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| 03:47.48 | snadge | i just stopped asterisk.. moved Master.csv to Master.csv.bak and restarted it |
| 03:47.52 | snadge | should it not recreate Master.csv ? |
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| 06:31.56 | snadge | so with auth by ip.. you can only use one ip address? |
| 06:32.17 | snadge | im kinda wondering if you can use allowguest=yes and permit/deny lines to work around that |
| 06:32.28 | snadge | or whether allowguest=yes is a global setting, not per trunk |
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| 07:12.22 | jkroon | why would asterisk set ODBCSTATUS to success if it get's a SELECT command denied to user from the underlying mysqld driver? |
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| 07:38.09 | Onyx47 | hey all, quick question: is there a way to manipulate which contact is dialed first when using PJSIP? basically, I want to dial a specific contact from queue if available, and dial the other one if not (the plan is to have them share the same AOR) |
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| 07:54.47 | file | No, there is not. Contacts within an AOR have no individually identifying information except the contact URI which is just how to reach something |
| 07:56.30 | Onyx47 | darn... well, thanks for the info, I can at least stop wasting time trying to get it working |
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| 09:24.50 | Onyx47 | Ok, so I went the other route and defined two AORs for an endpoint and I'm running into a weird issue... if I set max_contacts to 1 in each AOR trying to register a second device it errors out on the first one and tells me it will exceed the maximum |
| 09:25.59 | Onyx47 | if I set it to 2 both devices register to same AOR, which is not what I wanted since I intended to inspect DEVICE_STATE on AOR level, if that's even possible (some of the stuff on wiki suggests it is) |
| 09:26.08 | Onyx47 | it's like the second AOR gets ignored? |
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| 09:59.13 | dan_j | Hi. Is there any way to get voicemail to playback in reverse order? IE newest first? |
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| 10:15.37 | Tiffon | There is some cli command to disable SIP Trunk? |
| 10:17.17 | Tiffon | I want to turn on / off settings of a SIP trunk without using the web environment |
| 10:17.34 | Tiffon | Thank you |
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| 10:33.12 | mirela666 | Tiffon, I think you can use sip unregister trunk-name |
| 10:44.49 | Tiffon | ok mirela666 Perfect, thanks |
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| 11:12.18 | mirela666 | cheers |
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| 12:29.32 | [TK]D-Fender | Tiffon, Depends on you definition of "turn off" |
| 12:31.12 | mirela666 | true |
| 12:31.17 | Tiffon | disable trunk |
| 12:31.26 | [TK]D-Fender | Again that is vague |
| 12:31.46 | [TK]D-Fender | unregistering at best may stop the provider from sending you calls. |
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| 12:31.53 | [TK]D-Fender | it does nthing for outgoing calls from your side |
| 12:32.41 | [TK]D-Fender | Also, you'll still be able to match calls if they send them to you without your registering (which is rarer) |
| 12:32.49 | Tiffon | sorry for my English and asterisk, I am interested not attempt to register the trunk |
| 12:33.06 | [TK]D-Fender | If you're using a GUI... then use the GUI |
| 12:33.18 | [TK]D-Fender | That's what it's there for. |
| 12:34.33 | Tiffon | perfect with gui / web, I wanted to know do the same from cli. no problem thanks |
| 12:34.56 | wasanzy | getting this error: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Driver's SQLAllocHandle on SQL_HANDLE_HENV failed |
| 12:35.11 | wasanzy | when I tried to write cdr data to mysql |
| 12:35.44 | wasanzy | but in another case, is it the best way to go instead? http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql |
| 12:38.05 | Samot | No. |
| 12:38.15 | Samot | voip-info.org is out of date. |
| 12:38.15 | [TK]D-Fender | Tiffon, that command you were given earlier will work for chan_sip only to unregister an INCOMING registration, nothing to do with trunks |
| 12:38.31 | Samot | wasanzy: This a new install? |
| 12:38.57 | [TK]D-Fender | Tiffon, You will have to wait for your registration to expire, and remove if from your config |
| 12:39.04 | wasanzy | Samot: yes, I followed this: http://www.mikeslab.net/?p=363 |
| 12:39.34 | wasanzy | but I want to use this now? http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql |
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| 12:40.36 | Samot | Because that error is generally due to libraries not being found. |
| 12:41.03 | wasanzy | Samot: what library? |
| 12:41.05 | Samot | wasanzy: First, use the Asterisk wiki for this stuff. Someone's blog may not be covering the right version of Asterisk. |
| 12:41.09 | Samot | ODBC. |
| 12:41.47 | wasanzy | Samot: haven't found asterisk wiki for using obdc |
| 12:42.07 | wasanzy | should I use this? http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql |
| 12:42.26 | Samot | No. |
| 12:42.29 | Samot | For the third time. |
| 12:42.33 | Samot | It's out dated. |
| 12:42.37 | wasanzy | ok |
| 12:42.44 | Samot | You mean you couldn't find this? https://wiki.asterisk.org/wiki/display/AST/Getting+Asterisk+Connected+to+MySQL+via+ODBC |
| 12:43.10 | file | my kingdom for useful Confluence searching |
| 12:43.23 | Samot | That was google. |
| 12:43.34 | file | yeah |
| 12:43.46 | wasanzy | ok thanks |
| 12:43.52 | Samot | Much like how I found out what that error message means that he posted. |
| 12:44.00 | Samot | I Google'd it. |
| 12:45.11 | Samot | wasanzy: What does "cdr show status" or "odbc show all" output? |
| 12:45.14 | Samot | Both of those? |
| 12:45.30 | Samot | s/?/. |
| 12:46.00 | wasanzy | ok a min |
| 12:47.39 | wasanzy | https://paste.linux.community/view/2dc503cf |
| 12:48.39 | Samot | Well you definitely have an ODBC connection issue. |
| 12:48.53 | wasanzy | ok |
| 12:49.01 | wasanzy | /usr/lib64/libmyodbc5.so |
| 12:49.01 | Tiffon | Thank you [TK]D-Fender |
| 12:49.05 | wasanzy | I have that |
| 12:52.22 | Samot | Have you done: isql <DSN> <User> <Pass>? |
| 12:55.50 | wasanzy | I didn't pass user and pass |
| 12:55.56 | wasanzy | let me try that again |
| 13:01.34 | wasanzy | ah, am good now. |
| 13:01.45 | wasanzy | thanks for the support Samot |
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| 14:00.19 | *** join/#asterisk equilibrio (45465aca@gateway/web/cgi-irc/kiwiirc.com/ip.69.70.90.202) |
| 14:01.56 | equilibrio | Hello, I would like to know if there is a way to recompile only the VoiceMail module. I have installed the version 13.10.0 of asterisk and patched it with the Cisco Patch for the phones. The most recent one. Everything works well. But it is on a virtual machine and I heard its better to recompile while using the NATIVE BUILD_DISABLED option. The th |
| 14:01.56 | equilibrio | ing is that I use FreePBX with Asterisk and the way to upgrade and patch with the Cisco patch is to use rpm. What is the way to disable the Native build option with rpm? |
| 14:02.35 | equilibrio | As for the Voicemail module right now, I have an asterisk server working great, only thing missing is the voicemail, so I was wondering if there was a way to just re-install that module. If I try to pre-load app_voicemail.so it just doesn't work |
| 14:02.57 | equilibrio | I need to be able to re-compile selectively just the voicemail module, is it possible from the tat.gz file? |
| 14:03.06 | equilibrio | tar.gz I mean |
| 14:05.42 | WIMPy | Strictly speaking it's not impossible, but don't dother to tro to do mixed installations. It's unlikely to work. |
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| 14:12.19 | equilibrio | Hmmm |
| 14:12.24 | equilibrio | So its better I re-install then |
| 14:12.50 | equilibrio | because that VoiceMail trouble happened after the patch, but the the machine has been uploaded to a VMWare ESXi server and I have to downgrade compatibility |
| 14:13.21 | equilibrio | so I think its better I re-install the asterisk server and do the compatibility downgrade before installing a patch and rebuilding asterisk |
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| 14:39.42 | drmessano | equilibrio: are you using the FreePBX distro? |
| 14:44.29 | Samot | If this is the Cisco patch for FreePBX, it's only applicable to Asterisk 11. |
| 14:44.42 | Samot | You've stated you have Asterisk 13. |
| 14:49.16 | equilibrio | Samoa: no the guy made it for asterisk 13 now |
| 14:49.31 | equilibrio | Samot: sorry didn't mean Samoa, damn auto-correct :P |
| 14:50.06 | Samot | For 13.10? |
| 14:50.09 | equilibrio | yes |
| 14:50.11 | equilibrio | see here |
| 14:50.12 | equilibrio | http://docs.acsdata.co.nz/asterisk-cisco/patching-asterisk.shtml |
| 14:50.16 | equilibrio | have it running since months |
| 14:50.21 | equilibrio | he finally updated it |
| 14:51.26 | equilibrio | its just that when you build it, its better to use the info from the freepbx page about the cisco patch, the rpm way to upgrade and just change all the values to the version 13.10.0 |
| 14:51.29 | equilibrio | have a nice day! |
| 14:51.31 | Samot | Except he updated it for Asterisk. |
| 14:51.46 | Samot | ?!? |
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| 15:04.30 | Gaster | In context of attended transferring what is the difference between `transferdigittimeout` and `featuredigittimeout` options in features.conf? |
| 15:06.43 | WIMPy | The time you've got to enter the feature code and to enter the destination. |
| 15:07.23 | Gaster | Oh, right. Thank you! |
| 15:08.24 | Gaster | The wording in sample config's comments made this unobvious. |
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| 15:54.01 | Gaster | Are "=" and "=>" 100% equivalent in Asterisk .conf files? |
| 15:55.03 | jrun_ | insecure=port,invite , how do i translate that to pjsip? |
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| 16:06.57 | igcewieling | Gaster: no. But they seem to be in extensions.conf and maybe voicemail.conf |
| 16:06.58 | rmudgett | Gaster: Yes '=' and '=>' are exactly the same. The code does make note of which was used so if the config is saved later the same token is written. |
| 16:07.46 | igcewieling | rmudgett: port=5060 is the same as port=>5060 ? |
| 16:07.56 | rmudgett | yes |
| 16:08.00 | igcewieling | cool. |
| 16:08.34 | *** part/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 16:08.48 | rmudgett | In early asterisk their may have been an intention to treat them differently but nothing came of it. |
| 16:08.57 | rmudgett | s/their/there/ |
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| 16:13.50 | Gaster | Wiki says something about "Objects", but there is no explaination what they are. https://wiki.asterisk.org/wiki/display/AST/Objects |
| 16:14.06 | Gaster | :( |
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| 16:31.24 | equilibrio | Hello, I got a problem with my asterisk v13.10.0 with FreePBX installed. I get the VMBOXESISTSSTATUS=success |
| 16:31.33 | equilibrio | but... get that: pbx.c:2814 pbx_extension_helper: No application 'VoiceMailMain' for extension (from-internal, *97, 118) |
| 16:32.18 | equilibrio | I am trying to make it work since a while now, can't seem to be able to. Even from new install, after patching asterisk with v13.10.0 of the cisco patch, the voicemail doesn't work, even before the patch... sorry voicemail works but not voicemailmain |
| 16:32.19 | [TK]D-Fender | Then you have failed to load a module to provide the app |
| 16:32.36 | equilibrio | is there a way to enable to VoiceMailMain? because app_voicemail.so is loaded |
| 16:32.40 | [TK]D-Fender | Whoice app_voicemail normall provides |
| 16:32.41 | equilibrio | I can also show voicemail users |
| 16:32.55 | [TK]D-Fender | So maybe your patch broke something in the code |
| 16:33.00 | equilibrio | I know... that is what is strange. app_voicemail.so is loaded |
| 16:33.02 | [TK]D-Fender | And that one isnèt getting registered |
| 16:33.33 | equilibrio | app_voicemail.so Comedian Mail (Voicemail System) 0 Running core |
| 16:33.46 | equilibrio | the patch did not edit the voicemail module, that is why it is strange |
| 16:34.19 | equilibrio | is there a way to see a sub-module of a module. Like that app_voicemail.so that is supposed to provide with VoiceMailMain |
| 16:34.33 | equilibrio | VoiceMail is okay, just that freaking VoiceMailMain :P |
| 16:47.10 | igcewieling | equilibrio: IIRC they are the same module. |
| 16:47.48 | igcewieling | are you are 100% sure there are no extra spaces in the dialplan? |
| 16:54.55 | equilibrio | yes |
| 16:55.05 | equilibrio | I just tried core show application voicemail and it doesn't work |
| 16:55.12 | equilibrio | but the app_voicemail.so module is loaded |
| 16:55.21 | equilibrio | is there a way to just rebuild that voicemail application and load it? |
| 16:55.39 | equilibrio | I think there is a way to .configure and make menuselect only voicemail |
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| 16:59.06 | igcewieling | that only time I've seen that is when someone does something stupid like exten => same,n, VoicemailMain |
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| 17:05.27 | equilibrio | lol |
| 17:05.32 | equilibrio | well im getting that while im debugging |
| 17:05.36 | equilibrio | [2016-09-26 12:59:24] WARNING[8573] pbx_app.c: Already have an application 'VoiceMailMain' |
| 17:05.53 | equilibrio | then it unregisters the already running VoiceMailMain application + the one that started to run or something |
| 17:05.55 | equilibrio | this is weird |
| 17:08.15 | equilibrio | I am getting that too: app_voicemail_imapstorage.c: Couldn't find mailbox 1001 in context default |
| 17:08.33 | equilibrio | mailbox 1001 is configure for context default.. anyways |
| 17:10.14 | [TK]D-Fender | Sounds like you may be running multiple VM apps |
| 17:10.22 | [TK]D-Fender | which isnèt doable |
| 17:12.05 | equilibrio | yeah lol |
| 17:12.12 | equilibrio | but the main problem |
| 17:12.21 | equilibrio | is the fact that it doesn't show in core show application voicemail |
| 17:12.46 | equilibrio | the app_voicemail.so is loaded and shows in core show module but no voicemail or voicemail main in core show application |
| 17:15.10 | [TK]D-Fender | make sure you don't have any other related so's in your modules folder |
| 17:20.05 | igcewieling | sounds more and more like a bad voicemail.conf |
| 17:31.28 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
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| 18:15.07 | igcewieling | wonders if frying a veggie burger in lard is a sin. |
| 18:16.04 | equilibrio | dont think so :P |
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| 18:35.04 | *** part/#asterisk jrun_ (~jrun@unaffiliated/glphvgacs) |
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| 18:57.54 | pjensen00 | Uh oh. |
| 18:58.23 | pjensen00 | They ruin everything! |
| 18:58.34 | pjensen00 | I even have stickers printed. :( |
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| 19:00.27 | pjensen00 | I'm taking it back. I'm taking that phrase back. |
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| 19:41.09 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/09), 11.23.1 (2016/09/08), Standard: 14.0.0 (2016/09/26); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 20:50.10 | Kobaz | it's astricon2016 if anyone needs it.. found it on the schedule page |
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| 21:25.36 | *** part/#asterisk equilibrio (45465aca@gateway/web/cgi-irc/kiwiirc.com/ip.69.70.90.202) |
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| 23:25.26 | file | There are also signs around |
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