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| 01:27.39 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.1 (2018/11/14), Security Only: 15.6.2 (2018/11/14); 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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| 01:56.32 | life_of_e | Early Christmas present coming: an HT318. I'll be playing with the hook flash during my short vacation time. :) |
| 01:58.09 | life_of_e | Grandstream ignored my emails asking about it and they also ignored my phone calls (having left multiple messages over the past three weeks with no return calls) |
| 02:24.32 | Samot | Would it be because they don't have an HT318? |
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| 03:16.51 | life_of_e | 813, dyslexic I am today |
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| 11:00.52 | phrearch | hi |
| 11:02.00 | phrearch | i'm trying to install asterisk 16 with webrtc support. when i try to connect a client, i get this warning though: no protocols out of 'sip' supported |
| 11:02.32 | phrearch | tried with the cyber mega phone(jssip) demo and sipjs |
| 11:09.58 | phrearch | load_modules: res_statsd declined to load. a lot seems to depend on this one. |
| 11:27.52 | phrearch | ok, got it. rebuild with 16.0.0-rc-3, used basic-pbx modules.conf minus load = res_pjsip_registrar_expire.so(not there). statsd loads, but still got the same error, until i added statsd.conf. websocket connections are accepted now. |
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| 11:51.17 | FuriousGeorge | hi everyone |
| 11:51.51 | FuriousGeorge | if i have to dump 200 callfiles to playback a sound file, would it be prudent to stagger them, rather than dump them all at the same time? |
| 11:52.24 | FuriousGeorge | i have a modern xeon processor with 4 cores allocated for the asterisk container, along with about 2 gigs of ram |
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| 12:23.05 | FuriousGeorge | in one case i would have to open 234 channels in a reasonable short amount of time for a public address |
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| 13:52.55 | Samot | phrearch: why are you using an RC release when there is a Stable version? |
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| 15:21.29 | craigify | Does anyone in here have experience with working on caller ID scam lists? Is is similar to the providers that maintain databases of email spammer's ip addresses? Those at least, you can identify and request removal. I've got a client who'se caller ID is one one or more of these lists. |
| 15:22.57 | aoeui | Find the service blacklisting and contact them |
| 15:23.39 | aoeui | I've requested removal from truecnam/nomorobo and they've removed numbers, don't know about other services, and there's many |
| 15:23.55 | drmessano | You have to send a postcard to Ward Mundy at his beach house in Tampa. |
| 15:24.03 | craigify | thank you aoeui. I must point out that you are wearing your obvious pants right now :) |
| 15:24.22 | craigify | see I figured you had to send a postcard |
| 15:24.30 | craigify | and it had to have the right secret stamp on it |
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| 16:03.29 | aoeui | some people here need Mr. Bovious |
| 16:03.41 | aoeui | Obvious |
| 16:04.29 | aoeui | it's really the same as with email blacklisting, you need the DNSRBL and request removal |
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| 16:17.00 | igcewieling | Twillo has an API to "score" the spammyness of a call. https://pro.whitepages.com/blog/feature-friday-spam-score/ does something similar. Don't know if that helps. |
| 16:17.39 | igcewieling | Whitepages.com wants you to fill out a form before you can even download a whitepaper on the service. screw that. |
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| 16:47.29 | jkroon | hi all, with res_odbc, if I utilize multiple invocations of a specific res_odbc_func dialplan function, is it possible to group the write queries as a single commit? |
| 16:50.07 | aoeui | I use nomorobo with Twilio |
| 16:50.42 | aoeui | you can request a whitelist there - and there are about a dozen services on Twilio that provide telemarketer spam scores or metrics |
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| 17:18.06 | igcewieling | ha! "machine learning âa thing companies often do to distance themselves from any responsibility for the actions taken by their algorithms" |
| 17:19.40 | jeffspeff | lol |
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| 19:38.28 | jrun | reading up on this: https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI |
| 19:39.56 | jrun | i'm using libwebsocket instead of wscat. i'm getting connected to the server but i don't see any indication in asterik's logs. do i need to enable debug or somethiing? |
| 19:42.00 | jrun | Asterisk 16.0.1 built |
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| 19:51.15 | jrun | nevermind, logging facility was wrong. |
| 19:56.39 | jrun | well, i only see HTTP connection logged but not websocket. |
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| 20:55.01 | FuriousGeorge | hey everyone |
| 20:56.11 | zigggggy | hi FuriousGeorge! |
| 20:56.32 | FuriousGeorge | i need to generate up to 240 call files to invoke playback(). should i just dump them all in there in one shot, fearlessly? i have a debian container running asterisk. i have 2 cores of a modern xeon and 2 gb of RAM dedicated to it |
| 20:57.37 | FuriousGeorge | or would it be better to divide it into 10 dumps, and pausing a few seconds between each of them |
| 20:58.12 | FuriousGeorge | correction: i have four cores |
| 20:58.32 | Samot | I don't see a problem outside of it being a container... |
| 20:58.41 | Samot | I've never put a PBX system on a container. |
| 20:58.48 | FuriousGeorge | Samot: i can do a kvm instance as well |
| 20:59.07 | Samot | But 250 calls isn't a real big deal. |
| 20:59.29 | FuriousGeorge | Samot: ive done small pbx in containers. nothing that has ever tried to open a couple of hundred plus channels simultaneously |
| 20:59.38 | MrTAP | FuriousGeorge: are all of these endpoints on a single network at the same location? |
| 20:59.54 | FuriousGeorge | MrTAP: there is one network connection to an fxs channel bank |
| 21:00.07 | FuriousGeorge | actually, correction: up to 7 banks |
| 21:00.11 | MrTAP | roger. if they were SIP devices on one network, multicast paging would be the efficient way to do that |
| 21:00.22 | MrTAP | no good with that setup though |
| 21:00.28 | FuriousGeorge | MrTAP: alas, there appears to be no way to do it. |
| 21:00.37 | Samot | What? |
| 21:00.49 | Samot | You want to send 240 calls out.. |
| 21:00.54 | Samot | There's numerous ways to do it. |
| 21:01.11 | FuriousGeorge | i was going to generate 240 call files to app playback |
| 21:01.15 | Samot | So? |
| 21:01.26 | Samot | I do this with 1800 calls at a time. |
| 21:01.28 | FuriousGeorge | so, nothing, based on what you said above |
| 21:01.28 | Samot | Over the PSTN |
| 21:01.48 | Samot | Not on a 4 core / 2GB system.. |
| 21:01.49 | FuriousGeorge | s/"i was going to"/"i am going to" |
| 21:01.56 | Samot | But that is going to be fine for 240 calls |
| 21:02.04 | Samot | If it's not, it's a VM...give it more resources. |
| 21:02.11 | FuriousGeorge | it's actually 4 core. and i can always buy more ram, though id rather not |
| 21:02.17 | Samot | But what you want to do is completely achievable. |
| 21:02.23 | FuriousGeorge | sweet |
| 21:02.33 | Samot | It's a common thing done. |
| 21:03.04 | Samot | If this is to mimic an "announcement" we went over this before... |
| 21:03.26 | Samot | Even if the gateways auto answer on the SIP side it still needs to do the FXS side and that adds lag. |
| 21:03.36 | Samot | So for what you need to do, this is the best option. |
| 21:03.50 | FuriousGeorge | it is. i just wanted to clarify that 240 simultaneous playback() would be ok. i was only clarifying the implementation |
| 21:04.01 | Samot | Yes. |
| 21:04.06 | Samot | That's nothing more that 240 calls |
| 21:04.16 | Samot | s/that/than/ |
| 21:04.29 | FuriousGeorge | seems like a big number to me, coming from my experience of never having more than 2 or 3 calls going |
| 21:04.39 | FuriousGeorge | but i can see that it is not a lot to ask of modern hardware |
| 21:04.58 | file | really you just have to try this stuff within the environment and see... |
| 21:05.11 | FuriousGeorge | we are not transcoding video on every channel... does asterisk 15 do that yet, btw? |
| 21:05.26 | file | no version of Asterisk transcodes video. |
| 21:05.42 | FuriousGeorge | file: it's just that this was requested after the fact, the system is already in place, and i want to minimize testing or potential downtime |
| 21:06.02 | FuriousGeorge | an abundance of caution, if you will |
| 21:06.07 | Samot | Then set up another VM |
| 21:06.16 | file | this is not something you minimize testing on |
| 21:06.22 | Samot | ^^^^ |
| 21:06.29 | Samot | You have to make sure the system can take the load |
| 21:06.58 | Samot | Without it pooping itself on top of everything else it is already doing. |
| 21:07.19 | Samot | The system that does 1800+ calls like this at a time...it is all that system does. |
| 21:07.40 | FuriousGeorge | this would effectively be all that system is doing, while it is doing it |
| 21:07.57 | Samot | How many times do you expect to make this dump? |
| 21:08.03 | Samot | Is it a one at a time thing? |
| 21:08.07 | FuriousGeorge | a handful of times per day, or less |
| 21:08.31 | Samot | Is it a user generated thing? |
| 21:08.39 | Samot | Or is it a automated thing? |
| 21:08.53 | FuriousGeorge | "please remove any vehicles from the parking lot for snow removal" |
| 21:09.00 | Samot | Ie Alice calls in, makes recording, call dump is done with that recording? |
| 21:09.07 | FuriousGeorge | yeah |
| 21:09.18 | Samot | What if Alice and Bob call in at the same time to do this? |
| 21:09.24 | Samot | Are you sending 240 and then 240 |
| 21:09.29 | Samot | Or will you end up with 480? |
| 21:09.45 | FuriousGeorge | alice works in security, and has the only phone |
| 21:09.52 | Samot | OK. |
| 21:10.04 | FuriousGeorge | i suppose i could implement a lock using astdb, if i added another phone |
| 21:10.11 | FuriousGeorge | right? |
| 21:10.33 | FuriousGeorge | "please wait, while alice completes her announcement" |
| 21:11.03 | FuriousGeorge | don't see why not |
| 21:11.05 | FuriousGeorge | thanks guys |
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| 21:13.56 | FuriousGeorge | just occurred to me: if i were ever going to use this to play files recorded by other means, then i would want to make sure it was an 8khz @ 8bit wav file, in order to avoid transcoding, correct? |
| 21:14.14 | FuriousGeorge | (or, whatever format/codec the gateways are using) |
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| 23:58.51 | \dev\cache | Hi. |
| 23:59.05 | \dev\cache | Anyone try the Yeastar GSM Gateway |