into many many parts of Asterisk.   If you are trying to reduce the number of configs or modules, res_parking is the LAST thing you should try that with.
17:41.16[TK]D-Fender<Samot> I'm not sure SFO counts as "support" <- Seriously Fucked Over :)
17:41.33paule32oh
17:41.44[TK]D-Fender<paule32> is that context automatically created? <- yes
17:41.50[TK]D-Fenderbecause of your parking config
17:42.02[TK]D-Fender<paule32> ah, ok, i had thought that is defined in a config <- it is
17:42.38SamotBetween the sample files and the wiki, these questions can be answered.
17:42.47igcewielingHere is my modules.conf, you'll need to modify it to suit your needs.
17:42.50igcewielinghttps://pastebin.com/8zbJ43WU
17:43.10igcewielingcomment out any odbc stuff first
17:43.34Bud82Hello all! Since 14.4.0 it seems that Asterisk no longer uses the external_media_address to correctly set the IP when run behind NAT with pjsip. There is a change ( https://issues.asterisk.org/jira/browse/ASTERISK-26879 ) that says if local_net is not defined, ignore the external_media_address. But local_net -is- defined.
17:44.06Bud82It has worked perfectly in 14.3.0
17:46.00filethat's not actually what it says, it makes the code skip the localnet check if localnet is not defined
17:46.59fileBud82: I haven't seen any issue reports about that - and we've got quite a few users who do that, so I'd suggest filing an issue report with the full SIP log (pjsip set logger on) and configuration
17:47.00Bud82When the call comes in, no audio is heard (from Asterisk) because Asterisk asks to connect to a local IP for RTP.
17:47.07SamotBud82: Show an internal call using the external_media_address in it.
17:47.19Bud82Internal (on the same LAN) it works.
17:47.27SamotOK
17:47.31SamotLet me rephrase then.
17:47.40fileyou can also provide it here and Samot or others can take a look before filing
17:47.42filegoes for dog walk
17:47.51Bud82Can I provide it in private ?
17:47.54SamotBud82: Show a call that is using the wrong media IPs.
17:47.58Bud82Sure
17:48.01Samot~pb
17:48.01infobothmm... 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.
17:48.30SamotNo PMs
17:48.37SamotIn the channel
17:49.26*** join/#asterisk jetlag (~jetlag@c-71-226-222-56.hsd1.nj.comcast.net)
17:50.18SamotBud82: I said no PMs
17:50.23SamotAnd to use pastebin.
17:50.28SamotPlease follow instructions.
17:50.30Bud82sorry, I'll check that
17:51.42Bud82http://pastebin.ca/3813826
17:52.41Bud82The SIP messages use the correct IP's, the RTP stream are announced with the wrong IP's
17:53.08Samot<--- Received SIP request (765 bytes) from UDP:94.211.59.254:54291 --->
17:53.13Bud82(the internal ones I'd like to rewrite using external_media_address, as works OK in 14.3.0)
17:53.26SamotContact: <sip:7001@94.211.59.254:54291;transport=UDP>
17:53.32Samot^^ So 7000 is a local extension?
17:53.37Bud82yes
17:53.45fileYou need to provide the config too
17:53.51Samot^^^^
17:54.04Bud82the transports part ?
17:54.12SamotBecause 7001 is making the call from an External IP
17:54.15igcewielingsip.conf of course.
17:54.20Samotso why is the phone sending an External IP?
17:54.24SamotWell pjsip.conf
17:54.26igcewielingor whatever config file pjsip uses.
17:54.30Bud82because it's not on the same network
17:54.36Samot......
17:54.37SamotWhat
17:54.49Bud82it's an android phone connected over the internet to the Asterisk PBX
17:54.53SamotOK
17:54.56SamotSo it's NOT LOCAL
17:55.00SamotIt's REMOTE
17:55.11SamotSo of course it's going to use the external media ips for it's side of the call
17:55.30Bud82You mean local as in local number to the pbx
17:55.42SamotNo
17:55.47SamotI mean local as in a local IP
17:55.55SamotOn the same networks that are listed in local_net
17:56.04Bud82yes, then it is REMOTE indeed
17:56.15SamotSo it should be using the external_media_address
17:56.20Bud82so it would need to make use of external_media_address for sure
17:56.40SamotSo the channel for 7001's call into the PBX
17:56.50SamotRemote  so use external_media_addresss
17:57.03Samotthe extension it is calling, local IP on the same network as Asterisk,
17:57.10SamotSo that channel is going to use private IPs
17:57.16Bud82http://pastebin.ca/3813828
17:57.17Samotthen Asterisk is going to bridge them together.
17:57.29Bud82the relevant config in pjsip.conf
17:57.44Samotlocal_net=192.168.178.0/24 <-- If the device isn't on this IP range..
17:58.00SamotIt's going to use external_media_address, end of story.
17:58.09Bud82and it doesn't
17:58.16Bud82it did before
17:58.18SamotI just saw it
17:59.12Bud82I'm unable to figure out what is wrong with the commit, so hopefully you can find out with this info a little ?
17:59.43paule32i can play english, i download german (de) sound files, but they will not be play
18:00.03paule32i wrote "language=de" in sip.conf
18:00.06paule32reload
18:00.12paule32not help
18:00.38*** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net)
18:01.18SamotSo the peer for the devices
18:01.24SamotBoth their settings
18:03.08Bud82For this test only, it doesn't work when calling an IVR application in Asterisk (unchanged config to 14.3.0).
18:03.58Bud82RTP ports are open, all client devices have appropiate settings for use of Asterisk in NAT environment.
18:04.15SamotShow the settings
18:05.05Bud82Zoiper has "use RPORT for signalling" on as well as "use RPORT for media"
18:05.18*** join/#asterisk tzafrir (~tzafrir@bzq-179-40-170.cust.bezeqint.net)
18:05.20SamotI'm going to ask one more time.
18:05.37SamotShow the settings for these devices for there pjsip config.
18:06.20Bud82It's inside a mysql table using the ODBC connector, I'll show you what is set.
18:07.14paule32ha
18:07.18paule32i am good
18:07.31paule32it work fpr multiple languages
18:07.41paule32excelent software
18:07.44SamotSo you're using RealTime for your extensions?
18:07.56paule32me, yes
18:08.08SamotNo. Bud82
18:08.29Bud82Yes
18:09.14Bud82@Samot: http://pastebin.ca/3813830
18:09.16paule32it is a little bit funny, i call the echotest, a female person speaks, but the call don't end/hangup
18:09.40[TK]D-FenderIt hangs up ... when you hang up
18:09.52Samottransport-udp-nat       7002   7002       internal       all    ulaw,alaw   yes        yes     nl
18:10.09SamotTo: <sip:9000@office.online4you.nl>;tag=ab8dbba8-7281-4b8c-8e80-0e6fbcca63a3
18:10.15SamotI said the two extensions in the call
18:10.36Samot7001 and 9000
18:10.52paule32[TK]D-Fender: yes, but i define hangup in dialplan
18:10.57Bud829000 is an application in extensions.conf I've created over the past 2 months, not able to copy that in full
18:11.15Bud82however, it's not even relevant as it works perfectly in 14.3.0 without the local_net change
18:11.28SamotThen you will need to open a bug.
18:12.09Samot-- Executing [9000@internal:5] Answer("PJSIP/7001-00000002", "") in new stack
18:12.21Samot^^^ Because if this is an application, what is is doing with the call
18:12.40Bud82just playing a few files asking for input, etc.
18:12.55*** join/#asterisk miralin (~Thunderbi@85.115.248.226)
18:12.59Bud82but they can't be heard since 14.4.0 because the RTP replies an internal LAN address
18:13.04Samothttps://www.irccloud.com/pastebin/n2LsA6yN/
18:13.08SamotWhere?
18:13.12SamotI see the channel answered.
18:13.14SamotAnd that's it
18:13.18SamotNo playbacks
18:13.33SamotJust a a digit timeout
18:13.52SamotAnd a response timeout
18:13.56SamotI don't see digits entered.
18:14.00SamotI don't see anything else happening.
18:14.04Bud82it plays allright probably, but my phone can't connect to the RTP stream
18:14.15SamotThere's NO PLAYBACK
18:14.18Bud82so Asterisk is doing its thing OK
18:14.29SamotThere would be something in the verbose log showing it
18:14.35SamotPlayback() or Background()
18:14.36SamotSomething.
18:14.44SamotIt would show the files being loaded and playing back
18:14.48SamotNothing exists
18:15.14SamotShow the context for 9000
18:16.48Bud82Sorry, I can't just publish that as a pastebin in the public channel. I see that I've hung up too fast in the copied conversation between 7001 and 9000
18:16.54SamotOK
18:16.56SamotWell then..
18:16.59SamotThere's not RTP
18:17.08SamotBecause nothing is saying there needs to be RTP
18:17.12SamotThere's no playback
18:17.23SamotBackground(), Progress() nothing.
18:17.32SamotNot a single thing that would require RTP happening.
18:18.02Bud82There's a lot of PlayBack in the app
18:18.07Bud82just not in this pastebin
18:18.08SamotSure
18:18.11SamotWhere is it?
18:18.15SamotIt should have happened in the call
18:18.21Bud82it does
18:18.26SamotI don't see it.
18:18.29SamotLook at your pastebin.
18:18.44SamotDo you see it being executed in the dialplan?
18:18.53Bud82yes, I know, I've hung up in the client too so for you to see all the playbacks
18:19.02Bud82yes, I see it each time
18:19.06Bud82that's not the issue
18:19.06SamotOK
18:19.08SamotI'm done.
18:19.14SamotYou fail to follow instructions.
18:19.17SamotOpen a bug.
18:19.21SamotOr go back to 14.13
18:19.21Bud82the issue is the RTP response from Asterisk ignoring external_media_address
18:19.30SamotI asked for a call
18:19.38SamotYou gave me incomplete output
18:19.53SamotNothing that you've shown proves RTP was started in that call.
18:20.04Bud82the reason for this failure seems to be caused by the recent issue that "fixed" it
18:20.11SamotThen open a bug.
18:20.17Samotissues.asterisk.org
18:24.21Bud82There's no account creation there ?
18:24.43filelook to the left
18:25.05SamotIn the huge f'ing letters.
18:25.06SamotTo create an account, please visit signup.asterisk.org.
18:25.14Bud82it is ;)
18:25.38Bud82Thanks @Samot for being so very helpful
18:26.04fileyou WILL need to provide a complete log with SIP traces and complete configuration for that
18:26.50SamotWITHOUT sanitizing it.
18:29.54paule32is that valid:  exten => 100,4,Echo
18:29.56paule32?
18:30.37paule32when i comment out these line, the caller hangs up
18:30.54paule32when, not, the caller run in silence
18:31.54paule32exten => 100,4,Echo  ; don't work
18:32.03paule32;exten => 100,4,Echo  ; work
18:34.48[TK]D-Fender<paule32> ;exten => 100,4,Echo  ; work <- doesn't work
18:34.51[TK]D-Fenderthat does NOTHING
18:34.56[TK]D-Fenderit's commented out
18:35.29[TK]D-Fenderbefore running echo you need to ANSWER the channel and do something to establish RTP setup (playback,e tc)
18:35.39[TK]D-FenderEven if it's silence/1
18:36.03SamotFunny.
18:36.09SamotI think I was just saying that
18:36.55*** join/#asterisk matrix1233 (~matrix123@41.230.44.234)
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18:38.59paule32[TK]D-Fender: i would say, if i include the text/cmd (exten => 100,4,Echo)  then it wait for answer?
18:39.35paule32i comment out the line, and the caller hangs up - and the callee
18:39.45[TK]D-FenderThre is no Callee
18:39.45paule32so both sides hang up
18:39.50[TK]D-Fenderthere is no other person.
18:39.57[TK]D-FenderThis is ONE channel executing DIALPLAN
18:40.10[TK]D-FenderYou don't seem to understand what you are doing here....
18:41.27paule32look: http://codepad.org/9PvZy60o
18:41.40paule32this work
18:42.02paule32both sides hangup
18:42.13[TK]D-Fenderther is no "both sides"
18:42.21[TK]D-FenderThis is ONE person <-------------
18:42.48paule32line 5, is uncomment the call is pending
18:43.15[TK]D-Fenderisn't seeing an actual problem anywhere
18:43.35paule32but me
18:43.49[TK]D-FenderWHERE"S THE CALL SHOWING THE ACTUAL PROBLEM?
18:44.00[TK]D-Fendershow the ****failure****
18:44.22[TK]D-FenderYour description is not useful and you are not SHOWING a failure
18:46.58paule32here my steps:   1. person one call asterisk server 100,  2. voicefile is play,  3. the call /asterisk close the line,   4. soft phone hangs in silence.  without the Echo line,  the person one, and the asterisk hang up the call,  else, only asterisk close the line
18:47.47[TK]D-Fender<[TK]D-Fender> WHERE"S THE CALL SHOWING THE ACTUAL PROBLEM?
18:47.51paule32where person one still in call process
18:47.59[TK]D-FenderThere is no 2nd person
18:48.32paule32halejuljia
18:48.42paule32the 1 person is asterisk
18:48.49[TK]D-FenderIT ISN'T A PERSON
18:48.49paule32the 2 person is me
18:49.09[TK]D-FenderDIAL calls something.  then there are 2 actual parties bridged by *
18:49.13paule32i initiate an echotest
18:49.16[TK]D-Fender<[TK]D-Fender> <[TK]D-Fender> WHERE"S THE CALL SHOWING THE ACTUAL PROBLEM?
18:49.48paule32it is not showing, but i have the problem here
18:50.26[TK]D-Fender[TK]D-Fender> <[TK]D-Fender> <[TK]D-Fender> WHERE"S THE CALL SHOWING THE ACTUAL PROBLEM?
18:55.11paule32http://codepad.org/INbwMt8A
18:58.26*** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net)
18:58.45paule32is Echo an application, that only for wait of user interaction?
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19:07.38igcewielingpaule32: is English your second language?
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19:14.26[TK]D-Fenderpaule32> is Echo an application, that only for wait of user interaction? <- you don't seem to understand what it IS.
19:14.33[TK]D-Fenderit's there to echo what you SAY back at you
19:14.37[TK]D-Fenderjust to test audio <-
19:14.52[TK]D-FenderRead the instructions
19:15.07[TK]D-Fender"core show application echo" <-
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20:06.38fc_rbHello, I think this is the correct place to ask this, but please correct me if I am wrong. I'm hoping to generate hints based on a portion of the subscriber's identity, but it is unclear if there are any variables available for use besides ${EXTEN} when reaching the subscribecontext. Can anyone offer insight into the available variables? Thanks!
20:08.47[TK]D-FenderWhat others have you tried?
20:10.34fc_rbI tried CONTEXT, just to see if I could get anything besides EXTEN, but it is empty as well.
20:11.03fc_rbWe have Devices registered as ###-ORGUNIT
20:11.12fc_rband I was hoping to use the ORGUNIT to generate the hint
20:11.30[TK]D-FenderShow your code and the attempt
20:11.41[TK]D-Fender~pb
20:11.41infobotsomebody said pastebin was 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.
20:11.42[TK]D-Fender^^^
20:20.43fc_rbhttps://pastebin.ca/3813868
20:20.53fc_rbThanks for taking a look!
20:21.33[TK]D-FenderAnd the subscribe attempt
20:22.03[TK]D-FenderAnd we should also be testing to mak sure that entire function works in a normal call as well.
20:23.58*** join/#asterisk tuxian (~tuxian@igilmour.plus.com)
20:27.20fc_rbSo in production we use [subscribe-ORGUNIT] as the subscribe_context. That is becoming unmanageable though so we are trying to get it working dynamically instead
20:28.06[TK]D-FenderI'm about to run out of time...
20:28.11[TK]D-Fenderbefore heading home
20:28.16[TK]D-FenderWhile will take a bit..
20:28.27[TK]D-Fenderso I need some solid debug as requested quick..
20:28.38Samot4:30?
20:28.41SamotPsssh.
20:28.43SamotCushy job.
20:31.12*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
20:32.59[TK]D-Fender...and checkout time.
20:33.00[TK]D-FenderBBL
20:33.35fc_rbI'm not sure what additional debug is needed, I thought the details in the pastebin were pretty clear
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