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| 00:18.43 | *** 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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| 03:40.28 | Samot | With the mailbox= setting, can that be set to watch all the mailboxes in a certain context or would I have to specify each box with the context? |
| 03:41.25 | Samot | So can it be mailbox=@default or will it end up being mailbox=vm1@default,vm2@default etc.. |
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| 05:41.45 | Samot | Are there going to be any transcoding paths added for opus? At least g711 <--> opus? |
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| 05:47.52 | drmessano | Yes |
| 05:48.00 | drmessano | Was added in 13 |
| 05:48.14 | Samot | https://www.irccloud.com/pastebin/wyL3OQSh/ |
| 05:48.19 | drmessano | 13.12.1 or something |
| 05:49.29 | Samot | Asterisk 14.3.0 <-- That's where that came from. |
| 05:49.43 | Samot | Let me update. |
| 05:50.06 | drmessano | Working fine here |
| 05:50.17 | drmessano | Do you have the module? |
| 05:50.51 | drmessano | Im sure like g729, it does passthrough without the codec_opus.so |
| 05:51.48 | drmessano | I wont bother to screen dump, but I just ran the same on 13.12.1 and I have paths to everything except SILK |
| 05:51.57 | Samot | Hrm. |
| 05:52.09 | Samot | It's listed in show codecs |
| 05:52.28 | Samot | Let me make sure it's loaded. |
| 05:52.33 | drmessano | module show like opus |
| 05:53.02 | Samot | res_format_attr_opus.so Opus Format Attribute Module 1 Running core |
| 05:53.09 | drmessano | Nope |
| 05:53.12 | drmessano | codec_opus.so OPUS Coder/Decoder 1 Running extended |
| 05:53.12 | drmessano | format_ogg_opus.so OGG/Opus audio 0 Running core |
| 05:53.12 | drmessano | res_format_attr_opus.so Opus Format Attribute Module 1 Running core |
| 05:53.43 | Samot | Got it..thought it installed with 14.. |
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| 06:01.49 | Samot | https://www.irccloud.com/pastebin/VbaSYiMB/ |
| 06:02.01 | Samot | Hey, check out what happens when you follow _all_ of the steps. |
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| 06:25.20 | drmessano | Thats badass |
| 06:25.31 | drmessano | Glad I could help. Thank you, drive through |
| 06:29.36 | drmessano | Uhhhh.. there seems to be a problem with the ceiling fan |
| 06:30.35 | Samot | What's that? |
| 06:30.58 | Samot | It's standing still but you're spinning? |
| 06:33.08 | drmessano | It's Beavis and Butthead |
| 06:33.19 | drmessano | Closing Time |
| 06:33.27 | drmessano | How do you boys explain this mess? |
| 06:33.30 | drmessano | Uhhhh.. there seems to be a problem with the ceiling fan |
| 06:33.44 | drmessano | When they were tossing food into the fan |
| 06:35.01 | Samot | Well it's evil, wicked, mean and nasty (Don't step on the grass, Sam) And it will ruin our fair country (Don't be such an ass, Sam) Well, it will hook your Sue and Johnny (You're so full of bull, Sam) All will pay that disagree with me (Please give up you already lost the fight, alright) |
| 06:35.24 | CRCinAU | so here's one |
| 06:35.32 | CRCinAU | why does 'core show translation' show this: |
| 06:35.33 | CRCinAU | https://paste.fedoraproject.org/paste/2hMXwUH10tZC8Xroz7zV8V5M1UNdIGYhyRLivL9gydE= |
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| 06:36.13 | Samot | It's a chart of translation times |
| 06:36.35 | CRCinAU | but so many slin's? |
| 06:36.45 | lorsungcu | its easter |
| 06:36.52 | lorsungcu | after sunday all slins qwill be forgiven |
| 06:37.35 | CRCinAU | and no mention of g723 or g729 or g719 or speex or g726? |
| 06:37.53 | Samot | Do you have them installed? |
| 06:37.58 | Samot | All of those are licensed codecs. |
| 06:38.08 | Samot | It also doesn't have opus. |
| 06:38.20 | Samot | slin is the playback format. |
| 06:38.29 | CRCinAU | hrrrm |
| 06:38.47 | Samot | So when you do Playback(hello) |
| 06:39.16 | drmessano | When you said you havent used Asterisk in years |
| 06:39.23 | Samot | and you have .g722, .ulaw and .wav it will determine which version is the best based on your codec and it coverts to that slin rate |
| 06:39.24 | drmessano | Did you mean *never*? |
| 06:39.26 | drmessano | Be honest |
| 06:39.53 | drmessano | or did you get banned from here before Asterisk could actually make calls |
| 06:39.57 | drmessano | That would be legit too |
| 06:40.19 | CRCinAU | drmessano: are you talking to me or someone else? |
| 06:41.12 | drmessano | CRCinAU: Who do you think? |
| 06:41.24 | drmessano | lol |
| 06:42.26 | CRCinAU | no idea - I have the man flu and it hurts to even read the screen at the moment - let alone think properly :\ |
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| 06:43.38 | CRCinAU | facepalms having 64bit modules helps... 64bit system can't load 32 bit modules. |
| 06:44.13 | CRCinAU | but yeah, I got banned here around asterisk 1.2 days when it was forking into several versions. |
| 06:45.15 | drmessano | Considering how difficult it is to get banned from here, your performance must have been extroadinary |
| 06:45.27 | CRCinAU | my arguement was that it'd make things a pain for developers and it should proceed as even number = stable, odd = devel |
| 06:45.35 | CRCinAU | remember how we did kernel stuff back then? |
| 06:46.36 | CRCinAU | but hey, it worked itself out over time - so meh. |
| 06:46.55 | lorsungcu | i hear that. darn kernels. |
| 06:47.03 | CRCinAU | iirc it was around the time we had 1.2 / 1.4 / 1.6 / 1.8 on the go at the same time |
| 06:47.21 | CRCinAU | that's if memory serves... was a *long* time ago. |
| 06:48.02 | lorsungcu | youre forgetting 1.69 |
| 06:48.11 | drmessano | well in 1.6.x.x they stopped added major new features to the release |
| 06:48.18 | drmessano | well in 1.6.x.x they stopped adding major new features to the release |
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| 06:48.54 | drmessano | Anything that would break existing dialplans or API |
| 06:50.02 | CRCinAU | my memory is that it was a time of great turbulance for non-asterisk based developers |
| 06:50.25 | drmessano | I think that's an exaggeration |
| 06:50.53 | CRCinAU | more that lots of people weren't ready for it - and external addons failed etc etc |
| 06:51.06 | CRCinAU | now its probably just more a ton of #ifdef etc |
| 06:51.22 | drmessano | That's an exaggeration |
| 06:51.33 | CRCinAU | I haven't really paid much notice other than download / ./configure / make install these days |
| 06:52.08 | Samot | You realize that as of 1.10 all they did was drop the 1. |
| 06:52.15 | drmessano | I remember a few limited occasions where something was fixed or changed within a branch that, while well publicized caused a few people to complain |
| 06:53.01 | CRCinAU | that being said, I was using chan-sccp-b for just about as long... but that's adapted now, so eh. |
| 06:53.03 | drmessano | Saying that new point releases would just break the shit out of things was a gross exaggeration |
| 06:53.23 | CRCinAU | if you only used SIP, you were fine. |
| 06:54.12 | CRCinAU | I still use chan_sccp-b to this day - and it even works over IPv6, so that's kinda cool. |
| 06:54.45 | drmessano | Most of that was people looking for a reason to hate on Asterisk |
| 06:54.58 | drmessano | There were several things in play |
| 06:55.25 | CRCinAU | eh, I just think it was which side of the fence you were stuck on and if you were affected or not. |
| 06:56.28 | drmessano | For example, sure.. there were changes on a couple of occasions that broke something for someone. But also, as a developer, if you're patching something into Asterisk EVEN TO THIS DAY.. you have to keep up with point releases. If you don't, your extensive patching just doesn't work. So it was convenient to blame Digium for everything that stopped working |
| 06:57.00 | drmessano | Like chan-sccp-b |
| 06:57.06 | drmessano | Because I used it too |
| 06:57.11 | drmessano | and every time it broke |
| 06:57.52 | CRCinAU | out of intrest, why was it never upstreamed? |
| 06:57.52 | drmessano | It had nothing to do with Digium and everything to do with the patches just failing because that's how patching works (or fails) when core files change |
| 06:58.05 | CRCinAU | I feel it would make sense to replace the skinny stuff with it |
| 06:58.16 | drmessano | Because Asterisk had chan_skinny, which was infinitely more stable |
| 06:58.25 | drmessano | Not as many features.. but far more stable |
| 06:58.36 | drmessano | Which Digium also was blamed for.. |
| 06:59.06 | drmessano | Because it wasn't the chan-sccp-b developers fault the code was shitty |
| 06:59.26 | drmessano | #sarcasm |
| 07:01.58 | CRCinAU | anyway, not that any of it matters these days. |
| 07:02.41 | drmessano | No, but there's no reason to gaslight anyone over it either. |
| 07:03.21 | drmessano | Asterisk has come a long way, but a lot of people assumed a lot of things that were just simply incorrect |
| 07:03.50 | CRCinAU | eh - I was just answering your question... tbh I don't really care now. |
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| 09:02.26 | file | Moo |
| 09:13.10 | detha | filed under Moo |
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| 09:19.59 | ChannelZ | meow |
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| 11:04.40 | CRCinAU | woof? |
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| 17:06.37 | igcewieling | I think the problem with SCCP/Skinny is that Cisco hated all non-Cisco PBXs and didn't want people to use their phones on non-Cisco platforms. That isn't the case as much today, since SIP clearly won over Skinny and they sell SIP phones and PBXs. |
| 17:07.16 | igcewieling | My outlook was that if they didn't want my money, I didn't want their products. |
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| 17:12.03 | Mr_Pleb_Mgoo | wait their new range is on SIP ? |
| 17:12.12 | igcewieling | new range? |
| 17:12.51 | igcewieling | Their Linksys products don't do Skinny/SCCP |
| 17:12.59 | Mr_Pleb_Mgoo | oh |
| 17:13.04 | Mr_Pleb_Mgoo | so not the 9000 series phones |
| 17:14.19 | igcewieling | I don't see those on a google search, just Ciscoo Nexus switches. |
| 17:15.09 | igcewieling | Ah, there it is. Searching on Linksys 9000 worked. |
| 17:15.58 | Mr_Pleb_Mgoo | sorry its the 8000 series, typod before |
| 17:16.02 | igcewieling | Mr_Pleb_Mgoo: I was referring to Cisco's Call Manager products. |
| 17:16.13 | Mr_Pleb_Mgoo | http://www.cisco.com/c/en/us/products/collaboration-endpoints/unified-ip-phone-8800-series/index.html |
| 17:16.34 | Mr_Pleb_Mgoo | so those things are SIP friendly? because the late iterations of the 7000's were a nightmare |
| 17:17.15 | igcewieling | Ah, those phones are from long after they bought Linksys so don't apply to my comments. |
| 17:17.41 | Samot | Unless it is an SPA series the phone is designed with the CUCM as the primary "system" for the phones. |
| 17:17.55 | igcewieling | Samot: thanks for the info. |
| 17:18.41 | Samot | Acquiring those lines of devices answered the "SIP first" problem for Cisco and their devices. |
| 17:21.23 | sekil | 7800 have SIP fw |
| 17:21.46 | Samot | Right. |
| 17:21.52 | Samot | So do the 79xx |
| 17:22.05 | Samot | They were designed to work with the CUCM first and foremost. |
| 17:22.06 | igcewieling | Most Cisco phones had a SIP load, even if it was all Call Manager centric and terrbily documented and comlicated to set up. |
| 17:22.07 | sekil | that is...they have SIP primary fw for cucm |
| 17:22.09 | sekil | Samot: no |
| 17:22.12 | Samot | SIP is an afterthought for those devices. |
| 17:22.18 | Samot | OK. |
| 17:22.20 | sekil | Samot: 7800 have SIP 3rd party fw officialy |
| 17:22.32 | Mr_Pleb_Mgoo | yeah all the 7000's have sip firmware but the late models have such a crippled featureset with the SIP firmware there's no real point in using them over something like the SPA phones |
| 17:22.37 | sekil | Samot: and it works quite well..not like 79xx |
| 17:22.50 | sekil | Samot: which officially supported Cisco only |
| 17:22.57 | Samot | I'm sure they work just fine. |
| 17:23.23 | igcewieling | Long ago I standardized on Polycom phones, they want my money and even provide useful documentation. Unlike Cisco's phones at the time. |
| 17:23.30 | sekil | the thing is..when you move to 3rd party..you loose a licence to Cucm |
| 17:23.45 | Samot | .... |
| 17:23.53 | Samot | Which Cicso phones? |
| 17:24.04 | sekil | I have 7821 |
| 17:24.08 | igcewieling | 79xx are the ones I have specific experience with. |
| 17:24.16 | Samot | Oh yeah, those are garbage. |
| 17:24.19 | Samot | IMO. |
| 17:24.23 | Samot | I hated using them. |
| 17:24.37 | igcewieling | I assume they were OK if used with CUCM |
| 17:24.46 | Mr_Pleb_Mgoo | the first gen 7900's were easy enough |
| 17:24.46 | sekil | 79x0 worked ok...79x1 had issues |
| 17:24.51 | Mr_Pleb_Mgoo | the later ones were horrendous |
| 17:24.59 | Samot | It was easy to fix. |
| 17:25.01 | igcewieling | I LIKE some of the ideas of SCCP and MGCP, but they lost out to SIP. |
| 17:25.04 | Samot | Just not use them. |
| 17:25.23 | sekil | even Cisco is moving to SIP for CUCM |
| 17:25.48 | sekil | I don't think 78xx have SCCP fws...maybe some.. |
| 17:25.57 | sekil | I think 7821 only has SIP fws |
| 17:26.10 | igcewieling | sekil: That is about 15 years too late to make any difference to me. |
| 17:26.45 | sekil | well.. |
| 17:28.32 | igcewieling | Hmmm..That's a little before I got so fed up with Digium cards that I switched to Sangoma. |
| 17:31.54 | igcewieling | I have at least 80 customer sites, some sites have more than one card, so we might have 100 - 125 Sangoma cards and a thousand or more polycom phones. Very little will convince me to move away from those products, though Digium tries once in a while. BTW: OLD Digum cards had issues, I don't have experiece with modern Digium cards -- I've heard they are quite good. |
| 17:32.17 | Mr_Pleb_Mgoo | cards as in DACs? |
| 17:33.12 | sekil | telcos don't offer SIP? |
| 17:36.10 | igcewieling | Cards as in T-1 or Analog ports. sekil: We offer SIP, but some customers don't want SIP. |
| 17:37.17 | igcewieling | Most customers want backup lines, usually analog, for when their SIP service or Internet service go down. |
| 17:37.54 | sekil | that's interesting.. |
| 17:37.56 | CRCinAU | heh - I have 2 x 7970, 1 x 7960, 1 x 7940... only the 7940 uses SIP... |
| 17:38.05 | CRCinAU | the rest use chan_sccp-b |
| 17:38.42 | Mr_Pleb_Mgoo | my condolences |
| 17:38.53 | CRCinAU | works perfectly for me - and I'm still not sure if there's a phone with as decent speakerphone setup around that isn't mucho expensive. |
| 17:40.08 | igcewieling | A 7970 isn't expensive? |
| 17:40.19 | igcewieling | ...er 7970 |
| 17:40.46 | CRCinAU | you get the 7970's for like $50AUD these days |
| 17:41.00 | CRCinAU | I think I paid $100 for mine around 5-6 years ago? |
| 17:41.16 | sekil | those old phones cost nothing nowadays |
| 17:42.08 | sekil | CRCinAU: hey..what's like there down under? |
| 17:43.15 | Mr_Pleb_Mgoo | dark |
| 17:43.24 | Mr_Pleb_Mgoo | dat almost 2AM feel |
| 17:43.28 | CRCinAU | yeah - I should be in bed. |
| 17:43.33 | CRCinAU | 3:43am here :\ |
| 17:43.42 | Mr_Pleb_Mgoo | rip in eastern states |
| 17:46.02 | CRCinAU | anyhow - I should piss off to sleep... having the man flu sucks and I've been in bed most of the day - so my sleeping pattern is cactus anyway :\ |
| 17:46.21 | CRCinAU | couple that with a 4 day weekend, and this is going to be an intersting week lol |
| 17:46.30 | Samot | sekil: Most deliver of services by Telcos/Cable companies is SIP and they convert it. |
| 17:47.31 | sekil | Samot: in europe it's mostly voip to customer |
| 17:47.39 | CRCinAU | yeah, I need to go to bed... was looking at ebay going "Woooo, a new 7970GE for $45... that's not bad... I could use that... No. Go to sleep" |
| 17:47.53 | sekil | Samot: PRI/BRI is being phased out...there are analog via SIP gws |
| 17:48.09 | CRCinAU | late night ebaying is much worse than late night informercials. |
| 17:48.20 | CRCinAU | o/ |
| 17:49.40 | Samot | Here in the US, Comcast offers "PRIs" |
| 17:49.52 | Samot | Completely SIP. |
| 17:54.31 | Samot | As much of a proponent of SIP that I am, I have no hesitation dropping analog or digital circuits into a client location for the right reasons and needs. |
| 17:55.28 | Mr_Pleb_Mgoo | i find analog gateways tend to create more problems than they solve |
| 17:55.34 | Samot | Alarm/elevator lines are a prime example. |
| 17:55.46 | Samot | I find the opposite. |
| 17:56.10 | Samot | I use gateways over physical cards. |
| 17:56.12 | Mr_Pleb_Mgoo | granted, i also find im always inheriting something horrendous |
| 17:56.27 | Mr_Pleb_Mgoo | like SPA9000/400 combos |
| 17:56.54 | Mr_Pleb_Mgoo | or some ancient audiocodes box |
| 17:57.00 | Samot | I use SPA8000's for hotels that still have FXS based systems. |
| 17:57.07 | Samot | And NeoGates. |
| 17:58.08 | Samot | Even when I replace a hotel's FXS based system with an IP PBX, still have to use FXS gateways. |
| 17:58.25 | Samot | For the rooms. |
| 17:58.44 | Samot | Unless I can get in during construction, like I'm doing with two new ones.. |
| 17:59.03 | *** join/#asterisk matrix1233 (~matrix123@197.2.121.123) |
| 17:59.05 | Samot | Which has all the rooms doing IP Phones on PoE switches. |
| 18:00.09 | Samot | The worst part about replacing the FXS PBX systems in hotels.. |
| 18:00.45 | Samot | All the cards are 16 channel cards in the system, which means the 66 blocks are never fully used. |
| 18:04.57 | sekil | yeah I used 8000s for hotels |
| 18:06.14 | Samot | Well since the 6.x firmware, the SPA8K will do SIP trunks. |
| 18:06.51 | Samot | You can do up to 4 trunks split how you want over the FXS ports. |
| 18:08.21 | sekil | yeah...I needed one only |
| 18:08.27 | Samot | Yup. |
| 18:08.30 | Samot | That's all I do. |
| 18:09.05 | Samot | NeoGates offer the same thing. |
| 18:09.18 | Samot | They just have 4 to 32 port options. |
| 18:09.42 | sekil | there is Patton too |
| 18:10.53 | Samot | Yup. |
| 18:11.12 | Samot | I've used Patton and I've used Adtrans. |
| 18:11.31 | Samot | I would prefer to use Adtrans but cost restraints stop that. |
| 18:11.50 | Samot | NeoGates are decent, they hold up well. |
| 18:12.10 | Samot | The 32 port version has 2 RJ-50 connectors.. |
| 18:12.45 | sekil | there is Grandstream too :) |
| 18:12.54 | sekil | although not recommended :) |
| 18:12.55 | Mr_Pleb_Mgoo | *shudder* |
| 18:13.01 | Samot | I won't use Grandstreams. |
| 18:13.30 | Mr_Pleb_Mgoo | the newer audiocodes gear is pretty solid in my experience |
| 18:14.12 | Mr_Pleb_Mgoo | never heard of neogates |
| 18:14.53 | Samot | Well it's an Asian PBX company that is trying to make a mark in the US. |
| 18:14.59 | Samot | Their PBXes are horrible. |
| 18:15.22 | Mr_Pleb_Mgoo | sounds a bit like Planet |
| 18:15.27 | Samot | They run an modified version of Asterisk 1.6 but they are moving up. |
| 18:15.35 | Samot | However, the FXS gateways.. |
| 18:15.42 | Samot | For being FXS gateways are just fine. |
| 18:15.47 | Mr_Pleb_Mgoo | Planet's early stuff was good but shits gone downhill lol |
| 18:16.07 | Samot | Plus, I can use Asterisk debugging in them. |
| 18:16.41 | Samot | So unlike the SPA8K, I can watch a verbose log and a sip debug. |
| 18:17.25 | sekil | Samot: that's probably rebranded Yeastar...it has 1.6 as well |
| 18:17.36 | Samot | It is Yeastar. |
| 18:17.43 | sekil | Samot: the pbx..it's terrible |
| 18:17.45 | Samot | The NeoGate is their gateway brand. |
| 18:17.54 | Samot | 2:14:54 PMÂ <Samot>Â Their PBXes are horrible. |
| 18:18.13 | Samot | They tried to hire me to fix them. |
| 18:18.54 | sekil | that sounds bad |
| 18:19.12 | Samot | It's like anything with Asterisk. |
| 18:19.35 | Samot | They wrote subpar dialplan and functionality.. |
| 18:19.56 | Samot | They had no real idea on the US market they were trying to get into. |
| 18:20.23 | sekil | this pbx dies all of the sudden on some sip packets |
| 18:20.35 | sekil | chan_sip stops working |
| 18:21.00 | Mr_Pleb_Mgoo | dat old asterisk feel |
| 18:21.12 | Samot | I have _one_ in a client location. |
| 18:21.33 | Samot | Very, very low use client. |
| 18:21.42 | drmessano | sekil: Is that PBX exposed to the internet? |
| 18:21.47 | Mr_Pleb_Mgoo | i have PBX's on 1.8 (kill me) that randomly decide they will no longer close / terminate channels / sessions, so ill find one locked up with like 700 active channels lol |
| 18:21.49 | Samot | Uhm. |
| 18:21.51 | Samot | No. |
| 18:21.51 | sekil | drmessano: no |
| 18:21.56 | drmessano | Interesting |
| 18:22.12 | Samot | Oh sekil. |
| 18:22.14 | sekil | drmessano: at least I don't think so..I don't really care.. |
| 18:22.27 | Samot | That's the attitude. |
| 18:22.30 | drmessano | I know it was fairly common until recently that chan_sip didn't handle scans so well |
| 18:22.52 | sekil | drmessano: I offered my service to get rid of that...they said no..so.. |
| 18:23.36 | drmessano | heh |
| 18:23.41 | drmessano | Seems to be a going thing |
| 18:23.43 | Mr_Pleb_Mgoo | so is pj_sip recommended over chan_sip yet |
| 18:23.54 | Samot | Depends on your needs. |
| 18:24.01 | drmessano | Not unless you need it |
| 18:24.44 | Samot | That being said... |
| 18:24.52 | Samot | PJSIP is the new kid.. |
| 18:24.55 | Mr_Pleb_Mgoo | ive not really looked into the differences although i do seem to recall pj_sip will handle multiple registers for the same peer |
| 18:25.02 | Mr_Pleb_Mgoo | or something like that |
| 18:25.05 | drmessano | Kinda sorta yes |
| 18:25.05 | Samot | So future development is all about the new kid. |
| 18:25.10 | drmessano | It has AORs |
| 18:25.10 | Mr_Pleb_Mgoo | havent touched it in a few years |
| 18:25.17 | drmessano | Which are nice |
| 18:25.21 | Samot | Yes. |
| 18:25.22 | Mr_Pleb_Mgoo | AOR? |
| 18:25.54 | Mr_Pleb_Mgoo | no idea what an aor is |
| 18:26.19 | Samot | Address of Record. |
| 18:26.31 | drmessano | Address Of Record.. basically keeps track of those peers now |
| 18:26.37 | drmessano | Assigns them an ID |
| 18:26.40 | Mr_Pleb_Mgoo | ah |
| 18:26.41 | Samot | Allowing a single SIP account to have multiple locations in memory |
| 18:26.45 | Samot | Yup. |
| 18:26.50 | Mr_Pleb_Mgoo | that reminds me |
| 18:27.38 | Mr_Pleb_Mgoo | what do you do with chan_sip if you have different remote peers sitting on the same IP address? i've seen some funky stuff happen where thats setup |
| 18:27.49 | Samot | What do you mean? |
| 18:28.11 | Samot | You mean having 5 phones at a remote location that connect to Asterisk? |
| 18:28.36 | drmessano | Use a decent router |
| 18:28.40 | Samot | ^^^ |
| 18:28.41 | drmessano | Not some ISP provided shit |
| 18:28.45 | drmessano | It's called having proper NAT |
| 18:28.49 | Mr_Pleb_Mgoo | no NAT |
| 18:29.05 | drmessano | Ok.. so you need to explain "Funky stuff" then |
| 18:29.08 | Samot | Are they registering? |
| 18:29.13 | Samot | Or peering? |
| 18:29.23 | drmessano | and how are they sitting on the same IP address? |
| 18:29.28 | Samot | ^^ |
| 18:29.41 | Mr_Pleb_Mgoo | Server A with two trunks to Server B (each trunk has a different account code), what seems to happen is Server A starts getting the auth on the two trunks confused and one of the trunks will just fail to auth |
| 18:30.04 | Samot | So you have two PBXes peering together? |
| 18:30.06 | Mr_Pleb_Mgoo | SIP trunks that is |
| 18:30.14 | Samot | Do they register? |
| 18:30.15 | drmessano | Because you're using the wrong type |
| 18:30.17 | Mr_Pleb_Mgoo | two asterisk boxes, not acting as PBX's |
| 18:30.21 | Mr_Pleb_Mgoo | yeah they register |
| 18:30.27 | Mr_Pleb_Mgoo | i believe they're set to 'friend' |
| 18:30.29 | drmessano | Look up type=peer vs type=user |
| 18:30.32 | drmessano | Thats the problem |
| 18:30.41 | Samot | With different listening ports? |
| 18:30.43 | drmessano | Of course they are getting mixed up |
| 18:30.47 | drmessano | They are matching each other |
| 18:31.02 | drmessano | friend creates a user and a peer for each one |
| 18:31.10 | drmessano | and they both match differently |
| 18:31.17 | drmessano | So no way to differentiate |
| 18:31.17 | Mr_Pleb_Mgoo | Samot not sure off the top of my head but i would doubt it |
| 18:31.33 | Samot | Well if you have two trunks from the same system.. |
| 18:31.41 | Samot | Going to the same remote system.. |
| 18:31.50 | drmessano | https://www.voip-info.org/wiki/view/Asterisk+sip+type |
| 18:32.06 | Samot | ^^^ Follow that. |
| 18:32.20 | Samot | Or what he said, at least. |
| 18:32.21 | Mr_Pleb_Mgoo | yeah, i figured chan_sip wouldn't like it, its not something i have setup just something i've noticed in yet another inherited system lol |
| 18:32.34 | drmessano | chan_pjsip won't like it either |
| 18:32.40 | drmessano | You have an improper config |
| 18:32.50 | Samot | SIP doesn't like it. |
| 18:32.56 | drmessano | You're creating 4 objects, not 2 |
| 18:33.00 | drmessano | and you're matching on all 4 of them |
| 18:33.01 | Mr_Pleb_Mgoo | thats what i like to hear :p |
| 18:33.21 | drmessano | We call that a "Charlie Foxtrot" |
| 18:33.42 | Mr_Pleb_Mgoo | so i'd want to configure type=user in this situation and just auth the trunks? |
| 18:34.56 | drmessano | That would be my suggesting |
| 18:34.59 | drmessano | That would be my suggestion |
| 18:35.26 | drmessano | The only benefit of type=user is when you _want_ to match on username regardless of IP the calls originate from. <-- From the Wiki |
| 18:35.49 | Mr_Pleb_Mgoo | yeah that is exactly what it needs to do |
| 18:35.54 | drmessano | 'type=peer' is _never_ matched on username for incoming calls, only |
| 18:35.54 | drmessano | matched on IP address/port number (unless you use insecure=port or higher). |
| 18:35.59 | Mr_Pleb_Mgoo | might have to suggest that change at some point |
| 18:36.06 | drmessano | You're matching on user AND IP |
| 18:36.13 | Mr_Pleb_Mgoo | not sure how that fuck-up happened in the first place |
| 18:36.14 | drmessano | Since you have friend |
| 18:36.19 | drmessano | Which is both objects |
| 18:36.28 | drmessano | So it's yes and yes, oh and yes and yes |
| 18:36.35 | Samot | They didn't understand the type= options. |
| 18:36.54 | Mr_Pleb_Mgoo | i should point out, the trunks have different usernames, but will have the same peer IP |
| 18:37.11 | Mr_Pleb_Mgoo | will = do * |
| 18:37.31 | drmessano | That and you're creating a user and a peer on each end |
| 18:37.41 | drmessano | Which matching both ways |
| 18:37.47 | drmessano | Which is matching both ways |
| 18:37.51 | Mr_Pleb_Mgoo | but that explains why the log is full of auth errors about getting the wrong username attempting to auth against the trunk |
| 18:38.00 | Samot | Yup. |
| 18:38.03 | drmessano | lol yes |
| 18:38.10 | Mr_Pleb_Mgoo | :~) |
| 18:38.36 | Mr_Pleb_Mgoo | yet another thing for me to fix |
| 18:38.43 | Mr_Pleb_Mgoo | man i tell you what i am never short of work |
| 18:38.58 | drmessano | Youre preaching to the choir |
| 18:39.03 | Samot | ^^^^^ |
| 18:39.05 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 18:39.08 | Samot | Big time. |
| 18:39.17 | Samot | Funny though... |
| 18:39.34 | Samot | My kid asked my about two hours ago "Do you think you're the best at your job?" |
| 18:39.34 | Mr_Pleb_Mgoo | the last major work i did with asterisk i actually made a (fairly primitive but rather reliable) email to fax / fax to email system |
| 18:39.41 | Samot | I said "No." |
| 18:39.50 | Samot | He asked "Why?" |
| 18:40.26 | Samot | I said "I spent most of my time fixing things that people who thought they where 'The best at their jobs' did" |
| 18:40.41 | Mr_Pleb_Mgoo | kek |
| 18:40.59 | Mr_Pleb_Mgoo | amen |
| 18:41.09 | Mr_Pleb_Mgoo | every time i look at this realtime system i cry |
| 18:41.26 | Mr_Pleb_Mgoo | there's some good ideas in there, but some of them are implemented in such a horrendous way |
| 18:41.32 | drmessano | I think only PFSense users have a higher percentage of unqualified admins |
| 18:41.56 | Mr_Pleb_Mgoo | asterisk realtime tought me an important lesson though |
| 18:42.07 | Mr_Pleb_Mgoo | Postgres > MySQL/Maria |
| 18:42.16 | drmessano | At least people trying Asterisk know what a phone looks like, 95% of the time |
| 18:42.29 | drmessano | Im not so sure about PFSense users and basic cabling |
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