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| 02:21.20 | cmendes0101 | is there still a cli command to see the performance of asterisk converting between codecs? |
| 02:21.45 | cmendes0101 | for playing audio files |
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| 10:24.13 | sziszi | hey guys! |
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| 10:28.02 | sziszi | I have a problem with AGI scripts. I wrote a bash AGI script which returns 'SET VARIABLE xy "OK"' if it's successful (it has two parameters). I reimplemented this stuff in another language but it's not working. Running them from console, both produces the same output, both logs the input vars, and both looks good to me. What I'm missing? what can I do to debug this further? |
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| 11:03.10 | danielyk | When will Asterisk 13.13.2 be released? |
| 11:05.08 | wdoekes | 13.14.0, unless there is a security issue or severe breakage |
| 11:07.59 | danielyk | Okay, because I am waiting for release of following patch: https://issues.asterisk.org/jira/browse/ASTERISK-26603 |
| 11:08.22 | danielyk | Where can I see in which version the patch will be released? |
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| 11:19.38 | dan_j | Hi. I'm really struggling with what I think is a bug but can't figure out the issue in order to report it. I've got an AGI script which allows the user to record a voicemail greeting outside of app_voicemail. The greetings are stored in a realtime db. In v11, it works perfectly. In v13, any greetings recorded by the AGI script cause Asterisk to deadlock when |
| 11:19.38 | dan_j | they are played back by Voicemail(). The channel doesnt hang up and the CLI stops responding to commands. IE' channel request hangup' does nothing and nothing happens when I enter 'core stop now'. |
| 11:19.53 | dan_j | Any idea how I can diagnose it? I've been at it for days now. |
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| 11:28.15 | wdoekes | danielyk: you can't see in which version it will be released until it's marked as such (merge into 13.x.0 release branch and tagged in JIRA). but it generally is the upcoming version. and when? I don't know, but I guess soon |
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| 11:34.28 | dan_j | My issue must be a bug in app_voicemail somewhere. I've taken the same .wav file and updated all the messages stored in the database with the same .wav file. |
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| 11:35.09 | dan_j | VoicemailMain() can play back the wav file, but the same wav file can't be played by Voicemail() as a greeting. |
| 11:36.50 | sziszi | dan_j, at my company we dont use voicemail, we built our recording context |
| 11:37.07 | dan_j | Yeh. I have a feeling that I'm going to have to do that. |
| 11:37.22 | dan_j | sziszi: where do you store your messages? |
| 11:37.41 | sziszi | dan_j, in FS, plain wav files |
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| 11:38.01 | dan_j | How do you deal with MWI? |
| 11:38.41 | dan_j | IE. how do the endpoints know that there is a message? |
| 11:39.05 | sziszi | they dont know |
| 11:39.31 | sziszi | most of our customers doesnt even know they got phone via IP |
| 11:39.53 | sziszi | we send these wav files in emails as an attachment |
| 11:41.11 | sziszi | they're notified, they can listen to it whenever they want, and if they pay extra they can access to their storage over FTP/webdav/etc |
| 11:41.23 | dan_j | Some clients want to listen to voicemail on the phone so i'll have to figure out how to control the MWI myself. |
| 11:41.52 | sziszi | it's a bummer :( |
| 11:42.13 | dan_j | Yep. |
| 11:42.54 | dan_j | Still havent given up on app_voicemail though. |
| 11:43.31 | sziszi | I'd probably write a context where they can listen to their stuff (numberplan+ playback() ) |
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| 11:44.13 | sziszi | when they done h,1,mv /vm_received/* /vm_listened |
| 11:44.16 | sziszi | umm |
| 11:44.19 | sziszi | with system() |
| 11:44.30 | sziszi | not pretty, but it works |
| 11:49.29 | dan_j | As the messages are already in mysql realtime, and they are accessed by multiple asterisk servers, it's probably easier to leave them in the database. |
| 11:55.16 | sziszi | in that case yes |
| 11:55.30 | sziszi | we dont have _any_ redundant services |
| 11:56.07 | sziszi | because running two same stuff in parallel is wasting money |
| 11:56.44 | Rac-on | dan_j: it is not that hard to call a random script from your own 'voicemail'-context that sends a MWI message to a phone |
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| 11:58.55 | Rac-on | dan_j: some inspiration: http://saevolgo.blogspot.nl/2012/07/asterisk-behind-kamailio-voicemail-mwi.html |
| 12:00.21 | dan_j | sziszi: I have two mysql servers using drbd replication and pacemaker. |
| 12:01.17 | sziszi | dan_j, that would be the best for us too, it's sad I dont have the right to make these decisions :( |
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| 12:11.13 | sziszi | okay, I resolved my issue |
| 12:11.25 | sziszi | freakin workarounds :( |
| 12:13.55 | dan_j | Rac-on: thanks |
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| 12:32.58 | MacroMan | Taking a look at at v14. Where has the install_prereq script gone? |
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| 12:45.06 | ruied_ | Hello. I'm trying to use python to make .call files to asterisk but I'm having an error that I'm not getting. I have installed pycall but it seems python can't access callfile.py |
| 12:45.27 | ruied_ | http://pastebin.com/9R88MtRf |
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| 13:26.12 | DivideBy0 | MacroMan: ./contrib/scripts/install_prereq |
| 13:26.32 | MacroMan | DivideBy0, THanks. Was just about to say I was looking in the wrong place |
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| 14:31.30 | EnrgySmth | On this example: http://www.voip-info.org/wiki/index.php?page_id=1120 |
| 14:32.43 | EnrgySmth | The outgoing call format at the top - Channel defines the number you are dialing - if it is internal SIP extension it would just be SIP/EXT# ? |
| 14:33.12 | EnrgySmth | And what does the Extension: s line represent? |
| 14:33.21 | Samot | What are you actually trying to do? |
| 14:33.31 | Samot | Before you use somethhing from a site that hasnt been current in years. |
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| 14:35.15 | EnrgySmth | I have a Raspberry Asterisk basic config going, with one extension on an SPA112 (Ext 200). I want to be able to externally trigger having the system call that extension and play a pre-recorded message |
| 14:35.50 | [TK]D-Fender | <EnrgySmth> And what does the Extension: s line represent? <- dialplan extension |
| 14:36.02 | EnrgySmth | An the first page of the Discussion tab there is a comment about changes for Asterisk 1.4.11 |
| 14:36.07 | [TK]D-Fender | <EnrgySmth> The outgoing call format at the top - Channel defines the number you are dialing - if it is internal SIP extension it would just be SIP/EXT# ? <- this is exactly what you'd put in a Dial() |
| 14:36.23 | [TK]D-Fender | Stop reading that WIKI. |
| 14:36.29 | [TK]D-Fender | It is almost the LAST place to look. |
| 14:36.35 | Samot | Totally. |
| 14:36.41 | EnrgySmth | Ah, ok - sorry |
| 14:36.42 | Samot | Maybe in 2006. |
| 14:36.52 | EnrgySmth | thanks guys |
| 14:37.02 | [TK]D-Fender | Mind you call files haven't really changed at all |
| 14:37.44 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files |
| 14:37.58 | dan_j | I'm using PHPAGI and can't seem to get it working on my v11 test system. When I do wait_for_digit, the cli shows 'The FD we were waiting for has something waiting. Waitfordigit returning numeric 1' but I never entered anything. Any ideas? |
| 14:38.07 | EnrgySmth | This is a project for my Grand Daughter. I tested last night and she is too young to get her to pick up the line before hitting numbers |
| 14:40.24 | EnrgySmth | So I am changing direction and trying to set up something where her grandma can click a link on a PHP webpage and it remotely triggers the system to call the 200 extension. |
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| 14:48.31 | [TK]D-Fender | Plenty easy |
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| 15:10.26 | dan_j | It appears wait_for_digit is broken as get_data works fine. |
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| 15:14.17 | Samot | Are you handling the output properly? |
| 15:16.24 | Samot | wait_for_digit returns results in ASCII |
| 15:16.31 | Samot | You have to convert it. |
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| 15:18.12 | Samot | $returned = $agi->wait_for_digit(3000); |
| 15:18.12 | Samot | $digit = chr($returned['result']); |
| 15:18.18 | Samot | ^^^ Works without issue. |
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| 15:21.56 | jayjay883 | When an incoming call is transferred/forwarded to an external number, depending on the number/operator it is transferred to I get no ring tone before the call is answered. Any options to enable this for any destination ? |
| 15:26.44 | [TK]D-Fender | "any destination" is not a thing |
| 15:27.13 | [TK]D-Fender | Look at the call. If the callee side does not indicate ringing, then * will not pass that state on to your calling channel. |
| 15:27.42 | Samot | Isn't Progress() the answer? |
| 15:27.54 | [TK]D-Fender | If the calling channel has already been asnwered first * won't generate ringing tones in that case. There is a Dial() options tto specifically force ringing to be indicated |
| 15:27.57 | Samot | To have ringing on the calling channel while stuff happens on the called channel.. |
| 15:28.29 | Samot | Or even if there isn't a called channel. |
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| 15:30.17 | Samot | Sorry, now I'm just thinking out loud about Progress(). |
| 15:30.42 | Samot | But yeah, what [TK]D-Fender said. Done in the Dial(). |
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| 15:41.09 | jayjay883 | Tha channel hasnt been answered |
| 15:42.09 | jayjay883 | Ill try progress ;) |
| 15:42.10 | Samot | It's still done in Dial() |
| 15:42.16 | Samot | No, for this you want Dial(). |
| 15:42.25 | Samot | core show application dial |
| 15:43.01 | jayjay883 | Both r(tone) and R is default, so what else is it ? |
| 15:43.26 | Samot | Show a call that doesn't get ringback. |
| 15:43.34 | Samot | sip set debug on |
| 15:43.39 | Samot | or pjsip set logger on |
| 15:43.50 | Samot | whichever tech you are using. |
| 15:44.09 | Samot | ~pb |
| 15:44.09 | infobot | [pastebin] 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. |
| 15:45.48 | jayjay883 | Samot: I cant really do that right now. The guy is really pissed and I unfortunately dont have any other testnumbers to try ;) However I can call the number from my asterisk user, be forwarded and hear the ring tone. Its if I call from an external mobile operator -> asterisk -> forward to mobile operator that doesnt work. |
| 15:46.15 | jayjay883 | And only to some forwarded numbers, not all |
| 15:47.21 | Samot | Like TK said, you have to see if the other side is indicating ringing. |
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| 15:49.49 | jayjay883 | ok tnx |
| 15:49.58 | [TK]D-Fender | Step 1 : LOOK AT THE CALL |
| 15:50.01 | Samot | indications.conf <-- you have this setup correctly? |
| 15:50.05 | [TK]D-Fender | forget that |
| 15:50.09 | [TK]D-Fender | may not be in play at all |
| 15:50.10 | dan_j | Samot: I'm not even trying to use the output. I'm just testing to see if it stops the execution. |
| 15:50.11 | [TK]D-Fender | Step 1 : LOOK AT THE CALL |
| 15:50.46 | dan_j | Samot: Heres my code. http://stackoverflow.com/questions/41896876/asterisk-phpagi-issue |
| 15:51.40 | [TK]D-Fender | dan_j, Show the full call with AGI debug |
| 15:51.46 | dan_j | ok. 1sec |
| 15:52.05 | [TK]D-Fender | dan_j, You should never have posted less than this |
| 15:52.58 | dan_j | https://www.irccloud.com/pastebin/h2UCFBhS/ |
| 15:53.43 | dan_j | Do you want sip debug or anything else included? |
| 15:53.49 | jayjay883 | samot: yeah but this is multinational. The [general] section refers to a single country. Ill check that when I get the sdp data tnx |
| 15:54.00 | [TK]D-Fender | that does not look like Asterisk AGI debug |
| 15:54.16 | [TK]D-Fender | We should see the AGI command call... |
| 15:54.19 | [TK]D-Fender | and we don't |
| 15:54.53 | Samot | $test = $agi->wait_for_digit(); <-- You have to set a wait time or it waits indefinitely |
| 15:55.54 | dan_j | Samot: This is a test. I want it to wait indefinitely for this test, but if you look at the output, its not waiting. It's just terminating. |
| 15:56.13 | dan_j | The reason I'm running this test is that I can't get stream_file to work. |
| 15:56.29 | dan_j | But get_data does work. |
| 15:56.56 | Samot | Oh wait, |
| 15:57.00 | Samot | You need -1 |
| 15:57.15 | Samot | If you want to wait indefinitely. |
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| 15:58.06 | Samot | What is stream_file doing that it is not supposed to? |
| 15:58.16 | Samot | You play a file that can be intterupted by a digit. |
| 15:58.22 | Samot | You play a file that can be interupted by a digit. |
| 15:58.41 | dan_j | https://www.irccloud.com/pastebin/2gJXl8VD/ |
| 15:58.52 | dan_j | [TK]D-Fender: Sorry, forgot to turn on agi debug |
| 15:59.21 | dan_j | Stream file doesnt seem to do anything. No warnings etc. I'll do a test with streamfile to show you. |
| 15:59.26 | igcewieling | the error appears to be fairly obvious |
| 16:00.02 | igcewieling | <SIP/peer1-0000000d>AGI Rx << #!/usr/bin/php |
| 16:00.02 | igcewieling | <SIP/peer1-0000000d>AGI Tx >> 510 Invalid or unknown command |
| 16:00.03 | Samot | <SIP/peer1-0000000d>AGI Rx << #!/usr/bin/php <-- WTH? |
| 16:00.11 | Samot | Exactly. |
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| 16:02.12 | dan_j | https://www.irccloud.com/pastebin/qbJJWBXV/ |
| 16:04.18 | dan_j | I added #!/usr/bin/php to phpagi.php to try to fix things, but that didnt help. Removed it and now it's working. |
| 16:04.58 | dan_j | Putting back the rest of my code to see if thats working now. In short, I'm trying to build my own voicemail system because v13 doesnt like my agi code that allows users to record a message outside of app_voicemail. |
| 16:05.50 | Samot | Is there a reason you're not using app_voicemail? |
| 16:07.32 | Samot | I mean, I can get not wanting the users to directly access app_voicemail because of what can be done in it. |
| 16:07.52 | dan_j | I've got an AGI script that allows users to record a voicemail greeting outside of app_voicemail. It is used to allow junior staff to record the greeting without giving them access to the messages. |
| 16:07.53 | Samot | But that can be mitigated.. |
| 16:08.01 | Samot | OK. |
| 16:08.05 | dan_j | In v11, it works fine, in v13, the greeting deadlines asterisk. |
| 16:08.07 | dan_j | deadlocks |
| 16:08.51 | dan_j | I cant figure out the issue so cant report it as a bug. So my only choice at the moment due to time constraints is to write my own. |
| 16:08.53 | [TK]D-Fender | #!/usr/bin/php <---- where's the -q ? |
| 16:09.06 | dan_j | [TK]D-Fender: I've fixed that one. |
| 16:09.28 | dan_j | 16:04 D<dan_j> I added #!/usr/bin/php to phpagi.php to try to fix things, but that didnt help. Removed it and now it's working. |
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| 16:21.15 | Samot | dan_j: So if you record a file via your AGI... |
| 16:21.33 | Samot | You can play it back immediately, no issue.. |
| 16:21.45 | Samot | But once it's in the DB, deadlock? |
| 16:22.16 | dan_j | If I record the file via my AGI and save it to the DB, I can then (via AGI) extract it from the DB to a temporary file and play it back using stream_file. |
| 16:22.33 | dan_j | But if I try to play it back using Voicemail(), it deadlocks in v13 |
| 16:22.54 | dan_j | The identical code in v11 works fine. |
| 16:23.49 | dan_j | deadlocks = completely. the CLI stops answering comments that I enter such as 'channel request hangup' to hang up the deadlocked channel, and 'core stop now' doesnt do anything. |
| 16:24.02 | dan_j | s/comments/commands |
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| 16:28.13 | dan_j | Im offline for 24 hours. Will have to deal with this tomorrow evening. Thanks for the help as always! |
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| 16:31.06 | WIMPy | When building DAHDI I get an empty Modules.symvers leading to unusabl modules. Does that ring a bell to anyone? Or is 4.8.15 too new? |
| 16:31.37 | igcewieling | 4.8.15? |
| 16:31.54 | WIMPy | Linux |
| 16:37.23 | jayjay883 | Samot: Regarding the no ring tone part. Looks like the SDP packet "180 Ringing" is not received when its not working, only "183 session progress" is received. Using pjsip. |
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| 16:53.31 | *** mode/#asterisk [+o gtjoseph] by ChanServ |
| 16:57.16 | drmessano | #FreeShia |
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| 17:13.50 | freebs | relax guy |
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| 17:19.29 | drmessano | I cant until he's released |
| 17:20.13 | Samot | We will not be divided. |
| 17:20.38 | Samot | But maybe his jail stint will get us Holes 2. |
| 17:20.40 | drmessano | WE WILL NOT BE FORKED |
| 17:21.27 | Samot | #ShivingShia |
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| 17:21.33 | drmessano | lol |
| 17:21.50 | drmessano | What a douchebag |
| 17:22.17 | drmessano | Could be worse |
| 17:22.36 | drmessano | I'm thinking about starting my own protest |
| 17:22.42 | drmessano | Against unreadable fonts |
| 17:22.56 | drmessano | WE WILL NOT BE UHH WHAT DOES THAT SAY |
| 17:22.58 | Samot | A part of me feels that he would be a douchebag but I would probably still hang out with him. |
| 17:23.19 | drmessano | #WeWillNotBeUhWhatDoesThatSay |
| 17:23.40 | drmessano | #FreeHelvetica |
| 17:24.31 | [TK]D-Fender | #SupportWingDdings |
| 17:24.35 | [TK]D-Fender | #SupportWingDings |
| 17:25.04 | drmessano | YES |
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| 21:29.04 | *** topic/#asterisk by gtjoseph -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.13.1 (2016/12/08), 11.25.1 (2016/12/08), Standard: 14.2.1 (2016/12/08); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 21:29.38 | *** topic/#asterisk by gtjoseph -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.13.1 (2016/12/08), 11.25.1 (2016/12/08), Standard: 14.2.1 (2016/12/08); 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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| 21:55.16 | *** join/#asterisk CrashSys (~Kumba@rrcs-97-76-33-146.se.biz.rr.com) |
| 21:55.26 | CrashSys | I don't suppose anyone knows how to debug this: kernel: traps: asterisk[4396] trap invalid opcode ip:4ab2cf sp:7ffd2f9e07c0 error:0 in asterisk[400000+230000] |
| 21:55.39 | CrashSys | It's Asterisk 11.25.1, or is supposed to be |
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| 21:59.40 | igcewieling | CrashSys: is it running in a virtual machine |
| 21:59.41 | igcewieling | ? |
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| 22:09.36 | CrashSys | No, physical hardware |
| 22:09.41 | CrashSys | on virtualbox is loads just fine |
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| 22:10.05 | CrashSys | This recently started happening about a month ago. It's built on OBS |
| 22:10.11 | CrashSys | err OpenSuSE Build Studio. |
| 22:10.20 | CrashSys | But I'm trying to figure out what instruction is the problem so I can recompile it |
| 22:10.49 | CrashSys | Trying to look up instructions to use gdb to find out |
| 22:11.44 | igcewieling | did you change the CPU or was it compiled on a different VM. |
| 22:12.14 | CrashSys | It was compiled by a repository that builds multiple architectures |
| 22:12.39 | CrashSys | Similar do building a DEB or RPM package |
| 22:12.50 | igcewieling | compile it on the vm it runs on |
| 22:13.26 | CrashSys | That won't help others trying to update their system from the repository |
| 22:14.54 | CrashSys | I get this from strace: ILL_ILLOPN Illegal operand |
| 22:15.03 | CrashSys | err --- SIGILL {si_signo=SIGILL, si_code=ILL_ILLOPN, si_addr=0x4ab2cf} --- |
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| 22:16.24 | vader- | drmessano so i got that tekSIP thing working... hehe... works exactly how i want it to and i was able to break the one line off from our SIP trunk... ran into a downside... TekSIP doesn't really handle voicemail so i have to figure out a solution... probably just going to roll FreePBX, run all my numbers into that and then connect to the freepbx with my |
| 22:16.24 | vader- | phone and then probably try and use zoiper on my cell phone attached to freepbx to handle all the different numbers... |
| 22:36.18 | drmessano | Whatever, dude |
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