IRC log for #asterisk on 20191007

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01:43.44jmordicaHi all, I know 11 isn’t supported anymore but we are not able to upgrade yet, however I’m trying to take advantage of connection pooling using the newest UnixODBC 2.3.7 (which I have working and Pooling = Yes in my odbc config). Do I need to also set pooling = yes I’m res_odbc.conf or can I just rely on unixODBC to handle the pooling without
01:43.45jmordicaadditional asterisk config?
01:46.14jmordicabasically I’m wondering if by setting pooling = yes in res_odbc.conf that means asterisk is handling pooling instead of unixODBC. I’d rather have odbc handle it
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09:11.07javi404is there something better than freepbx as an easy to setup asterisk option?
09:12.36wraytheDepends on your requirements @javi404. FreePBX is relatively easy to set up though
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10:03.50wraytheAre there any major changes planned for 17?
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10:17.49filesome ARI improvements, some video improvements, some internal tweaks
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13:46.28*** topic/#asterisk by file -> Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.1 (2019/09/05) 16.5.1 (2019/09/05); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22
13:54.46SamotDoes the March survey still need to be in the topic?
13:56.50filegood point
13:57.20*** topic/#asterisk by file -> Join us at AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.1 (2019/09/05) 16.5.1 (2019/09/05); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22
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14:47.35igcewielingHmmm...our techs arrived on-site before noon.  today might be a good day to buy a lottery ticket.
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15:36.55pyro25hey everyone !
15:37.32pyro25for asterisk 13, because res_pjsip didn't support SIPADDHEADER in ARI (to add headers), a patch was created so we can use PJSIPADDHEADER
15:38.10pyro25I tried in asterisk 16 what it was meant to be from the start: setting PJSIP_HEADER(add,P-Header) in POST variables
15:38.28pyro25so far, without success
15:38.59pyro25I started looking to see if the *13 patch could be applied to *16, I could do it but the code changed quite a bit
15:39.14pyro25so before I dig this, I'm wondering if anyone have some news :P
15:47.18SamotSIPADDHEADER will never work on PJSIP regardless of your Asterisk version. It only works on Chan_SIP channels.
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15:50.36seanbrightpyro25: are you using PJSIP_HEADER on the correct channel?
15:50.41seanbrightyou need to use it from a pre-dial handler
15:50.46seanbrightnot before a call to Dial
15:50.52Samot^^ There is that too.
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15:51.32seanbrightpyro25: https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER
15:52.00seanbrightthe formatting on that page is awful, but look for the section that starts with "If you call PJSIP_HEADER in a normal dialplan..."
15:52.32seanbrightit shows how to use a pre-dial handler
15:53.12Samotpyro25: You'll need to pass the variables you want in the headers from the ARI POST that can be passed in the b() option as arguments in the GoSub.
15:53.26seanbrightdidn't mean to link to the asterisk 12 documentation, but it should be the same for 13 & 16
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15:57.47pyro25yeah, I'm working on the ARI POST arguments, it feels it would be so much easier to be straight in the dialplan ^,^
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15:58.31pyro25Samot: I don't understand the GoSub part you are talking about
15:59.02seanbrightpyro25: you'd have to look at the link i recommended to understand
15:59.22pyro25thanks mate
15:59.22SamotI was adding to seanbright's comments about the pre-dial handler.
15:59.36MLCWhat does the ChanIsAvail() command check for a PJSIP channel? Doesn't seem to check the registration.  I don't see any SIP traffic when I execute that.
15:59.46SamotThe b() DIAL option executes a GoSub.
16:00.11[TK]D-FenderMLC, is a channel is in progress for it or not.
16:00.18SamotMLC: Registration has nothing to do with channels.
16:00.23pyro25aah, right
16:00.50pyro25but then, I doubt I have a similar Dial() / b() handler using ARI originate ?
16:01.06SamotWhat are you originating?
16:01.12SamotA call to the PBX that stays in the PBX?
16:01.26SamotOr a call to the PBX that dials another device/endpoint?
16:01.37pyro25it dials another endpoint
16:01.42pyro25then bridges them
16:02.13SamotOK, so then the Originate is going to Dial() another channel.
16:02.18pyro25yep
16:02.24SamotSo when you Dial() that new channel you want to use the b() option.
16:02.37SamotTo apply the headers on the new channel the Dial() is creating.
16:03.01MLC[TK]D-Fender and Samot: I'm adding some SIP lines in test mode before porting our toll free number.  In my currently outbound call logic I use ChanIsAvail to see if a PRI line is available, and if not I use a POTS line. Will that same logic work for trying the PJSIP lines first?
16:03.31seanbrighti think the issue is that ARI /channels is doing the dial for you
16:03.35seanbrighthe's never actually calling Dial
16:03.48SamotIt's an Originate.
16:03.49pyro25seanbright: correct
16:04.00pyro25there's no Dial() in ARI, only POST variables
16:04.05SamotThe channel needs a Context,Exten,Priority or an App
16:04.05seanbrightso you'd want to post variables in the body of the request
16:04.16pyro25yeah
16:04.19SamotThat's why I said he needs to send the headers as vars.
16:04.31pyro25the docs are so fuzzy on that part
16:04.50seanbrightsomething like...
16:04.50seanbright{
16:04.50seanbright<PROTECTED>
16:04.50seanbright<PROTECTED>
16:04.52seanbright<PROTECTED>
16:04.55seanbright<PROTECTED>
16:04.58seanbright}
16:05.07seanbrighti _think_
16:05.23SamotThat's going to apply it on the channel the originate spawns.
16:05.28SamotWhich is in the PBX.
16:05.37seanbrightwhich is correct, no?
16:05.46SamotUntil something else is done. If it's calling another endpoint it's spawning a new channel.
16:06.08seanbrightah, ok
16:06.10SamotWell, I would need to see the full POST.
16:06.19SamotTo see how this other channel is being called.
16:06.26seanbrightyes.
16:07.03SamotBecause the endpoint could be PJSIP/Alice or it could be Local/test@outbound\n
16:08.18pyro25seanbright: my code looks pretty much like your's
16:08.19pyro25let
16:08.22pyro25let's see
16:13.09pyro25body:
16:13.28pyro25that returns a 200 OK
16:15.01pyro25lol, I think the answer to this question is going to be in C :P
16:15.02pyro25https://github.com/asterisk/asterisk/commit/c6d755de11e98c1f6f33b1b35db9725f64f72053
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16:15.12pyro25I just fell on that commit about ARI and channel variables
16:15.34seanbrighti don't follow
16:15.42seanbrightyou didn't show us the post body
16:16.10pyro25yeah, look at "body:"
16:16.20seanbright?
16:16.29seanbrightthere is nothing after "body:"
16:16.34seanbrightit is literally "body:"
16:16.37pyro25<--- ARI request received from: 172.29.0.4:46518 --->
16:16.40pyro25originator: 1570464686.28
16:16.44seanbright~pb
16:16.44infobotpastebin is probably 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.
16:16.51seanbrightpastebin your post body
16:16.55pyro25ha, oops
16:17.13pyro25is it a /pb command ?
16:17.25seanbright...
16:17.33seanbright~pb
16:17.33infobotrumour has it, 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.
16:17.40seanbrightread this ^^^
16:18.03seanbrighti think we're poorly communicating
16:18.09seanbrighti want you to go to one of those URLs
16:18.22seanbrighti want you to paste your POST body there
16:18.25seanbrightit will generate a URL
16:18.28seanbrightthen you paste the URL here
16:18.48seanbrightplease do that now
16:19.16pyro25only the last one seems to work
16:19.20pyro25first links are dead
16:20.32pyro25https://pastebin.com/RPQrWaPk
16:20.42pyro25ha, works :-)
16:20.44seanbrightyes, that is wrong
16:21.25seanbrighthttps://pastebin.com/raw/M3b6wsLp
16:21.29seanbrightit needs to look like that
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16:26.19pyro25yes !
16:26.36seanbrightyes.
16:26.40seanbrightno C required
16:26.49pyro25:b
16:27.01pyro25it's a bit odd in ari-py then
16:27.14pyro25variables={"variables" : {"PJSIP_HEADER(add,P-Header-Id)": "1234567890"}}
16:27.29pyro25I don't know if there's another use for variables
16:27.37pyro25either, I'm glad I don't need a compile for this
16:27.39pyro25thanks mate !
16:27.47pyro25either way*
16:29.12seanbright>  Latest commit
16:29.12seanbright1647d79
16:29.14seanbrighton Dec 19, 2014
16:29.32seanbrightassuming you're using github.com/asterisk/ari-py
16:29.38seanbrightmight be time to find a better library
16:30.20pyro25yeah, it got unmaintained for quite a while
16:30.30pyro25everyone has had to fix python3 support on their own
16:31.13pyro25it's the last commit, but I remember it doesn't work
16:34.36pyro25do you use another yourself ?
16:34.45seanbrighti don't use ARI
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17:24.21seanbrightsigh
17:24.58seanbrightignore that - wrong channel
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17:36.05DovidDoes astdb slow down if I add to it 200k endtries?
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17:42.42SamotNo.
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17:49.46seanbrightmaybe
17:53.24sibiriawhat's the cause of 16.5.1 spamming a lot of errors from the module loader regarding pjsip, despite res_pjsip.so and chan_pjsip.so being "noload" in modules.conf?
17:53.29SamotWell it would be like any other SQLlite database.
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17:53.37sibiriai can't recall this being the case in 16.2
17:54.09SamotAre you noloading all the other PJSIP modules?
17:54.29SamotUnlike chan_sip.so, there are numerous pjsip_*.so modules.
17:54.40sibiriano. using the same config as in 16.2, where none of this is spammed when i noload chan_pjsip and res_pjsip
17:54.42Samoter res_pjsip*.so
17:55.26sibiriait's as if 16.2 keeps silent when the dependency is set to "noload"
17:56.39sibiria16.3*
17:57.53sibirianot an issue, really...
18:02.47Stefan26Sangoma Technologies Corporation bought Digium?
18:03.08sibiriaStefan26: about a year ago
18:03.10fileyes, like... yeesh... a year ago?
18:03.16filemaybe bit more, I forget
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18:05.06sibiriacan modules.conf glob?
18:05.13sibiriaie. noload => res_pjsip*
18:05.58sibiriaseems like it
18:09.08sibiriawarps the error spam to a single warning "Some non-required modules failed to load"
18:33.15seanbright~pb
18:33.15infoboti guess 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.
18:33.33seanbrightwhoever has infobot access should remove the bad ones (the first 3 are dead)
18:33.47seanbrightpastebin.com works
18:37.26Stefan26does anyone have practical experience of running asterisk dockerized?
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18:39.10bluejaypophello team, I'm trying to get the BUSY SIP status, when somebody rejects the call. How can it be done in dialplan ? I am reading about AST_DEVICE_BUSY status.
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18:47.47[TK]D-Fenderbluejaypop, "DIALSTATUS"
18:47.57[TK]D-Fenderthe basic channel variable....
18:53.13bluejaypop<PROTECTED>
18:56.09bluejaypopyes it works, you're a life saver.
19:01.15Stefan26yeah if you don't set the right busy status, someone might die!
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20:57.01sibiriamedia_cache.c is incredibly debug-spammy in 16.5. it throws two repeated debug messages aout "Failed to obtain media at" for exactly everything that does exist
20:57.22sibiriaeven for the path monitor() wants to record audio to
20:57.36sibiriathing flew off the handle
21:06.54sibiriahttps://pastebin.com/raw/A62fnsDZ
21:06.57seanbrightdebug is spammy by definition
21:07.05sibiriaright but should it be doing this?
21:07.48sibiriai can get it trying various file formats for BackGround/Playback and being chatty about that, but for Monitor()?
21:08.15filesame API, same behavior
21:08.17seanbrightif you feel there is a bug, you can report the issue on https://issues.asterisk.org/jira
21:08.29seanbrightbut as far as i know, that code has not changed in recent releases
21:15.22sibiriai guess it's not really a bug. it probably just isn't aware of the access mode when being fed part of what whatever ends up in an fopen() or whatever
21:15.47sibiriabut repeating itself seems a bit odd
21:16.02seanbrightalso, why are you using Monitor?
21:16.05seanbrightit's 2019
21:16.18sibiriaold dialplan, it's being moved to mixmonitor :)
21:19.54seanbrightso ast_fileexists() looks for the file twice
21:20.26seanbrightactually, potentially multiple times
21:20.52seanbrightso it's not a bug, it's an implementation detail
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