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| 00:19.45 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.1 (2018/06/11), Standard: 15.4.1 (2018/06/11); 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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| 01:15.48 | jpsharp | I'm having an odd problem with chan spy + confbridge. I have person A and person B in a confbridge and they're talking without a problem. Person C uses chan spy to whisper to person B. While this is going on, audio from person B is dropping in and out and sounds muffled. I'm not sure where to start troubleshooting this. |
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| 11:06.05 | z0rb | Hi |
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| 11:44.35 | z0rb | https://pastebin.com/cdG5TP1u |
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| 11:44.48 | z0rb | can anyone take a look at this and tell me why my calls are going into congestion |
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| 12:42.22 | Samot | [Jun 13 16:27:21] -- SIP/CanadaOne-0000004c is circuit-busy |
| 12:42.22 | Samot | [Jun 13 16:27:21] == Everyone is busy/congested at this time (1:0/1/0) |
| 12:42.28 | Samot | Your provider is busy |
| 12:42.33 | Samot | Or congested. |
| 12:42.45 | [TK]D-Fender | answered in #freepbx |
| 12:43.01 | Samot | I just got back. |
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| 13:38.14 | apocsDK | Hi everyone - Someone who can guide me how to set up queues so that member call counter is preservered even after a core reload - Thanks in advance :) |
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| 13:42.07 | [TK]D-Fender | apocsDK, you can't |
| 13:43.34 | apocsDK | [TK]D-Fender: Hmm So only the call sum for the queue is saved? |
| 13:44.56 | [TK]D-Fender | Whatever is persistent is persistent. It isn't an option |
| 13:47.16 | apocsDK | Then how does asterisk reliably honor the strategy=wrandom if it throws away the call stats? The thing is, that I have a config that reloads the queues every 15 minutes (a new member might have connected) |
| 13:48.10 | [TK]D-Fender | I'm not sure I understand the question |
| 13:48.18 | [TK]D-Fender | if it's random then how does call counter matter? |
| 13:50.31 | apocsDK | Its wrandom not random :) |
| 13:50.44 | apocsDK | The members are weighted :) |
| 13:51.09 | [TK]D-Fender | [TK]D-Fender> Whatever is persistent is persistent. It isn't an option <- that's what we've got |
| 13:51.22 | [TK]D-Fender | So anything more sounds like a feature request that should added to the tracker |
| 13:56.52 | apocsDK | Found the answer; need to create the members as dynamic members. That way, their call stats are not wiped on reload :) |
| 13:57.34 | [TK]D-Fender | Shouldn't be a difference for static VS dynamic.... |
| 13:57.50 | [TK]D-Fender | Sounds odd |
| 13:58.00 | apocsDK | There is. I just tested it :) |
| 13:58.10 | [TK]D-Fender | ok... |
| 13:58.46 | apocsDK | I'm not sure what the difference is though :) But the stats stays when it's a dynamic member :) |
| 13:59.53 | apocsDK | Thank you for taking your time to answer me, [TK]D-Fender :) |
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| 14:34.51 | Ast001 | Can you tell me more descriptive what is "Maximum allowed time of incoming registrations (seconds)" ? |
| 14:35.17 | [TK]D-Fender | How much more of a description is needed? |
| 14:35.28 | Ast001 | if it is 3600 what happens after that peer is losing registration ? |
| 14:35.58 | [TK]D-Fender | If they come in asking to refresh every 4 hours and you limit them to 2 hours... then you respond "No, your registration expires in 2 hours" |
| 14:36.25 | [TK]D-Fender | I ask for A. They say I can only have up to B. If they don't respect what I tell them then too bad |
| 14:36.42 | [TK]D-Fender | So they should be following what the server insists it is holding them to |
| 14:36.49 | [TK]D-Fender | the wording is completely clear |
| 14:37.26 | Ast001 | I have one sip peer who is losing registration and get "Unregistered" in cli and after some time registers again and I am trying to prevent that. |
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| 14:38.59 | [TK]D-Fender | Do you see them attempting to register again? Do you see the exchange that was held in the first place? |
| 14:39.34 | Ast001 | I see 4-6 unegister messges then it registered |
| 14:40.00 | Ast001 | but that happens to often |
| 14:40.46 | Ast001 | https://pastecode.xyz/view/f356cee4 |
| 14:40.49 | [TK]D-Fender | if it is specifically unregistering on purpose then look at the client |
| 14:41.14 | [TK]D-Fender | And you're not looking at SIP debug for this |
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| 14:42.59 | Ast001 | I can look at sip debug, i thought it is connected with reg time min max expiry etc... I keep default values for that in Asterisk, and it happens no matter I set 10 60 or 1 on that sip device. |
| 14:43.49 | [TK]D-Fender | You're not looking |
| 14:44.04 | [TK]D-Fender | Don't say "I thought" when you aren't even seeing what's happening |
| 14:44.22 | [TK]D-Fender | Go enable SIP debug and take a proper look at this |
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| 14:46.56 | igcewieling | *grumble* If Verizon installed non-flamable fiber maybe their lines would not fail because of an "underground fire" 8-| |
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| 14:50.19 | [TK]D-Fender | combustion != sublimation :) |
| 14:50.35 | [TK]D-Fender | they just vaporized! |
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| 15:15.13 | dan_j | Hello. |
| 15:15.30 | dan_j | How do you enable DTMF features for queue agents? |
| 15:16.31 | [TK]D-Fender | Depends which ones |
| 15:16.37 | dan_j | I've got this defined in features.conf but it is not executing when a call goes through a queue and the called agent dials *1 |
| 15:16.39 | dan_j | NewRecordApp => *1,self/both,Macro,ToggleRecording |
| 15:17.42 | dan_j | Just realised, __DYNAMIC_FEATURES is not set when dialling the queue. Is it that simple? |
| 15:18.01 | [TK]D-Fender | That variable exists for a reason.... |
| 15:18.07 | [TK]D-Fender | You didn't set it before the queue? |
| 15:18.31 | dan_j | I wasnt sure it was inherited through the queue to the agent like normal non-queued calls are. |
| 15:18.34 | dan_j | thanks for the confirmation |
| 15:19.04 | [TK]D-Fender | I'd start with that then move on to using local channels for your agents |
| 15:20.38 | dan_j | noted. thanks for the further tip |
| 15:23.03 | dan_j | Works. Perfect. Thanks! |
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| 18:51.36 | TechSmurf | There a good site out there for comparing phones from different vendors? |
| 18:52.37 | TechSmurf | particularly including older models that might be getting sold in bulk on ebay these days... |
| 18:53.25 | igcewieling | TechSmurf: not really. just get Polycom and stop worrying about it. |
| 18:55.47 | jpsharp | Yep. Polycom or grandstream. Ciscos are too much hassle if you have to change firmware. |
| 18:56.47 | TechSmurf | No love for digium phones? |
| 18:56.54 | igcewieling | Both Digium and Sangoma have phones too. If I'd not standardized on Polycom back in 2002 I might be using Digium or Sangoma phones. |
| 18:57.05 | TechSmurf | nods |
| 18:57.45 | TechSmurf | Bought a bunch of cheap linksys-cisco SPAs years back... regretted it ever since... trying to avoid a repeat ;) |
| 18:57.46 | jpsharp | I've honestly never used any of the digium phones. Everyone i've dealt with wants cheap cheap cheap phones, so I end up getting polycom or grandstream off Fleabay. |
| 18:58.26 | igcewieling | TechSmurf: the Polycom Soundpoint IP phones have *TERRIBLE* html/xhtml support. The VVX phones have a more complete browser on the phone. |
| 18:59.09 | TechSmurf | jpsharp: first gen digium is now a fleabay option |
| 18:59.33 | jpsharp | I've never once had anyone need the html/xhtml support. People just want phones that ring and dial out. |
| 18:59.38 | TechSmurf | still not as cheap as polycom/gs |
| 18:59.57 | TechSmurf | I'm not even sure what I'd use html/xhtml support for :\ |
| 18:59.58 | igcewieling | The SoundPoint IP 3xx phones are great to give to people you hate. |
| 19:00.29 | igcewieling | TechSmurf: phone apps |
| 19:00.29 | TechSmurf | afaik my people just want phones, and I don't want to learn nortel. |
| 19:00.49 | jpsharp | The IP 3xx take so...damn..long....to.....boot. |
| 19:00.50 | TechSmurf | been dealing with keeping a nortel system afloat for 12 years |
| 19:01.12 | jpsharp | i'm sorry :) |
| 19:01.13 | TechSmurf | while implementing a bit of voip here and there preparing for the day they give me monies for phones. |
| 19:01.51 | jpsharp | Whichever phone you pick, make sure you can build a provisioning system for it. Managing phones manually sucks shit. |
| 19:01.52 | TechSmurf | now we're getting ready to move to a new building and I'm going "you know we're not taking the nortel... we need the new phones now..." |
| 19:02.30 | TechSmurf | jpsharp: I've been using.. uhm.. the free epm to some sort of degree of success... |
| 19:03.54 | jpsharp | If you use polycom, asterisk has a basic provisioning system built in. |
| 19:04.37 | jpsharp | Or if you do something crazy and use FreePBX, it has provisioning platforms for lots of other phones. |
| 19:07.29 | TechSmurf | nods |
| 19:07.37 | TechSmurf | I've bounced around distros a bit |
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| 19:07.57 | TechSmurf | as a one-man department, I don't have time to do asterisk the hard way.. |
| 19:08.44 | TechSmurf | trix was neat for the month or so they were still developing it after I deployed it... then years of waiting for updates that never came... |
| 19:09.19 | TechSmurf | *now was neat for the year or so they were still supporting it after I switched... then it went into decay-mode too. |
| 19:09.36 | TechSmurf | so yeah, I have at this point given up and moved to freepbx :P |
| 19:10.06 | jpsharp | It's fine as long as you just want a fairly basic office pbx. Anything else and it kinda gets in its own way. |
| 19:10.17 | TechSmurf | nods |
| 19:10.51 | TechSmurf | I'm familiar with GUIs buggering up my custom work in other platforms |
| 19:11.20 | TechSmurf | but yeah.. we really just need phones. |
| 19:11.46 | TechSmurf | without having to deal with nortel-style configuration. |
| 19:14.08 | jpsharp | How many do you need? |
| 19:14.25 | TechSmurf | 18-24 |
| 19:20.44 | TechSmurf | hmm. soundpoints sure are cheap today tho.. |
| 20:19.11 | igcewieling | The Soundpoint IP 501+ and 601 (you don't want the 500) are the models I like. |
| 20:19.27 | igcewieling | 501+ meaning 5xx except 500 |
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| 20:51.11 | Samot | SoundPoints are super cheap because they are EOL'd. |
| 20:56.33 | igcewieling | IIRC only the VVX models support IPv6 |
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| 22:05.13 | rhl | Hi, I want to run asterisk and hook it up to google voice. I have fedora 28 and I have installed asterisk package. looking for a tutorial or someone to help. can someone point me at the right spot? |
| 22:05.41 | rhl | if there is a docker container for this, that would be aswesome too |
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| 22:47.59 | TechSmurf | igcewieling: Sadly, I've given up on my ipv6 requirement for new hardware. The phone vlan can stay ipv4 forever. |
| 22:49.43 | TechSmurf | I think gigabit switches are on my must-have list, just in case I need to use it, so the 560 is ok, but it looks like even the 550 is out... |
| 22:56.25 | TechSmurf | esp when 560 is ~20/ea on ebay in lot quantities. |
| 22:57.23 | TechSmurf | but then polycom falls short on line presence indicators for the office folks... so I can polycom half the building and then... something else for the office. |
| 22:58.25 | TechSmurf | Not promoting this as a store, but they seem to have a *huge* selection of phones and some good feature filtering options. https://www.pcliquidations.com/c158---polycom?criteria=3716%2C3720 |
| 23:00.21 | TechSmurf | just helped me narrow polycom to 560/670 only |
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