| 00:05.58 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 00:14.16 | Samot | #DAHDITimingMatters |
| 00:15.35 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 00:18.37 | drmessano | Sorta along the same lines as the other day |
| 00:18.59 | drmessano | If you're on a modern OS, timerfd will be your timing source |
| 00:19.22 | drmessano | You're just using DAHDI for audio mixing |
| 00:19.52 | drmessano | So even less of a need for the seldom needed CONFIG_HZ_1000 |
| 00:30.36 | drmessano | Guess pf didn't help him |
| 00:30.40 | drmessano | Samot: ^ |
| 00:30.59 | Samot | Poor guy |
| 00:31.06 | drmessano | calm down |
| 00:31.13 | Samot | relax fella |
| 00:31.14 | drmessano | pf is fine, guy |
| 00:31.28 | freexer | but meetme uses dahdi right? |
| 00:32.03 | Samot | Yes. |
| 00:32.11 | Samot | So, good luck with that. |
| 00:32.32 | freexer | So still no need for CONFIG_HZ_1000? |
| 00:32.54 | Samot | I don't know. I haven't used meetme in a very, very long time. |
| 00:33.21 | freexer | oh i see |
| 00:33.31 | freexer | derp...meetme uses dahdi just for audio mixing |
| 00:33.33 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-asgtqzntxsbygmvk) |
| 00:33.35 | freexer | not timing |
| 00:34.09 | freexer | that right drmessano...I know now on a virtual machine I run I never did that and it all works |
| 00:34.14 | freexer | using kvm though |
| 00:35.03 | drmessano | Won't be a problem |
| 00:38.25 | freexer | I love you drmessano. I know that shouldn't be said out loud...still |
| 00:38.51 | drmessano | It's ok, dude.. Just no butt stuff |
| 00:41.30 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 00:43.35 | freexer | lol |
| 00:43.42 | freexer | wife says the same thing |
| 00:44.03 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 00:45.57 | drmessano | Ouch.. Just buy a bigger bottle of Patron |
| 00:46.15 | drmessano | She'll either be compliant or barking like a dog |
| 00:46.21 | drmessano | Could go either way, really |
| 00:50.38 | *** join/#asterisk monkey_ (~monkey@155.143.124.243) |
| 00:51.13 | monkey_ | Hello |
| 00:52.32 | monkey_ | anyone there ? |
| 00:53.33 | WIMPy | Where? |
| 00:54.01 | *** join/#asterisk Bordr (~Bordr@c-75-70-116-232.hsd1.co.comcast.net) |
| 00:56.49 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 00:57.30 | *** join/#asterisk monkey_ (~monkey@155.143.124.243) |
| 00:57.55 | monkey_ | hello |
| 00:58.21 | freexer | drmessano I'll take your advice...again. :) |
| 00:58.43 | drmessano | lol |
| 01:01.32 | monkey_ | Hello, is it possible for asterisk IP PBX to forward DID to mobile phone ? |
| 01:02.17 | [TK]D-Fender | You process your calls howver you want |
| 01:02.26 | [TK]D-Fender | however* |
| 01:02.50 | [TK]D-Fender | Call hits your server. If you want to dial out then you dial out. |
| 01:03.20 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 01:04.23 | monkey_ | I need a DAHDI right ? |
| 01:04.40 | monkey_ | I only want to use the server as voip |
| 01:04.46 | [TK]D-Fender | no |
| 01:04.56 | [TK]D-Fender | DAHDI is a choice |
| 01:05.12 | [TK]D-Fender | if you are using a line and have a DAHDI card you want to use it via |
| 01:05.22 | [TK]D-Fender | You use what you decide to use |
| 01:05.47 | [TK]D-Fender | A call may arrive via an ITSP and you might choose to use a phsyical analog line via an interface on your server to send the call out |
| 01:05.49 | [TK]D-Fender | Or not |
| 01:05.59 | [TK]D-Fender | It's your decision |
| 01:06.27 | monkey_ | how can I divert virtual voip number to mobile phone number ? |
| 01:06.36 | monkey_ | or shall I say forward |
| 01:07.43 | WIMPy | Those termes are all blurred anyway since telephony was abandoned. |
| 01:07.52 | WIMPy | But your question is rather vague. What do you have? What do you want? |
| 01:08.39 | [TK]D-Fender | monkey_, Call comes in... you Dial out. |
| 01:08.42 | [TK]D-Fender | the end |
| 01:08.46 | [TK]D-Fender | it's your dialplan. |
| 01:08.50 | [TK]D-Fender | DIAL <------- |
| 01:08.54 | [TK]D-Fender | Call out. |
| 01:09.00 | WIMPy | Do you already have that "DID"? Where? Or do you want to get one? |
| 01:09.18 | [TK]D-Fender | There is no such thing as "virtual voip number"" |
| 01:09.21 | [TK]D-Fender | That isn't a thing. |
| 01:09.59 | [TK]D-Fender | A telco has DID's assigned to it. You pay them for a service. to have calls to that number sent to you. If that is via a VoIP protocol, that is just one option. |
| 01:10.05 | [TK]D-Fender | So is a boring analog line |
| 01:10.30 | [TK]D-Fender | So is a digitial telco circuit like ISDN/PRI |
| 01:11.15 | monkey_ | I want : 1) all incoming calls to voip line. 2) I have a few voip number that I can forward them one by one to mobile phone, so if someone call the voip number 1, it would divert them to mobile 1 |
| 01:11.33 | [TK]D-Fender | <PROTECTED> |
| 01:11.46 | [TK]D-Fender | do you HAVE a service you are paying for already that routes a DID to you? |
| 01:12.02 | WIMPy | If you want to divert all calls, just tell your provider. Done. |
| 01:12.27 | monkey_ | not yet. I don't have a provider |
| 01:12.35 | [TK]D-Fender | Go pick one |
| 01:12.36 | [TK]D-Fender | <[TK]D-Fender> it's your dialplan. |
| 01:12.36 | [TK]D-Fender | <[TK]D-Fender> DIAL <------- |
| 01:12.39 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 01:12.53 | [TK]D-Fender | -------> DIAL <------------ |
| 01:13.02 | [TK]D-Fender | that is what you use to call something from your dialplan |
| 01:13.04 | [TK]D-Fender | ~book |
| 01:13.07 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 01:13.07 | monkey_ | Do I have to have a provider ? |
| 01:13.12 | WIMPy | Then check if you can get that DID for a mobile in your area. |
| 01:13.27 | [TK]D-Fender | <monkey_> Do I have to have a provider ? <- how else do you think you get a phone #? |
| 01:13.35 | [TK]D-Fender | You don't get magic service out of thin air |
| 01:13.52 | [TK]D-Fender | Do you get free cellphone service that isn't associated with a specific company? |
| 01:14.01 | [TK]D-Fender | SOMEBODY is givin you this service |
| 01:14.24 | monkey_ | so if I use asterisk as IP PBX for voip calls handling, I must find a provider and purchase DID ? |
| 01:14.46 | [TK]D-Fender | No. |
| 01:14.52 | [TK]D-Fender | Do you WANT a DID? |
| 01:16.03 | monkey_ | isn't DID for voip ? |
| 01:16.10 | Samot | No. |
| 01:16.17 | Samot | Direct Inward Dial |
| 01:16.19 | [TK]D-Fender | a DID is a PHONE NUMBER |
| 01:16.21 | WIMPy | And waht do you want to do yourself? Anything at all? |
| 01:16.45 | [TK]D-Fender | My CELL PHONE has an account .... that has a PHONE NUMBER |
| 01:16.57 | [TK]D-Fender | My fax machine is on an analog line ... THAT has a phone number |
| 01:17.09 | [TK]D-Fender | Do you want a NUMEBR people can call that will LAND on your PBX? |
| 01:17.47 | monkey_ | yes |
| 01:17.59 | WIMPy | Why? |
| 01:18.09 | [TK]D-Fender | And you'd like that call to ARRIVE at your server via VoIP? |
| 01:18.12 | WIMPy | What do you want to do there? |
| 01:18.47 | monkey_ | I want the callers to dial a phone number that can reach asterisk |
| 01:19.08 | monkey_ | that asterisk have IVR attached to it |
| 01:19.23 | WIMPy | I get the impression that you don't really know what you want. |
| 01:19.33 | [TK]D-Fender | [TK]D-Fender> And you'd like that call to ARRIVE at your server via VoIP? <-- |
| 01:19.40 | WIMPy | Ok, now we're getting somewhere. |
| 01:19.43 | [TK]D-Fender | He doesn't have the words yet. |
| 01:19.46 | [TK]D-Fender | but we're getting there |
| 01:20.19 | WIMPy | So what is that IVR supposed to do? |
| 01:20.52 | monkey_ | The IVR will allow them to route calls |
| 01:22.47 | monkey_ | so if they want to forward mobile num to another mobile num they can. If they want to forward a landland num to mob num they can and so on |
| 01:23.16 | WIMPy | Err, who can forward what? |
| 01:23.53 | monkey_ | person A can forward a phone number to his mobile |
| 01:24.43 | WIMPy | So you wan to set up a service for customers giving them number they can route elsewhere? |
| 01:24.52 | monkey_ | yes |
| 01:25.15 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 01:25.26 | [TK]D-Fender | <[TK]D-Fender> [TK]D-Fender> And you'd like that call to ARRIVE at your server via VoIP? <-- |
| 01:25.35 | Samot | Oh god. |
| 01:25.38 | Samot | I so called this. |
| 01:25.38 | [TK]D-Fender | we don't seem to be registering on this one yet |
| 01:25.46 | Samot | There's no point. |
| 01:25.48 | WIMPy | You definitely need to consult someone into this. That's not a beginners project. That kind of service usually involves a lot of legal stuff as well. |
| 01:25.54 | Samot | He wants to be a provider. |
| 01:26.15 | Samot | We shouldn't encourage anything but hiring someone and learning. |
| 01:26.51 | Samot | Starting with learning what the words mean. |
| 01:26.55 | WIMPy | You probably even need a licence to set up such service. |
| 01:27.05 | Samot | In the US? |
| 01:27.07 | Samot | HAHAHAHA |
| 01:27.22 | Samot | There's no regulation on VoIP in North America. |
| 01:27.38 | monkey_ | sounds scary |
| 01:27.43 | WIMPy | Don;t know where, yet. But how would he get the numbers? |
| 01:28.07 | Samot | He would contact a wholesale provider or wholesale carrier. |
| 01:28.15 | monkey_ | I don't know yet. I guess I am just throw ideas here and see what's the response like ;) |
| 01:28.28 | Samot | VoIP Innovations, PeerOne, Vitelity are prime examples of a wholesale provider. |
| 01:28.38 | Samot | The response is, learn about Telephony first. |
| 01:28.51 | Samot | This isn't web hosting. It's a public utility. |
| 01:28.55 | WIMPy | Consult someone or forget about it until you're in to it. |
| 01:29.06 | *** join/#asterisk xnaron (~xnaron@S0106b4750e5de3b2.ed.shawcable.net) |
| 01:29.17 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 01:29.23 | [TK]D-Fender | Before thinking of offering phone services... you should actually understand them. |
| 01:29.40 | monkey_ | true |
| 01:29.41 | Samot | 8:26:53 PMÂ <Samot>Â Starting with learning what the words mean. |
| 01:29.54 | [TK]D-Fender | <[TK]D-Fender> <[TK]D-Fender> [TK]D-Fender> And you'd like that call to ARRIVE at your server via VoIP? <-- |
| 01:30.04 | WIMPy | Most people don't know the words. |
| 01:30.47 | Samot | People who are actually skilled and knowledgeable to be doing this, yes they do. |
| 01:31.23 | WIMPy | They are on the edge of extiction. |
| 01:31.41 | monkey_ | why extinction ? |
| 01:32.24 | Samot | Because people don't learn anymore. |
| 01:32.43 | monkey_ | if you don't learn, how can you innovate ? |
| 01:32.52 | WIMPy | Because a LOT of terms are used outside their meaning in the VoIP days. And Asterisk terminology is a little special on top of that. |
| 01:32.55 | igcewieling | exactly. |
| 01:33.03 | Samot | Like? |
| 01:33.11 | WIMPy | We are post innovation. |
| 01:34.04 | WIMPy | Marketing digs out old stuff to sell it as new. Costs less than innovation. And the customers surely won't notice anyway. |
| 01:35.05 | Samot | I still want to know what telephony terms are being used outside their meaning in VoIP... |
| 01:35.19 | igcewieling | trunk |
| 01:35.44 | igcewieling | that is the best example 8-| |
| 01:35.57 | WIMPy | DID, trunk, forwarding for a start. |
| 01:35.57 | drmessano | But a SIP Trunk sounds really cool |
| 01:36.37 | drmessano | Like I have a 50 pair cable coming in over my DSL line I am using for my internet connection |
| 01:36.43 | Samot | DID? |
| 01:37.16 | WIMPy | Does such a thing as DID even exist in the NANP area? |
| 01:37.34 | Samot | It's the Direct Inward Dial number. |
| 01:37.45 | WIMPy | When ther's talk about DIDs it's clearly about NDID. |
| 01:39.03 | Samot | DID is DDI |
| 01:39.16 | Samot | It is for numbers that route over the same "trunk" |
| 01:39.18 | Samot | PRI, etc. |
| 01:39.28 | WIMPy | And does that exist? |
| 01:39.32 | Samot | Yes. |
| 01:39.40 | Samot | DDI <-- European version. |
| 01:39.50 | WIMPy | Never heard about it so far. |
| 01:39.51 | Samot | Direct Dial-In |
| 01:40.04 | Samot | Those are Telecom terms. |
| 01:40.11 | Samot | That moved to VoIP. |
| 01:40.31 | Samot | A SIP Trunk is nothing more than a PRI that has unlimited channel abilities. |
| 01:40.46 | Samot | When it comes down the "channelizing" it. |
| 01:40.52 | WIMPy | Yes, but the americans usually use automativ operators unstead of DDI, don't they? |
| 01:41.05 | igcewieling | it varies. |
| 01:41.19 | WIMPy | Well, DDI is something the ITSPs really struggle with. |
| 01:41.30 | Samot | It''s not a US telecom term. |
| 01:41.32 | Samot | DID is. |
| 01:42.38 | Samot | In the NANP the 10 digits break down to the region (NPA), exchange (CO), station (User) |
| 01:42.43 | igcewieling | I work for a small CLEC (VoIP and Resale) in the USA. Many companies have pbxs with only analog support, so we do analog handoff with no "DIDs" |
| 01:42.45 | WIMPy | I mean it's pretty obvious that the use of DID isn't too wiede spread with the limited adress range. |
| 01:43.06 | Samot | CLEC? |
| 01:43.16 | Samot | You have your own SPID? |
| 01:43.45 | WIMPy | Is that what tehy're there for? |
| 01:43.56 | Samot | Being a CLEC is a big deal. |
| 01:44.04 | Samot | It means you're connecting directly to the PSTN |
| 01:44.11 | igcewieling | I am not involved in the Resale side. |
| 01:44.14 | Samot | 100% regulated by the Feds. |
| 01:44.17 | igcewieling | There are several types of clecs. |
| 01:44.42 | Samot | So you're not Tier 1 |
| 01:44.51 | igcewieling | I never claimed to be tier 1. |
| 01:44.59 | Samot | I never said you did. |
| 01:45.00 | *** part/#asterisk monkey_ (~monkey@155.143.124.243) |
| 01:45.02 | Samot | I was clarifying. |
| 01:45.47 | igcewieling | there are resale clecs and facilities based. We are resale only. I don't deal with that side. We also have a VoIP side of the company which I deal with. |
| 01:46.58 | igcewieling | The voIP side uses Asterisk and Adtran CPEs of various types. We don't have a lot of direct SIP handoff to customers, most are analog or PRI handoff to customers. |
| 01:49.55 | Samot | Same thing I did, part of the reason I left. |
| 01:50.49 | Samot | The "old school" guys, mainly sales, refused to learn and in some cases talk nicely about SIP based services. |
| 01:52.21 | Samot | It took me a year to get the toll free and international off TDM routes when I got there. That took going over the CTO and showing the CFO the numbers. |
| 01:52.22 | igcewieling | The VoIP side also does FreePBX systems, I don' count those as "sip handoff". |
| 01:52.38 | Samot | Are they using SIP trunks? |
| 01:52.50 | igcewieling | If you *must* use that term, yes. |
| 01:53.08 | Samot | OK SIP Peers.. |
| 01:53.16 | Samot | Registrations, whatever. |
| 01:53.26 | Samot | It's just a delivery method. |
| 01:53.28 | igcewieling | at last count, we had about 70 of them. |
| 01:53.28 | WIMPy | Accounts |
| 01:53.44 | Samot | You can deliver SIP Trunks with 24 channels. |
| 01:53.50 | Samot | You can give them 50 channels. |
| 01:54.18 | igcewieling | Is that a question or a statement? |
| 01:54.23 | Samot | Or just 2. The beauty is, you don't need to add more wiring or expand 66 blocks... |
| 01:54.28 | Samot | That's a statement. |
| 01:54.33 | WIMPy | That's the hilarious thing about SIP accounts. You almost always have a maximum channel limit. |
| 01:54.49 | Samot | When it's "unlimited" yes. |
| 01:54.56 | voipmonk | "" ! right |
| 01:55.17 | igcewieling | for us, the "limit" is usually the bandwidth to the customer. |
| 01:55.18 | WIMPy | Why isn't it? That's COULD be an advantage of using SIP. |
| 01:55.40 | Samot | Why isn't what? |
| 01:56.04 | WIMPy | Why isn't it unlimited? |
| 01:56.16 | Samot | Im talking about usage. |
| 01:56.26 | Samot | They limit channels to control usage. |
| 01:56.28 | WIMPy | And yes, available bandwidth can often be an issue. |
| 01:56.38 | WIMPy | Yes. Why? |
| 01:56.40 | voipmonk | <PROTECTED> |
| 01:57.01 | Samot | Why? business reasons. |
| 01:57.28 | WIMPy | That's just combining the cons of both technologies. |
| 01:57.31 | Samot | You dont have to limit the amount of channels on a SIP trunk.. |
| 01:57.40 | Samot | Most SIP providers don't. |
| 01:57.49 | Samot | They just charge you for usage. |
| 01:58.07 | Samot | VoIP providers are mimicking traditional phone service. |
| 01:58.23 | WIMPy | That would be nice, but isn't the usual thing. |
| 01:58.27 | Samot | Connecting analog phones... |
| 01:58.45 | WIMPy | Yes, and they are doing it pretty badly. |
| 01:58.51 | Samot | Some are. |
| 02:03.40 | drmessano | Depends what you buy |
| 02:04.20 | drmessano | An $8 "Unlimited DID" is limited to 2 channels and about 3000 mins. For $8, you don't get unlimited channels |
| 02:04.39 | drmessano | When you pay metered, there is no practical limit |
| 02:05.04 | drmessano | Unless some provider has a failsafe of x number of channels |
| 02:05.28 | voipmonk | well then u reach their cps limits |
| 02:05.36 | drmessano | I had SIP service with a provider that gave us 32 channels for X dollars per month |
| 02:05.47 | drmessano | But that's how we paid.. it wasn't metered |
| 02:05.54 | drmessano | So basically "It depends" |
| 02:06.42 | Samot | CPS != active channels |
| 02:07.11 | WIMPy | I think you mean CAPS. |
| 02:07.18 | voipmonk | cps = calls per second |
| 02:07.20 | Samot | Calls Per Second. |
| 02:08.11 | Samot | CPS impacts your channels but channels do not impact your CPS. |
| 02:11.28 | *** join/#asterisk wolfmitchell (~wolfmitch@unaffiliated/wolfmitchell) |
| 02:11.41 | Samot | ASR x ACD x CPS (CAPS) x .6 = avg simultaneous channels |
| 02:13.50 | wolfmitchell | For some reason, when I call an extension I have set up on my Asterisk instance, Festival TTS isn't sending any audio unless I do playtones before it |
| 02:14.35 | WIMPy | Did you Answer() the channel? |
| 02:14.41 | wolfmitchell | yep |
| 02:14.55 | wolfmitchell | uploading my extensions.conf now |
| 02:14.59 | wolfmitchell | https://screenshits.nofla.me/2017-03-04_21-14-57.txt |
| 02:15.23 | wolfmitchell | ... probably should have changed the text in it to not identify the company i'm doing this for but w/e |
| 02:16.06 | wolfmitchell | there changed the text to not identify the company :P |
| 02:18.49 | Samot | So is it not playing back at all if there are no tones first or does it playback and there's just no audio? |
| 02:19.19 | voipmonk | play a second of silence before the tts, see how that works for u |
| 02:19.22 | [TK]D-Fender | Playtones would have nothing specific to do with it |
| 02:19.27 | [TK]D-Fender | It jsut happens to setup audio |
| 02:19.37 | [TK]D-Fender | So would a Playback(silence/0) |
| 02:19.42 | Samot | ^^ right |
| 02:19.46 | wolfmitchell | lemme try that |
| 02:19.48 | [TK]D-Fender | Answer() alone often doesn't |
| 02:19.55 | Samot | That's what I was getting at. Is it because it starts audio. |
| 02:20.12 | wolfmitchell | if I do a Playtones then StopPlaytones it doesn't work, though |
| 02:20.17 | Samot | No. |
| 02:20.27 | Samot | 9:19:40 PMÂ <[TK]D-Fender>Â So would a Playback(silence/0) |
| 02:20.28 | Samot | ^^ |
| 02:20.35 | Samot | You need to start the audio stream. |
| 02:20.45 | *** join/#asterisk adnauseum (uid126405@gateway/web/irccloud.com/x-etmtfkqohkczdnqt) |
| 02:20.57 | Samot | No need for Playtones(). |
| 02:21.31 | wolfmitchell | yeah, just silence for 3s then hangs up |
| 02:21.37 | wolfmitchell | and the TTS plays for like 8-10s |
| 02:21.41 | Samot | Show a call |
| 02:21.44 | wolfmitchell | so idk if it's playing it |
| 02:21.47 | Samot | asterisk -rvvvvvvvvv |
| 02:21.50 | Samot | sip set debug on |
| 02:21.53 | Samot | ~pb |
| 02:21.53 | infobot | i guess pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 02:21.55 | Samot | show the results. |
| 02:22.16 | Samot | or pjsip set logger on if you're using PJSIP as the tech. |
| 02:22.49 | adnauseum | 2 people on smartphones using same asterisk server, both use tls/srtp. How long approx. would it take for someone to decode the conversations assuming no access to the asterisk box |
| 02:23.59 | WIMPy | Simple answer: It depends on the available processing power. |
| 02:25.35 | adnauseum | assume its an agency that is trying to listen to your calls. they have tons of money to buy whatever computer that want |
| 02:25.52 | Samot | 10 seconds |
| 02:25.53 | [TK]D-Fender | Assumptions are for idiots |
| 02:25.59 | Samot | Let's assume they have all the power in the world. |
| 02:26.16 | [TK]D-Fender | https://www.google.ca/#q=how+hard+is+it+to+decode+srtp&* |
| 02:26.18 | [TK]D-Fender | GOOGLE |
| 02:26.23 | adnauseum | Samot: that is really not real |
| 02:26.26 | [TK]D-Fender | This shit has been brought up plenty |
| 02:26.51 | Samot | Neither is your question. |
| 02:27.14 | Samot | Well your assumption we are supposed to make in order to answer it. |
| 02:27.35 | WIMPy | If they had serious trouble decoding it, they'd make sure it wouldn't be used. |
| 02:27.39 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 02:27.48 | wolfmitchell | Samot: Internet having issues now and I lost connection to my server, stuck on my phone's 4g and my znc for now |
| 02:28.00 | Samot | First they would need to capture the RTP traffic. |
| 02:28.19 | wolfmitchell | so i cant grab logs :/ |
| 02:28.33 | wolfmitchell | Ill look at it when I get internet again |
| 02:28.34 | adnauseum | Samot: just trying to get an idea of how long it might take if someone was able to grab the packets of the call but had absolutely no access to the astersisk box |
| 02:28.46 | Samot | First they would need to capture the RTP traffic. |
| 02:28.55 | Samot | So they would need to do it on the PBX or on the device. |
| 02:29.01 | Samot | At that point, it's no longer encrypted. |
| 02:29.15 | adnauseum | lets assume the person had access to the cell carrier your smartphone was using for data |
| 02:29.22 | Samot | Or they would need access to one of the boxes the RTP passes through. |
| 02:29.28 | Samot | Why? |
| 02:29.34 | Samot | What is the reasoning for this? |
| 02:29.37 | adnauseum | i will try this again. |
| 02:29.40 | WIMPy | That's more like a fact than an assumption. |
| 02:29.56 | Samot | Without hypotheticals. |
| 02:30.16 | Samot | Or insane ones. |
| 02:31.35 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 02:32.02 | adnauseum | samot, you and I are on a call. My extension is on your server and your extension is on your server, there is no itsp involved. Just your data account with a carrier and my data account with carrier. both of us use tls/srtp. However, there is someone who works with one of our carriers that can capture the packets of our call. How long for that person to |
| 02:32.02 | adnauseum | break the encryption? |
| 02:32.37 | Samot | I don't know. |
| 02:32.59 | WIMPy | That's asking for wild guesses. |
| 02:33.09 | [TK]D-Fender | [TK]D-Fender> https://www.google.ca/#q=how+hard+is+it+to+decode+srtp&* |
| 02:33.09 | [TK]D-Fender | <[TK]D-Fender> GOOGLE |
| 02:33.15 | adnauseum | lets assume for second the person trying to intercept the call works for some sort of spy agency |
| 02:33.16 | WIMPy | But if you want mine, I'd go for a few hours. |
| 02:33.19 | Samot | No. |
| 02:33.29 | WIMPy | max |
| 02:33.34 | Samot | This is just asinine. |
| 02:33.35 | drmessano | lol |
| 02:33.43 | adnauseum | no its not |
| 02:33.44 | [TK]D-Fender | Of course it is |
| 02:33.51 | Samot | Why are you even asking this? |
| 02:33.54 | [TK]D-Fender | vague garbage in hypothecial scenarios with no parameters |
| 02:33.59 | adnauseum | cant say at this point |
| 02:34.06 | drmessano | ..... |
| 02:34.10 | Samot | Or course not. |
| 02:34.17 | Samot | Go Facebook vague shit. |
| 02:34.18 | [TK]D-Fender | And we dont' care |
| 02:34.19 | Samot | Not here. |
| 02:34.21 | drmessano | You know what |
| 02:34.22 | [TK]D-Fender | Go goole this for yourself |
| 02:34.41 | drmessano | Alex Jones called me |
| 02:34.41 | adnauseum | i did a search before asking here, and really, found nothing |
| 02:34.48 | drmessano | He said LNB is paranoid |
| 02:34.55 | drmessano | Really |
| 02:35.11 | Samot | Hey WIMPy! Look, a case in point for your "doing it horribly wrong" statement. |
| 02:35.13 | [TK]D-Fender | My search told me LOTS |
| 02:35.14 | drmessano | He said "That guy is a total paranoid blank blank blank" |
| 02:36.01 | drmessano | I was like "Whoa, Alex.. calm down" |
| 02:36.11 | adnauseum | i am surprised at your answers. |
| 02:36.13 | drmessano | But he kept going on and on |
| 02:36.19 | Samot | Why? |
| 02:36.22 | adnauseum | thought you'd really know or have a close idea |
| 02:36.29 | drmessano | We do |
| 02:36.34 | Samot | Of what a hackers abilities are? |
| 02:36.35 | adnauseum | this is not rocket science |
| 02:36.45 | igcewieling | not every question needs to be answsered. |
| 02:36.46 | adnauseum | its actually a simple question |
| 02:36.53 | Samot | We told you |
| 02:36.58 | Samot | It all depends |
| 02:36.58 | adnauseum | no you did not |
| 02:37.01 | adnauseum | on? |
| 02:37.02 | Samot | We dont have all the variables. |
| 02:37.06 | Samot | They abilities |
| 02:37.09 | Samot | Their resources |
| 02:37.10 | WIMPy | Why not just call the NSA and ask them? |
| 02:37.24 | adnauseum | we dont know their abilities |
| 02:37.29 | Samot | how strong your encryption is... |
| 02:37.46 | Samot | Dude, if someone at a CARRIER is recording your calls for no reason.. |
| 02:37.51 | Samot | It's illegal. |
| 02:37.57 | Samot | you should be talking to authorities. |
| 02:38.01 | adnauseum | Samot: asterisk with both extensions using tls and srtp nothing more |
| 02:38.02 | Samot | Not on IRC. |
| 02:38.08 | Samot | So? |
| 02:38.25 | Samot | If the DATA carrier is intercepting the data and getting the RTP stream... |
| 02:38.56 | adnauseum | yes assume itercepter is able to get RTP |
| 02:39.04 | Samot | It still all depends. |
| 02:39.16 | adnauseum | on what? |
| 02:39.21 | Samot | Are they just running a capture program and have no idea how to deal with encrypted data |
| 02:39.27 | adnauseum | ohh |
| 02:39.29 | adnauseum | ok |
| 02:39.33 | adnauseum | assume FBI |
| 02:39.53 | Samot | Then how much resources they put into it. |
| 02:40.03 | [TK]D-Fender | https://www.youtube.com/watch?v=m2ENydEdJBU |
| 02:40.07 | adnauseum | all they have |
| 02:40.07 | Samot | Now we're just going with conspraicy theories. |
| 02:40.11 | [TK]D-Fender | Funny I can even YOUTUBE search this is SECONDS |
| 02:40.12 | Samot | Why? |
| 02:40.15 | [TK]D-Fender | And you "can't find anything" |
| 02:40.26 | Samot | Why in the world would the FBI be putting all they have into a call of your clients?! |
| 02:40.30 | drmessano | adnauseum: 2 Answers |
| 02:40.33 | [TK]D-Fender | 15 minutes |
| 02:40.35 | [TK]D-Fender | maybe less |
| 02:40.35 | Samot | Unless that client is doing something illegal.. |
| 02:40.39 | adnauseum | i did NOT say my clients |
| 02:40.41 | [TK]D-Fender | Fucking give up computers. |
| 02:40.43 | Samot | Or you. |
| 02:40.47 | drmessano | 128-bit AES = Longer than the length of the universe |
| 02:40.48 | Samot | Doesnt matter. |
| 02:40.53 | adnauseum | so lets get off me. In general would be more accurate |
| 02:40.58 | Samot | No. |
| 02:41.00 | drmessano | #2.. If there is an AES backdoor, which there is, no time |
| 02:41.12 | Samot | This is just speculation and wild guessing. |
| 02:41.17 | drmessano | So if its a US government entity, done |
| 02:41.19 | adnauseum | I hardly thing the FBI wants my pizza client |
| 02:41.21 | [TK]D-Fender | 15 miuntes |
| 02:41.28 | [TK]D-Fender | that's what I see a PILE of videos demoing |
| 02:41.29 | adnauseum | wow |
| 02:41.34 | adnauseum | 15 minutes? |
| 02:41.34 | [TK]D-Fender | that is your HIGH bar at this point |
| 02:42.07 | drmessano | Thats a MiTM attack |
| 02:42.15 | [TK]D-Fender | carrier IS in the middle |
| 02:42.28 | [TK]D-Fender | since that's who we're talking about |
| 02:42.44 | [TK]D-Fender | But this hypothetical spy shit is a waste of our time |
| 02:42.50 | adnauseum | [TK]D-Fender: by carrier you are talking about the DATA carrier for the smartphone correct? |
| 02:42.59 | Samot | Yes. FFS. |
| 02:43.04 | [TK]D-Fender | As is wasteing in on someone who can't GOOGLE |
| 02:43.08 | adnauseum | since there is NO pstn in teh middle |
| 02:43.10 | drmessano | MiTM attack is pretty easy |
| 02:43.14 | Samot | It's DATA |
| 02:43.15 | adnauseum | ok thanks |
| 02:43.17 | adnauseum | right |
| 02:43.20 | [TK]D-Fender | <adnauseum> [TK]D-Fender: by carrier you are talking about the DATA carrier for the smartphone correct? <- ANYONE in the fucking path |
| 02:43.25 | adnauseum | it is Samot |
| 02:43.25 | [TK]D-Fender | It's hitting YOUR server |
| 02:43.28 | Samot | That's what I said. |
| 02:43.35 | [TK]D-Fender | ANY fucker IN BETWEEN |
| 02:43.35 | drmessano | Because you don't need to decrypt the data |
| 02:43.56 | adnauseum | again, I said there is NO access to the asterisk box |
| 02:44.13 | drmessano | Just insert yourself in the call the the endpoints negotiate with you |
| 02:44.24 | adnauseum | its very obvious if the interceptor broke into the box, there is NO protection period |
| 02:44.34 | drmessano | Box access doesnt matter |
| 02:44.42 | drmessano | I just need access to the connection |
| 02:44.46 | drmessano | Which the carriers have |
| 02:44.51 | adnauseum | yes |
| 02:44.56 | Samot | Pointless. |
| 02:44.57 | drmessano | So 15 mins |
| 02:45.02 | adnauseum | wow |
| 02:45.14 | adnauseum | so tls/srtp is basically useless |
| 02:45.18 | Samot | What? |
| 02:45.23 | drmessano | ROFL |
| 02:45.28 | Samot | We're talking about MiTM |
| 02:45.32 | drmessano | If youre the one carrying the fucking data, yes |
| 02:45.34 | Samot | Where they have ACCESS to the PATH |
| 02:45.53 | drmessano | They have access to the wire |
| 02:45.54 | adnauseum | this is more complicated than i thought it would be |
| 02:46.00 | adnauseum | i am very glad i asked |
| 02:46.02 | Samot | What, Health Care? |
| 02:46.05 | drmessano | Your carrier IS the middle |
| 02:46.07 | WIMPy | I've been saying it's useless for a long time. Noone wants to believe. |
| 02:46.08 | Samot | Oh VoIP... |
| 02:46.10 | adnauseum | yes it is |
| 02:46.16 | adnauseum | sip to sip call |
| 02:46.19 | adnauseum | no pstn |
| 02:46.23 | Samot | We know |
| 02:46.25 | drmessano | Doesnt matter |
| 02:46.27 | adnauseum | ok |
| 02:46.28 | Samot | the DATA carrier is involved. |
| 02:46.32 | drmessano | Youre using your carrier DATA |
| 02:46.33 | adnauseum | yes it is |
| 02:46.35 | adnauseum | yes |
| 02:46.38 | adnauseum | 100% |
| 02:46.39 | drmessano | They are in the middle of the DATA |
| 02:46.44 | adnauseum | correct |
| 02:46.45 | Samot | So yes, they can do this. |
| 02:46.48 | adnauseum | ok |
| 02:46.49 | drmessano | THEY ARE THE MIDDLE |
| 02:46.51 | adnauseum | thank you |
| 02:47.37 | drmessano | Sniffing it with wireshark is another story |
| 02:47.48 | Samot | Why a cell carrier would be tapping your data.. |
| 02:47.51 | Samot | For no reason... |
| 02:47.52 | drmessano | So youre pretty safe from some asshole at an internet cafe |
| 02:47.53 | [TK]D-Fender | PARANOIA |
| 02:47.55 | Samot | Beyond me. |
| 02:48.04 | drmessano | But if youre worried about the carrier, forget it |
| 02:48.27 | Samot | Might as well take the battery out and throw it in the trunk. |
| 02:48.30 | drmessano | If the actor has access to the carrier, this all goes to shit |
| 02:49.07 | drmessano | I wouldnt call it useless |
| 02:49.25 | drmessano | You DO have to be able to insert yourself in the middle |
| 02:49.59 | drmessano | But in your example, we've satisfied that |
| 02:50.16 | [TK]D-Fender | There ARE stupid questions. |
| 02:50.22 | [TK]D-Fender | BIGLY |
| 02:51.23 | drmessano | SRTP really isn't the problem |
| 02:51.42 | drmessano | It's the fact that SIP is so easy to MiTM |
| 02:51.54 | drmessano | So why wouldn't SRTP? |
| 02:58.25 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 03:13.06 | adnauseum | from what I have quickly search for mitm, they dont really talk about two sip phones using tls and srtp. If the intruder was unable to break into server connected two SIP callers (using extensions on same server) but did have access to verizon where the intruder could capture the packets of the call, the rtp is still encrypted |
| 03:21.11 | drmessano | You can spoof that |
| 03:21.14 | *** join/#asterisk chendy (~alexc@121.34.129.196) |
| 03:21.25 | drmessano | insert a proxy, done |
| 03:22.02 | drmessano | That's called MAN IN THE MIDDLE |
| 03:22.30 | drmessano | If I break into the server, I am not a man in the middle |
| 03:22.46 | drmessano | I have access to the unencrypted audio |
| 03:23.18 | drmessano | I wouldn't consider that MiTM at all |
| 03:23.29 | drmessano | and that's not what anyone is referring to |
| 03:43.25 | adnauseum | drmessano: let me try this a different way. You want to to have call with another extension on your system. The other extension is several thousand miles from you. You and your buddy want to make a sip call, no ITSP, no telco involved. What would you do to secure it. |
| 03:44.39 | *** join/#asterisk setham (~setham@unaffiliated/setham) |
| 03:44.47 | adnauseum | drmessano: when i say no telco, I mean you are not dialing a typical phone number, you only dial your buddies extension |
| 03:46.09 | [TK]D-Fender | You just said a PROVIDER is going to spy on this |
| 03:46.21 | [TK]D-Fender | They are in the FUCKING MIDDLE |
| 03:46.48 | [TK]D-Fender | And can proxy the whole thing invisibly |
| 03:47.48 | adnauseum | [TK]D-Fender: so what you';re saying is, iiuc, the call can easily be 'listened' to, the srtp means jack shit |
| 03:48.04 | [TK]D-Fender | Not if they are in the FUCKING MIDDLE of it. |
| 03:48.14 | [TK]D-Fender | I didn't just say SPYING on it., |
| 03:48.18 | [TK]D-Fender | there is in the FUCKING PATH |
| 03:48.24 | [TK]D-Fender | A is NOT talking to B |
| 03:48.32 | [TK]D-Fender | C is in the FUCKING MIDDLE |
| 03:48.44 | [TK]D-Fender | And niether side knows SHIT |
| 03:49.00 | adnauseum | how then does C actually hear the words in plain english |
| 03:49.07 | [TK]D-Fender | IT ECHOS IT |
| 03:49.10 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 03:49.25 | [TK]D-Fender | Decodes what IT needs and STILL passes it on |
| 03:49.25 | adnauseum | in plain fucking english? |
| 03:49.34 | adnauseum | wow |
| 03:49.39 | [TK]D-Fender | You ship a box. |
| 03:49.42 | [TK]D-Fender | I GRAB IT. |
| 03:49.48 | adnauseum | so there is no privacy |
| 03:49.49 | [TK]D-Fender | I open it and read everything. |
| 03:49.55 | [TK]D-Fender | then *I* repack the box and pass it on |
| 03:49.59 | adnauseum | wow |
| 03:50.16 | [TK]D-Fender | What part of "in the middle" are you having trouble with' |
| 03:50.25 | [TK]D-Fender | DIRECTLY IN THE WAY. |
| 03:50.27 | adnauseum | plenty obviously |
| 03:50.31 | [TK]D-Fender | FULL FUCKING HIJACK |
| 03:50.42 | adnauseum | ok |
| 03:50.52 | [TK]D-Fender | You asked about the ***ISP** being the perpetrator. |
| 03:51.03 | adnauseum | how then if possible, to prevent that middle attack |
| 03:51.06 | [TK]D-Fender | Well they ARE the fucking provider of your traffic and they can FUCK YOU as much as they want |
| 03:51.49 | [TK]D-Fender | Go do your own research and stop wasting our time |
| 03:52.11 | adnauseum | i dont think its a waste of anyone's time |
| 03:52.17 | [TK]D-Fender | It is |
| 03:52.22 | adnauseum | perhaps you dont want to |
| 03:52.26 | [TK]D-Fender | You say "can't find anything" |
| 03:52.30 | adnauseum | but its far from waste of time |
| 03:52.37 | [TK]D-Fender | 5 seconds on Google and youtube turns up a shit-ton of guides |
| 03:52.45 | [TK]D-Fender | INSTRUCTIONS |
| 03:52.52 | adnauseum | right |
| 03:52.55 | adnauseum | google |
| 03:52.57 | adnauseum | yes |
| 03:52.58 | [TK]D-Fender | And you're wondering if the INTERNET PROVIDER has access |
| 03:53.06 | [TK]D-Fender | They have SERVER FARMS AND YOUR TRAFFIC |
| 03:53.10 | adnauseum | i had this all wrong |
| 03:53.13 | adnauseum | bad |
| 03:53.19 | [TK]D-Fender | Nothing you send goes ANYWHERE without their say-so |
| 03:56.36 | Samot | This is still going on? |
| 04:04.43 | drmessano | Sadly, yes |
| 04:04.47 | drmessano | He doesn't get it |
| 04:04.54 | drmessano | Forget the fucking call |
| 04:05.00 | drmessano | They own the CONNECTION |
| 04:05.14 | drmessano | I don't see how that's hard to understand |
| 04:05.24 | drmessano | Yeah I have a fucking internet on your phone |
| 04:05.36 | drmessano | that's the problem |
| 04:05.54 | drmessano | Verizon is your ISP |
| 04:06.07 | drmessano | when you have a cell phone you have an ISP |
| 04:06.15 | drmessano | Your soft phone uses that ISP |
| 04:06.21 | drmessano | They are in the middle |
| 04:06.45 | drmessano | They can fuck with your packets like I can if I grab your Ethernet cable |
| 04:06.48 | drmessano | Done. |
| 04:07.06 | [TK]D-Fender | Yeah I wasted more time in PM |
| 04:07.27 | drmessano | He's so fucking worried about SRTP |
| 04:07.30 | Samot | Just another reason why I want VoIP to be regulated. |
| 04:07.32 | drmessano | AND THE PBX |
| 04:07.43 | drmessano | And access to the PBX |
| 04:07.53 | drmessano | And the call being ext to ext on the PBX |
| 04:07.56 | [TK]D-Fender | PBX doesn't even matter |
| 04:08.00 | drmessano | Who gives a fuck |
| 04:08.01 | [TK]D-Fender | the traffic goes through them |
| 04:08.04 | drmessano | It's the data |
| 04:08.09 | [TK]D-Fender | nothing goes out they can't proxy and screw with |
| 04:08.13 | drmessano | Yep |
| 04:08.24 | [TK]D-Fender | And numb-nuts here has no clue |
| 04:08.30 | drmessano | If the provider is compromised you're screwed |
| 04:09.10 | [TK]D-Fender | [TK]D-Fender> My WATCHING you hand a box to person B is not the same as you handing ME the box to deliver to person B for you and then I fucking open it and repack a NEW box for them saying "it's from A" |
| 04:11.04 | drmessano | The funny part is |
| 04:11.04 | lorsungcu | it's lnb |
| 04:11.26 | drmessano | If I get a packet capture of an SRTP call |
| 04:11.33 | drmessano | Im going to be at it a while |
| 04:11.51 | drmessano | But Literally one to the nth billion times faster is just get in the middle |
| 04:12.31 | drmessano | I cant make the mailman open a package for me with brute force |
| 04:12.42 | drmessano | But tell him "Im Steve" and he hands it to me, opened |
| 04:13.13 | drmessano | Its secure as hell |
| 04:13.30 | drmessano | As long as you dont insert yourself in any way |
| 04:15.04 | drmessano | Also throw in that SHA1 is now a target |
| 04:15.18 | [TK]D-Fender | #openexploitssaywhaaaat |
| 04:35.08 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 04:49.26 | wolfmitchell | What path does Asterisk look for sounds at by default? (I'm using the debian package for it, if that helps) |
| 04:52.10 | wolfmitchell | nvm, found it |
| 05:44.35 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:47.23 | ChannelZ | OK so totally random OT question |
| 05:48.47 | ChannelZ | Awhile back - months and months ago - someone here was talking about doing custom notifications to their phone (Android I think) with some app |
| 05:49.52 | drmessano | Was the app Pushover? |
| 05:49.57 | ChannelZ | It might have been drmessano, or penguin, or .... |
| 05:50.06 | ChannelZ | well I think you just answered my question :D |
| 05:50.19 | ChannelZ | Do you use that? |
| 05:50.37 | drmessano | Oh yes.. I use the hell out of it.. We have it deployed at the office as well |
| 05:51.08 | ChannelZ | ok.. Yeah that was probably it. I just remember mention of it and thought to myself "hmm I gotta check that out sometime" and then completely forgot what it was |
| 05:51.15 | ChannelZ | Thanks dr :) |
| 05:51.21 | drmessano | Well |
| 05:51.26 | drmessano | All these months later |
| 05:51.40 | drmessano | Not only do I still love it, but I keep finding new uses |
| 05:51.48 | drmessano | So add that to the prior convo |
| 05:52.01 | ChannelZ | Good. |
| 05:53.43 | ChannelZ | Just reading over the site.. so it's basically just $5 for the app? |
| 05:54.20 | drmessano | $5 per platform.. Buy it once, use it forever |
| 05:55.13 | drmessano | I paid $15.. I mostly have iOS stuff.. so that covered that.. I also wanted to try the browser/desktop version, so I paid the $5.. and then I tested for a while on an Android tablet, and I paid $5 there too |
| 05:55.21 | drmessano | But one platform, just the $5 |
| 05:55.39 | ChannelZ | Hmm nice. I shal sign up with the trial and play. Thanks again |
| 06:00.55 | *** join/#asterisk chendy (~alexc@121.34.129.196) |
| 06:50.31 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 07:25.29 | *** join/#asterisk overyander (~Jeff@209.141.208.197) |
| 07:26.22 | overyander | Where are voicemail greetings stored when using odbc based voicemail & voicemail_messages? |
| 07:29.06 | Samot | Where does the ODBC config say they are? |
| 07:29.24 | Samot | They're in a database. |
| 07:29.32 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 07:30.54 | overyander | i see where there's a table for messages and another table for vm account settings. the table for message storage didn't look like it was setup for also storing the users greeting, away and busy messages |
| 07:32.18 | Samot | Sorry, I overlooked the greetings part. |
| 07:32.28 | Samot | Those are not stored in the database. Only messages are. |
| 07:32.55 | overyander | ok. thanks |
| 07:33.18 | Samot | The greetings are probably in /var/spool/asterisk/voicemail |
| 07:33.37 | Samot | Or wherever the directory is set to. |
| 07:33.59 | overyander | yeah, just trying to think of how to centralize that for multiple servers. |
| 07:35.01 | Samot | Send them there. |
| 07:35.10 | Samot | You control the call. |
| 07:35.33 | Samot | Instead of firing off VoiceMail() send a Dial() to the VM server. |
| 07:35.50 | overyander | i wasn't planning to have a separate vm server |
| 07:36.08 | Samot | Whatever server becomes the "central" server. |
| 07:36.29 | Samot | If you want to centralize it, one server is going to be the main voicemail server. |
| 07:37.08 | Samot | It's probably why they choose ODBC to store the messages. |
| 07:37.12 | Samot | All in a database. |
| 07:37.41 | Samot | There's nothing stopping you from storing the greetings in the database. |
| 07:38.26 | overyander | i was planning to have 2+ asterisk boxes and 2 mysql servers in multi-master mode. each * box would have a local mysql server for ps_contacts. dialplan would be the same among all the * servers. all the servers would point to the same 2 mysql servers for all the realtime and db stuff. |
| 07:38.46 | Samot | You can store the greetings in a database table. |
| 07:39.20 | Samot | I just think you have to do some leg work to put them there and pull them.. |
| 07:39.35 | Samot | dan_j has done it I think. |
| 07:41.17 | Samot | So I think you can achieve what you are looking for. I just don't know the steps for it. Never really did ODBC based voicemail. |
| 07:41.33 | overyander | ok. me neither, just read up on it |
| 07:41.55 | overyander | i *think* my plan is solid, just time to start putting it together. |
| 07:41.56 | Samot | But the messages are just stored in a blob datatype. |
| 07:42.06 | Samot | So you can do the same thing with the greetings. |
| 07:42.17 | Samot | You can even use ODBC to connect back to it. |
| 07:45.16 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 07:46.05 | overyander | one thing i haven't figured out yet is how to share queues across servers. doesn't * keep the details of the queue strategy in the local db? for example, if using rrmemory, the last member used is stored locally. so, if you have the same queue defined on 2 servers, and calls coming in on both systems for load balancing how is server A going to know where server B left off at in the queue? |
| 07:46.46 | Samot | As far as I know, you cant share them across servers.. |
| 07:47.24 | Samot | I wouldn't load balance a queue. |
| 07:48.19 | *** join/#asterisk Oatmeal (~Suzeanne@c-68-45-50-54.hsd1.nj.comcast.net) |
| 07:48.24 | Samot | It would make doing reporting a nightmare, one thing that comes to mind. |
| 07:48.48 | overyander | i'm trying to set this up so that the phone system has at least 2 servers running in unison. a call can come in to either of them and get routed properly, a phone can register to either and the user won't have a clue. if one goes down then all is good and things just keep working. |
| 07:49.11 | Samot | How many users? |
| 07:49.28 | Samot | Why wouldn't you run this is some sort of failover? |
| 07:49.33 | overyander | right now just ~400 |
| 07:49.44 | Samot | Load balancing calls is more work than you want for this. |
| 07:49.54 | Samot | OK, so just have them all on a server... |
| 07:50.26 | Samot | Have another server running as a warm spare that is updated/syncd |
| 07:50.52 | overyander | i know a single server can handle that fine, but for redundancy, balancing, growth reasons, we can't just limit it to one server |
| 07:51.09 | overyander | i was trying to come up with something a bit more dynamic that could grow or shrink on demand |
| 07:51.21 | Samot | Having one queue load balanced over multiple servers is going to be a pain. |
| 07:51.31 | Samot | Run VMs. |
| 07:51.39 | Samot | Grow them as needed. |
| 07:51.45 | overyander | we do run vm's |
| 07:51.56 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 07:52.23 | overyander | we also have 15 offices scattered across the US and about 150 people who work from home. the system can't be tied to just one location |
| 07:53.15 | Samot | That's not true but OK. |
| 07:53.31 | Samot | Queues can be run in real time as for as the queues and the members themselves. |
| 07:53.33 | overyander | what's not true? |
| 07:53.55 | Samot | That one location can't handle all the calls with proper backup/failover |
| 07:54.02 | overyander | it can |
| 07:54.13 | overyander | yes, until something takes out that physical location |
| 07:54.20 | overyander | then everyone's screwed |
| 07:54.22 | Samot | That's why you have a BACKUP |
| 07:54.26 | Samot | In another location. |
| 07:55.23 | Samot | I have no idea how you would know agents states across a single queue on multiple servers... |
| 07:55.34 | Samot | How they would even log into multiple servers for a single queue.. |
| 07:55.46 | overyander | we don't use logins |
| 07:55.50 | Samot | OK |
| 07:55.54 | overyander | just very simple queues |
| 07:55.57 | Samot | How do they register their phones? |
| 07:56.08 | Samot | How will you tell the queues these agents are available? |
| 07:56.26 | Samot | How will the servers know the locations of the phones to send calls to? Or get their states from? |
| 07:56.37 | overyander | time rule for when the office hours are. if nobody is at a certain desk/phone then the call goes to the next one. |
| 07:56.55 | Samot | How does server A, B and C know that Phone X exists? |
| 07:56.58 | Samot | And where it is? |
| 07:57.30 | Samot | How does Phone X register to servers A, B and C? |
| 07:58.10 | overyander | all of that part has already been figured out |
| 07:58.20 | Samot | So how do you do it? |
| 07:58.46 | Samot | How does server A and B know not so send a call to Phone X because it's on a call on Server C? |
| 07:59.24 | Samot | How do you run reporting on any of this? |
| 07:59.41 | Samot | You'll have calls logged across multiple servers... |
| 08:00.30 | overyander | server A doesn't care if the phone is already on a call at C. it attempts to dial and then does what's necessary based on the dialstatus received back. |
| 08:00.40 | Samot | ? |
| 08:00.42 | Samot | Why? |
| 08:00.48 | Samot | Why are you trying to call a busy agent? |
| 08:01.26 | Samot | What happens when two users in the queue, on two different servers, try the same agent? |
| 08:01.33 | overyander | dude, the queue part is what i said i had NOT FIGURED OUT. the regular calling stuff IS FIGURED OUT. |
| 08:01.44 | Samot | I'm not talking about regular calling. |
| 08:01.54 | Samot | I'm talking about the queues. |
| 08:02.00 | overyander | you started asking how servers know that phones exist and how to contact them, that's when i said that part is figured out |
| 08:02.10 | Samot | Because your agents are on phones. |
| 08:02.16 | Samot | Statically assigned to a queue |
| 08:02.20 | Samot | That exists on three servers. |
| 08:02.23 | Samot | For example. |
| 08:02.51 | Samot | How does that phone register with two servers, log the agent in and keep the status of the agent for calls in the queue? |
| 08:03.11 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
| 08:03.58 | overyander | 1) I never said anything about a phone registering with multiple servers. 2) I already said we don't do agent logins. |
| 08:04.11 | Samot | How do your agents get calls? |
| 08:04.35 | Samot | How in the hell does a call get set to an agent if you don't have phones registered? |
| 08:04.53 | Samot | How does the queue know that phone is even available to take calls? |
| 08:04.56 | overyander | a phone can be registered without having anything to do with agents or queues |
| 08:05.28 | Samot | How does Server A know what phones/agents are registered on the system and thus available to get calls? |
| 08:06.43 | Samot | Both servers need to know that phone/agent Y is registered/online and available to take a call. |
| 08:07.07 | overyander | [02:01:26] overyander:dude, the queue part is what i said i had NOT FIGURED OUT. |
| 08:07.16 | Samot | Right.. |
| 08:07.18 | Samot | I GET THAT |
| 08:07.21 | Samot | This are the PIT FALLS |
| 08:07.26 | Samot | Of trying to LOAD BALANCE |
| 08:07.33 | overyander | then why do you keep asking me the same question about it? |
| 08:07.33 | overyander | lol |
| 08:07.40 | Samot | Because you haven't answered. |
| 08:07.50 | Samot | Or acknowledged that you've considered it. |
| 08:07.55 | Samot | Any of those items. |
| 08:08.26 | Samot | This is why I am saying.. |
| 08:08.34 | Samot | Primary server..with live backups.. |
| 08:08.46 | Samot | In geo redundant locations. |
| 08:08.49 | overyander | [01:46:05] overyander:one thing i haven't figured out yet is how to share queues across servers. doesn't * keep the details of the queue strategy in the local db? for example, if using rrmemory, the last member used is stored locally. so, if you have the same queue defined on 2 servers, and calls coming in on both systems for load balancing how is server A going to know where server B left off at in the queue? |
| 08:09.15 | overyander | that's when i acknowledged those pitfalls |
| 08:09.23 | Samot | 2:53:34 AM S<Samot> Queues can be run in real time as for as the queues and the members themselves. |
| 08:09.37 | Samot | But just their information.. |
| 08:09.57 | Samot | Not the actual agent states. |
| 08:11.14 | Samot | So *IF* you set up the same queue on server A and B... |
| 08:11.18 | Samot | That's fine. |
| 08:11.25 | Samot | You can tell it to have the same agents. |
| 08:11.28 | Samot | But now.. |
| 08:11.37 | Samot | Your agents need to register to EACH server. |
| 08:12.34 | Samot | Which is basically impossible for a single account. |
| 08:15.26 | Samot | You would at least need some sort of proxy/sbc between the phones and the servers. THAT might be able to send updates and information to both servers. Because then it would be Phone --> Proxy/SBC ---> Server A & B |
| 08:16.12 | Samot | But the phones would only talk to the Proxy/SBC and it would send updates to the servers and the servers would talk to the proxy/sbc which would handle each message and send it to the phone. |
| 08:16.21 | Samot | In theory. |
| 08:20.17 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:20.34 | overyander | what is the "state_interface" field in database queue_members table for? |
| 08:28.04 | Samot | Not 100%, I haven't used RealTime that much and never queues with it. |
| 08:30.12 | Samot | Oh, it's where to Dial() |
| 08:30.40 | Samot | So if it was device/agent 100 it would be SIP/100 or PJSIP/100, depending on your tech choice. |
| 08:30.42 | overyander | wouldn't that be 'interface'? |
| 08:31.23 | Samot | Where are you seeing this? |
| 08:31.25 | Samot | Link it |
| 08:31.50 | overyander | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html and the linked doc http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html#ACD_id288726 |
| 08:33.49 | Samot | None of this is RealTime based. |
| 08:34.04 | Samot | state_interface is the state of the device, which is stored in memory. |
| 08:34.48 | overyander | the field is generated by the db scripts |
| 08:35.12 | overyander | it must just be static db if it's not true "realtime" |
| 08:38.42 | Samot | I don't know, I don't state_interface in any queue member table schema in the docs I've found so far. |
| 08:38.46 | Samot | So I have no idea. |
| 08:41.56 | overyander | it's in the alembic script for the config setup |
| 08:43.14 | overyander | do you know if you can specify a context for a queue to use when dialing? or do you have to dial TECH/PEER directly? |
| 08:44.28 | Samot | The queue will use local channels to try to dial the devices. |
| 08:46.55 | *** join/#asterisk aness (~aness@2a02:fe1:3102:a700:47c:ba4c:2728:9c17) |
| 08:54.51 | overyander | i think i just solved the problem Samot |
| 08:55.03 | Samot | OK |
| 08:55.46 | overyander | queues will be on a separate server. queue members are configured as dummy sip peers; these same dummy accounts also exist on the main * servers. when queue app calls sip/queue1001 the exten is registered to main server... main server answers in the appropriate context and routes the call to the appropriate phone i.e. extensions,1001 |
| 08:56.22 | overyander | it's a bit ugly but i think that would work. |
| 09:02.53 | overyander | keeping it as simple as possible. you'd have Server A and Server Q. all phones register to server A. All PSTN calls come in to server A. When server A needs to send something to a queue, it sends it to Server Q which has the queue rules, members, etc. The members of the queus will be more like quasi-peers. Server Q will have a queue "foobar" with queue members SIP/queue_foobar_1000 and SIP/quue_foobar_1001. the peers queue_foobar_1000 and |
| 09:02.53 | overyander | <PROTECTED> |
| 09:03.32 | overyander | you'd have to get pretty fancy with hints and stuff. |
| 09:05.31 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 09:07.42 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 09:15.06 | *** join/#asterisk Maliuta_ (maliutamat@gateway/shell/matrix.org/x-paphihvqryclzqvz) |
| 09:15.11 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 09:15.12 | overyander | Samot, thoughts? ^^ |
| 09:18.15 | lorsungcu | overyander: what happens when either server fails? |
| 09:19.44 | overyander | lorsungcu, Server A is actually part of a larger cluster of active-active/master-master servers, so no issue there. server q will have a standby failover |
| 09:20.16 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:5d91:b93e:8b61:ff76) |
| 09:26.46 | *** join/#asterisk defsdoor (~andy@cpc35-sutt4-2-0-cust184.19-1.cable.virginm.net) |
| 12:07.38 | *** join/#asterisk chendy (~alexc@121.34.129.196) |
| 12:18.03 | *** join/#asterisk miralin (~Thunderbi@194.8.128.50) |
| 12:38.58 | *** join/#asterisk miralin1 (~Thunderbi@194.8.128.50) |
| 12:53.05 | *** join/#asterisk miralin (~Thunderbi@194.8.128.50) |
| 13:10.20 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 13:18.09 | *** join/#asterisk miralin (~Thunderbi@194.8.128.50) |
| 13:39.27 | *** join/#asterisk miralin (~Thunderbi@194.8.128.50) |
| 13:55.28 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 14:04.41 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 14:06.52 | *** join/#asterisk j0hnd0e (~Mutter@24.48.86.58) |
| 14:13.46 | *** join/#asterisk chendy (~alexc@121.34.129.196) |
| 14:18.23 | Samot | Sounds like a lot of work for a hack solution. |
| 14:24.36 | WIMPy | Is that the summary of the whole VoIP thing? |
| 14:25.26 | Samot | There's nothing wrong with VoIP. |
| 14:26.10 | WIMPy | Wow |
| 14:26.45 | Samot | There's not. |
| 14:27.44 | WIMPy | That's what I call ultimate devotion :-) |
| 14:28.05 | Samot | It's what I call being practical. |
| 14:29.02 | Samot | SIP interconnects play a big part of how Telecom is handled now. Even the RBOCs/ILECs use SIP for their connections between each other. |
| 14:29.29 | WIMPy | I'm not doubting that. |
| 14:30.33 | Samot | SIP or TDM, doesn't matter, if you're shitty at providing voice services the delivery method isn't going to overly matter. |
| 14:31.17 | WIMPy | It matters a lot. One of thme just makes it impossible to provide good service. |
| 14:31.30 | Samot | How? |
| 14:31.55 | Samot | You build a horrible "copper" network and have shitting PRI connections/CPE's etc... |
| 14:31.56 | WIMPy | Have you ever compared them? |
| 14:32.05 | Samot | I've worked for a Tier I CLEC |
| 14:32.15 | Samot | They bought my ITSP so they could expand their SIP offerings |
| 14:32.17 | Samot | And network. |
| 14:32.18 | Samot | Yes. |
| 14:32.26 | Samot | I've compared both, worked with both, done both. |
| 14:33.15 | WIMPy | Well, most people have stopped believing in VoIP. And there are just too many reasons to mention. |
| 14:34.01 | Samot | I'm not really sure who that is since business hasn't really stopped growing and I replace a lot of copper based services. |
| 14:34.20 | WIMPy | But luckily it's really old stuff so there's hope we get something new some time. |
| 14:46.18 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 14:51.45 | *** join/#asterisk aness (~aness@cm-84.209.155.137.getinternet.no) |
| 15:11.49 | avb | WIMPy: :) dont argue with an expert |
| 15:12.35 | WIMPy | Oh, yes. We all love experts. |
| 15:13.19 | avb | as im usually telling 'i love asterisk and voip as much as I hate them' |
| 15:13.21 | avb | :) |
| 15:14.16 | WIMPy | It's like Windows 95. :-) |
| 15:14.18 | *** join/#asterisk newtonr (~newtonr@173-17-133-211.client.mchsi.com) |
| 15:14.18 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:14.57 | avb | last bits what webrtc added to make it work with sip drived me crazy |
| 15:15.37 | avb | before i thought that im starting to understand sdp :) |
| 15:31.14 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 15:51.26 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
| 16:04.40 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 16:18.51 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 17:09.47 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 17:21.02 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 19:32.53 | *** join/#asterisk newtonr_ (~newtonr@173-17-133-211.client.mchsi.com) |
| 19:32.53 | *** mode/#asterisk [+o newtonr_] by ChanServ |
| 19:53.16 | *** join/#asterisk miralin (~Thunderbi@194.8.128.50) |
| 20:10.16 | adnauseum | WIMPy: i think socialism got to you |
| 20:24.14 | *** join/#asterisk Tristan-Speccy (~tristan@2001:1af8:4010:a021:2::) |
| 20:30.15 | *** join/#asterisk Tristan-Speccy (~tristan@antares.whatbox.ca) |
| 21:45.28 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 21:49.52 | *** join/#asterisk j0hnd0e (~Mutter@24.48.86.58) |
| 22:01.30 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:11.01 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
| 23:20.58 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |