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| 01:43.50 | cptmorgan | is it possible to setup the voicemail for a phone number as a different phone numbers mailbox? basically want to share the same voicemail for two phone numbers |
| 01:53.27 | Samot | Sure, you're just pointing to the mailbox in the dialplan anyways. |
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| 02:52.17 | igcewieling | cptmorgan: "core show application voicemail" |
| 02:52.30 | cptmorgan | Samot: not as an extension. I want two different numbers to ring but both numbers go to the same mailbox |
| 02:53.29 | Samot | cptmorgan: I understood. What I said holds true. |
| 02:53.52 | Samot | cptmorgan: You can send calls from both numbers to the same mailbox |
| 02:55.40 | cptmorgan | Samot: can you point me to some documentation? im new to asterisk and took over managing a office with an old asterisk server |
| 02:57.46 | Samot | wiki.asterisk.org |
| 02:58.38 | Samot | https://github.com/asterisk/asterisk/tree/master/configs/samples |
| 03:04.01 | cptmorgan | exten => 1234567890,1,Voicemail(u${different_voicemail}) ? something like this |
| 03:09.11 | Samot | Well the u option comes after. |
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| 03:30.49 | cptmorgan | this looks like really old version of asterisk. im seeing an old config in here that show exten => 1234567890,1,Voicemail(u123456780). Is that older syntax or something? version 1.6.2.24 :/ |
| 03:34.29 | Samot | Oh man. 1.6 is beyond old and unsupported. |
| 03:34.54 | drmessano | Only like 12 years old or so |
| 03:40.03 | Samot | So old that I would have to refresh on things. |
| 03:42.05 | drmessano | So old, I told it to act its age and it segfaulted |
| 03:42.35 | cptmorgan | this old config has multiple sections. does it matter what section this is in? i could technically put this under [default] and it would be used? they have multiple sections under default. |
| 03:43.16 | Samot | This is real honest advice... |
| 03:44.02 | Samot | As they manager of this office, manage yourself an Asterisk person. |
| 03:44.14 | Samot | It will save you a lot of headache. |
| 03:44.46 | cptmorgan | fuck, i got reading to do. thanks for the direction :) |
| 03:45.24 | Samot | That version of Asterisk is really unsupported. |
| 03:45.38 | Samot | And wasn't 1.6.2 kinda buggy? |
| 03:46.28 | Samot | I mean if I recall, we skipped all of 1.6.x |
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| 04:09.52 | cptmorgan | Samot: well its been running for 11 years without a hitch :/ |
| 04:11.34 | cptmorgan | Samot: can you help me with one last thing? do i need to create a section for the peer the call is coming from? like [peer_name] and have the dialplan under that? |
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| 11:12.04 | altker128 | Hey guys. I the "97 to access voicemail "standard" Asterisk or is this something FreePBX inserted into a dial plan? |
| 11:12.10 | altker128 | *is the |
| 11:13.04 | file | there is no standard, it's completely up to the configured dialplan - which would be FreePBX |
| 11:15.07 | altker128 | From the Asterisk CLI how would I see whatever special *xx codes might be defined? |
| 11:15.14 | altker128 | Would it show up in show dialplan ? |
| 11:15.44 | file | yes. |
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| 11:35.05 | sibiria | when issuing 'core stop when convenient', asterisk will eventually stop waiting for calls to finish and just return with exit code 0 without terminating the main asterisk instance |
| 11:35.14 | sibiria | is there a way to control this behavior? |
| 11:37.34 | sibiria | and once it gives up, the main asterisk instance seems to forget what it was told to do |
| 11:37.48 | sibiria | (new calls can again start up) |
| 11:39.58 | sibiria | once it has given up, re-issuing 'core stop when convenient' will return instantly with exit code 0. none of the waiting that happens the first time |
| 11:40.15 | sibiria | as if the state is pending and lingering somewhere, despite not having any effect |
| 11:45.55 | sibiria | sorry, i mean "gracefully" |
| 11:55.22 | file | when convenient doesn't stop new calls |
| 11:55.30 | file | gracefully stops new calls |
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| 13:03.30 | igcewieling | any idea what might cause this? [2021-04-20 08:45:46] WARNING[6274]: db.c:304 db_execute_sql: Error executing SQL (COMMIT): database is locked |
| 13:03.48 | Samot | Your database is locked. |
| 13:03.57 | igcewieling | Where is the key? |
| 13:04.04 | Samot | Go look at the database. |
| 13:04.37 | igcewieling | They are pretty. astdb.sqlite3 |
| 13:04.53 | igcewieling | assuming that IS the database they are talking about. |
| 13:05.31 | Samot | I don't know. What was being done when that happened? |
| 13:05.50 | igcewieling | the most recent is playing voicemail |
| 13:06.47 | sibiria | sqlite is not write-concurrent between separate processes. if you're doing any of this for example in AGI scripts then you'll face this problem |
| 13:07.04 | sibiria | (assuming plenty of write access) |
| 13:07.05 | Samot | Well I'm guessing your sqlite db is messed up. |
| 13:10.38 | nbjoerg | and read-concurrent only if it is WAL-mode |
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| 13:12.42 | sibiria | for reads-only it is always fully concurrent across any number of processes, but WAL provides concurrency for processes that only read while another process is writing |
| 13:13.36 | sibiria | it's overall a better operational mode |
| 13:14.32 | Samot | igcewieling: you can try copying it to a backup. Thst will remove the lock. |
| 13:14.47 | Samot | The try copying it back over. |
| 13:14.54 | Samot | Then* |
| 13:19.33 | igcewieling | I stopped freepbx, copied astdb.sqlite3, deleted the original, then copied the copy to the original name, started freepbx |
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| 13:22.06 | Samot | Might need two restarts. Is it working? |
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| 13:24.50 | igcewieling | at this point, I'll have to wait until they complain. too many active calls now. |
| 13:29.07 | sibiria | you can try enable WAL on the database |
| 13:29.31 | sibiria | it's a permanent change that will be reflected onto each process accessing it |
| 13:31.09 | Samot | Well |
| 13:31.17 | Samot | Are you seeing lockup errors in the logs? |
| 13:31.23 | Samot | I mean, you did check that right? |
| 13:32.26 | seanbright | astdb has some known performance problems, so if anyone wanted to take a shot at fixing them, that would be great |
| 13:32.37 | seanbright | patches welcome, as the kids are saying |
| 13:34.45 | igcewieling | I've never head of WAL |
| 13:35.01 | igcewieling | I'm using freepbx. I'm not doing anything. |
| 13:36.05 | sibiria | you don't have to alter the application for it. it's a setting in the sqlite db file itself |
| 13:36.31 | sibiria | https://sqlite.org/wal.html |
| 13:38.22 | igcewieling | while there were no active calls, I dumped the databsse, deleted the original and recreated the db from the backup |
| 13:38.46 | Samot | And are the lock errors gone? |
| 13:39.56 | igcewieling | that did not work at all. |
| 13:40.23 | Samot | Well I said it could take up to two restarts. |
| 13:40.38 | Samot | So you're still getting the lock errors? |
| 13:41.52 | igcewieling | Sorry, I just fixed the "astdb is not a databse" error which was caused by the databsse I created by the restore. |
| 13:42.01 | igcewieling | now I'm waiting for lock errors |
| 13:43.16 | igcewieling | no errors so far, but no more calls yet |
| 13:44.06 | sibiria | is there a journal file stuck there? |
| 13:44.06 | igcewieling | A call came in and no error so far. |
| 13:44.18 | sibiria | i.e. a file with -journal suffix |
| 13:44.19 | igcewieling | sibiria: what might such a file be named? |
| 13:44.28 | igcewieling | nope |
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| 13:46.12 | igcewieling | still no more errors |
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| 14:04.14 | igcewieling | Samot: why two restarts? Does asterisk only fix the issue half way on the first startup? |
| 14:04.53 | Samot | Honestly, I really don't know. Not something I have to deal with a lot so I just did some quick googling. |
| 14:05.38 | igcewieling | *nod* I've never seen any issues with astdb on FreePBX since the switch to sqlite. |
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| 14:15.08 | sibiria | shutting down asterisk and anything else that may hold a file lock - and removing eventual journal/wal files relating to the astdb file - is all that it takes |
| 14:17.59 | nbjoerg | you shouldn't remove journal/wal files |
| 14:18.07 | nbjoerg | that in fact can corrupt the database |
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| 14:33.57 | sibiria | can happen only with a hot journal if sqlite crashes or is killed in the middle of it |
| 14:34.18 | sibiria | but that is a risk nevertheless |
| 14:35.14 | sibiria | corrupt data in the wal/journal, which sqlite will try to "replay" and then corrupt the db, is a bigger risk |
| 14:35.43 | sibiria | very tiny risk, but the technical possibility is there |
| 14:36.10 | nbjoerg | sibiria: if you remove the wal it will not be able to correctly rollback the pending transactions |
| 14:36.38 | Samot | Well, it could just be me but I have had astdb mess up more in pure Asterisk than FreePBX |
| 14:36.41 | nbjoerg | that's a much higher chance of data corruption than a process writing to the db writing garbage |
| 14:37.19 | sibiria | it goes without saying that you should never fiddle with the wal/journal files while something is operating on the db file |
| 14:37.48 | nbjoerg | the way to clean it up would be using the sqlite cli |
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| 14:52.01 | igcewieling | Samot: I don't really use astdb on non-freepbx boxes. |
| 14:52.44 | Samot | You sure? |
| 14:54.40 | igcewieling | I have 2 entries on "database show". One for pbx/UUID and my single registered device. All other devices are static IPs and don't register. |
| 14:54.52 | igcewieling | so, yes, I'm sure. |
| 14:55.16 | Samot | Well I say that because it's what is used by Asterisk. |
| 14:55.25 | Samot | So sure, you have a very minimal install. |
| 15:00.09 | igcewieling | My vanilla Asterisk boxes just route calls. |
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| 17:11.46 | THECobra | having an issue with an AGI script dialing a new call, the flow is - agi calls number1 and then we set some vars in the dialplan and then make an agi call to initialize the call within our system and then that same AGI script makes a second call to number2 when the call is answered it destroys the second call and sends it to the bridge which in |
| 17:11.46 | THECobra | turn invokes the channel hangup. The problem i am facing is i need to follow calls2 to the real hangup and then return the total call time variable this was a change within asterisk around version 11/12 this works as it is in 1.6 but not in 16 due to the change |
| 17:12.15 | THECobra | does anyone have any ideas on how to implement this so that i can see the totalcall time for call2 |
| 17:16.15 | Samot | I've read that three times and I'm still a bit lost. |
| 17:16.35 | igcewieling | sounds like you are calling an agi from inside another agi? |
| 17:19.19 | THECobra | sorry was trying to be as brief as possible |
| 17:20.07 | THECobra | ok so step on i have a perl script that checks for new calls that need to happen in a database if a call is there it uses AMI to originate a call |
| 17:21.28 | THECobra | then the dialplan sets variables and calls out to AGIÂ initialize the call internally so it sets what trunk to use and what callerid to set to the second call etc |
| 17:22.10 | THECobra | after the agi script does all that it calls a go sub which dials the second channel to number2 |
| 17:22.59 | THECobra | since gosub is not async it waits for the dial to be completed to move on in the agi script itself and then tries to collect the variables like totalcaltime |
| 17:24.07 | THECobra | $self->agi->get_variable("DIALEDTIME") is currently returning null because based on what i have read that channel essentially disappears once its put in the simple_bridge |
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| 17:24.28 | THECobra | so that var is returned null and the AGI script continues |
| 17:25.09 | THECobra | does that make more sense? |
| 17:25.30 | igcewieling | Set a shared variable containing the timestamp of the start of the call. |
| 17:26.08 | igcewieling | You can also set a hangup handler which might help. |
| 17:27.20 | THECobra | tried that, the issue is that when the dial plan dials and the end party answers it moves the call into the bridge and that original channel is i guess for a better word abandoned it will run the hangup hadler but the call is still on going |
| 17:28.06 | THECobra | even when i push a hangup handler to the channel it is not carried to the newly created channel that is in the simple_bridge |
| 17:28.26 | igcewieling | the my first suggestion might be better |
| 17:29.44 | THECobra | i can set the variable but not sure how i would be able to access it since the agi script moved past that point with a null variable since the dial completed |
| 17:31.12 | THECobra | when that second call is disconnected it doesnt run a hangup handler that i can set because once its moved to the bridge all of that is lost |
| 17:32.53 | THECobra | i even tried dynamically creating ConfBridge'sand moving calls into it but the result appears to be the same the call details for the second call are not accessible |
| 17:34.00 | igcewieling | "Implements a shared variable area, in which you may share variables between |
| 17:34.00 | igcewieling | channels.The variables used in this space are separate from the general namespace of the |
| 17:34.00 | igcewieling | channel and thus ${SHARED(foo)} and ${foo} represent two completely different |
| 17:34.00 | igcewieling | variables, despite sharing the same name." |
| 17:34.07 | igcewieling | hmm..that formatted poorly. |
| 17:36.28 | THECobra | https://pastebin.com/HZC4z5sF |
| 17:36.53 | THECobra | that is the dialplan the that the go sub for call2 uses |
| 17:38.56 | THECobra | the dial should stall the dialplan until hangup so the variables are available on hangup but since it this is a second call and the asterisk core moves it to a bridge it immediately calls the hangup with null variables |
| 17:41.27 | igcewieling | Sorry, I check things like DIALEDTIME in hangup handlers. |
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