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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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| 01:20.10 | salviadud | I'm trying to debug a cisco phone I converted to sip |
| 01:20.37 | salviadud | I 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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| 10:44.37 | RomanDcoz | Hello All, I am just streaming audio file using "$agi->stream_file("1495017496-1574156409", '#');" but file not playing and direct hangup the call. |
| 10:44.37 | RomanDcoz | Here is the logs : https://pastebin.com/qe39D9fD |
| 10:44.47 | RomanDcoz | Can any one please guide me ? I am stuck here. |
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| 11:08.00 | Dirk23 | Hi. is it possible (for monitoring) to log in to asterisk cli, send sip show peers (parse the output) and exit cli? |
| 11:10.50 | Dirk23 | with a script |
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| 11:19.27 | dadrc | asterisk -rx 'sip show peers' |
| 11:20.36 | dadrc | Dirk23: â |
| 11:20.53 | Dirk23 | jap, found that one too. Thnx |
| 11:21.47 | Martin` | ah that is easy :D |
| 11:21.58 | Dirk23 | indeed |
| 11:24.51 | RomanDcoz | <PROTECTED> |
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| 11:27.18 | Martin` | Hm don't know, never did that before with an agi |
| 11:31.48 | RomanDcoz | Thanks. If any one facing same kind of issue then please suggest me. |
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| 12:45.16 | paule32 | hello |
| 12:45.25 | paule32 | i would like to implement an echo test |
| 12:45.29 | paule32 | i get this: |
| 12:45.35 | paule32 | [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 |
| 12:45.35 | paule32 | [May 17 14:41:56] WARNING[24075][C-00000000 |
| 12:45.54 | paule32 | but the file exists |
| 12:46.30 | file | <PROTECTED> |
| 12:46.52 | paule32 | i add gsm |
| 12:46.59 | paule32 | after filename |
| 12:47.03 | file | don't |
| 12:47.13 | file | Asterisk will automatically find the best file it can |
| 12:47.16 | paule32 | exten => 81,3,Playback(/var/lib/asterisk/sounds/de/demo-echotest.gsm) |
| 12:47.55 | paule32 | when i use: |
| 12:47.55 | paule32 | exten => 81,3,Playback(demo-echotest) |
| 12:48.00 | paule32 | the same error |
| 12:48.06 | paule32 | file not found |
| 12:48.19 | file | remove the ".gsm" from your first one, do a dialplan reload, and try again |
| 12:49.02 | *** join/#asterisk drathir (~kamiljk8@unaffiliated/drathir) |
| 12:49.12 | paule32 | http://codepad.org/FNrI276v |
| 12:49.23 | paule32 | this was the original |
| 12:51.03 | paule32 | when reload in cli |
| 12:51.06 | paule32 | [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'. |
| 12:52.52 | file | is the extension '81' reachable from the context 'default'? |
| 12:55.21 | *** join/#asterisk drathir (~kamiljk8@unaffiliated/drathir) |
| 12:56.01 | [TK]D-Fender | Clearly not... |
| 12:56.40 | [TK]D-Fender | paule32> 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-Fender | Don't pollute your tests by changing *2* things at a time. |
| 12:58.45 | paule32 | http://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-Fender | and your earlier sample seemed to imply it's in a DE folder but we have no proof what the current language is |
| 13:00.20 | paule32 | german |
| 13:00.24 | [TK]D-Fender | And we don't see the files themselves |
| 13:00.33 | [TK]D-Fender | I don't see PROOF that the language is set <- |
| 13:00.52 | paule32 | http://codepad.org/kahl7pb0 |
| 13:01.04 | [TK]D-Fender | Show a new call with the folder dump |
| 13:01.25 | [TK]D-Fender | exten => _3X,n,Dial,SIP/${EXTEN}|55|Ttr <- no good |
| 13:01.32 | paule32 | http://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-Fender | Show the new call with proper verbose and the sound folder "ls -la" dump |
| 13:03.01 | *** join/#asterisk scgm11_ (~scgm11@r186-50-105-44.dialup.adsl.anteldata.net.uy) |
| 13:03.34 | paule32 | http://codepad.org/LfQB0elM |
| 13:04.07 | *** join/#asterisk tuxian_ (~tuxian@194.12.3.78) |
| 13:04.55 | [TK]D-Fender | waits for the other half |
| 13:07.50 | paule32 | http://codepad.org/q7P8chGF |
| 13:08.43 | [TK]D-Fender | Fix your ownerships <- |
| 13:09.14 | [TK]D-Fender | 75% = ROOT |
| 13:09.19 | paule32 | the Dial plan, instead | a comma? |
| 13:09.45 | [TK]D-Fender | [TK]D-Fender> exten => _3X,n,Dial,SIP/${EXTEN}|55|Ttr <- no good |
| 13:09.58 | *** join/#asterisk drathir (~kamiljk8@unaffiliated/drathir) |
| 13:10.05 | [TK]D-Fender | exten => _3X,n,Dial(SIP/${EXTEN},55,Ttr) <- good |
| 13:10.33 | paule32 | thank you |
| 13:10.54 | paule32 | and context ankommend, same? |
| 13:11.15 | paule32 | exten => 621,1,Dial,SIP/30|30|r |
| 13:11.23 | [TK]D-Fender | Same sort of thing |
| 13:11.44 | paule32 | is that old convention (the parts from me) ? |
| 13:12.04 | [TK]D-Fender | yes |
| 13:12.07 | [TK]D-Fender | VERY old |
| 13:12.20 | [TK]D-Fender | 10 years... |
| 13:12.24 | paule32 | uh |
| 13:12.35 | paule32 | and _3X .. |
| 13:13.04 | [TK]D-Fender | What about it? |
| 13:13.22 | paule32 | other convention, or correct |
| 13:13.45 | [TK]D-Fender | ? |
| 13:13.57 | paule32 | what means _ |
| 13:14.12 | [TK]D-Fender | ~book |
| 13:14.12 | infobot | Asterisk: 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-Fender | Read up on your dialplan basics |
| 13:14.30 | paule32 | thank you |
| 13:14.35 | [TK]D-Fender | pattern <- |
| 13:14.53 | paule32 | i have no isdn, a dsl phone line |
| 13:15.30 | paule32 | can i do my self outgoing calls to other pbx - in other region |
| 13:15.44 | paule32 | the good old telephony |
| 13:16.04 | paule32 | i have a 56k modem card, is it enough if possible? |
| 13:16.17 | *** join/#asterisk drathir (~kamiljk8@unaffiliated/drathir) |
| 13:16.45 | [TK]D-Fender | no |
| 13:17.38 | [TK]D-Fender | DAHDI supports a very specific set of chipsets for analog interfaces. Generic modems are no good |
| 13:17.51 | paule32 | i 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) |
| 13:18.45 | paule32 | and now, i read here, isdn |
| 13:18.59 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:18.59 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:19.12 | [TK]D-Fender | Not with a cheap modem |
| 13:20.09 | [TK]D-Fender | There are several choices for analog, BRI, PRI, CAS, etc |
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| 13:28.03 | paule32 | http://codepad.org/a5tmMyHG |
| 13:28.10 | paule32 | something wrong |
| 13:28.30 | paule32 | with wich number, i have to log in? |
| 13:28.36 | paule32 | i get only |
| 13:28.47 | paule32 | QouteCom |
| 13:29.04 | paule32 | woops i mean, i get online at home server |
| 13:29.11 | *** join/#asterisk drathir (~kamiljk8@unaffiliated/drathir) |
| 13:29.38 | paule32 | and get the messages above |
| 13:30.01 | [TK]D-Fender | ? |
| 13:30.05 | paule32 | http://codepad.org/a5tmMyHG |
| 13:30.11 | paule32 | i call 81 |
| 13:30.20 | paule32 | and get the message in cli |
| 13:30.21 | paule32 | http://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-Fender | Means what it says |
| 13:30.38 | [TK]D-Fender | There is no match for 81 in [default] |
| 13:30.43 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:30.43 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:30.44 | [TK]D-Fender | there is not hing to guess about |
| 13:30.55 | [TK]D-Fender | it is telling exactly what was requested, and where it is looking for it |
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| 13:31.39 | paule32 | [default] |
| 13:31.39 | paule32 | include => lokal |
| 13:31.39 | paule32 | include => echotest |
| 13:31.46 | [TK]D-Fender | http://codepad.org/kahl7pb0 |
| 13:31.49 | [TK]D-Fender | THIS showed 81 |
| 13:32.07 | [TK]D-Fender | http://codepad.org/a5tmMyHG <- Here is see ***600*** |
| 13:32.12 | [TK]D-Fender | These are NOT the same at all |
| 13:32.50 | paule32 | so far i read is 600 default by standard installation |
| 13:32.54 | paule32 | that i have |
| 13:33.14 | paule32 | the standard number of echotest |
| 13:33.14 | [TK]D-Fender | You showed 1 part of code, and the dialplan execution does NOT match |
| 13:33.21 | paule32 | woops |
| 13:33.22 | paule32 | sorry |
| 13:33.55 | paule32 | http://codepad.org/PGaH0EAa |
| 13:35.17 | [TK]D-Fender | "dialplan show" <- |
| 13:36.07 | [TK]D-Fender | And 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-Fender | Code doesn't come out of thing air... |
| 13:37.15 | paule32 | http://codepad.org/KOpLxTVW |
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| 13:37.32 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 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-Fender | Get rid of LUA & AEL |
| 13:38.59 | paule32 | that not my config |
| 13:39.03 | paule32 | that is default |
| 13:39.05 | [TK]D-Fender | it clearly blew away your default context |
| 13:39.09 | [TK]D-Fender | You left that there |
| 13:39.11 | [TK]D-Fender | it is screwing you |
| 13:39.13 | [TK]D-Fender | remove it |
| 13:39.16 | paule32 | ok |
| 13:39.22 | paule32 | moment please |
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| 14:24.56 | paule32 | jens@rechner ~ $ sudo asterisk -r |
| 14:24.56 | paule32 | Privilege escalation protection disabled! |
| 14:24.56 | paule32 | See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. |
| 14:24.56 | paule32 | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 14:25.07 | paule32 | i delete the .ctl file |
| 14:25.14 | paule32 | but asterisk will not start |
| 14:25.33 | [TK]D-Fender | it is complaining that it does nto exist... |
| 14:25.38 | [TK]D-Fender | and you tell us you DELETED IT |
| 14:25.42 | [TK]D-Fender | Do you see a problem with this? |
| 14:25.48 | paule32 | hehe |
| 14:25.49 | paule32 | ook |
| 14:26.25 | *** join/#asterisk drathir (~kamiljk8@unaffiliated/drathir) |
| 14:26.52 | paule32 | rechner jens # touch /var/run/asterisk/asterisk.ctl |
| 14:26.53 | paule32 | rechner jens # asterisk -r |
| 14:26.57 | [TK]D-Fender | So 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-Fender | You can't just TOUCH the file |
| 14:27.21 | [TK]D-Fender | it is supopsed to hold the PID for the running * process <- |
| 14:27.49 | paule32 | yeah |
| 14:27.55 | paule32 | how to start asterisk? |
| 14:28.05 | paule32 | service asterisk start ? |
| 14:28.24 | [TK]D-Fender | probably |
| 14:28.32 | [TK]D-Fender | depending on your install method & distro |
| 14:28.39 | paule32 | linux mint |
| 14:28.42 | paule32 | same problem |
| 14:28.52 | [TK]D-Fender | of course that assumes it isn't already running |
| 14:29.18 | [TK]D-Fender | And by douing "touch" like that... guess who the OWNER is? |
| 14:29.24 | [TK]D-Fender | You keep screwing things with permissions.... |
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| 14:58.04 | igcewieling | I miss the days of interesting questions. |
| 14:59.30 | *** join/#asterisk matrix1233 (~matrix123@41.230.44.234) |
| 15:03.04 | *** join/#asterisk drathir (~kamiljk8@unaffiliated/drathir) |
| 15:03.29 | Samot | Well I just got one. |
| 15:03.52 | Samot | Someone just emailed me about helping them with their 3rd party Node.js SIP client.. |
| 15:04.20 | Samot | That uses SUBSCRIBE/NOTIFY for CallerID. Which is odd, to say the least. |
| 15:10.02 | *** join/#asterisk drathir (~kamiljk8@unaffiliated/drathir) |
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| 16:25.44 | DanQuinney | ha 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) |
| 16:40.31 | carlosdienstmann | hello 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 ()] |
| 16:42.22 | carlosdienstmann | anyone 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 |
| 16:42.41 | carlosdienstmann | for master.csv it worked nice |
| 16:46.49 | paule32 | hello |
| 16:47.16 | paule32 | how can i remove the items, so only my own rules are loaded: |
| 16:47.17 | paule32 | http://codepad.org/EafFDaNT |
| 16:47.18 | paule32 | ? |
| 16:47.57 | [TK]D-Fender | Stop loading those modules or trash their configs |
| 16:48.04 | [TK]D-Fender | [pbx_ael |
| 16:49.24 | [TK]D-Fender | And I no longer see your extensions.conf in there at all |
| 16:58.34 | paule32 | [TK]D-Fender: i have download the latest version |
| 16:58.42 | paule32 | compile and install it |
| 16:58.49 | [TK]D-Fender | ... |
| 16:58.50 | paule32 | no more old stuff |
| 16:59.39 | [TK]D-Fender | Then you've wasted out time in the earlier debugging. |
| 16:59.57 | [TK]D-Fender | You 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-Fender | And are making this a guessing game |
| 17:09.10 | paule32 | i have a connection now, and echotest work |
| 17:09.46 | paule32 | [ Context 'parkedcalls' created by 'res_parking/default' ] |
| 17:09.46 | paule32 | <PROTECTED> |
| 17:09.53 | paule32 | in which file is that ? |
| 17:10.58 | igcewieling | paule32: that is one of the contexts which are more trouble than it is worth. |
| 17:11.55 | igcewieling | paule32: features.conf |
| 17:12.23 | [TK]D-Fender | I 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 |
| 17:13.14 | igcewieling | [TK]D-Fender: interesting. What config file it is it then? |
| 17:15.40 | igcewieling | Actually, I'm stuck in 2014-10-23, the day before Asterisk 13 was released. |
| 17:15.59 | igcewieling | I suppose that would not make as interesting a song title. |
| 17:16.21 | igcewieling | Long live Asterisk 11! |
| 17:17.29 | Samot | Well at least until Oct 25th. |
| 17:18.24 | paule32 | thanks |
| 17:18.29 | paule32 | you are super heros |
| 17:19.03 | paule32 | is it possible to call mobile phones? |
| 17:20.24 | paule32 | also i can't see contexts 700 in the conf |
| 17:20.29 | paule32 | features.conf |
| 17:20.36 | paule32 | it is a file from the compilation |
| 17:20.43 | paule32 | by "make samples" |
| 17:20.53 | paule32 | all text is commented |
| 17:21.21 | Samot | features.conf only plays a part of Park if you want a feature code to do the transfer for you to a slot |
| 17:21.33 | Samot | Otherwise you transfer the call to the lot or a specific slot. |
| 17:21.57 | paule32 | yes, but it is default sample text |
| 17:22.07 | paule32 | so i think i can delete the file |
| 17:22.14 | Samot | Why? |
| 17:22.27 | Samot | Are you saying you don't want feature codes? |
| 17:22.33 | Samot | Or that you don't want Parking? |
| 17:22.52 | paule32 | i would like learn the procedures of asterisk |
| 17:23.06 | paule32 | so, the basics are echotest |
| 17:23.06 | Samot | So then deleting them would be a waste. |
| 17:23.17 | Samot | Because you can't learn about them if they aren't there. |
| 17:25.25 | paule32 | yeah, 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 |
| 17:26.40 | Samot | Settings really haven't changed much over the releases. |
| 17:26.51 | Samot | New features/settings have been added.. |
| 17:27.06 | Samot | But even deprecated settings still function. |
| 17:27.17 | paule32 | yeah |
| 17:28.21 | paule32 | now, i have the source code, and i am not have to deal with old repos |
| 17:28.35 | paule32 | i work under linux, but lazy sysadmin |
| 17:28.37 | paule32 | :-) |
| 17:29.13 | Samot | file: 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. |
| 17:29.25 | file | yes. |
| 17:29.36 | Samot | OK. |
| 17:32.38 | igcewieling | Asterisk 11 will not suddenly break when support expires. |
| 17:32.57 | file | yes, and no |
| 17:33.10 | file | as newer compiler releases occur older versions can sometimes no longer build |
| 17:35.04 | Samot | I'm not sure SFO counts as "support" |
| 17:35.22 | Samot | But yeah, 11 isn't just going to go belly up. |
| 17:36.19 | rmudgett | That's why people still run DOS 3.3 |
| 17:36.31 | Samot | Or Asterisk 1.8 |
| 17:36.33 | Samot | Or 1.4 |
| 17:36.40 | file | or why rmudgett runs antiquated versions of Ubuntu |
| 17:36.45 | file | :P |
| 17:36.47 | paule32 | [ Context 'parkedcalls' created by 'res_parking/default' ] |
| 17:37.05 | paule32 | is that context automatically created? |
| 17:37.25 | *** join/#asterisk matrix1233 (~matrix123@41.230.44.234) |
| 17:37.30 | paule32 | i wipe data, reload asterisk, and can't see progress |
| 17:38.23 | igcewieling | file: 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. |
| 17:38.30 | paule32 | file: you don't like ubuntu? |
| 17:38.41 | file | paule32: I use Ubuntu |
| 17:38.55 | paule32 | o, sorry misread |
| 17:38.56 | file | rmudgett just holds on to the version he is currently on for far too long |
| 17:39.11 | igcewieling | paule32: yes, it is automatically created |
| 17:39.11 | paule32 | have read the debian info |
| 17:39.19 | *** join/#asterisk Bud82 (5ed33bfe@gateway/web/freenode/ip.94.211.59.254) |
| 17:39.24 | igcewieling | Why are you still wasting your time on this? |
| 17:39.43 | paule32 | with wipe data? |
| 17:39.52 | igcewieling | with trying to remove that context. |
| 17:40.17 | paule32 | ah, ok, i had thought that is defined in a config |
| 17:40.32 | Samot | The wiki and the sample files tell you |
| 17:40.41 | Samot | The the partext is created automatically. |
| 17:41.00 | igcewieling | res_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 :) |
| 17:41.33 | paule32 | oh |
| 17:41.44 | [TK]D-Fender | <paule32> is that context automatically created? <- yes |
| 17:41.50 | [TK]D-Fender | because of your parking config |
| 17:42.02 | [TK]D-Fender | <paule32> ah, ok, i had thought that is defined in a config <- it is |
| 17:42.38 | Samot | Between the sample files and the wiki, these questions can be answered. |
| 17:42.47 | igcewieling | Here is my modules.conf, you'll need to modify it to suit your needs. |
| 17:42.50 | igcewieling | https://pastebin.com/8zbJ43WU |
| 17:43.10 | igcewieling | comment out any odbc stuff first |
| 17:43.34 | Bud82 | Hello 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. |
| 17:44.06 | Bud82 | It has worked perfectly in 14.3.0 |
| 17:46.00 | file | that's not actually what it says, it makes the code skip the localnet check if localnet is not defined |
| 17:46.59 | file | Bud82: 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 |
| 17:47.00 | Bud82 | When the call comes in, no audio is heard (from Asterisk) because Asterisk asks to connect to a local IP for RTP. |
| 17:47.07 | Samot | Bud82: Show an internal call using the external_media_address in it. |
| 17:47.19 | Bud82 | Internal (on the same LAN) it works. |
| 17:47.27 | Samot | OK |
| 17:47.31 | Samot | Let me rephrase then. |
| 17:47.40 | file | you can also provide it here and Samot or others can take a look before filing |
| 17:47.42 | file | goes for dog walk |
| 17:47.51 | Bud82 | Can I provide it in private ? |
| 17:47.54 | Samot | Bud82: Show a call that is using the wrong media IPs. |
| 17:47.58 | Bud82 | Sure |
| 17:48.01 | Samot | ~pb |
| 17:48.01 | infobot | hmm... 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. |
| 17:48.30 | Samot | No PMs |
| 17:48.37 | Samot | In the channel |
| 17:49.26 | *** join/#asterisk jetlag (~jetlag@c-71-226-222-56.hsd1.nj.comcast.net) |
| 17:50.18 | Samot | Bud82: I said no PMs |
| 17:50.23 | Samot | And to use pastebin. |
| 17:50.28 | Samot | Please follow instructions. |
| 17:50.30 | Bud82 | sorry, I'll check that |
| 17:51.42 | Bud82 | http://pastebin.ca/3813826 |
| 17:52.41 | Bud82 | The SIP messages use the correct IP's, the RTP stream are announced with the wrong IP's |
| 17:53.08 | Samot | <--- Received SIP request (765 bytes) from UDP:94.211.59.254:54291 ---> |
| 17:53.13 | Bud82 | (the internal ones I'd like to rewrite using external_media_address, as works OK in 14.3.0) |
| 17:53.26 | Samot | Contact: <sip:7001@94.211.59.254:54291;transport=UDP> |
| 17:53.32 | Samot | ^^ So 7000 is a local extension? |
| 17:53.37 | Bud82 | yes |
| 17:53.45 | file | You need to provide the config too |
| 17:53.51 | Samot | ^^^^ |
| 17:54.04 | Bud82 | the transports part ? |
| 17:54.12 | Samot | Because 7001 is making the call from an External IP |
| 17:54.15 | igcewieling | sip.conf of course. |
| 17:54.20 | Samot | so why is the phone sending an External IP? |
| 17:54.24 | Samot | Well pjsip.conf |
| 17:54.26 | igcewieling | or whatever config file pjsip uses. |
| 17:54.30 | Bud82 | because it's not on the same network |
| 17:54.36 | Samot | ...... |
| 17:54.37 | Samot | What |
| 17:54.49 | Bud82 | it's an android phone connected over the internet to the Asterisk PBX |
| 17:54.53 | Samot | OK |
| 17:54.56 | Samot | So it's NOT LOCAL |
| 17:55.00 | Samot | It's REMOTE |
| 17:55.11 | Samot | So of course it's going to use the external media ips for it's side of the call |
| 17:55.30 | Bud82 | You mean local as in local number to the pbx |
| 17:55.42 | Samot | No |
| 17:55.47 | Samot | I mean local as in a local IP |
| 17:55.55 | Samot | On the same networks that are listed in local_net |
| 17:56.04 | Bud82 | yes, then it is REMOTE indeed |
| 17:56.15 | Samot | So it should be using the external_media_address |
| 17:56.20 | Bud82 | so it would need to make use of external_media_address for sure |
| 17:56.40 | Samot | So the channel for 7001's call into the PBX |
| 17:56.50 | Samot | Remote so use external_media_addresss |
| 17:57.03 | Samot | the extension it is calling, local IP on the same network as Asterisk, |
| 17:57.10 | Samot | So that channel is going to use private IPs |
| 17:57.16 | Bud82 | http://pastebin.ca/3813828 |
| 17:57.17 | Samot | then Asterisk is going to bridge them together. |
| 17:57.29 | Bud82 | the relevant config in pjsip.conf |
| 17:57.44 | Samot | local_net=192.168.178.0/24 <-- If the device isn't on this IP range.. |
| 17:58.00 | Samot | It's going to use external_media_address, end of story. |
| 17:58.09 | Bud82 | and it doesn't |
| 17:58.16 | Bud82 | it did before |
| 17:58.18 | Samot | I just saw it |
| 17:59.12 | Bud82 | I'm unable to figure out what is wrong with the commit, so hopefully you can find out with this info a little ? |
| 17:59.43 | paule32 | i can play english, i download german (de) sound files, but they will not be play |
| 18:00.03 | paule32 | i wrote "language=de" in sip.conf |
| 18:00.06 | paule32 | reload |
| 18:00.12 | paule32 | not help |
| 18:00.38 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 18:01.18 | Samot | So the peer for the devices |
| 18:01.24 | Samot | Both their settings |
| 18:03.08 | Bud82 | For this test only, it doesn't work when calling an IVR application in Asterisk (unchanged config to 14.3.0). |
| 18:03.58 | Bud82 | RTP ports are open, all client devices have appropiate settings for use of Asterisk in NAT environment. |
| 18:04.15 | Samot | Show the settings |
| 18:05.05 | Bud82 | Zoiper 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) |
| 18:05.20 | Samot | I'm going to ask one more time. |
| 18:05.37 | Samot | Show the settings for these devices for there pjsip config. |
| 18:06.20 | Bud82 | It's inside a mysql table using the ODBC connector, I'll show you what is set. |
| 18:07.14 | paule32 | ha |
| 18:07.18 | paule32 | i am good |
| 18:07.31 | paule32 | it work fpr multiple languages |
| 18:07.41 | paule32 | excelent software |
| 18:07.44 | Samot | So you're using RealTime for your extensions? |
| 18:07.56 | paule32 | me, yes |
| 18:08.08 | Samot | No. Bud82 |
| 18:08.29 | Bud82 | Yes |
| 18:09.14 | Bud82 | @Samot: http://pastebin.ca/3813830 |
| 18:09.16 | paule32 | it 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-Fender | It hangs up ... when you hang up |
| 18:09.52 | Samot | transport-udp-nat 7002 7002 internal all ulaw,alaw yes yes nl |
| 18:10.09 | Samot | To: <sip:9000@office.online4you.nl>;tag=ab8dbba8-7281-4b8c-8e80-0e6fbcca63a3 |
| 18:10.15 | Samot | I said the two extensions in the call |
| 18:10.36 | Samot | 7001 and 9000 |
| 18:10.52 | paule32 | [TK]D-Fender: yes, but i define hangup in dialplan |
| 18:10.57 | Bud82 | 9000 is an application in extensions.conf I've created over the past 2 months, not able to copy that in full |
| 18:11.15 | Bud82 | however, it's not even relevant as it works perfectly in 14.3.0 without the local_net change |
| 18:11.28 | Samot | Then you will need to open a bug. |
| 18:12.09 | Samot | -- Executing [9000@internal:5] Answer("PJSIP/7001-00000002", "") in new stack |
| 18:12.21 | Samot | ^^^ Because if this is an application, what is is doing with the call |
| 18:12.40 | Bud82 | just playing a few files asking for input, etc. |
| 18:12.55 | *** join/#asterisk miralin (~Thunderbi@85.115.248.226) |
| 18:12.59 | Bud82 | but they can't be heard since 14.4.0 because the RTP replies an internal LAN address |
| 18:13.04 | Samot | https://www.irccloud.com/pastebin/n2LsA6yN/ |
| 18:13.08 | Samot | Where? |
| 18:13.12 | Samot | I see the channel answered. |
| 18:13.14 | Samot | And that's it |
| 18:13.18 | Samot | No playbacks |
| 18:13.33 | Samot | Just a a digit timeout |
| 18:13.52 | Samot | And a response timeout |
| 18:13.56 | Samot | I don't see digits entered. |
| 18:14.00 | Samot | I don't see anything else happening. |
| 18:14.04 | Bud82 | it plays allright probably, but my phone can't connect to the RTP stream |
| 18:14.15 | Samot | There's NO PLAYBACK |
| 18:14.18 | Bud82 | so Asterisk is doing its thing OK |
| 18:14.29 | Samot | There would be something in the verbose log showing it |
| 18:14.35 | Samot | Playback() or Background() |
| 18:14.36 | Samot | Something. |
| 18:14.44 | Samot | It would show the files being loaded and playing back |
| 18:14.48 | Samot | Nothing exists |
| 18:15.14 | Samot | Show the context for 9000 |
| 18:16.48 | Bud82 | Sorry, 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 |
| 18:16.54 | Samot | OK |
| 18:16.56 | Samot | Well then.. |
| 18:16.59 | Samot | There's not RTP |
| 18:17.08 | Samot | Because nothing is saying there needs to be RTP |
| 18:17.12 | Samot | There's no playback |
| 18:17.23 | Samot | Background(), Progress() nothing. |
| 18:17.32 | Samot | Not a single thing that would require RTP happening. |
| 18:18.02 | Bud82 | There's a lot of PlayBack in the app |
| 18:18.07 | Bud82 | just not in this pastebin |
| 18:18.08 | Samot | Sure |
| 18:18.11 | Samot | Where is it? |
| 18:18.15 | Samot | It should have happened in the call |
| 18:18.21 | Bud82 | it does |
| 18:18.26 | Samot | I don't see it. |
| 18:18.29 | Samot | Look at your pastebin. |
| 18:18.44 | Samot | Do you see it being executed in the dialplan? |
| 18:18.53 | Bud82 | yes, I know, I've hung up in the client too so for you to see all the playbacks |
| 18:19.02 | Bud82 | yes, I see it each time |
| 18:19.06 | Bud82 | that's not the issue |
| 18:19.06 | Samot | OK |
| 18:19.08 | Samot | I'm done. |
| 18:19.14 | Samot | You fail to follow instructions. |
| 18:19.17 | Samot | Open a bug. |
| 18:19.21 | Samot | Or go back to 14.13 |
| 18:19.21 | Bud82 | the issue is the RTP response from Asterisk ignoring external_media_address |
| 18:19.30 | Samot | I asked for a call |
| 18:19.38 | Samot | You gave me incomplete output |
| 18:19.53 | Samot | Nothing that you've shown proves RTP was started in that call. |
| 18:20.04 | Bud82 | the reason for this failure seems to be caused by the recent issue that "fixed" it |
| 18:20.11 | Samot | Then open a bug. |
| 18:20.17 | Samot | issues.asterisk.org |
| 18:24.21 | Bud82 | There's no account creation there ? |
| 18:24.43 | file | look to the left |
| 18:25.05 | Samot | In the huge f'ing letters. |
| 18:25.06 | Samot | To create an account, please visit signup.asterisk.org. |
| 18:25.14 | Bud82 | it is ;) |
| 18:25.38 | Bud82 | Thanks @Samot for being so very helpful |
| 18:26.04 | file | you WILL need to provide a complete log with SIP traces and complete configuration for that |
| 18:26.50 | Samot | WITHOUT sanitizing it. |
| 18:29.54 | paule32 | is that valid: exten => 100,4,Echo |
| 18:29.56 | paule32 | ? |
| 18:30.37 | paule32 | when i comment out these line, the caller hangs up |
| 18:30.54 | paule32 | when, not, the caller run in silence |
| 18:31.54 | paule32 | exten => 100,4,Echo ; don't work |
| 18:32.03 | paule32 | ;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-Fender | that does NOTHING |
| 18:34.56 | [TK]D-Fender | it's commented out |
| 18:35.29 | [TK]D-Fender | before running echo you need to ANSWER the channel and do something to establish RTP setup (playback,e tc) |
| 18:35.39 | [TK]D-Fender | Even if it's silence/1 |
| 18:36.03 | Samot | Funny. |
| 18:36.09 | Samot | I think I was just saying that |
| 18:36.55 | *** join/#asterisk matrix1233 (~matrix123@41.230.44.234) |
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| 18:38.59 | paule32 | [TK]D-Fender: i would say, if i include the text/cmd (exten => 100,4,Echo) then it wait for answer? |
| 18:39.35 | paule32 | i comment out the line, and the caller hangs up - and the callee |
| 18:39.45 | [TK]D-Fender | Thre is no Callee |
| 18:39.45 | paule32 | so both sides hang up |
| 18:39.50 | [TK]D-Fender | there is no other person. |
| 18:39.57 | [TK]D-Fender | This is ONE channel executing DIALPLAN |
| 18:40.10 | [TK]D-Fender | You don't seem to understand what you are doing here.... |
| 18:41.27 | paule32 | look: http://codepad.org/9PvZy60o |
| 18:41.40 | paule32 | this work |
| 18:42.02 | paule32 | both sides hangup |
| 18:42.13 | [TK]D-Fender | ther is no "both sides" |
| 18:42.21 | [TK]D-Fender | This is ONE person <------------- |
| 18:42.48 | paule32 | line 5, is uncomment the call is pending |
| 18:43.15 | [TK]D-Fender | isn't seeing an actual problem anywhere |
| 18:43.35 | paule32 | but me |
| 18:43.49 | [TK]D-Fender | WHERE"S THE CALL SHOWING THE ACTUAL PROBLEM? |
| 18:44.00 | [TK]D-Fender | show the ****failure**** |
| 18:44.22 | [TK]D-Fender | Your description is not useful and you are not SHOWING a failure |
| 18:46.58 | paule32 | here 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? |
| 18:47.51 | paule32 | where person one still in call process |
| 18:47.59 | [TK]D-Fender | There is no 2nd person |
| 18:48.32 | paule32 | halejuljia |
| 18:48.42 | paule32 | the 1 person is asterisk |
| 18:48.49 | [TK]D-Fender | IT ISN'T A PERSON |
| 18:48.49 | paule32 | the 2 person is me |
| 18:49.09 | [TK]D-Fender | DIAL calls something. then there are 2 actual parties bridged by * |
| 18:49.13 | paule32 | i initiate an echotest |
| 18:49.16 | [TK]D-Fender | <[TK]D-Fender> <[TK]D-Fender> WHERE"S THE CALL SHOWING THE ACTUAL PROBLEM? |
| 18:49.48 | paule32 | it 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? |
| 18:55.11 | paule32 | http://codepad.org/INbwMt8A |
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| 18:58.45 | paule32 | is Echo an application, that only for wait of user interaction? |
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| 19:07.38 | igcewieling | paule32: is English your second language? |
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| 19:14.26 | [TK]D-Fender | paule32> 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-Fender | it's there to echo what you SAY back at you |
| 19:14.37 | [TK]D-Fender | just to test audio <- |
| 19:14.52 | [TK]D-Fender | Read the instructions |
| 19:15.07 | [TK]D-Fender | "core show application echo" <- |
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| 20:06.38 | fc_rb | Hello, 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-Fender | What others have you tried? |
| 20:10.34 | fc_rb | I tried CONTEXT, just to see if I could get anything besides EXTEN, but it is empty as well. |
| 20:11.03 | fc_rb | We have Devices registered as ###-ORGUNIT |
| 20:11.12 | fc_rb | and I was hoping to use the ORGUNIT to generate the hint |
| 20:11.30 | [TK]D-Fender | Show your code and the attempt |
| 20:11.41 | [TK]D-Fender | ~pb |
| 20:11.41 | infobot | somebody 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 | ^^^ |
| 20:20.43 | fc_rb | https://pastebin.ca/3813868 |
| 20:20.53 | fc_rb | Thanks for taking a look! |
| 20:21.33 | [TK]D-Fender | And the subscribe attempt |
| 20:22.03 | [TK]D-Fender | And we should also be testing to mak sure that entire function works in a normal call as well. |
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| 20:27.20 | fc_rb | So 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-Fender | I'm about to run out of time... |
| 20:28.11 | [TK]D-Fender | before heading home |
| 20:28.16 | [TK]D-Fender | While will take a bit.. |
| 20:28.27 | [TK]D-Fender | so I need some solid debug as requested quick.. |
| 20:28.38 | Samot | 4:30? |
| 20:28.41 | Samot | Psssh. |
| 20:28.43 | Samot | Cushy job. |
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| 20:32.59 | [TK]D-Fender | ...and checkout time. |
| 20:33.00 | [TK]D-Fender | BBL |
| 20:33.35 | fc_rb | I'm not sure what additional debug is needed, I thought the details in the pastebin were pretty clear |
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