IRC log for #asterisk on 20170503

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00:27.10*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); 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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00:47.42drmessanoav0cad0:  no
00:47.49drmessanoIs this a remote endpoint?
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13:50.26*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); 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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15:03.53syadnom_hi all.  anyone know if there are sound files in opus available...
15:07.17igcewieling1syadnom_: not yet.  thanks for volunteering.
15:07.47syadnom_lol
15:08.03syadnom_I'm not sure HOW this would be done for opus w/ asterisk
15:08.54syadnom_because in asterisk you define the codec pretty explicitly, ie opus 24kbps loss 10% complexity 8.  Would you need sound files for each entry in codecs.conf?
15:10.55igcewieling1No idea.  What is this recent fetish for Opus?
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15:15.30syadnom_it's magically, that's what the fetish is about lol
15:15.39syadnom_igcewieling1, ever played with it?
15:18.00igcewieling1syadnom_: No, because it is so new in Asterisk, almost nothing supports it.  Sounds like a non-starter.
15:18.25syadnom_xlite, bria, grandstream, yealink, polycom all support opus
15:19.28syadnom_polycom's VVX500+ have for quote a long time.  Grandstream's 8xx series ATA's have since they were released.  Yealink very recently in their 'S' series, 5x, and a few other random devices.
15:19.40igcewieling1syadnom_: Softphones suck.  We don't use them.  Polycom's support for Opus is limited to recent models.   g722 is supported by anything we use which supports wideband codecs.
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15:20.33syadnom_but g722 is 'just' wideband.  It doesn't do anything for random network conditions, which is why most people have dedicated or high end links for business SIP.
15:21.27syadnom_opus REALLY changes that.  You can have really good sounding voice even with consumer connections that lose a few packets or buffer bloat makes for some late arrivals.
15:21.45igcewieling1Yup, 95% of our customers have QoS from their site to our NOC, tough they are not totally dedicated to VoIP.
15:22.04lorsungcurelying on the codec to mitigate your shitty internet will not end well
15:23.00DanQuinneysyadnom_: got a couple of phones to play with opus, we're moving to * 13 after running 1.8 for far too long
15:23.29syadnom_DanQuinney, I'm on 13.14.  I moved up from 11 for opus support
15:23.31igcewieling1I suppose if you are a hobbyist Opus might work well or maybe a small company where you can totally controll all aspects of the setup.
15:23.49syadnom_I don't get that statement
15:23.51igcewieling1Does Asterisk transcode Opus?
15:24.14syadnom_yes
15:24.30DanQuinneyI want to move to v14 for the nice kamailio subscription support
15:24.32syadnom_in later 13 releases and in 14 it does
15:28.31DanQuinney13.14 had a great bug where pjsip subscriptions never expired, so in the end after a couple of days we'd see 100,000+ NOTIFY's for each and every call
15:28.58syadnom_I'm not using pjsip
15:29.52DanQuinneyhow come?
15:30.53syadnom_my kit isn't completely compatible yet.  working on it
15:31.03DanQuinneyAh ok
15:31.47syadnom_I understand that everyone doesn't have to be excited about opus, but I don't get the naysaying
15:32.14DanQuinneyI think I'm the next 12-24 months it'll become really popular
15:32.38DanQuinneyand it doesn't cost us anything other than time supporting it
15:33.10syadnom_opus is wide-band, capable of handling significant network issues, and is gaining support on popular hardware.  Even digium's new phones support opus.
15:34.48syadnom_I see no audible indication when switching between the primary and secondary internet connections in a failover scenario.  with g711/722 there is a noticeable gap in the audio.  That's a pretty solid gain.
15:35.56igcewieling1syadnom_: I don't consider "my internet sucks" as a valid reason to use a codec.    I'm glad it it works for you.   How many phones do you support?
15:36.14syadnom_~600 or so.
15:36.22syadnom_but only a small number are opus
15:36.52syadnom_I have a ton of vvx3/4xx units out there, using siren7 for those.
15:37.01igcewieling1I'd think you'd be skeptical and bitter after that many phones.   Are your users saints or something?
15:37.22syadnom_why would I be bitter?
15:38.02igcewieling1syadnom_: one word: users
15:38.20syadnom_sure, but that's every job ever, can't let them get you down.
15:39.53syadnom_biggest issue I typically get is crackly audio.  Every customer I've moved to opus, that complaint vanishes.
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15:40.52syadnom_went from crackly audio when they hit their internet connection to smooth audio even if you hammer it with torrents and whatever downloads you can.
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15:42.54igcewieling1Maybe Adtran will support is someday.
15:43.32syadnom_Sangoma is working on it, but they say it will have to come on the next-gen hardware.  Not enough CPU in their current gear to handle it and no hardware path available either.
15:43.47igcewieling1We use Adtrans to convert between SIP (our side) and analog and PRI (customer PBX side).   Hiding a crappy internet connection would be handy.
15:44.41syadnom_yeah, I'm wanting an SBC so I can do opus as the transit codec, but then g722 on prem.
15:45.00syadnom_I don't do a ton of trunking to PBXs
15:45.33igcewieling1Our Sangoma transcoder cards don't support Opus either.
15:45.47syadnom_yeah, next-gen units supposedly will
15:46.06igcewieling1I love those cards.  Install the card and never have a transcoding issue again.
15:47.27syadnom_There is hardware support for opus in many of the new ARM chips, might mean faster time to market for transcoders...
15:48.08SamotYou know..
15:48.17SamotEveryone had this same hardon for G722
15:48.37syadnom_yeah, and g722 is pretty widely supported and used today....
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15:49.28igcewieling1Verizon SIP wholesale and Level 3 SIP wholesale don't support G722.
15:49.38SamotMost carriers don't.
15:49.39SamotWhy?
15:49.41SamotPSTN
15:49.48SamotStill got to transcode that baby
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15:49.59syadnom_almost no wholesalers support 722, or really anything except g711 and sometimes g729
15:50.04igcewieling1Samot: I use g722 between phones.
15:50.09SamotYeah
15:50.12SamotSo do I
15:50.20SamotBut when it gets to my media gateway for the PSTN
15:50.26syadnom_yeah, for on-prem systems it's g722 all the way
15:50.26SamotIt goes out and comes in g711
15:51.10igcewieling1We have G729 handoff to the carriers for reasons too long to go into.
15:51.32SamotG711 and G729 are carrier standards.
15:51.59igcewieling1We have G729 handoff to the carriers instead of ulaw for reasons too long to go into.
15:52.14Rac-onmaybe it depends on the location, i wouldnt call g729 standard here
15:52.15syadnom_I'm not a g729 fan myself.
15:52.20Rac-onbecause of quality and licensing issues
15:52.24syadnom_Rac-on, where is here?
15:52.33igcewieling1IIRC Cell carriers are slowly supporting g722
15:52.36SamotRac-on: LECs have g729 licenses.
15:52.38Rac-oneurope, nl specifically
15:52.43syadnom_ah, ok.
15:52.48SamotRac-on: In the US
15:52.53Rac-onalmost 100% g711 here, because that is what pstn does
15:52.53syadnom_us also
15:53.00SamotBecause 15 years ago bandwidth was an issue.
15:53.14SamotActually the PSTN doesn't do g711
15:53.27SamotIt's just the closest and least transcoding needed.
15:53.33Rac-ontrue
15:54.05syadnom_In the US, PSTN switches are g711
15:54.42syadnom_Bell used g711 and most phone companies are former bell systems.
15:54.54SamotAnd T.40
15:55.01Rac-onthese days i reserve g729 for people on wifi, since it deals with that much better compared to g711
15:55.07igcewieling1syadnom_: assuming, the call hit a PSTN gateway.  Not all calls do.
15:55.08Rac-onthat about the only valid use-case imho
15:55.41SamotWell
15:55.45SamotThe PSTN is the PSTN
15:56.01SamotWhether or not the transport is IP or copper is a different story.
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15:56.08syadnom_I find g729 much more prone to 'robot voice' .  g711 likes to just leave gaps or sound a little crackly, but g729 goes all '70s robot in a hurry
15:56.47Samotg729  is lower compression.
15:56.59igcewieling1For example, When we call a Time Warrner Cable (residential service) the call goes us -> Verizon -> TWC -> customer.   There doesn't have to be a g711 component to the call.
15:57.35SamotBut it's still the PSTN.
15:57.42sawgoodWhy not just leave things at G711 uLaw?  (why compress)?
15:57.44Rac-onigcewieling1: it is hard to predict wether or not the will be POTS (see what i did there Samot ?) in the flow
15:57.48syadnom_there doesn't have to be, but many lecs will keep everything pcm or g711 because when they built out the transcoding costs were high
15:57.53sawgoodhellow by the way: been a while!
15:58.26syadnom_sawgood, I actually don't care to compress that much, I want good packet loss concealment.
15:59.14sawgoodI was thinking: for a while: that I would leave the PBX and phones at default (G711) and then any compresssion (could be done) at the edge by a gateway appliance
15:59.20syadnom_with g729, you are basically just getting compression.  gains in audio quality are only from smaller packets being more deliverable.
15:59.23igcewieling1Samot: I think of "The PSTN" as switches like 5ESS or DMS100 and that sort if thing.   A rather limited view I suppose.
15:59.25sawgoodthis way: nothing needs to be touched on the phones or PBX (making) the compression easier?
16:00.03SamotThe "Switch"part isn't referring to hardware.
16:00.07syadnom_opus, well, it takes similar CPU to do it at 64kbps vs 24kbps etc so there is no disadvantage to compression.  plus, you get those smaller packets.
16:00.15SamotIt's referring to switching the calls between carriers.
16:00.16igcewieling1Samot: I understand that.
16:00.34sawgoodright, but now (today) there is Opus (codec) (forgot that) sweet old Asterisk 14!
16:01.08syadnom_sawgood, so pick a side, we are having a lovely debate about the usefulness of opus, with plenty of tangents thrown in!
16:01.22syadnom_all because I asked if there were any opus audio files for asterisk lol
16:01.36syadnom_instigator
16:01.44SamotOK well then to say a call between TWC and Verizon doesn't touch the PSTN because it's over IP is wrong.
16:01.44sawgoodsyadnom_: that is a good saying, "pick a side" "get off the fence" (right on)
16:01.46Rac-onthats the internet :p
16:01.59sawgoodmost of our stuff is using Asterisk 13 (and) some 11
16:02.31sawgoodhey, I get to sit through the Switchvox training (4 hours) for a cert (soon) ....
16:02.56sawgoodI was super happy about this: until we learned (you get) no Asterisk CLI (abliity) under Switchvox (and) I didn't so much get that ...
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16:03.23syadnom_really?
16:03.43sawgoodyeah: SSH is "cut off"
16:04.19sawgoodno Asterisk CLI (and) maybe I got it wrong, but I asked a few times (And) I saw others online asking why this is the case (no SSH) and no Asterisk touching
16:05.36sawgoodeither way: we will accept it and build around it, etc
16:06.52SamotThat's the compromise when using Switchvox.
16:07.12SamotIt comes preconfigured to do specific things in specific ways so you don't have to.
16:07.32SamotAllowing you to "muck" around in the system backend means you can break that.
16:07.41syadnom_well, a lot of people have no business in the asterisk cli anyway lol
16:07.43SamotIt's just a support/management nightmare.
16:07.49SamotSure they do.
16:07.59SamotFor troubleshooting and debugging what they are doing.
16:08.04sawgoodSamot:  We were hoping to have Asterisk CLI for the easier things like:  troubleshooting a new SIP circuit: or for watching when a phone does not register (seeing) those Asterisk output live
16:08.12SamotBut again, you make a compromise when choosing an appliance.
16:08.24SamotSwitchvox has debug tools.
16:08.26fileyou can see Asterisk logs and do packet captures from the Switchvox web UI
16:08.33Samot^^^^
16:08.51SamotThe main reason for the CLI = debug
16:08.56SamotSo they gave you a way to do that
16:09.02sawgoodI'm going to get into it (and) use those tools for Switchvox
16:09.24SamotThis is like installing FreePBX over just Asterisk.
16:09.31sawgoodha!
16:09.36SamotYou're making a compromise for how much YOU get to control.
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16:09.47SamotAnd HOW it's controlled.
16:10.11sawgoodhaving Titatium support to call Huntsville (will be super cool) when that is desired
16:14.23sawgooddumber question:  how come AsteriskNOW (even) with Asterisk 11/13 (is not) used or pushed as much?
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16:20.08zafi don't think it's worth the effort of maintaining an entire linux distribution just to support the app
16:20.26sawgooddo you mean the AsteriskGUI logic?
16:21.01[TK]D-FenderAsteriskGUI has been dead for more than half a decade.
16:21.08SamotAsteriskNOW is not maintained by Digium.
16:21.15SamotIt's basically FreePBX distro.
16:21.37sawgoodyeah I sort of knew that, but I had not seen it in YEARS ... (Even) though it has a recent release with 11/13
16:22.07syadnom_freepbx was bought by sangoma, so I think that soured digium on it
16:22.08SamotIt hasn't gone anywhere.
16:22.15Samot?
16:22.32SamotFreePBX _maintains_ the AsteriskNOW distro.
16:22.38sawgoodis that right?
16:22.41SamotIt has FreePBX 13 on it.
16:22.50sawgoodinteresting ...
16:23.05SamotSo I'm not sure how you're coming to this conclusions?
16:23.12syadnom_I don't believe freepbx does maintain asterisknow...
16:23.17SamotIs it not used as much by people? No, probably not.
16:24.03sawgoodAsteriskGUI was the PBX logic (and) if that switch to Sangoma FPBX (I) did not know that (or I forgot)
16:24.08sawgoodthanks for the catchup!
16:24.30SamotAsteriskNow version 10.13.66-17
16:24.37syadnom_Hasn't AsteriskNOW been FreePBX based for a long time?
16:24.41SamotYou know what 10.13.66-17 is?
16:24.48SamotFreePBX Distro release.
16:25.10sawgoodha (right on)
16:25.10SamotIt's the exact same distro, different GUI skin.
16:26.14sawgoodHow are you liking the D80 phone?  We will get some soon ...
16:26.32SamotI don't use Digum phones.
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16:35.41sawgoodThe Switcvox (switchboard) has Google Maps build in as a widget for incoming phone calls (showing) call lookup and location
16:35.56sawgoodI thought that was pretty cool for a built in widget of the PbX logic
16:39.18SamotYes
16:50.23zafi've got some d80s
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