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| 00:18.42 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12), Standard: 15.5.0 (2018/07/12); 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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| 03:10.57 | forgotmynick | Hello. Is anyone having trouble with google voice? I just noticed all of a sudden asterisk is showing itâs connected but not picking up any incoming calls. It rings and then goes to google voicemail. Itâs showing itâs connected and thereâs nothing in the log files (no new data is populated unless I ring non google voice numbers). Iâm also ringing from a non google number when testing it. Any thoughts? |
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| 04:29.07 | drmessano | forgotmynick: Yeah, it's dead |
| 04:29.14 | drmessano | chan_motif no longer works |
| 04:29.36 | drmessano | Google removed the ability to use Google Voice via XMPP |
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| 08:24.26 | kosc_job | Hello! How can I remove user (operator) from queue with Asterisk Manager API? |
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| 16:32.49 | DanFromUK | Hi. How can I maintain DYNAMIC_FEATURES on transfers? I've used __DYNAMIC_FEATURES but that doesn't appear to help. |
| 16:35.39 | DanFromUK | The call originates from a queue and agentA answers. DYNAMIC_FEATURES works for them. Then when they transfer to agentB, they are unable to use the DYNAMIC_FEATURES that were originally set. |
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| 16:46.18 | DanFromUK | I suppose my question is, is it possible to inherit the DYNAMIC_FEATURES variable upon attended and blind transfers? |
| 16:51.56 | DanFromUK | Seems to only be an issue with specific endpoints. eg. Bria Stretto seems to cause the issue |
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| 17:35.50 | Kobaz | this is something i occasionally run into... i don't recall how to deal with it |
| 17:36.05 | Kobaz | polycom soundpoint ip... make a mac-directory.xml file on the provisioning server |
| 17:36.28 | Kobaz | and the phone refuses to download it, until you go to the directory physically on the phone... add an entry, save and reboot |
| 17:36.39 | Kobaz | then now it will pull the mac-directory.xml file off the server |
| 17:37.04 | Kobaz | but before that you cannot get it to use the directory |
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| 17:39.58 | [TK]D-Fender | It only reads changed on reboot. |
| 17:40.07 | [TK]D-Fender | this is not a live manipulation kind of thing and never was |
| 17:42.54 | Kobaz | yes i know it's not live |
| 17:42.58 | Kobaz | only ldap would be |
| 17:43.00 | Kobaz | but! |
| 17:43.18 | Kobaz | the polycom does not attempt to download mac-directory.xml, until it has been once manually updated on the phone itself |
| 17:43.38 | Kobaz | you can reboot it all you like, it will never load the directory xml |
| 17:44.01 | Kobaz | this used to be a problem on vvx phones in early firmware versions... it's since been fixed starting in 5.6.0 i believe |
| 17:44.16 | Kobaz | i don't think they fixed it for soundpoint... i'm not sure if there's any workarounds |
| 17:46.40 | Kobaz | i wonder if there's some sort of <device> setting |
| 17:46.50 | Kobaz | that gets updated internally when you manually save a directory entry |
| 17:47.35 | igcewieling | I'm not aware of any workaround for the Soundpoint IPs either. |
| 17:49.46 | igcewieling | On recent firmware versions on VVX phones there is a setting voIpProt.SIP.specialEvent.checkSync.downloadDirectory="1" to fix the issue |
| 17:49.54 | Kobaz | ah |
| 17:50.23 | Kobaz | i didn't even have to set that |
| 17:50.28 | Kobaz | once you update to 5.6.0 it just works |
| 17:51.53 | igcewieling | It might default to 1, I force it to be set at all installs I do. The contact directory stupidity is one of the few things I really hate about Polycom phones. |
| 17:52.06 | Kobaz | yeah it's super dumb |
| 17:52.13 | Kobaz | i set up ldap once and it worked well |
| 17:52.20 | Kobaz | it's on a test system somewhere |
| 17:52.26 | Kobaz | i should get back to that |
| 17:52.56 | Kobaz | see |
| 17:53.05 | Kobaz | i just saved a direcory entry manuallyu |
| 17:53.10 | Kobaz | did a check-sync notify |
| 17:54.18 | Kobaz | and now it's downloading the directory... now i can update the xml to whatever i want and it'll pull it |
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| 17:57.19 | Kobaz | well not on check-sync.. but on reboot |
| 18:00.03 | Kobaz | i bought my first yealink desk ip phone |
| 18:00.16 | Kobaz | i've been using the ip-dect cordless set for a while, been pretty happy |
| 18:00.17 | DanFromUK | Regarding my issue, I can see that __DYNAMIC_FEATURES=NewRecordApp is set on the call with the Call ID in the logs of C-00002e08. But after transfer, when C-00002e08 dials the feature code, it doesn't work. Seemingly forgotten the feature's availability. |
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| 18:17.18 | DanFromUK | Ok, this only happens when calls are sent through a queue. Otherwise, it's fine. __DYNAMIC_FEATURES is set prior to executing Queue, but that doesn't help with transfers. |
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| 18:44.18 | tuxd00d | Kobaz: The Yealink T29G is really nice for an entry level color display. The desk phones are much easier to use than the W52H or W56H. |
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| 18:55.03 | Kobaz | DanFromUK: you'll need to hook into the masquerade to keep the dynamic feature going |
| 18:55.41 | Kobaz | __ inheritance isn't well defined for transfers and stuf |
| 18:55.58 | Kobaz | in theory it should pass through for dynamic feautres, sounds like it could be a bug |
| 18:56.35 | Kobaz | DanFromUK: for all that type of stuff like star codes, i wind up writing an AMI application |
| 18:56.36 | DanFromUK | Yeh. I posted on the mailing list following some tests. After attended transfer, DYNAMIC FEATURES is still set. It just doesn't work. |
| 18:56.50 | DanFromUK | What am I googling for regarding your hook suggestion? |
| 18:57.12 | Kobaz | what asterisk version? |
| 18:57.16 | DanFromUK | 13 |
| 18:57.30 | Kobaz | you would need to listen for transfer/masquerade type stuff in AMI or Stasis |
| 18:57.38 | Kobaz | i'm not actually super familiar with the event stream for 13 |
| 18:57.46 | Kobaz | i do a lot of work in 1.8 and 11 still |
| 18:57.52 | Kobaz | since our platform hasn't been migrated yet |
| 18:57.56 | DanFromUK | erggg. was hoping you werent going to say that |
| 18:58.07 | Kobaz | or |
| 18:58.13 | Kobaz | are you using chan_sip ? |
| 18:58.42 | DanFromUK | was a long time ago when i wrote our call recording AMI client to force call recording to start on the right channel. can't remember any of my C skills. |
| 18:58.47 | DanFromUK | nope. pjsip |
| 18:58.49 | Kobaz | actually, that wouldn't matter, most of the masqerade stuff is core and not channel driver related |
| 19:01.19 | DanFromUK | I wonder what would happen if I set dynamic features in the dialplan that is used when calling the 2nd agent during the transfer. |
| 19:01.25 | Kobaz | but yeah if you look at all the fixup code |
| 19:01.28 | Kobaz | in main/channel.c |
| 19:01.58 | Kobaz | that sound be a good place to start looking for anything lacking on handlind dynamic features upon masquerade |
| 19:02.17 | Kobaz | and then features.c |
| 19:02.43 | Kobaz | you could add some debugging to see if anything is actually running, maybe it's actually enabled but failing some checks and refusing to execute |
| 19:03.20 | Kobaz | DanFromUK: you can play around with predial callbacks |
| 19:03.33 | Kobaz | like make sure that you set the dynamic feature on the right channel |
| 19:04.05 | Kobaz | b and B |
| 19:04.39 | DanFromUK | Unfortunately, its set on the same channel, both prior when it works and after the transfer when it doesn't. No others have it set. |
| 19:04.58 | Kobaz | sounds like a features/res_features issue then |
| 19:05.54 | DanFromUK | Ok, thanks. I'll hunt around and see if I can spot anything obvious. |
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| 19:08.01 | geckos | Hi everybody. I need to do a performance test on an Asterisk machine. Does any body have a guide or document about it? |
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| 20:16.45 | Kobaz | anyone have any experience with the new new vvx 450 |
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| 20:52.59 | *** topic/#asterisk by bford -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12) 16.0.0-rc1 (2018/08/08), Standard: 15.5.0 (2018/07/12); 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 |
| 20:53.52 | Samot | oooooooooo |
| 20:53.59 | Samot | 16RC1 |
| 20:55.32 | *** join/#asterisk zigggggy (~bbcde3e@unaffiliated/zigggggy) |
| 20:59.01 | Samot | Allow wrapuptime to be set for each queue member <<- Like that. |
| 21:07.06 | kfife | 450 looks nice Kobaz |
| 21:07.11 | kfife | Not seen it |
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| 21:10.08 | kfife | It's perfect for me though because I'm a 'Knowledg Worker" |
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| 21:25.29 | kfife | Do most people still install Asterisk onto CentOS? |
| 21:28.10 | jpsharp | Mine's a mix of Ubuntu server and Gentoo. |
| 21:28.23 | kfife | Selinux? Root? |
| 21:28.39 | jpsharp | I usually take selinux out back and shoot it. |
| 21:28.53 | kfife | Do you run as root? |
| 21:29.39 | jpsharp | On the ubuntu boxes, yes. Gentoo no. The Gentoo boxes predate me here, but the ubuntu boxes I built. |
| 21:29.54 | kfife | So not too hard to run as not-root? |
| 21:29.58 | kfife | That's good. |
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| 21:30.23 | kfife | Out of curiosity, wHy not SELinux? |
| 21:30.32 | jpsharp | Set the appropriate settings, start it as root, and it'll switch to the user/group you set. |
| 21:31.02 | jpsharp | Because it gets in my way more often than not. |
| 21:31.09 | kfife | roger that. |
| 21:31.58 | kfife | I recall in the past you had to chown a few dozen folders to the asterisk user. Now you can specify the user to run as in asterisk.conf or some such place? |
| 21:32.53 | jpsharp | You do have to make sure the folders have the right permissions, but you set the "runuser" and "rungroup" in asterisk.conf |
| 21:33.12 | jpsharp | Or you can specify the username/group on the command line. |
| 21:33.22 | kfife | Very Helpful. Thank you. |
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| 21:34.30 | jpsharp | No problem. |
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| 21:58.30 | minipudding | Hello. I'm having problems setting up pjsip (using pjsip wizard) to allow incoming calls anonymously without any user/pass authentication. When I try to call from my softphone it throws me a "failed login" message, and the Asterisk console keeps saying "no matching endpoint found" and "failed to authenticate" |
| 21:58.40 | minipudding | This is my config: https://pastebin.com/raw/jQJLrHW0 |
| 21:58.49 | minipudding | It's Asterisk 13.20.0 |
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| 22:25.59 | kfife | Does anyone know if the supplemental power connector on the DAHDI cards requires BOTH 5V and 12V to be energized, or can it run on 12V only? |
| 22:31.34 | kfife | DOC is ambiguous, but I found the PWR2400B, which is 12V only. Solved. |
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