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| 00:00.43 | ChannelZ | I believe you should use the CHANNEL function these days and the different channel types have slightly different things. See 'core show function CHANNEL' |
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| 00:33.31 | gk | hello, I have a problem with asterisk 1.8: it does not reconnect to the upstream SIP server (freeswitch) after that upstream server restart |
| 00:34.02 | gk | I believe asterisk thinks that it reconnected and stops further attempts but it is not registered in upstream and nothing works |
| 00:34.40 | gk | if nothing happens to the upstream SIP server then the session is perfectly OK for days |
| 00:35.15 | gk | no NAT between, no DNS involved (SIP upstream set by IP), freebsd |
| 00:35.55 | gk | is it a known problem? any idea where should I look? |
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| 01:25.29 | ChannelZ | How does the upstream server know the asterisk server? Does it have a static IP/hostname, are you required to register to it, etc |
| 01:26.32 | ChannelZ | If you have to register, then you'd have to just set yours to periodically reregister. |
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| 11:37.15 | Erico | Hi mates |
| 11:37.29 | Erico | is there a ConfBridge command like the MeetMeAdmin? |
| 11:37.53 | Erico | Mute or Kick users from Dialplan? |
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| 12:10.37 | Erico | \ |
| 12:10.56 | Erico | is this channel even active? |
| 12:12.19 | mjordan | Erico: patience. It's kinda early in the US. |
| 12:12.29 | mjordan | Also: that information is pretty easy to find on the wiki. |
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| 12:14.27 | mjordan | ConfBridge takes a slightly different approach to controlling users. Admin users have DTMF commands to control how they manipulate users. That's documented both in the module configuration (https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge) and on the Confbridge page (https://wiki.asterisk.org/wiki/display/AST/ConfBridge) |
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| 12:15.09 | mjordan | outside of the admin DTMF menu options - which you create yourself in confbridge.conf - ConfBridge generally wants you to use AMI to control the conference. |
| 12:15.23 | mjordan | Hence, most of the analogous commands are AMI actions. |
| 12:15.44 | mjordan | all of which are documented here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+AMI+Actions |
| 12:16.17 | [TK]D-Fender | Or CLI commands... which you can call from the dialplan |
| 12:16.26 | [TK]D-Fender | (with a few seconds of thought) |
| 12:17.57 | Erico | alright got it .. so no command like MeetMeAdmin, i will be using AGI/AMI i guess |
| 12:18.42 | [TK]D-Fender | or CLI |
| 12:19.01 | [TK]D-Fender | "confbridge" <tab> |
| 12:19.26 | Erico | ya but i need it from Dialplan |
| 12:20.52 | [TK]D-Fender | Yes and you can GET to it from there too... |
| 12:20.57 | [TK]D-Fender | [08:16][TK]D-Fender(with a few seconds of thought) |
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| 12:41.56 | mjordan | I'm not really sure why AMI is so hard in that context. If all you need to do is run a single AMI action, that's about as hard as invoking an AGI. |
| 12:42.10 | mjordan | or running a CLI command from the dialplan, for that matter. |
| 12:42.15 | mjordan | shrugs |
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| 12:57.37 | [TK]D-Fender | mjordan: Which is what I suggested |
| 12:57.56 | [TK]D-Fender | mjordan: Mind you that implies shelling out which I'm OK with... |
| 12:58.14 | [TK]D-Fender | mjordan: Mind you we should ahve a dialplan app so save that hackery. |
| 12:58.35 | mjordan | that would be nice. |
| 12:58.45 | mjordan | albeit, there are some funky things that could happen there. |
| 12:59.19 | mjordan | permission escalation issues that the System and SHELL app/function are explicitly noted as having |
| 12:59.27 | mjordan | but it'd be worth looking into |
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| 13:19.46 | DivideBy0 | up-to-date and quality asterisk docs are the wiki and the book. are there any others? |
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| 13:22.59 | nbjoerg | reminds me, there was a nice typo in the asterisk book |
| 13:23.04 | nbjoerg | 80x24 pixel terminals :) |
| 13:26.16 | WIMPy | Isn't the standard 84x48 pixels on Terminals? |
| 13:29.05 | [TK]D-Fender | DivideBy0: Don't know of anyone else spending time to make a 3rd repository of such information. |
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| 13:37.12 | mjordan | quite a lot of work just trying to keep those two correct and updated :-) |
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| 13:43.10 | DivideBy0 | I'm sure. Thanks guys |
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| 15:22.04 | wasanzy | hello, using queue techniques, how can I change the music on hold audio? |
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| 15:23.25 | [TK]D-Fender | "Queue" isn't a "technique" |
| 15:23.33 | [TK]D-Fender | Want to set the MoH? READ THE sample config |
| 15:23.39 | [TK]D-Fender | tHE SETTINGS ARE ALL DESCRIBED IN THERE |
| 15:24.00 | wasanzy | I want to change the default audio to something else |
| 15:24.19 | [TK]D-Fender | So change the MoH class. |
| 15:25.50 | wasanzy | under [general] ? |
| 15:26.10 | [TK]D-Fender | under your QUEUE |
| 15:27.08 | wasanzy | ok |
| 15:30.01 | wonderworld | hi, i want to use the pickupaexten feature from features.conf. i understand that i would give the calling channel a pickupgroup in the dialplan. what i didn't get is how to assign the extension that wants to pickup an incoming call to that pickupgroup? tnx |
| 15:35.38 | [TK]D-Fender | wonderworld: Um... could you rephrase what you're actually trying to accomplish? |
| 15:35.50 | wonderworld | ok let me try |
| 15:36.56 | WIMPy | wonder what the "features" have their place in that. |
| 15:37.21 | wonderworld | pickupexten feature in features.conf..... |
| 15:37.29 | wonderworld | should allow me to do that |
| 15:37.50 | wonderworld | i can set the pickupgroup for incoming calls in the dialplan with a channel variable |
| 15:38.15 | WIMPy | I have no idea what that could be good for. Usually you have an extension to do a pickup. |
| 15:38.16 | wonderworld | but how would i set the pickup group for the peer that would like to "steal" the call? |
| 15:38.41 | WIMPy | Usually as Paramerter to The Pickup Application. |
| 15:39.15 | wonderworld | this is what i am refering to: https://wiki.asterisk.org/wiki/display/AST/Call+Pickup |
| 15:39.47 | [TK]D-Fender | wonderworld: no such thing as setting the incoming group for incoming calls. |
| 15:39.52 | [TK]D-Fender | wonderworld: ALL calls are incoming |
| 15:39.58 | [TK]D-Fender | wonderworld: that is PER CHANNEL |
| 15:40.01 | wonderworld | i want to do this: Requesting to pickup a call is done by two basic methods.2) by dialing the pickupexten configured in features.conf. |
| 15:40.44 | wonderworld | so the wiki article tells me how to set the pickupgroup on the channel |
| 15:41.13 | wonderworld | but i don't know how to set the pickup group for the person that dials *8 on the phone to pickup the call |
| 15:41.47 | wonderworld | (as configured by the pickupexten feature in features.conf) |
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| 15:43.29 | [TK]D-Fender | clarify the timing you are expecting to do this in |
| 15:44.25 | wonderworld | sorry, didn't get you...which timing? |
| 15:44.46 | [TK]D-Fender | Describe your USAGE scenario |
| 15:44.55 | [TK]D-Fender | EXACTLY where are you where you intend to trigger this,. |
| 15:45.06 | [TK]D-Fender | What else is happening just PRIOR, and what do you exect to do AFTER? |
| 15:46.00 | wonderworld | ok. everyone in the office should be able to answer calls from their own phones, even if another phone is ringing. they should be able to pickup the calls for colleagues who are not there without walking to their phones. |
| 15:47.11 | wonderworld | phone b rings, person sitting in front of phone a should be able to pickup the call for b |
| 15:47.28 | [TK]D-Fender | that doesn't sound like features.conf material at all |
| 15:47.42 | wonderworld | maybe is misread the wiki article? |
| 15:47.55 | [TK]D-Fender | And all you should have to do is have the incomiong channel have that value set and the group already set on your other devices |
| 15:47.59 | wonderworld | it says: 2) by dialing the pickupexten configured in features.conf. |
| 15:48.32 | WIMPy | was serious about his question. Why does pickup exist as a "feature"? It doesn't make any sense to me. |
| 15:48.34 | wonderworld | ok, thats my problem. how do i set this pickup group on the other device? |
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| 15:49.09 | [TK]D-Fender | dial an extension that calls the Pickup application |
| 15:49.09 | wonderworld | sip.conf? |
| 15:49.10 | WIMPy | 'core show application Pickup' |
| 15:49.25 | WIMPy | Make an extension, not a "feature". |
| 15:49.58 | wonderworld | what would be the downside of using the "feature" method? |
| 15:50.19 | [TK]D-Fender | features.conf = crap you do while on ANOTHER call. |
| 15:50.20 | WIMPy | You need to be on a call already to use "features". |
| 15:51.03 | wonderworld | ok thanks. so the wiki article is wrong? or would it work while being in another call? |
| 15:51.15 | WIMPy | When did I last mention that "features" is a really really bad name for a feature? |
| 15:51.55 | wonderworld | pickupexten isn't even listed as an option in my dist features.conf file |
| 15:52.15 | WIMPy | That what we just told you, but even if you are on a call already, I don't see any sens in using "features". |
| 15:52.39 | WIMPy | +e |
| 15:56.32 | [TK]D-Fender | wonderworld: We are not talking about features.conf at all. |
| 15:56.36 | [TK]D-Fender | wonderworld: Stop looking there. |
| 15:56.43 | [TK]D-Fender | wonderworld: DIALPLAN apps.... |
| 15:56.52 | [TK]D-Fender | you are doing a NEW call from your phone to do the pickup |
| 15:57.00 | [TK]D-Fender | make an extension accordingly. |
| 15:59.15 | wonderworld | ok, will do. tnx |
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| 16:16.07 | *** join/#asterisk polysics (~polysics@93-38-164-94.ip71.fastwebnet.it) |
| 16:16.40 | polysics | hello! dev question: is there any way to set the BridgeuniqueId on brdiges on Ast 12+? |
| 16:21.08 | mjordan | polysics: bridges created through ARI can have their uniqueID set on creation. |
| 16:21.22 | mjordan | can't recall what version that went in on; probably an early 13 or very late 12. |
| 16:21.36 | polysics | mjordan: no way to do that through AMI? even indirectly, like setting BridgeCreator or something? |
| 16:21.38 | rmudgett | It was a very late 12 but before 13 |
| 16:21.49 | mjordan | polysics: you can't create bridges directly in AMI. |
| 16:22.23 | mjordan | Bridge AMI action kind of tries to provide that, but only in a very limited way. |
| 16:22.34 | mjordan | and no, it doesn't expose an attribute for that. |
| 16:23.33 | polysics | "kind of" or "not at atll"? :D |
| 16:26.00 | polysics | the ability to set any kind of identifier over AMI Bridge would greatly help |
| 16:26.12 | polysics | but if it's not htere, it's not htere :) |
| 16:26.18 | polysics | also double typo. |
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| 16:39.19 | drab | morning |
| 16:40.34 | [TK]D-Fender | Somewhere I'm sure... |
| 16:40.47 | drab | over the rainbow |
| 16:41.16 | drab | what are people looking for self-manged asterisk hosting normally get? like for a "normal" linux server you can either get a VPS from say Linode, put your machine in some DC or maybe go in the cloud... |
| 16:41.41 | drab | a bunch of googling shows lots of cloud or things like rentapbx, sort of c-panel like things, all managed |
| 16:42.03 | [TK]D-Fender | drab: Those are CLOSED solutions |
| 16:42.11 | [TK]D-Fender | typically with restrictive GUI's |
| 16:42.16 | drab | I couldn't find something along the lines of a VPS that I could manage myself with a barebone system but instead of just a network card also had FXO |
| 16:42.20 | drab | right, I want none of that |
| 16:42.21 | [TK]D-Fender | VPS = whatever you want it to be |
| 16:42.49 | mjordan | if you want an analog card, you're probably going to not find much. |
| 16:43.10 | drab | ok |
| 16:43.14 | [TK]D-Fender | depends on a point of view there.... |
| 16:43.23 | [TK]D-Fender | define your actual needs... then start looking |
| 16:46.03 | drab | needs a pretty simple, non profit here, so $$ is a huge factor, with a bunch of centers in diff places. Need to talk from center to center and have a bunch of public phone numbers for ppl to call in, voicemail, a couple conf bridges, IVR and a rollover, pretty standard I'd say |
| 16:46.13 | drab | the major thing I don't have a clue is actual hw rec |
| 16:46.26 | drab | as Id on't quite understand when/how intensive transcoding are and how much bw is used per call |
| 16:46.29 | doop | drab: you can do all that without POTS |
| 16:46.37 | doop | use an ITSP |
| 16:46.46 | drab | I've seen some offerings advertising number of CPUs = number of calls |
| 16:46.56 | drab | and warning about compression and bw |
| 16:47.11 | drab | tryign to read up on that and run some measuraments locally and play with codecs |
| 16:47.16 | doop | drab: what's your maximum number of simultaneous calls? |
| 16:48.15 | drab | doop: 4x bigger centers, 1x smaller ones |
| 16:48.32 | [TK]D-Fender | drab: 4 calls? |
| 16:48.33 | drab | conf calls would get bigger tho, since they'd pull in small centers |
| 16:48.34 | doop | does that mean four calls for the bigger centers? |
| 16:49.01 | drab | yes sorry,4 simultaneous calls, ie 4 people from diff departments making calls for different purposes |
| 16:49.31 | doop | how many centers do you have? |
| 16:49.35 | drab | part of the limit depends on costs really, they could do with more, but those are min rec based around the price of ~$24/m I've sen for a trunk |
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| 16:50.14 | drab | 7, 2 big, 5 small ones |
| 16:50.44 | doop | so about 13-24 calls out to the PSTN simultaneously? |
| 16:51.27 | doop | or do your numbers include calls between centers |
| 16:51.33 | drab | roughly yeah |
| 16:51.39 | drab | no, wasn't including calls between centers |
| 16:51.50 | doop | you'll be fine with a VPS and an ITSP |
| 16:52.08 | drab | I assumed those could be "free" as Asterisk to asterisk, ie don't need to go via the trunk. Am I mistake? |
| 16:52.13 | drab | mistaken* |
| 16:52.22 | doop | the ones within your centers can be "free" yes |
| 16:53.46 | doop | i would try g.711/ulaw before trying any codecs with lots of compression -- see how that works for you |
| 16:54.10 | doop | unless your centers are on super slow DSL or something |
| 16:54.46 | doop | but yeah you'll be fine |
| 16:57.18 | mjordan | probably could get by with the cheapest server from digital ocean |
| 16:57.24 | mjordan | but that's just my 2 cents. |
| 16:57.26 | doop | $5/month |
| 16:57.39 | file | I'm a fan of https://wable.com/ myself |
| 16:57.47 | DivideBy0 | any reason not to choose amazon for one of those servers? |
| 16:57.59 | doop | amazon is pricier? |
| 16:58.08 | mjordan | file: $6??? Mr. Money-Banks. |
| 16:58.19 | mjordan | although, for that extra dollar, you do get 1GB of RAM |
| 16:58.23 | mjordan | so, touche |
| 16:58.46 | mjordan | yeah, that's actually a much better deal than digital ocean. I should have asked you again before I panicked and decided I needed a server for a demo. |
| 16:59.19 | file | the $8 tier is useful because there's a "powerboost" which adds an additional 2 CPUs/VPS, 4GB of extra RAM, and 30GB of space |
| 16:59.35 | file | so if you need a large instance - fine, if you need a few smaller ones, fine |
| 16:59.37 | file | same price. |
| 17:00.08 | file | does not work for them |
| 17:00.24 | file | arguably works for mjordan |
| 17:04.49 | mjordan | only arguably. |
| 17:04.56 | mjordan | technically, you don't |
| 17:05.26 | file | indeed |
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| 17:53.21 | cusco | I have been using ovh, for nearly 2 years now, a cheap dedi unmanaged |
| 17:53.26 | cusco | kimsufi.com |
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| 18:18.33 | Jayka | hello everyone! |
| 18:20.17 | Jayka | I am having an issue which I was hoping someone can advise me on how to solve. When calling some endpoint, and then cancelling the call right away while it is still ringing I see that Asterisk is sending Bye instead of Cancel, despite the fact that the call has not been established. This is causing the endpoint to continue ringing indefinitely. Any workarounds? |
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| 18:26.43 | [TK]D-Fender | Jayka: Show us the actual call... |
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| 22:31.15 | monsterco | do you know of any AGI file that send basic call details? I don't want to spend time coding one. I want basic caller ID and other call details to be sent in a POST request |
| 22:32.22 | WIMPy | What about using curl directly? |
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| 22:56.34 | jmetro | Any opinions here on best conference phone? I'm partial to polycom starfish's but wondering if there are other options that sound as good / pick up audio as good as those. |
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| 23:07.36 | WIMPy | That's what Polycom has been know for, but there's also the Snom MeetingPoint now. |
| 23:09.22 | jmetro | I was pondering a Digium conference phone but I dont even see those on my voip suppliers catalog. I had a good experience with Snom in the past though |
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| 23:13.59 | jmetro | Gotta go, thanks though. |
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