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| 00:22.38 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:22.38 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); 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 |
| 03:58.41 | *** join/#asterisk jeffspeff (~Jeff@209.141.208.197) |
| 04:06.06 | *** join/#asterisk dbdave303 (~dbdave303@123.200.135.178) |
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| 04:06.52 | dbdave303 | Hi there - anyone had any luck with unsolicited mwi inbound from a pjsip trunk (office 365 is what I'm playing with) ? |
| 04:11.46 | Samot | Please restate your question as follows: "I am trying to accomplish ________. I have done ________, expecting ________ to happen, but am seeing ________." |
| 04:12.17 | dbdave303 | sure thing Samot |
| 04:17.58 | dbdave303 | I am trying to accomplish unsolicited WMI via NOTIFY/message-summary -> Messages-Waiting message delivery from a pjsip trunk (in this case office 365). I am unsure if this is even supported. I have pasted sip trace - https://pastebin.com/HUmUELbb. I am expecting a WMI state to be logged which will end up as a light on my desk phone via chan-sccp-b (confirmed working against chan-sip on another |
| 04:17.58 | dbdave303 | system), but am seeing a 501 not implemented being returned in response the the notify |
| 04:18.43 | Samot | Well.. |
| 04:18.51 | Samot | That's because chan-sccp-b |
| 04:19.11 | Samot | 501 is "I don't know what this is" |
| 04:19.14 | Samot | From the server. |
| 04:21.07 | dbdave303 | Message is being rejected inbound on the pjsip trunk before the device channel comes into the equation. Yes you are correct on the 501. I need to know if this is something pjsip supports before proceeding any further |
| 04:21.12 | Samot | [2017-07-05 02:37:03] DEBUG[7258]: res_pjsip/pjsip_distributor.c:266 find_dialog: Could not find matching transaction for Request msg NOTIFY/cseq=191 (rdata0x7fbbe019e120) |
| 04:21.16 | Samot | From the PJSIP side... |
| 04:24.50 | Samot | So that NOTIFY is coming in from Office 365? |
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| 04:28.32 | dbdave303 | correct - NOTIFY on MWI in coming from office365 |
| 04:29.22 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 04:29.30 | Samot | I don't think Asterisk expects NOTIFYs |
| 04:29.40 | dbdave303 | I previously did this successfully via chan_sip from MsExchangeUM (unsolicited MWI) but our business is moving to 365 |
| 04:29.42 | Samot | Since Asterisk would be the one to generate them generally. |
| 04:30.15 | Samot | Did you look in the pjsip.conf.sample to see how it's done via PJSIP |
| 04:30.41 | dbdave303 | as you've said - only doco I can find is about the sending of mwi, not ingress |
| 04:31.22 | Samot | res_pjsip: The functionality for outbound SIP subscription or handling of unsolicited NOTIFYs inbound to Asterisk is not available in res_pjsip yet. Internal infrastructure is built that would allow it, so if this is something you want to work on, please contact the Asterisk development community. |
| 04:31.32 | Samot | ^^ https://wiki.asterisk.org/wiki/display/AST/Message+Waiting+Indication |
| 04:31.34 | dbdave303 | sip notify can be used for all sorts of things - is there any way to intercept the message and pump it out to a script or something? |
| 04:31.54 | Samot | Now that was updated in 2014, not sure if that has changed in current versions and the Wiki is just lagging a bit with the changes. |
| 04:32.32 | Samot | Yes, NOTIFY is used to update subscribers. |
| 04:32.54 | Samot | So generally, as I said, Asterisk is the one to generate NOTIFYs |
| 04:33.08 | Samot | For voicemail, presence, text, etc. |
| 04:33.29 | Samot | NOTIFY can be used for keep-alives. |
| 04:33.35 | dbdave303 | right - but in a trunk example between two systems, notify is a two way street, potentially I can have subscriptions on both sides |
| 04:33.49 | Samot | Not with Chan_PJSIP |
| 04:33.54 | Samot | Unless something has changed. |
| 04:34.11 | dbdave303 | but is there any way to tell pjsip to listen for a message type and call something that I can deal with? |
| 04:34.19 | Samot | No. |
| 04:34.23 | Samot | Hence the 501 |
| 04:34.28 | Samot | You need to use Chan_SIP |
| 04:34.34 | Samot | If you are going to do this. |
| 04:35.16 | dbdave303 | No worries, I'll get cracking on that - I initially went to pjsip for some additional debug to get through a crypt issue I had prior to hitting this one - which I'm through now, so I'm confident I could get chan_sip to do this |
| 04:35.46 | Samot | chan_sip: outbound MWI subscriptions and receiving unsolicited MWI NOTIFY messages |
| 04:36.15 | Samot | I guess you can look at res_external_mwi for PJSIP |
| 04:36.44 | Samot | It's an API for remote systems to update Asterisk |
| 04:36.45 | dbdave303 | that's API, not through NOTIFY right? |
| 04:36.49 | Samot | yes. |
| 04:36.59 | dbdave303 | would be an option if the transport wasn't encrypted |
| 04:37.02 | Samot | Because like I showed in the scroll, it's not supported for PJSIP |
| 04:37.11 | Samot | No. |
| 04:37.16 | Samot | I linked the Wiki page |
| 04:37.28 | dbdave303 | No.? |
| 04:37.30 | Samot | I pasted the "PJSIP don't support this" section.. |
| 04:37.48 | Samot | PJSIP does not support external inbound NOTIFYs. |
| 04:38.00 | dbdave303 | yes right - if the transport wasn't encrypted I could listen on the wire for that notify type and poke the res_external_wmi api |
| 04:38.10 | Samot | No. |
| 04:38.16 | Samot | The API removes the need for the NOTIFY |
| 04:38.20 | Samot | That's the point |
| 04:38.20 | dbdave303 | understood |
| 04:38.24 | dbdave303 | but I can't control office365 |
| 04:38.30 | Samot | Right |
| 04:38.33 | Samot | So like I said... |
| 04:38.35 | Samot | Chan_SIP |
| 04:38.46 | Samot | That is what you need to use with Office365 to do this. |
| 04:39.12 | dbdave303 | thanks for your help - I'll go down the chan_sip route again |
| 04:41.20 | Samot | Np. |
| 04:42.19 | drmessano | np |
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| 05:51.43 | dbdave303 | Samot - have MWI working correctly in chan_sip - looks like you're right, limitation of res_pjsip |
| 05:52.15 | drmessano | dbdave303: You may get away with reporting it as a bug |
| 05:52.23 | drmessano | Since it works with chan_sip |
| 05:52.46 | drmessano | But they don't take feature requests, and if it's classified as such, stick with chan_sip, I guess |
| 05:53.11 | dbdave303 | bit of a shame - was really liking the structure and usability of pjsip |
| 05:53.30 | dbdave303 | it's a bit of a stretch to call it a bug, definitely not on the pjsip feature list |
| 05:59.21 | lorsungcu | There's a meme for this |
| 05:59.31 | lorsungcu | Can't call it a bug if it was never a feature |
| 06:05.52 | drmessano | dbdave303: Not so much |
| 06:06.01 | drmessano | It's clearly missing from the implementation |
| 06:06.09 | drmessano | and was supported by chan_sip |
| 06:06.25 | drmessano | Since chan_pjsip is the SIP channel driver moving forward |
| 06:06.34 | drmessano | Not a stretch to consider it a bug |
| 06:06.52 | drmessano | SIP is SIP.. you shouldn't have to be mindful of the channel driver |
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| 06:15.31 | dbdave303 | No issue at all, I'm happy to log it, I'll get something put forward over the next few days |
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| 06:28.43 | [TK]D-Fender | <lorsungcu> Any time there's bandwidth constraints <- same BW. If there's a PLC hit vs G.711 that'd be a separate thing, but raw BW = the same |
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| 06:58.08 | FarhaadN | call not hangup on digium dahdi card |
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| 07:00.06 | drmessano | FarhaadN: fxsls or fxsks? |
| 07:01.10 | FarhaadN | drmessano: fxsks |
| 07:01.43 | FarhaadN | what configuration need for that? |
| 07:01.56 | FarhaadN | https://wiki.freepbx.org/display/PC/Analog-+Hangup+Detection+Issues |
| 07:02.00 | FarhaadN | is this ok? |
| 07:03.40 | FarhaadN | https://pastebin.com/pzzRZt6a |
| 07:03.46 | FarhaadN | my configuration |
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| 07:08.50 | FarhaadN | add below command fix it : |
| 07:08.50 | FarhaadN | busydetect=yes |
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| 13:42.31 | Kobaz | [TK]D-Fender: poke |
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| 15:43.20 | khronos | Anyone work for any ITSP's in here?? |
| 15:43.48 | Samot | Are you taking a poll? |
| 15:43.50 | Samot | ?polls |
| 15:43.56 | Samot | ~polls |
| 15:43.56 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 15:45.59 | khronos | Samot: Are you asking me?? |
| 15:46.21 | Samot | Yes. |
| 15:47.04 | khronos | Samot: No, not at all. I'm wanting some advice on platform design. |
| 15:47.32 | Samot | And? |
| 15:47.48 | Samot | waits for an actual question with context. |
| 15:48.21 | khronos | Samot: Currently we are using Bicom for multitenant pbx services and we are feeling we have out grown what they can do for us. |
| 15:49.00 | [TK]D-Fender | Well we've already switched market segments |
| 15:49.09 | khronos | Samot: What switch venders should we be looking at after we have say 300 customers with 3500 to 4000 endpoints? |
| 15:49.24 | Samot | Well.. |
| 15:49.37 | Samot | Like TK just pointed out, you've already switched markets. |
| 15:49.59 | igcewieling | khronos: I've never heard of anyone else using Bicom. We use them. |
| 15:50.51 | Samot | khronos: What does the switch need to do? |
| 15:50.54 | igcewieling | I don't like them very much. They won't even provide support unless it is installed on Gentoo. |
| 15:52.09 | Samot | khronos: Actually, you probably need to hire someone that understands all this. |
| 15:52.20 | Samot | "Outgrown what they can do for us" is vague. |
| 15:52.23 | khronos | Samot: Provide multiple pbx systems on the same server, endpoint deployment, |
| 15:52.39 | Samot | You need to bring someone in. |
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| 15:53.05 | Samot | This isn't something that can be designed over IRC with random people that have no idea about your entire platform, needs or desires. |
| 15:53.07 | khronos | Samot: Basically their system seems to work well if you are keeping everything on one physical server, once you get past that it doesn't scale all that well. |
| 15:55.36 | igcewieling | Think of Bicom as a commercial FreePBX-like system which doesn't use FreePBX. |
| 15:56.34 | igcewieling | khronos: I'd try optimizing/upgrading the hardware instead of switching GUIs, |
| 15:56.43 | [TK]D-Fender | I used ScopServ at their start |
| 15:57.10 | [TK]D-Fender | They were targeted for large deployments |
| 16:00.40 | khronos | Thanks. |
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| 17:33.52 | lorsungcu | i've got a freepbx machine i'm having PJSIP troubles with. Not getting any response from that universe so i'll try here. |
| 17:34.36 | lorsungcu | asterisk 13.16 machine behind NAT, registering a remote endpoint also behind NAT using TLS |
| 17:35.14 | lorsungcu | registration works, but asterisk uses the local address of the endpoint for audio, regardless of options supplied |
| 17:36.25 | lorsungcu | here's the transport settings: |
| 17:36.29 | lorsungcu | https://www.irccloud.com/pastebin/8yGT0YpZ/ |
| 17:36.42 | lorsungcu | and the endpoint: |
| 17:36.44 | file | there was a bug in 13.15.0 and up which caused the local address to potentially not get swapped out, if you roll back does it resolve the problem? |
| 17:36.45 | lorsungcu | https://www.irccloud.com/pastebin/PxiKSWrN/ |
| 17:36.56 | lorsungcu | file: i can try. do you have a link to the bug? |
| 17:37.08 | file | https://issues.asterisk.org/jira/browse/ASTERISK-27024 |
| 17:37.51 | lorsungcu | thanks. i'll get a lab built and test it. do you see anything glaringly incorrect in this configuration? |
| 17:39.23 | file | no |
| 17:46.31 | *** join/#asterisk matrix1233 (~matrix123@41.230.43.114) |
| 17:47.17 | salviadud | Will pjsip eventually replace chan_sip for good? |
| 17:47.38 | file | there's no plans to remove chan_sip itself, and it has features that chan_pjsip does not |
| 17:49.04 | salviadud | I have to run chan_sip with a patch for cisco phones |
| 17:49.29 | salviadud | pjsip doesn't have that..., that's what I'm asking. |
| 17:49.37 | salviadud | why*, sorry |
| 17:49.45 | lorsungcu | salviadud: there's an easy solution to those problems |
| 17:49.55 | lorsungcu | you simply never use cisco phones |
| 17:49.57 | lorsungcu | ezpz |
| 17:50.24 | salviadud | hands lorsungcu a medal |
| 17:50.32 | lorsungcu | graciously accepts |
| 17:50.52 | lorsungcu | they do work well as wheel chocks, though. |
| 17:50.54 | salviadud | hehe, I would've bought digium phones. |
| 17:51.08 | salviadud | But they already had a bunch of ciscos without a cucm license. |
| 17:51.48 | lorsungcu | you can always sell them to [TK]D-Fender |
| 17:52.05 | [TK]D-Fender | $5 each, sure |
| 17:52.26 | salviadud | You might say they are crap, but, they kinda work. |
| 17:52.31 | lorsungcu | :D |
| 17:53.00 | salviadud | Asterisk does wonders imo |
| 17:56.59 | Samot | salviadud: The patch is third party, perhaps request he does a pjsip patch as well... |
| 17:57.11 | Samot | So like when Asterisk updates break his patch.. |
| 17:57.17 | Samot | You have to wait for him to fix it |
| 17:57.52 | salviadud | Samot, I am aware that upgrading will break chan_sip |
| 17:58.01 | Samot | No. |
| 17:58.06 | Samot | It doesn't break Chan_SIP. |
| 17:58.13 | Samot | It breaks the patch that you're using on Chan_SIP |
| 17:58.16 | Samot | There is a difference. |
| 17:58.36 | salviadud | Right, the patch won't work anymore |
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| 17:59.09 | salviadud | I'm currently running 11.25.1 |
| 17:59.22 | Samot | Well PJSIP isn't even available. |
| 17:59.37 | salviadud | for that version' |
| 17:59.55 | Samot | Right. |
| 17:59.59 | salviadud | is running more thirdy-party software on * |
| 18:00.07 | salviadud | I had to compile chan_sccp |
| 18:00.17 | salviadud | I have one of those sccp only devices... |
| 18:00.24 | salviadud | skinny doesn't cut it. |
| 18:00.30 | drmessano | Why are we talking about Cisco phones |
| 18:00.37 | drmessano | They're horrible. |
| 18:00.48 | salviadud | They are horrible to configure |
| 18:00.54 | salviadud | That I can agree 100% |
| 18:01.02 | drmessano | No, they are horrible to use |
| 18:01.18 | drmessano | Cisco phones are fine for 2003 |
| 18:01.25 | drmessano | But only 2003 |
| 18:01.30 | salviadud | Yeah, they're old |
| 18:01.40 | salviadud | I have a bunch of 7942g's |
| 18:01.40 | drmessano | Extinct |
| 18:01.45 | drmessano | I'm sorry |
| 18:01.55 | salviadud | A few 9971s |
| 18:02.00 | drmessano | Hope you feel better soon |
| 18:02.05 | drmessano | Maybe smoke some weed |
| 18:02.13 | drmessano | Or sell the Ciscos |
| 18:02.36 | salviadud | My agents complain about the phones being slow as hell |
| 18:02.50 | salviadud | the 7942s to be precise. |
| 18:03.04 | lorsungcu | thatd be because they are |
| 18:03.14 | salviadud | I'd love to sell them, but whose buying? |
| 18:03.21 | drmessano | Maybe i am the dumb one |
| 18:03.35 | drmessano | But they're old as shit. Slow. Unsupported. |
| 18:03.42 | drmessano | Why is this still a thing |
| 18:03.55 | salviadud | Like I said previously |
| 18:04.01 | salviadud | where I work |
| 18:04.16 | salviadud | We had an old avaya that had a dead fxo port |
| 18:04.41 | salviadud | It was analogue, sometimes we'd get some horrible noise in the phone call. |
| 18:04.50 | salviadud | Limited phones, it was hell. |
| 18:05.09 | drmessano | Ok |
| 18:05.10 | salviadud | And, we had a bunch of ciscos without a cucm license. |
| 18:05.11 | drmessano | So |
| 18:05.22 | salviadud | So, I made the boss buy a tdm card |
| 18:05.51 | salviadud | And revived the cisco phones, because it was the cheapest way to upgrade our pbx |
| 18:05.58 | salviadud | And retire the avaya |
| 18:06.12 | drmessano | Worst way, you mean |
| 18:06.53 | salviadud | We rather have every agent have a phone, rather than they share |
| 18:06.54 | drmessano | The correct answer was "Oh shit, let's buy some Polycoms off eBay |
| 18:07.00 | lorsungcu | salviadud: it's difficult to measure lost efficiency |
| 18:07.08 | lorsungcu | or lost goodwill from your users |
| 18:07.46 | lorsungcu | i mean, do what works |
| 18:07.50 | salviadud | And, I sorta saved us money from buying new sip phones that I'm sure will work a lot better. |
| 18:08.10 | drmessano | Or used |
| 18:08.13 | salviadud | Maybe we should sell the ciscos to somebody that doesn't know much about telephony |
| 18:08.17 | drmessano | Ip33o |
| 18:08.26 | drmessano | IP330 |
| 18:08.30 | salviadud | And then buy some digium or polycom |
| 18:09.26 | salviadud | The pain behind cisco is configuration, running a tftp server, if you want a directory on the phone, run some apache, nginx or lighttpd |
| 18:10.10 | lorsungcu | your users dont see any of that, and they're still complaining. |
| 18:10.35 | salviadud | Just my "smart" users sorta complain about the speed |
| 18:10.38 | lorsungcu | heh |
| 18:10.52 | salviadud | The ones from sales just know they're old |
| 18:10.56 | lorsungcu | does new firmware allow call rejection? |
| 18:10.57 | [TK]D-Fender | salviadud> The pain behind cisco is configuration, running a tftp server, if you want a directory on the phone, run some apache, nginx or lighttpd <- no more than with jsut about any other phone really. |
| 18:11.00 | lorsungcu | like, to VM |
| 18:11.23 | [TK]D-Fender | From what I've heard the 2 biggist issues depending on version seem to be an issue with conferencing (which I didn't get to test), and presence |
| 18:11.37 | salviadud | I have voicemail enabled but on timeout |
| 18:11.47 | lorsungcu | salviadud: so there is no call rejection |
| 18:11.54 | lorsungcu | or ring-silencing |
| 18:12.06 | salviadud | call rejection can be enabled if you enable do not disturb |
| 18:12.16 | lorsungcu | yes but during an incoming call |
| 18:12.24 | lorsungcu | you cannot press "reject" and redirect the call to VM |
| 18:12.34 | lorsungcu | is that still the case? |
| 18:12.43 | lorsungcu | i know it was in like ~2010/2011 |
| 18:13.08 | salviadud | It might be possible, but I have to look into it |
| 18:13.39 | salviadud | Cisco uses a function called iDivert |
| 18:13.52 | lorsungcu | yeah. that isn't present whatsoever in SIP firmware |
| 18:13.55 | lorsungcu | unfortunately |
| 18:14.06 | salviadud | and you could divert a call to VM, but, I can't get it to work because of the xml files related to it |
| 18:14.28 | salviadud | So, it's more of a "cisco" problem |
| 18:14.37 | lorsungcu | oh, it's all a 'cisco' problem |
| 18:14.49 | salviadud | And conferencing is fixed with the third-party patch [TK]D-Fender |
| 18:15.02 | salviadud | ad-hoc conferences do work |
| 18:15.13 | salviadud | I guess it uses confbridge |
| 18:18.33 | salviadud | So, if you have to deal with cisco, you can, but I'm sure it would be better to use something else that's made after 2007 |
| 18:18.54 | [TK]D-Fender | salviadud, ew. Phone-based or bust |
| 18:19.36 | lorsungcu | yep thats nasty |
| 18:19.42 | [TK]D-Fender | I'm not as worried about ad-hoc (2 independent calls able to be joined a-la-polycom), if it has the basic "call out and bridge 3rd partly locally" like every other phone |
| 18:20.42 | [TK]D-Fender | But Polycom's "JOIN" is gold. Recently discovered that my 335 doesn't seem to support it even under Menu > Features while on call |
| 18:22.45 | salviadud | [TK]D-Fender, it has that. Lets say I dial someone in the office, and we want to add someone from the pstn. I just hit the conference button, which is a button that has a logo with 3 ppl in it. The one that is in the call gets placed on hold, I dial out, I can immediately join the channels by hitting conf again, or I can swap between them and place either one on hold. |
| 18:22.52 | lorsungcu | [TK]D-Fender: 335 doesn't let you join two calls? |
| 18:23.24 | [TK]D-Fender | lorsungcu, Nope.... |
| 18:23.28 | salviadud | Its the magic of the sip cisco patch |
| 18:28.31 | salviadud | And speaking of hold music |
| 18:28.48 | salviadud | I have that song from fight club called Ikea Man |
| 18:29.00 | salviadud | it loops and loops, it's just 54 secs |
| 18:29.19 | salviadud | But if I personally put you on hold, it's battletoads pause music |
| 18:29.33 | salviadud | I wonder if there is a limit to music classes |
| 18:29.45 | salviadud | I fear that if I had too many, my pbx will crash |
| 18:32.05 | *** join/#asterisk TeknoJuce (~nonasuomy@kodi/staff/TeknoJuce) |
| 18:34.40 | [TK]D-Fender | Sounds like this patch doesn't exist merged into the main... |
| 18:39.55 | file | an individual maintains it and keeps it updated on an issue |
| 18:39.59 | file | it modifies chan_sip heavil |
| 18:40.02 | file | erm heavily |
| 18:40.47 | [TK]D-Fender | let me guess... didn't want to sign the waiver to allow it to be merged? |
| 18:42.39 | *** join/#asterisk polysics (~polysics@88.147.59.127) |
| 18:42.56 | polysics | hello1 I got a weird thing going on with a machine |
| 18:43.09 | file | no, initial comments were provided that breaking it up would make it easier to review but I don't think there was a followup set of reviews |
| 18:43.15 | polysics | asterisk is running, but I get "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" |
| 18:43.20 | polysics | on asterisk -rvvvvvv |
| 18:43.43 | polysics | <PROTECTED> |
| 18:43.56 | polysics | has to be a permission thing but sudo should work, should it? |
| 18:44.24 | [TK]D-Fender | show us the file... |
| 18:44.32 | igcewieling | polysics: running freepbx? |
| 18:44.54 | polysics | no, plain old asterisk 13, fresh compiled install |
| 18:45.12 | polysics | using the Ansible playbook you can find around, but it does not do anyhing special |
| 18:45.26 | igcewieling | usually that error means Asterisk isn't running |
| 18:45.41 | polysics | root 1239 0.1 0.0 115248 932 ? S 18:41 0:00 /bin/sh /usr/sbin/safe_asterisk |
| 18:45.53 | igcewieling | that isn't asterisk, that is a shellscript. |
| 18:46.54 | [TK]D-Fender | also doesn't see the file he asked for... |
| 18:46.59 | *** join/#asterisk matrix1233 (~matrix123@41.230.43.114) |
| 18:48.22 | polysics | which file? |
| 18:48.25 | polysics | the .ctl? |
| 18:48.32 | [TK]D-Fender | <polysics> /var/run/asterisk/asterisk.ctl does exist |
| 18:48.37 | [TK]D-Fender | <[TK]D-Fender> show us the file... |
| 18:49.14 | igcewieling | *grumble* Amazon customer service hasn't improved in the 15 years since I last needed it. |
| 18:49.22 | polysics | [TK]D-Fender: it's 0 bytes and empty, "cat: /var/run/asterisk/asterisk.ctl: No such device or addres" |
| 18:49.55 | [TK]D-Fender | empty means no PID # means no process to connect to |
| 18:50.16 | [TK]D-Fender | and from before you've only shown a shell script for (re)starting * was running, not * itself |
| 18:51.13 | polysics | ok, so I should try running that script manually, maybe |
| 18:51.33 | [TK]D-Fender | That script says nothing about why * isn't running ... |
| 18:51.38 | [TK]D-Fender | like it CRASHING in a loop, etc |
| 18:53.13 | polysics | running via -c works |
| 18:56.07 | polysics | I guess I can look at what the startup script is doing |
| 18:58.00 | polysics | Unable to open Asterisk database '/var/lib/asterisk/astdb.sqlite3': unable to open database file |
| 18:58.12 | [TK]D-Fender | That's a pretty big fail.... |
| 18:58.38 | polysics | file is there though |
| 18:58.45 | polysics | rw-r--r-- |
| 18:58.52 | [TK]D-Fender | permission / condition |
| 18:59.11 | [TK]D-Fender | There != usable |
| 19:00.14 | polysics | still flapping, most probably on that |
| 19:00.27 | polysics | the file is there, readable, and is a valid sqlite3 db |
| 19:02.21 | [TK]D-Fender | <polysics> rw-r--r-- <- alone doesn't say anything |
| 19:06.51 | polysics | what can say more? |
| 19:06.54 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 19:10.37 | polysics | [TK]D-Fender: SELinux had to be disabled |
| 19:11.30 | [TK]D-Fender | Which has always been the case with * |
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| 23:50.54 | lorsungcu | file: moving to 13.14.1 resolved the audio issue i had. thanks agian. |
| 23:51.26 | file | lorsungcu: RC for next release should land tomorrow with actual release hopefully next week - that will include the fix |
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| 23:52.18 | Samot | I take it this is present in the 14.x couterparts? |
| 23:52.25 | lorsungcu | file: we're at sangoma's mercy, generally. usually i dont stay quite so up to date because of this sort of thing. |
| 23:52.40 | file | Samot: yes |
| 23:52.50 | file | lorsungcu: ah - yeah, I have no power over that |