| 00:06.01 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1003:215:5dff:fe07:4806) |
| 00:08.01 | *** join/#asterisk nanoha-sama (~nanoha-sa@2001:470:e97f:1003:215:5dff:fe07:4806) |
| 00:09.30 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 00:09.31 | phix | igcewieling1: what did that do? |
| 00:09.36 | mattwj2002 | hi guys |
| 00:09.36 | mattwj2002 | :D |
| 00:09.52 | igcewieling1 | phix: it caused one way audio |
| 00:09.57 | phix | :D |
| 00:10.00 | phix | hai mattwj2002!!!! |
| 00:10.10 | mattwj2002 | I have asterisk on a raspberrypi |
| 00:10.11 | mattwj2002 | :P |
| 00:10.14 | phix | wooooo!!! |
| 00:10.18 | mattwj2002 | hai phix! |
| 00:10.24 | phix | now setup a samba4 DC on it |
| 00:10.30 | mattwj2002 | haha |
| 00:10.36 | mattwj2002 | and watch it start on fire :P |
| 00:10.40 | phix | heh |
| 00:10.52 | ketas | hmm |
| 00:10.59 | phix | sup ketas ? |
| 00:11.48 | mattwj2002 | man compiling is slow on the pi! |
| 00:11.50 | ketas | wondering if such embedded boards with ethernet are cheaper, i can put one into each lamp and wall switch |
| 00:11.53 | ketas | :P |
| 00:11.58 | phix | mattwj2002: cross compile then |
| 00:12.11 | mattwj2002 | cross compile? O.o |
| 00:12.19 | mattwj2002 | never heard of such a thing |
| 00:12.19 | ketas | whole new meaning to ethernet switch |
| 00:12.21 | phix | ketas: haha, they are pretty cheap, ~$40 AUD |
| 00:12.40 | mattwj2002 | they are cheap |
| 00:12.58 | mattwj2002 | I recommend a good 2.1 A ipad charger |
| 00:13.06 | phix | mattwj2002: yeah, cross compile, where you compile code on an intel computer to run on an ARM |
| 00:13.11 | mattwj2002 | and a nice case |
| 00:13.17 | mattwj2002 | O.o |
| 00:13.21 | mattwj2002 | is that even possible? |
| 00:13.23 | phix | yes |
| 00:13.28 | phix | it is called cross compiling :) |
| 00:13.37 | mattwj2002 | wicked |
| 00:13.39 | ketas | phix: well, surely they are cheaper than crazy expensive automation bus wall switches |
| 00:13.51 | phix | ketas: yeah those things are crazy |
| 00:13.52 | vastersk | anyone, how do you properly stop and restart asterisk? |
| 00:14.03 | phix | ketas: however the bus wall switches probably use less power |
| 00:14.06 | ketas | phix: but still bit too much for this purpose |
| 00:14.07 | mattwj2002 | core stop gracefully |
| 00:14.10 | mattwj2002 | I think |
| 00:14.16 | phix | vastersk: sudo service asterisk stop |
| 00:14.18 | ketas | phix: power -> heat, etc |
| 00:14.24 | phix | yup |
| 00:14.38 | phix | but I have already wanted to run apache in my wall socket :) |
| 00:14.47 | ketas | you can |
| 00:14.54 | mattwj2002 | :P |
| 00:15.04 | mattwj2002 | in the wall socket? |
| 00:15.05 | mattwj2002 | :P |
| 00:15.07 | ketas | yes |
| 00:15.10 | ketas | inside it |
| 00:15.13 | phix | I will hide them in my wall |
| 00:15.17 | vastersk | ok, thanks :) |
| 00:15.19 | mattwj2002 | you could! |
| 00:15.38 | ketas | i'm planning massive cat5 pulling |
| 00:15.43 | phix | my army of wall arms :) |
| 00:15.53 | ketas | wall-a |
| 00:16.06 | file | "friendly-scanner" amuses me |
| 00:16.08 | ketas | arm is in your arms |
| 00:16.18 | phix | wooo |
| 00:16.23 | ketas | if you use phone |
| 00:16.25 | ketas | :P |
| 00:16.55 | phix | oh yeah, my phone is an arm too! |
| 00:16.57 | phix | awesome |
| 00:17.06 | ketas | ARMy |
| 00:17.17 | phix | yup |
| 00:17.30 | ketas | army army |
| 00:17.33 | ChannelZ | I have an arm |
| 00:17.38 | ketas | i have two |
| 00:17.49 | ChannelZ | and I'm armed |
| 00:18.12 | ketas | i have phone and i'm not afraid to use it |
| 00:18.15 | igcewieling1 | file: calling it friendly scanner is like calling Genghis Kahn a friendly Mongol. |
| 00:18.20 | phix | Lets see, 2x wireless routers, 1 adsl router, 1 phone, 1 tablet, a printer, and soon to be an army of PIs :) |
| 00:18.26 | file | igcewieling1, quite |
| 00:18.30 | phix | lots of arms here |
| 00:18.36 | mattwj2002 | :D |
| 00:18.42 | ketas | https://en.wikipedia.org/wiki/Mobile_phone_throwing |
| 00:19.04 | mattwj2002 | I am killing my pi |
| 00:19.09 | ketas | why |
| 00:19.10 | phix | I prefer midget throwing |
| 00:19.11 | ketas | poor pi |
| 00:19.21 | file | what REALLY amuses me is this packet from it: |
| 00:19.22 | file | http://pastebin.com/9MUqZg0T |
| 00:19.24 | mattwj2002 | I screwed it up |
| 00:19.26 | mattwj2002 | :( |
| 00:20.01 | phix | file: haha |
| 00:20.26 | phix | mattwj2002: install Ubuntu or Debian onto your PI imo |
| 00:20.37 | ketas | calculate pi on pi |
| 00:20.43 | ChannelZ | while eating pie |
| 00:20.44 | phix | or are you using the offical one that you can purchase with it? |
| 00:20.51 | igcewieling1 | file: I saw something similar. our adtrans where complaining about invalid lines in the packet |
| 00:20.53 | phix | while looking at a pie chart |
| 00:21.06 | ChannelZ | program it in python |
| 00:21.08 | mattwj2002 | yeah I am doing a reinstall |
| 00:21.12 | mattwj2002 | raspbian |
| 00:21.18 | phix | python <3 |
| 00:21.23 | mattwj2002 | hey guys I have a question |
| 00:21.26 | phix | shoot |
| 00:21.31 | ketas | python will eat you |
| 00:21.33 | ChannelZ | but we are discussing pie |
| 00:21.38 | phix | hehe |
| 00:21.43 | mattwj2002 | has anyone got asterisk and google voice working together? |
| 00:21.49 | phix | never tried |
| 00:21.51 | ChannelZ | Sort of. |
| 00:21.56 | mattwj2002 | I am able to make outgoing calls but not incoming |
| 00:21.57 | phix | why would you want to do such a thing? |
| 00:22.06 | mattwj2002 | sort of ChannelZ? |
| 00:22.12 | mattwj2002 | free long distance! |
| 00:22.14 | ChannelZ | Well I don't use it that much |
| 00:22.20 | mattwj2002 | free incoming number |
| 00:22.33 | mattwj2002 | US only :) |
| 00:22.34 | phix | mattwj2002: In which country? |
| 00:22.38 | phix | yeah thought so |
| 00:22.39 | ChannelZ | and under 11/Motif I've had random issues with it |
| 00:22.40 | mattwj2002 | USA |
| 00:22.48 | phix | well that is useless to me :) |
| 00:22.52 | phix | I live in AU |
| 00:23.06 | mattwj2002 | your an Aussie? I am a Yankee nice to meet you |
| 00:23.07 | mattwj2002 | :P |
| 00:23.11 | phix | :D |
| 00:23.36 | phix | yes, pronouned ozzy not HAUS-EEE |
| 00:23.43 | ChannelZ | hussy |
| 00:24.34 | ChannelZ | Anyway, do the calls come in at all or you just can't get it to work afterwards? |
| 00:24.35 | mattwj2002 | :P |
| 00:24.56 | mattwj2002 | I don't know if they are even hitting the asterisk box |
| 00:25.01 | mattwj2002 | did you have to forward some ports? |
| 00:25.21 | phix | you behind NAT? |
| 00:25.27 | mattwj2002 | yeah |
| 00:25.35 | ChannelZ | well it uses tcp |
| 00:26.06 | mattwj2002 | ok |
| 00:26.22 | mattwj2002 | so what ports? |
| 00:26.23 | ChannelZ | you should turn on xmpp debug |
| 00:26.41 | ChannelZ | 5222 I think, it's in your config |
| 00:26.46 | mattwj2002 | ChannelZ: what version are you using? |
| 00:26.54 | ChannelZ | 11 |
| 00:26.57 | mattwj2002 | okay |
| 00:26.58 | ketas | bleh, why is it that company sites are total crap always |
| 00:27.23 | vastersk | when you start asterisk again, will it be sudo service asterisk start? |
| 00:27.39 | mattwj2002 | ChannelZ: do you know if it works with 1.8? |
| 00:28.14 | ChannelZ | Well, it used to.. but Google likes to f* with the protocol a lot so I dunno how well it works these days |
| 00:28.23 | ChannelZ | For that it's 'jabber debug' then |
| 00:28.30 | ChannelZ | or jabber set debug on |
| 00:28.40 | mattwj2002 | would you recommend 11 or 1.8? |
| 00:28.49 | mattwj2002 | I am rebuilding my box |
| 00:28.54 | ChannelZ | 11 unless you need to live in the past for some reason |
| 00:29.00 | mattwj2002 | ok :) |
| 00:29.05 | ChannelZ | (some other module, etc.) |
| 00:29.12 | mattwj2002 | I do like 80s music ;) |
| 00:29.25 | ChannelZ | The config is a little different between the two |
| 00:29.32 | [TK]D-Fender | If I knew how to play any more hair metal I'd have to buy stock in Revlon.... |
| 00:29.36 | mattwj2002 | yeah I am going to just research it |
| 00:29.50 | ChannelZ | IE jabber is gone, xmpp is in, chan_google or whatever it was is now Motif |
| 00:30.05 | file | there have been very few issue reports regarding google voice support |
| 00:30.12 | mattwj2002 | yeah I installed 1.8 |
| 00:30.16 | file | and from what I've heard it mostly "just works" for a lot of people, with Motif |
| 00:30.19 | mattwj2002 | and tried to upgrade to 11 |
| 00:30.22 | mattwj2002 | it wasn't happy |
| 00:31.02 | ChannelZ | It just behaves a little different, like calls come in a little differently than they did before |
| 00:31.38 | mattwj2002 | well I had a module issue |
| 00:31.45 | igcewieling1 | sure would be awsome if they fixed this bug: asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory |
| 00:31.53 | mattwj2002 | I figured I might as well just restall |
| 00:32.49 | mattwj2002 | *reinstall |
| 00:36.46 | ChannelZ | meh.. well my own gtalk seems not to be working at the moment |
| 00:37.00 | mattwj2002 | bummer |
| 00:37.27 | ChannelZ | oh no there it goes, the web client is just stupid |
| 00:38.38 | ChannelZ | * is just sitting there though. hmm |
| 00:40.48 | mattwj2002 | google down? |
| 00:40.49 | mattwj2002 | :P |
| 00:40.49 | ChannelZ | I get the jingly messages but it apparently doesn't understand them |
| 00:40.50 | mattwj2002 | jk |
| 00:41.43 | file | what are you trying to do? |
| 00:42.00 | ChannelZ | I was just testing gtalk since I haven't used it in awhile. |
| 00:42.06 | ChannelZ | See if it still worked |
| 00:42.29 | ChannelZ | I'm trying to even remember my google voice number to see if that works either. |
| 00:43.25 | mattwj2002 | you lose them if you don't use them |
| 00:43.28 | mattwj2002 | btw |
| 00:43.59 | *** join/#asterisk adeel (~adeel@64.229.153.204) |
| 00:44.00 | ChannelZ | I know they sent me a message about it awhile back asking if I still owned the number it was forwarding to |
| 00:44.32 | adeel | anyone ever happen to use the PAMI php class? http://marcelog.github.com/PAMI/index.html |
| 00:45.20 | ChannelZ | nope that ain't workin either. I get the XMPP message from google but * doesn't do anything. I must have something weird. |
| 00:51.38 | ChannelZ | dunno if it's the transport= in motif.conf or not.. the sample config is a bit confusing, it shows one as transport=google for "Google Talk" but then says Google Voice uses transport=google-v1 "the original Google Talk protocol" |
| 00:53.16 | *** part/#asterisk navaismo (~navaismo@189.241.118.172) |
| 00:53.19 | file | if * doesn't do anything then it's either 1. The account specified in motif.conf isn't correct or 2. You got hit by a bug with occurred if the connection bounced with Google, it has since been fixed |
| 00:53.27 | file | erm which |
| 00:53.36 | ChannelZ | huh switching it to google-v1 is working for both now. |
| 00:54.01 | mattwj2002 | google-v1? |
| 00:54.13 | file | there's 3 different Jingle protocols |
| 00:54.14 | ChannelZ | Yes for the transport= line in motif.conf |
| 00:54.24 | mattwj2002 | interesting |
| 00:54.31 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 00:54.44 | ChannelZ | Does Google use all 3 at random? :) |
| 00:54.46 | file | the standard, the slightly different Google standard, and the pre-standard "hey I'm not documented anywhere" Google |
| 00:54.53 | file | technically, yes |
| 00:55.02 | file | Google Voice use the pre-stand |
| 00:55.10 | file | gmail uses the slightly different Google standard |
| 00:55.13 | file | and other clients use the standard |
| 00:56.19 | ChannelZ | which google-v1 I guess seems to cover the first 2 |
| 00:56.30 | file | google-v1 is the pre-standard |
| 00:56.37 | ChannelZ | (both my GV number and calling through gmail seemed to work) |
| 00:56.46 | file | the code will also try to automatically figure out what the other side is using, and switch accordingly |
| 00:57.22 | ChannelZ | Does it default to something? I had it set as transport=google but I notice the Wiki sample setup doesn't show it at all. |
| 00:57.46 | mattwj2002 | ChannelZ: what wiki are you talking about? |
| 00:57.52 | file | ; If the target is not found in the roster the target will be used as-is and a session will be initiated using the |
| 00:57.52 | file | ; transport specified in this configuration file. If no transport has been specified the endpoint defaults to ice-udp. |
| 00:58.01 | ChannelZ | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 00:58.48 | ChannelZ | oh.. I need to re-copy all the dist configs into my sample dir :) |
| 00:59.10 | file | huh using my own nickname on my desktop triggered notifications to my cellphone and tablet, funn |
| 01:02.12 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 01:02.12 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 01:02.22 | mattwj2002 | O.o |
| 01:02.42 | mattwj2002 | don't kill me Qwell! |
| 01:05.50 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 01:07.41 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 01:09.31 | *** join/#asterisk mzb| (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 01:09.50 | mattwj2002 | god I hate IPv6 |
| 01:09.51 | mattwj2002 | :P |
| 01:09.59 | file | there's no place like [::1] |
| 01:10.04 | mattwj2002 | hehe |
| 01:10.49 | igcewieling1 | " My idea is to rip out Asterisk and install some key system from the mid 1980's which we get off ebay. That is what they actually want. " <-- customer is driving us up the wall |
| 01:10.58 | igcewieling1 | so there is my solution |
| 01:11.13 | mattwj2002 | what is wrong igcewieling1? |
| 01:11.26 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 01:11.27 | file | Joe! Pick up Line 1! |
| 01:11.35 | mattwj2002 | hehe |
| 01:11.59 | igcewieling1 | mattwj2002: Our sales people don't do a very good job of explaining before the sale that Asterisk is not a key system |
| 01:12.14 | *** join/#asterisk mzb- (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 01:12.32 | mattwj2002 | are they the fax type people too? |
| 01:12.40 | mattwj2002 | hey Joe can you fax me that |
| 01:12.41 | mattwj2002 | :P |
| 01:12.56 | mattwj2002 | I hate fax machines |
| 01:12.58 | igcewieling1 | mattwj2002: hey now, there are plenty of perfectly nice people who use fax. |
| 01:13.13 | igcewieling1 | And using fax is still *reasonable* |
| 01:13.32 | mattwj2002 | I guess |
| 01:13.35 | igcewieling1 | expecting every "line" to appear on every phone may have been reasonable 20 years ago, but it is not reasonable today. |
| 01:13.56 | mattwj2002 | hey guys have you meant my friend called "Mr. E-mail"? |
| 01:14.02 | mattwj2002 | *met |
| 01:14.11 | mattwj2002 | or worse yet |
| 01:14.32 | mattwj2002 | someone scans it using a really crappy fax machine and then e-mail it |
| 01:14.44 | mattwj2002 | I hate that :) |
| 01:15.16 | igcewieling1 | most people hate changing their bank and most people hate changing phone companies so the only customers our sales people seem to get are people who are unhappy with their current phone company, often for no other reason that the people are simply assholes and will be unhappy with any phone company. |
| 01:15.19 | *** join/#asterisk jhirley (~chatzilla@c-76-18-61-12.hsd1.fl.comcast.net) |
| 01:15.57 | mattwj2002 | so what are they saying exactly? |
| 01:16.16 | igcewieling1 | they want the same "line" to appear on different phones. |
| 01:16.58 | mattwj2002 | why? |
| 01:17.09 | igcewieling1 | this is not a new issue, but we have one customer who is complaining right now. |
| 01:17.23 | igcewieling1 | mattwj2002: because they want it to work like their old system |
| 01:17.34 | mattwj2002 | got ya |
| 01:18.34 | igcewieling1 | joe's line rings, you talk to the person, you put the call on hold, holler to joe in the office next door, hey you have a call! and then have joe press the button on his phone for that "line" and start talking. They are apparently too stubborn to learn to transfer or park calls |
| 01:18.54 | mattwj2002 | lol |
| 01:19.44 | igcewieling1 | my boss thinks everyone hates everyone else and don't want to talk to any of their coworkers and that is the real issue. |
| 01:20.15 | mattwj2002 | :P |
| 01:20.23 | mattwj2002 | sounds like a great place to work ;) |
| 01:20.35 | igcewieling1 | we are just their phone company/pbx vendor |
| 01:20.46 | mattwj2002 | no I mean your customer |
| 01:21.08 | mattwj2002 | sounds like a great place to work at your customer's company |
| 01:21.14 | igcewieling1 | *nod* |
| 01:21.18 | apb1963_ | simple solution (I think)... relabel the "Hol" button. |
| 01:21.21 | apb1963_ | oops |
| 01:21.23 | igcewieling1 | thankfully I don't usually have to interact with customers |
| 01:21.31 | apb1963_ | Relable the "Park" button... call it "Hold" |
| 01:22.06 | igcewieling1 | ah, we thought of something similar, but what they want would require a "personal" parking lot accessable from all phones. |
| 01:24.19 | mattwj2002 | igcewieling1: do you have a short user guide? |
| 01:24.26 | mattwj2002 | like a single page? |
| 01:24.36 | igcewieling1 | no idea, that is a sales thing |
| 01:24.59 | igcewieling1 | I believe we have several documents covering various things but nobody seems to read them. |
| 01:26.24 | mattwj2002 | good point |
| 01:26.39 | mattwj2002 | igcewieling1: if it makes you any better |
| 01:26.46 | mattwj2002 | *feel any better |
| 01:27.14 | mattwj2002 | I had to create a TV troubleshooting guide today |
| 01:27.26 | mattwj2002 | because don't know how to change the source on a TV :P |
| 01:32.10 | *** join/#asterisk vastersk (~vinscentp@124.6.136.142) |
| 01:34.22 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 01:35.40 | mattwj2002 | has anyone tried a digium phone? |
| 01:35.46 | mattwj2002 | are they any good ? :) |
| 01:36.24 | WIMPy | define "good" |
| 01:36.43 | mattwj2002 | well I meant that generally |
| 01:37.04 | mattwj2002 | what do you think of them? |
| 01:37.23 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 01:37.23 | mattwj2002 | they are not cheap :) |
| 01:37.38 | WIMPy | I find the configuration feels very basic. Usage is ok. They allow you to do what you need, but not neccessarily in the obvious way. |
| 01:37.53 | mattwj2002 | okay |
| 01:39.19 | WIMPy | So far they would be my second best choice. (within the SIP category) |
| 01:39.29 | apb1963_ | igcewieling1: I wonder... I'm assuming they're using phones with multiple buttons (lets say 5), where each button represents a "line" - or extension (say x101, x102, x103, x104, x105). Is that correct? Is it possible that each phone can register all 5 extensions? And then so everybody's device/phone rings when a call is for say x103... so they Fred puts the call on hold, and then Joe can go pickup x103 on his device? Does that make any sense and/ |
| 01:39.29 | mattwj2002 | what is your first? |
| 01:39.50 | WIMPy | The old Snoms (320/360/370). |
| 01:39.58 | apb1963_ | s/they Fred/Fred/ |
| 01:40.19 | igcewieling1 | apb1963_: that isn't really how asterisk (or SIP) works |
| 01:40.23 | [TK]D-Fender | apb1963_: No. |
| 01:40.27 | apb1963_ | Is it possible that each phone can register all 5 extensions? And then so everybody's device/phone rings when a call is for say x103... so they Fred puts the call on hold, and then Joe can go pickup x103 on his device? Does that make any sense and/or is do-able? |
| 01:40.49 | [TK]D-Fender | apb1963_: Thechincally "ye", BUT... they will all FIGHT for the registration and the LAST one wins (and will fight in circles). |
| 01:40.56 | igcewieling1 | apb1963_: problem 1: only one device can register to an account. |
| 01:40.57 | apb1963_ | weird how IRC only lets you have a few characters :/ |
| 01:41.00 | [TK]D-Fender | apb1963_: So forget your end result. Only ONE will ring |
| 01:41.23 | apb1963_ | account=extension? |
| 01:41.46 | apb1963_ | oh i see |
| 01:42.20 | [TK]D-Fender | apb1963_: apb1963_ Forget abou "key system" functionality |
| 01:42.31 | apb1963_ | feature request? :) |
| 01:43.02 | apb1963_ | [TK]D-Fender: consider it forgotten... since I have no idea what it is in the first place :) |
| 01:43.07 | WIMPy | apb1963_: Give them an introduction of new technology of the last century. |
| 01:43.30 | igcewieling1 | apb1963_: if it was easy it would have already been done. People have been asking for this for years. Asterisk's SLA features were supposed to provide something comparable, but the code doesn't seem to get a lot of love. |
| 01:43.31 | mattwj2002 | tell them to get rid of those CRTs too |
| 01:43.32 | mattwj2002 | :P |
| 01:43.46 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 01:43.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 01:44.04 | apb1963_ | What's wrong with CRTs? Everybody needs a little radiation. |
| 01:44.10 | mattwj2002 | :P |
| 01:44.24 | apb1963_ | igcewieling1: I never said it was easy :) |
| 01:44.28 | mattwj2002 | they actually give off small amounts of x ray |
| 01:44.34 | mattwj2002 | fyi |
| 01:44.36 | WIMPy | Nothing. So far, decent LCDs are only avalable in laptops, but not for the desktop. |
| 01:44.50 | apb1963_ | xray's don't count as radiation? |
| 01:45.02 | mattwj2002 | I didn't say that |
| 01:45.18 | igcewieling1 | LCDs just don't give that same pale at-death's-door pallor. |
| 01:45.23 | mattwj2002 | I was just saying x ray is one form of radition CRTs give off |
| 01:47.18 | apb1963_ | ok, it was just a thought. I'll go back to banging my head. Since I don't know the code I don't know why asterisk couldn't keep a list of devices registered for a particular extension.... I'm sure there's a good reason. |
| 01:47.48 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 01:47.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:47.50 | WIMPy | It's a 1:1 mapping. |
| 01:48.00 | mattwj2002 | why though? |
| 01:48.04 | apb1963_ | by design |
| 01:48.22 | mattwj2002 | so there are no shared line appearances? |
| 01:48.45 | igcewieling1 | apb1963_: by design done long long ago and considered by many people to be a mistake. |
| 01:48.56 | apb1963_ | ah, ok |
| 01:49.00 | [TK]D-Fender | mattwj2002: Correct |
| 01:49.05 | mattwj2002 | crazy |
| 01:49.18 | igcewieling1 | considered in hindsight, I'm sure everything thought it was a great idea at the time |
| 01:49.20 | apb1963_ | igcewieling1: so asterisk is the equivalent of a CRT? |
| 01:49.25 | *** join/#asterisk jetlag (~jetlag@pool-71-168-200-61.cmdnnj.east.verizon.net) |
| 01:49.37 | [TK]D-Fender | apb1963_: More like an Etch-A-Sketch really ;) |
| 01:49.41 | WIMPy | apb1963_: Maybe |
| 01:49.42 | igcewieling1 | apb1963_: no, key systems are equivalent of a CRT |
| 01:49.47 | apb1963_ | ah |
| 01:50.08 | igcewieling1 | A key system has a one to one mapping of phone lines to phone buttons, this makes no sense in the context of SIP |
| 01:50.21 | WIMPy | key systems = piece of slate |
| 01:50.32 | igcewieling1 | WIMPy: everyone seems to want the functionality |
| 01:50.55 | WIMPy | Everyone in the US maybe. |
| 01:51.02 | mattwj2002 | is there a technical reason? |
| 01:51.07 | apb1963_ | igcewieling1: I see. But... we're talking about not a one to one mapping, but a many to many mapping I think maybe? |
| 01:51.09 | WIMPy | I hadn't even heard of such stuff until I came here. |
| 01:51.11 | igcewieling1 | WIMPy: I stand corrected. |
| 01:51.48 | apb1963_ | WIMPy: We have indoor plumbing and brick houses. What's goin' on over there where you are? |
| 01:52.01 | igcewieling1 | apb1963_: you are sort of correct. one phone line gets patched into a button on multiple phones, but there is still a one to one mapping, when you press a button you ALWAYS get the same line. |
| 01:52.28 | WIMPy | apb1963_: We're in the final phase of switching off the PSTN. |
| 01:52.30 | apb1963_ | igcewieling1: On a per device basis. |
| 01:52.41 | mattwj2002 | where are you WIMPy? |
| 01:52.43 | WIMPy | Probably goin post-civilisation or simething. |
| 01:53.00 | apb1963_ | WIMPy: Nice. We still have a few people using outhouses. |
| 01:53.05 | mattwj2002 | no PSTN? |
| 01:53.08 | mattwj2002 | cries |
| 01:53.08 | WIMPy | Germany |
| 01:53.36 | mattwj2002 | you may take my freedom but you will never take my LANDLINE! :P |
| 01:54.07 | WIMPy | I guess you should avoid Europe then. |
| 01:54.11 | igcewieling1 | ISDN, which doesn't have a one-to-one mapping of number/"line", became popular in many parts of Europe in the 1990s (I think) and so I'm sure they got rid of their key systems then |
| 01:54.14 | apb1963_ | yeah I need my landline... I have no cellphone or mobile as some of you ummm....overseasians call it |
| 01:54.35 | mattwj2002 | WIMPy: I don't even have a landline |
| 01:54.50 | mattwj2002 | but the US still doesn't have full coverage nationwide |
| 01:55.00 | WIMPy | igcewieling1: They have only been popular for extremely small installations anyway. |
| 01:55.03 | igcewieling1 | I dont' have a landline either, but only because DSL is not available where I am |
| 01:55.25 | mattwj2002 | in the US cable is much faster where available |
| 01:55.27 | igcewieling1 | WIMPy: most of our new customers are those sorts of very small installations |
| 01:55.31 | mattwj2002 | I have 20 down and 4 up |
| 01:55.51 | apb1963_ | I have 10/768 |
| 01:56.08 | mattwj2002 | I need multiple megabits up |
| 01:56.15 | igcewieling1 | mattwj2002: I have 18 down / 3 up on cable, but if DSL was available I'd have ADSL as a backup to my cablemodem |
| 01:56.19 | apb1963_ | and I pay an arm & a leg for it |
| 01:56.33 | WIMPy | igcewieling1: What I wanted to say is that that concept hasn;t been popular for a lot longer than the appearance of ISDN with very few exceptions. |
| 01:56.36 | apb1963_ | and I have cable |
| 01:56.41 | mattwj2002 | backup igcewieling1? |
| 01:56.44 | mattwj2002 | for home? |
| 01:57.03 | mattwj2002 | your internet is that critical? |
| 01:57.03 | igcewieling1 | mattwj2002: I work from home and DSL when avaialble can be very cheap |
| 01:57.18 | mattwj2002 | I guess :) |
| 01:57.33 | igcewieling1 | my internet was down for most of the afternoon, had to switch to 4g to do ANYTHING |
| 01:57.48 | *** join/#asterisk deo_ (~deo@222.127.13.226) |
| 01:57.58 | mattwj2002 | we have wimax here too |
| 01:58.50 | mattwj2002 | but they cap it badly |
| 01:58.53 | apb1963_ | I'm considering switching to DSL. I can only get 6Mbps max, but (and I need to doublecheck this), the cost is about half. Considering I used to get by on 1Mbps no problem, I'll prolly be ok. |
| 01:59.09 | apb1963_ | shudders in memory |
| 01:59.32 | igcewieling1 | mattwj2002: 4g here is capped as well |
| 01:59.38 | mattwj2002 | I use to get by on dialup |
| 01:59.54 | mattwj2002 | technically my cable is capped too |
| 01:59.56 | mattwj2002 | 300 GB |
| 02:00.09 | igcewieling1 | when I lived at the campground I had satellite for web browsing and modem for ssh |
| 02:00.18 | apb1963_ | I used to get by on 9600K... but it was torture. |
| 02:00.28 | mattwj2002 | hehe |
| 02:00.38 | mattwj2002 | 9600 baud is better than nothing |
| 02:00.38 | igcewieling1 | I would not wish that setup on my ex-wife and I'd happily poison her. |
| 02:00.39 | mattwj2002 | :) |
| 02:00.45 | apb1963_ | so is 300 baud |
| 02:00.53 | mattwj2002 | true |
| 02:00.58 | apb1963_ | But you don't see me running for an 80's modem |
| 02:01.00 | WIMPy | Those were te days. |
| 02:01.14 | apb1963_ | That's when real men coded in ALGOL |
| 02:01.14 | mattwj2002 | oh come on |
| 02:01.19 | mattwj2002 | don't you love that sound? |
| 02:01.43 | WIMPy | Or the look of the sound? |
| 02:02.00 | WIMPy | Like getting garbage on the screen every time you had to cough? |
| 02:02.15 | apb1963_ | Yes! {{{{ |
| 02:02.23 | mattwj2002 | I had dialup briefly in 2006 |
| 02:02.44 | apb1963_ | {{{(3#{@x |
| 02:03.08 | apb1963_ | <screech> |
| 02:03.18 | mattwj2002 | :P |
| 02:03.47 | apb1963_ | You are in a maze of twistly little passages, all alike. |
| 02:04.16 | igcewieling1 | someone needs to code an IVR of that game, let people play it while on hold. |
| 02:04.32 | mattwj2002 | hehe igcewieling1 |
| 02:04.50 | igcewieling1 | apparently someone did http://www.internetnews.com/dev-news/article.php/3675671/Zork+Returns+Thanks+to+Open+Source+Asterisk+PBX.htm |
| 02:04.53 | mattwj2002 | what was that game called apb1963_? |
| 02:05.13 | apb1963_ | Adventure? |
| 02:05.35 | mattwj2002 | Zork Returns |
| 02:05.37 | mattwj2002 | I guess |
| 02:06.07 | mattwj2002 | heck that is 900 number material there |
| 02:06.08 | mattwj2002 | ;) |
| 02:06.38 | mattwj2002 | oh non profit only :( |
| 02:06.46 | apb1963_ | I think it was Zork when PC's came out |
| 02:07.00 | apb1963_ | but on mainframes I'm relatively sure it was called Adventure. |
| 02:07.27 | mattwj2002 | god CNN |
| 02:07.36 | mattwj2002 | is nothing but sex and guns tonight |
| 02:07.43 | apb1963_ | and then there was Moria. |
| 02:07.44 | lorsungcu | turn it off |
| 02:07.58 | mattwj2002 | I am considering it |
| 02:08.07 | lorsungcu | nothing that cnn will give you that you can't find elsewhere |
| 02:08.11 | lorsungcu | tv news is a joke. |
| 02:08.14 | *** part/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 02:08.20 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 02:08.33 | mattwj2002 | I was hoping they would say something but the March 1st deadline |
| 02:08.35 | apb1963_ | I'm always up for a laugh |
| 02:08.38 | mattwj2002 | not much |
| 02:08.47 | mattwj2002 | I have some new music cds |
| 02:08.57 | mattwj2002 | I think I'll listen to them instead of this trash |
| 02:09.17 | lorsungcu | you watch tv news and listen to physical media? |
| 02:09.40 | lorsungcu | that's pretty badass. |
| 02:09.48 | mattwj2002 | well they are converted to mp3s |
| 02:09.53 | lorsungcu | whre |
| 02:10.02 | mattwj2002 | where? |
| 02:10.04 | lorsungcu | whew. |
| 02:10.05 | lorsungcu | :p |
| 02:10.28 | mattwj2002 | I like a hard copy of my music |
| 02:10.33 | mattwj2002 | and full albums |
| 02:11.33 | mattwj2002 | shuts off CNN |
| 02:11.49 | lorsungcu | i used to |
| 02:11.59 | lorsungcu | would go buy tons of used cds |
| 02:12.04 | lorsungcu | then i just had a lot of cds. |
| 02:12.21 | mattwj2002 | ok? |
| 02:12.25 | mattwj2002 | why did you stop? |
| 02:12.45 | mattwj2002 | that is what I am currently doing |
| 02:12.50 | lorsungcu | album art pretty well quit being good, i was able to download/buy them electronically at the same or less cost |
| 02:13.03 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 02:13.19 | lorsungcu | they take up space, and i don't like the clutter |
| 02:13.36 | lorsungcu | feels way good to give them all to the free store :) |
| 02:13.46 | lorsungcu | and the RIAA can't get you for that one! |
| 02:13.55 | apb1963_ | http://en.wikipedia.org/wiki/Colossal_Cave_Adventure |
| 02:14.01 | mattwj2002 | copying CDs you own is legal |
| 02:14.10 | lorsungcu | but what if you copy it |
| 02:14.17 | lorsungcu | and give the physical copy away! |
| 02:14.22 | mattwj2002 | illegal |
| 02:14.24 | lorsungcu | and SOMEONE ELSE |
| 02:14.28 | lorsungcu | hears those musics. |
| 02:14.54 | lorsungcu | i am a felon :/ |
| 02:15.09 | apb1963_ | I believe that if you give away the CD you purchased that you no longer have the right to keep copies.... but I'm no lawyer. |
| 02:15.20 | mattwj2002 | yeah the RIAA can suck my balls :) |
| 02:15.29 | mattwj2002 | I do it legally |
| 02:15.34 | mattwj2002 | but I think it is a joke |
| 02:15.57 | lorsungcu | i just don't even bother anymore |
| 02:16.01 | mattwj2002 | ops sorry for the language |
| 02:16.06 | lorsungcu | http:///grooveshark.com |
| 02:16.15 | lorsungcu | has pretty much whatever i want |
| 02:16.23 | mattwj2002 | I have a subscription to grooveshark |
| 02:16.25 | mattwj2002 | spotify |
| 02:16.40 | lorsungcu | i used to to grooveshark, don't anymore |
| 02:16.43 | apb1963_ | shake your groove thing |
| 02:16.43 | lorsungcu | no need really.. |
| 02:16.54 | lorsungcu | exactly, apb1963_ |
| 02:16.59 | apb1963_ | I use pandora |
| 02:17.07 | mattwj2002 | oh yeah and pandora |
| 02:17.11 | lorsungcu | that is nice if you don't know what you want to hear |
| 02:17.14 | apb1963_ | and to a lesser extent, that other service... ummm... jango? |
| 02:17.23 | mattwj2002 | jango? |
| 02:17.27 | mattwj2002 | never heard of it |
| 02:17.32 | apb1963_ | i think that's what it's called |
| 02:17.35 | apb1963_ | been awhile |
| 02:18.07 | apb1963_ | and then there was Launch Music... which Yahoo then bought and incorporated. |
| 02:18.09 | mattwj2002 | nice |
| 02:18.28 | apb1963_ | I worked there briefly. |
| 02:19.02 | mattwj2002 | what ever happened to broadcast.com? |
| 02:19.16 | apb1963_ | There was yet another service that I don't remember the name... I also worked there briefly... I think it was started by Carol King or someone like that? |
| 02:19.25 | mattwj2002 | great it goes directly to yahoo.com |
| 02:19.58 | apb1963_ | yahoo is a monster |
| 02:20.14 | mattwj2002 | http://en.wikipedia.org/wiki/Broadcast.com |
| 02:20.18 | apb1963_ | they bought AT&T |
| 02:20.37 | mattwj2002 | broadcast.com I think use to have streaming music videos |
| 02:20.49 | mattwj2002 | like a mtv thing |
| 02:21.54 | mattwj2002 | not a youtube thing |
| 02:23.17 | apb1963_ | grooveshark looks pretty cool.. but I don't know the name of the song or artist I want to listen to :) |
| 02:23.39 | apb1963_ | unless it's Led Zeppelin |
| 02:24.47 | mattwj2002 | abba |
| 02:25.15 | apb1963_ | odd how they catagorize Def Leppard with Elton John |
| 02:26.32 | mattwj2002 | same era |
| 02:26.56 | mattwj2002 | then again |
| 02:27.03 | apb1963_ | oh I think I may like grooveshark |
| 02:27.16 | mattwj2002 | with that logical metalica could be categorized with Hanson |
| 02:27.24 | mattwj2002 | *logic |
| 02:27.27 | apb1963_ | we'll have to see what limitations they impose after 10 minutes or so. lol |
| 02:28.05 | mattwj2002 | apb1963_: on the pc I don't think there is |
| 02:28.25 | mattwj2002 | no ads last time I checked |
| 02:28.43 | mattwj2002 | too |
| 02:28.45 | apb1963_ | :-o |
| 02:28.55 | mattwj2002 | in the music I mean |
| 02:28.59 | mattwj2002 | I think there are some banners |
| 02:29.26 | apb1963_ | I would be amazed and a convert if so... unless I have to pick EVERY song |
| 02:29.47 | apb1963_ | that would get old |
| 02:29.50 | mattwj2002 | they have radio |
| 02:30.15 | apb1963_ | real radio? or just automated music picking - aka dj style? |
| 02:30.30 | mattwj2002 | radio you can skip |
| 02:30.34 | mattwj2002 | automated |
| 02:30.39 | apb1963_ | cool |
| 02:30.49 | mattwj2002 | http://grooveshark.com/#!/genres |
| 02:30.55 | apb1963_ | so far, I'm a convert |
| 02:31.16 | apb1963_ | yeah i'm the genres section... picked a song to listen to |
| 02:33.39 | apb1963_ | oh i'm liking this |
| 02:34.22 | mattwj2002 | :D |
| 02:34.38 | apb1963_ | now... not that i'm ready to do it... but how do I stream groovething into asterisk MOH? |
| 02:34.45 | apb1963_ | :) |
| 02:34.53 | mattwj2002 | that would be illegal |
| 02:34.56 | mattwj2002 | sends the RIAA! |
| 02:34.58 | mattwj2002 | :P |
| 02:35.02 | apb1963_ | oh |
| 02:35.13 | apb1963_ | well that's dumb |
| 02:35.24 | apb1963_ | I can listen but my "friends" can't? |
| 02:35.49 | mattwj2002 | someone in the room will have to explain the whole royality thing again |
| 02:35.53 | mattwj2002 | I forget how it works |
| 02:36.18 | mattwj2002 | but the RIAA is responsible for elevator on hold music |
| 02:36.30 | apb1963_ | hmmm... well lets see...I'm paying zero royalties... how about if I charge my friends double and send it in? Will that square things? |
| 02:36.41 | mattwj2002 | :P |
| 02:39.05 | igcewieling1 | apb1963_: I believe t is not illegal if you have groovesomething's permission. |
| 02:40.13 | mattwj2002 | apb1963_: pandora |
| 02:40.19 | mattwj2002 | has a business music service |
| 02:40.30 | mattwj2002 | I don't know if you can use it for music on hold and what not |
| 02:40.37 | mattwj2002 | another idea |
| 02:40.53 | mattwj2002 | assuming you live in the US |
| 02:41.03 | mattwj2002 | http://www.pandora.com/everywhere/business |
| 02:44.29 | mattwj2002 | according to this http://www.voneto.com/blogger/2012/10/30/music-on-hold-wait-it-costs-money/ you can use pandora |
| 02:44.38 | mattwj2002 | but I don't know if that is true |
| 02:44.58 | mattwj2002 | it has to be the pandora for business system though |
| 02:45.56 | mattwj2002 | http://www.timesdispatch.com/business/leading-edge-law-don-t-open-pandora-at-your-business/article_16834ecf-28ec-59e9-b0f5-4e59e8644ef3.html |
| 02:57.41 | apb1963_ | meh. Not that important. Just a passing "that would be cool" thought. |
| 02:58.34 | mattwj2002 | okay cool |
| 02:59.06 | apb1963_ | I mean maybe if I had a real business with real income it might be worth looking into, but with 2 calls a month.... nah |
| 03:01.15 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 03:01.23 | *** join/#asterisk Velluto (~stephen@207.210.6.97) |
| 03:02.53 | Velluto | Hey - I was wondering if anyone could help me quickly? I have an issue with registration packets not making it to Asterisk, but they are being picked up by tcpdump on the same server |
| 03:06.13 | apb1963_ | what makes you think they're not making it to asterisk? |
| 03:06.55 | Velluto | I have sip debugging watching the ip address, and there is nothing appearing for them |
| 03:07.17 | Velluto | There are 3 phones (NAT), which work on another server with the exact same config |
| 03:07.31 | igcewieling1 | Velluto: have you enabled sip debug in the cli? |
| 03:07.51 | apb1963_ | have you confirmed basic network connectivity? ping? |
| 03:08.13 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 03:08.33 | Velluto | igcewieling1: yeah, I'm monitoring the one of the ip addresses, and it shows up in tcpdump on port 5060 but never makes it to asterisk |
| 03:08.58 | igcewieling1 | Velluto: with sip debug enabled nothing shows up in the CLI? |
| 03:09.08 | Velluto | apb1963: yes, local phones are registering correctly, it's just the packets coming from nat that make it to the server but not asterisk |
| 03:09.19 | igcewieling1 | Velluto: ah. |
| 03:09.57 | igcewieling1 | you'll see the packets have an incorrect destination address. you have a nat issue. do you have localnet= and externip= set? |
| 03:10.06 | Velluto | igcewieling1: if I have sip debug on (without filter), every sip packet is shown, if I filter on the specific IP address, nothing appears (i get the IP address from tcpdump) |
| 03:10.24 | Velluto | localnet and externip are both set |
| 03:10.31 | Velluto | btw - I'm using asterisk 10.7.1 |
| 03:10.32 | igcewieling1 | what is the localnet set to? |
| 03:11.10 | Velluto | 192.168.2.0/255.255.255.0 |
| 03:11.48 | igcewieling1 | make sure to disable SPI and SIP ALG options on your router. |
| 03:12.49 | Velluto | if I point the router to our old asterisk server (also 10.7.1) these three phones register correctly - and the configs are the same (this is a failover server) |
| 03:13.06 | apb1963_ | if you grep for the IP address in the log with filtering off (so that every sip packet is shown).... you don't see the address show up in the log? |
| 03:13.35 | Velluto | let me do that - give me a second |
| 03:19.49 | Velluto | nope - the IP address does not show up in the logs with the filter off |
| 03:20.01 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 03:22.37 | igcewieling1 | what about the calling or called TN? |
| 03:22.43 | igcewieling1 | (showing up in the logs) |
| 03:23.02 | Velluto | what do you me calling? like Invite packets? |
| 03:23.08 | igcewieling1 | yes |
| 03:23.45 | Velluto | I tracked those as well, the packet gets forwarded correctly from the router, shows up in the tcp dump, but never makes it to asterisk |
| 03:27.21 | *** part/#asterisk poseidon (~joe@vps6967.inmotionhosting.com) |
| 03:27.24 | apb1963_ | so tcpdump is running on the same machine as asterisk right? |
| 03:27.42 | Velluto | yes, they are both running on the same machine |
| 03:27.48 | apb1963_ | and asterisk works for local phones right? |
| 03:27.55 | Velluto | tcp dump is watching port 5060 |
| 03:28.28 | apb1963_ | firewall? |
| 03:28.37 | Velluto | iptables and selinux are both disabled |
| 03:29.26 | apb1963_ | and you're definitely getting packets from local phones? |
| 03:29.53 | Velluto | yes - all phone internally are able to call eachother, and make it out the PRI |
| 03:30.05 | apb1963_ | pastebin the output of a registration that's working |
| 03:30.22 | Velluto | i just noticed tcpdump is pointing to the old hostname, when the failover kicked in it took the IP address from the old server, but it has a different hostname. Could that cause NAT devices not to function? |
| 03:30.59 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 03:30.59 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:31.19 | apb1963_ | ummm.... hostnames are associated with IP addresses... IP addressess are associated with machines. |
| 03:31.28 | apb1963_ | or NICs more accurately |
| 03:32.22 | Velluto | ok - let me see if I can get a register for you |
| 03:33.49 | apb1963_ | so... you're saying your new malfunctioning asterisk server (lets call it Fred) is using the IP address of the old functioning server (lets call it Wilma) ?? |
| 03:34.18 | Velluto | That is correct - because Wilma is no longer functioning |
| 03:34.49 | apb1963_ | So Fred is using Wilma's IP address because Wilma died. |
| 03:34.55 | Velluto | Correct |
| 03:35.44 | apb1963_ | is your router by any chance caching the MAC address? |
| 03:36.10 | Velluto | hmm... let me look - we're using a SonicWall if that helps |
| 03:37.00 | apb1963_ | nah, that makes no sense... you said the packets are showing up in tcpdump |
| 03:37.55 | Velluto | yeah - the router has the correct mac |
| 03:38.05 | apb1963_ | have you tried rebooting asterisk? Not that it would ever need it, but just for giggles. |
| 03:38.42 | Velluto | I have restarted Asterisk many (many) times, and restarted the physical server many times |
| 03:38.46 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 03:39.54 | apb1963_ | and no packets whatsoever (from the 3 magic phones) show up in asterisk logs? |
| 03:40.58 | Velluto | absolutely nothing - it's boggling my mind... |
| 03:41.08 | apb1963_ | verbosity is set to.... ? |
| 03:41.41 | ChannelZ | one million! |
| 03:41.53 | Velluto | verbosity set to 5 currently |
| 03:41.55 | apb1963_ | not enough |
| 03:42.02 | apb1963_ | one million I mean :) |
| 03:42.03 | ChannelZ | not enough for what? |
| 03:42.22 | apb1963_ | for me? |
| 03:42.25 | Velluto | here's the pastebin of my phones registration (internal) http://pastebin.centos.org/1429/ |
| 03:43.24 | ChannelZ | I thought they weren't registering? |
| 03:43.31 | ChannelZ | guesses he should read back further |
| 03:43.49 | Velluto | only NAT devices outside the firewall are not registering |
| 03:44.09 | Velluto | but the registration packets are making it to the tcpdump on the server |
| 03:45.05 | ChannelZ | netstat -alpn |grep 5060 |
| 03:45.14 | ChannelZ | is it listening/on the right interface? |
| 03:45.21 | apb1963_ | pastebin the relevant tcpdump |
| 03:45.55 | apb1963_ | ChannelZ it has to be... he's getting local registration |
| 03:46.39 | ChannelZ | unless it's only listening on a LAN IP and the forwarding is busted |
| 03:46.42 | apb1963_ | I'm wondering if his phones are pointed at the right port |
| 03:46.46 | Velluto | yes - it has the correct IP address and port |
| 03:46.52 | ChannelZ | but generally yes |
| 03:46.56 | apb1963_ | pb it |
| 03:47.15 | apb1963_ | he's got iptables and selinux turned off |
| 03:47.42 | ChannelZ | You said earlier you had sip debug turned on "for the IP" -- are you SURE you've not got the wrong thing? Just turn SIP debug on period |
| 03:47.48 | apb1963_ | more importantly... he's seeing the packets in tcpdum |
| 03:48.11 | apb1963_ | ChannelZ: He did that and then grepped in the logs for it. |
| 03:48.46 | apb1963_ | that's why I want to see the tcpdump |
| 03:49.01 | apb1963_ | confirm he's getting what he claims he's getting |
| 03:49.15 | ChannelZ | sounds like something else has taken a tcpdump |
| 03:49.20 | apb1963_ | lol |
| 03:49.42 | apb1963_ | you mean someone dropped the kids off at the pool? |
| 03:51.31 | Velluto | pastebin 1432 with the previous url |
| 03:51.35 | apb1963_ | assuming they're really showing up in tcpdump, then somehow packets are not making it from the interface to asterisk... or asterisk isn't talking about it. |
| 03:52.20 | Velluto | and that's what's confusing me |
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| 03:53.49 | Velluto | i just had a thought - it could be DNS |
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| 03:57.22 | apb1963_ | da heck is dat? That's not a packet. |
| 03:57.32 | apb1963_ | use -v |
| 03:57.41 | Velluto | I just went to try and ping google.com and it couldn't make it |
| 03:58.50 | apb1963_ | use traceroute |
| 03:59.09 | apb1963_ | use dnslookup |
| 04:00.45 | Velluto | ok - it's not DNS, it looks like the default gw is missing |
| 04:01.50 | Velluto | And that was it!! |
| 04:02.23 | Velluto | I'd like to thank all of you for your help! |
| 04:03.15 | apb1963_ | still weird. Not sure why a missing gateway would prevent it from reaching asterisk if in fact the packet was reaching the server in the first place. |
| 04:03.52 | apb1963_ | gateway is outgoing |
| 04:04.06 | Velluto | I'm thinking maybe Asterisk had trouble getting to the IP address based on the host name from the NAT'd device and crapped out without notice |
| 04:04.07 | apb1963_ | I presume |
| 04:04.25 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.124) |
| 04:04.52 | apb1963_ | so asterisk couldn't reach localhost? |
| 04:05.14 | apb1963_ | or rather it couldn't reach the external IP for that server |
| 04:05.14 | Velluto | no Asterisk couldn't reach the hostname of the NAT'd device to get its IP Address |
| 04:06.07 | Velluto | it was fine getting to itself, just couldn't get to google.com, which led me to believe it couldn't get to the shaw url in the tcpdump to translate that into an IP Address to register the device |
| 04:06.11 | apb1963_ | that would explain why the device wasn't getting a response... not why asterisk didn't receive the packet in the first place. |
| 04:06.35 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.173.50) |
| 04:06.39 | Velluto | maybe Asterisk did recieve the packet, but couldn't translate its host name, and there are no errors for that |
| 04:07.20 | Velluto | SIP relies on the IP address to register (I'm pretty sure), and without that IP address - it looks like it bailed |
| 04:07.28 | apb1963_ | needed a higher verbosity I think |
| 04:07.48 | Velluto | could be |
| 04:07.56 | apb1963_ | I knew a million wasn't enough :) |
| 04:08.22 | apb1963_ | so let that be a lesson to you young paduan |
| 04:08.39 | Velluto | well Thanks again - I'm documenting this for the future! |
| 04:08.45 | apb1963_ | I really need to get out more |
| 04:08.50 | apb1963_ | sure |
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| 06:10.07 | fulcan | I am using asterisk 1.4 from the vicidial iso and it is missing all of the sip commands. are they hidden or do I have to install/upgrade to a regular asterisk distro? |
| 06:10.35 | fulcan | No such command 'sip show peers' (type 'help sip show' for other possible commands) |
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| 06:41.40 | [TK]D-Fender | fulcan: "module load chan_sip.so |
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| 07:07.17 | resist0r | fulcan: I think 1.6.x and above started to implement the specification of sip in many of the commands you are likely used to now. You might try "show peers" (leaving out the sip) |
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| 07:08.22 | [TK]D-Fender | That one did not change. |
| 07:08.29 | [TK]D-Fender | chan_sip has not loaded for some reason |
| 07:10.17 | resist0r | waits to thanks [TK] |
| 07:10.22 | resist0r | thank* |
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| 08:29.47 | x1user | Anyone experienced with SMS? |
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| 08:30.29 | ChannelZ | HOW R U 2 DAY? |
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| 10:12.39 | bitwize | I have a problem with my Asterisk 1.8.4.4, under high load asterisk crashed and wont start again. Even if I restart the whole server asterisk fail to start. |
| 10:12.41 | bitwize | When i start asterisk with "asterisk -cvvvvv" the last message before returning to shell is "ERROR[2852]: cdr_mysql.c:566 my_load_module: Unable to query table description!! Logging disabled." |
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| 10:13.13 | Greenlight | Sounds like the mysql cdr backend is the problem |
| 10:13.53 | Greenlight | Do you *need* the CDRs ? |
| 10:13.54 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 10:15.17 | bitwize | actually I don't need the cdr's, I'll try to disable the cdr.. bbs |
| 10:15.37 | Greenlight | Yup - uneccissary overhead if you don't need them |
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| 10:17.40 | bitwize | Perfect, that solved my problem. Thanks Greenlight! |
| 10:18.18 | Greenlight | Glad to help |
| 10:18.28 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 10:18.41 | hrolf | Hi, is there any way to detect answer on analog lines? |
| 10:19.00 | hrolf | I enabled answerpolarity, but now I don't receive the answer event. |
| 10:19.09 | hrolf | What other options do we have? |
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| 10:22.36 | kchehab | i have a problem with srtp both have m=16 SAVP but 488 not acceptable here ,i guess its a nedia issue |
| 10:22.43 | kchehab | media |
| 10:23.20 | kchehab | is there any way to fix it as i try all codecs G711a/U g729 licenced abd crack all have the same scenario |
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| 10:42.05 | kchehab | any hint |
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| 10:49.01 | din3sh | hello all |
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| 11:28.22 | hrolf | Is there any way I can reject (not accept) calls which has the same CLI as any active call in the system? |
| 11:29.04 | hrolf | For instance, I would like to receive only one call from the same PRI (i.e. both have same CLI.) |
| 11:30.41 | Greenlight | hrolf: If I understand what you're trying to do, you could perhaps use GROUP_COUNT with the CallerID/CLI |
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| 11:42.56 | hrolf | Greenlight: How can it be done through GROUP_COUNT? |
| 11:45.15 | kchehab | guys |
| 11:45.17 | Greenlight | You're trying to prevent two incomming calls from the same number ? |
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| 11:45.57 | kchehab | i have a problem with srtp both have m=16 SAVP but 488 not acceptable here ,i guess its a media issue ,any one face such issue |
| 11:46.47 | Ice_Strike | Hi Greenlight |
| 11:46.54 | hrolf | Greenlight: Yes. |
| 11:46.56 | Greenlight | Morning Ice_Strike |
| 11:47.26 | Greenlight | hrolf: Then, looking at GROUP_COUNT, which part are you unsure of ? |
| 11:47.39 | hrolf | Greenlight: What is GROUP_COUNT? Is it a func? |
| 11:47.50 | Greenlight | hrolf: Yes |
| 11:47.56 | Greenlight | hrolf: Google is your friend :) |
| 11:48.31 | hrolf | Greenlight: Indeed! thanks I'll look into it. |
| 11:55.00 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 11:55.18 | nunne | anyone with experience with digium phones (d50/d70) on how to configure tone settings? |
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| 11:57.17 | ectospasm | nunne: you mean ringtone settings? |
| 11:57.34 | ectospasm | nunne: are you using DPMA? |
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| 11:59.17 | nunne | ectospasm: yeah, ringtone. would like european dialtones. this particular phone is connected to an old embedded asterisk 1.4. so not using DPMA |
| 11:59.55 | ectospasm | ringtones are not the same as European dial tones... |
| 12:01.00 | ectospasm | nunne: you can configure ringtones using the XML provisioning documentation: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=21463877 |
| 12:01.01 | nunne | ahh, sorry. read everything wrong. am a bit stressed :D i mean tone setting as in dial tone, busy tone, reorder tone etc.. |
| 12:01.35 | ectospasm | do you have any PSTN adapters? |
| 12:02.04 | ectospasm | normally you'd set the zone info with loadzone and defaultzone in /etc/dahdi/system.conf |
| 12:02.18 | ectospasm | but if you don't have any PSTN adapters, that won't be relevant |
| 12:03.35 | nunne | ectospasm: well, the sip phones generate their own tones. and i can't really find any tone settings in the wiki |
| 12:04.04 | nunne | ectospasm: plus we don't use ISDN nor analog. only SIP. |
| 12:04.11 | ectospasm | normally we don't set dial tones for the phones specifically. |
| 12:04.35 | eirirs | would be cool if we could make dog barking as dialtone on our sip phones |
| 12:06.33 | Greenlight | tt-monkeys ... |
| 12:06.40 | Greenlight | Bet then phones would get picked up quickly :) |
| 12:06.43 | *** join/#asterisk mattmonkeymagic (~matt_palm@213.106.234.250) |
| 12:06.46 | eirirs | haha |
| 12:06.52 | nunne | ectospasm: on all other sip phones you change the dial tone patterns on the phones or in provisioning. would be madness if the digium phones are locked to only use US dial tone patterns. |
| 12:06.58 | mattmonkeymagic | Hi everyone, anyone here that could help with a question about asterisk hanging on a reload?? |
| 12:07.18 | Greenlight | mattmonkeymagic: Ask the question and find out |
| 12:07.36 | ectospasm | nunne: like I said, that's usually set in Asterisk, through the proper tonezone settings. |
| 12:07.49 | kchehab | i have a problem with srtp both have UA's debug show on DSP m=16 SAVP but 488 not acceptable ,as UA1 cant call UA2 but UA2 can call UA1 ,i guess its a media issue ,any one face such issue ,as i try all codecs and all have tyhe same output |
| 12:08.51 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 12:08.51 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:09.13 | mattmonkeymagic | Hi Greenlight, I've got an issue with Asterisk where it appears to hang after clicking 'apply changes here'. Result is complete inability for any new calls in or out to be made until an asterisk stop start is issued |
| 12:09.57 | Greenlight | mattmonkeymagic: That sounds more like FreePBX perhaps, than Asterisk. Asterisk doesn't have an "Apply Changes" button. Is that what you're using ? |
| 12:10.36 | nunne | ectospasm: I'm not sure I follow. for ISDN/Analog I would assume so. (Plus i have loadzone=se set). But SIP phones always generate their own tone in the handset.. they never get dial tone from the PBX. |
| 12:10.56 | mattmonkeymagic | Hi Greenlight, yep sorry its Elastix, with asterisk 1.8.12.0 and FreePBX |
| 12:11.08 | *** join/#asterisk bitwize (~bitwize@h87-96-213-2.dynamic.se.alltele.net) |
| 12:11.25 | Greenlight | mattmonkeymagic: Thought it might be; although FreePBX isn't really supported here, that is a known issue. |
| 12:11.30 | Greenlight | ~freepbx |
| 12:11.30 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 12:11.50 | mattmonkeymagic | Ok, thanks guys! |
| 12:11.50 | Greenlight | mattmonkeymagic: From what I can tell turing off verbosity helps |
| 12:12.21 | mattmonkeymagic | yep, was also going to try and extend the max_execution time in php.ini for, see if that helps any |
| 12:12.23 | Greenlight | mattmonkeymagic: But really, on FreePBX installations with lots of extensions, you're best to do reloads out of hours |
| 12:13.09 | Greenlight | mattmonkeymagic: I think it's a problem on the Asterisk side, rather than the FreePBX web site. As I say, turn off verbosity "core set verbose 0", it apparently helps |
| 12:13.35 | mattmonkeymagic | I'll give that a go thanks Greenlight |
| 12:13.36 | ectospasm | nunne: if you've registered your Digium phones, I recommend opening a support case with Digium, so we can file a bug report/feature request on your behalf. |
| 12:13.56 | Greenlight | mattmonkeymagic: We found on those boxes where we had FreePBX running, once they got to that size, an "Apply Changes" had like a 30% failure rate - so unless it's a bloody urgent change, just wait till out of hours :) |
| 12:14.07 | ectospasm | ...since I don't see any ability to set the zone in the XML configuration. |
| 12:14.55 | mattmonkeymagic | ha Greenlight, unfortunately here, all changes are 'urgent' ;-) |
| 12:15.22 | ectospasm | mattmonkeymagic: your organization needs proper change management, then... |
| 12:15.53 | mattmonkeymagic | ectospasm:quite :-) |
| 12:16.29 | Greenlight | mattmonkeymagic: You can also try waiting for "quiet" times; lunch breaks etc |
| 12:16.55 | Greenlight | mattmonkeymagic: It's some kind of deadlock when under moderate to heavy load, with all that FreePBX crap |
| 12:22.24 | *** join/#asterisk davlefouAMD (~david@197.15.50.19) |
| 14:55.20 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 14:55.20 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.2.1 (2013/01/22), 10.12.1 (2013/01/22), 1.8.20.1 (2013/01/22), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 14:56.05 | igcewieling1 | leifmadsen: do you know what the max length of dial parameters is? |
| 14:56.45 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 14:57.05 | leifmadsen | igcewieling1: I documented it once |
| 14:57.07 | *** join/#asterisk emora (~emora@213.236.11.83) |
| 14:57.29 | leifmadsen | it depends on the version of asterisk, and it's more the length of the line in extensions.conf |
| 15:00.08 | danfromuk | leifmadsen: i'm using realtime and found that the app data is about 248 characters in 1.8 |
| 15:01.27 | leifmadsen | makes sense.... 248 + Dial( ) = 254 chars, plus probably a terminating char at 255 |
| 15:01.35 | leifmadsen | so somewhere around 256 chars |
| 15:02.00 | leifmadsen | I updating that in the latest Asterisk book somewhere, but cant' find it right now, and OFPS sucks for searching |
| 15:02.47 | phix | http://www.youtube.com/watch?v=TWfph3iNC-k |
| 15:03.09 | leifmadsen | file: ping! |
| 15:03.16 | file | what? |
| 15:03.19 | leifmadsen | <3 |
| 15:03.24 | leifmadsen | also I have a question |
| 15:03.40 | leifmadsen | Transcoding audio in Asterisk, possible when a video stream is involved? |
| 15:03.45 | file | yes |
| 15:03.57 | leifmadsen | interesting... always worked, or only worked as of version X? |
| 15:04.14 | file | should have always worked, but as video support is sort of ... hacked in there ... it could get confused |
| 15:04.42 | leifmadsen | ya, Asterisk 1.4 and 1.8... seem to have an issue transcoding ulaw->g729 when video is involved |
| 15:04.54 | leifmadsen | just wanted to see if it *should* work before digging in |
| 15:05.06 | file | it should. |
| 15:05.12 | leifmadsen | ok thanks! |
| 15:05.31 | file | the most likely problem to crop up is that the code doesn't remove the video part when trying to figure out translations |
| 15:05.42 | leifmadsen | gotcha |
| 15:06.02 | leifmadsen | I'll start the research process knowing there isn't something known that would break it absolute |
| 15:06.10 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 15:07.10 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
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| 15:12.05 | SuperNull | hey guys what do you use for call quality control ? |
| 15:13.12 | leifmadsen | I don't |
| 15:13.27 | leifmadsen | I just make sure I don't over provision my circuit |
| 15:13.44 | phix | gg leifmadsen |
| 15:14.15 | SuperNull | not a matter of that.. |
| 15:14.20 | SuperNull | were losing packets some where .. |
| 15:14.22 | SuperNull | i believe.; |
| 15:14.29 | SuperNull | not over provisioning.. |
| 15:14.45 | leifmadsen | losing packets is typically not something you can fix on your side, it's a network issue |
| 15:14.58 | SuperNull | derp. |
| 15:14.59 | leifmadsen | if they are just arriving out of order, you could use a jitterbuffer |
| 15:15.03 | SuperNull | <-- network engineer. |
| 15:15.14 | leifmadsen | do you control the network from one point to the other? |
| 15:15.17 | SuperNull | yep. |
| 15:15.24 | leifmadsen | you're not going over the internet then? |
| 15:15.33 | leifmadsen | I also don't run my voice network over the data network |
| 15:15.58 | SuperNull | technically yes, but its qosed over the small portion that isnt. We are in control of all legs regarding calls. or at least in such a way that we can ensure its good till PSTN |
| 15:16.13 | SuperNull | let me rephrase. |
| 15:16.29 | SuperNull | literally 10 feet off our 'internet' connection is the PSTN Nortel DMS. |
| 15:16.43 | SuperNull | and our 10gigabit link is not even 50% maxed. |
| 15:17.24 | SuperNull | i was looking at voipmonitor .. def takes a bit to set it up tho |
| 15:21.57 | *** part/#asterisk maetrik (maetrik@185.14.184.81) |
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| 15:38.50 | kalib | Hello guys. How can I check if a sip peer is in a call on my CLI? |
| 15:39.07 | kchehab | i have a problem with srtp both have m=16 SAVP at sdp message but 488 not acceptable here ,i guess its a media issue ,any one face such issue |
| 15:39.08 | [TK]D-Fender | kalib, "core show channels" |
| 15:39.31 | kalib | thanks.. |
| 15:39.36 | [TK]D-Fender | kchehab, Show it, don't "describe" it. |
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| 15:41.37 | kchehab | [TK]D-Fender my debug http://pastebin.com/i1qKFDxM |
| 15:42.17 | [TK]D-Fender | kchehab, And the full configs from both sides |
| 15:44.25 | SuperNull | TK you do any kind of quality monitoring ? |
| 15:45.29 | [TK]D-Fender | SuperNull, I buy only the best monitors.... preferrably LCDs.... |
| 15:46.17 | kchehab | [TK]D-Fender my config is here for clients and sip.conf http://pastebin.com/yLSEULpg |
| 15:46.52 | Greenlight | Is there a way to "clear" or "reset" the Asterisk internal database (the one that persists queues members etc between restarts) ? |
| 15:47.00 | Qwell | delete it |
| 15:47.07 | Greenlight | And it'll just be recreated ? |
| 15:47.11 | Greenlight | But empty |
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| 15:51.46 | malcolmd | correct |
| 15:52.20 | kchehab | [TK]D-Fender that your needs ? |
| 15:52.29 | SuperNull | TK you know what i ment bud ;) |
| 15:53.07 | Greenlight | Qwell, malcolmd: Thanks, that's nice and easy then. Where abouts is the actual data file stored ? |
| 15:53.24 | Qwell | /var/lib/asterisk/ |
| 15:53.58 | Greenlight | Ahh there we go - thanks again |
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| 15:56.09 | [TK]D-Fender | kchehab, Your debug doesn't match your configs. |
| 15:56.57 | [TK]D-Fender | kchehab, Your peers make no mention of codecs but it's clearly restricting to ALAW and that isn't the rules for [general]. I do not trust what you've shown me. |
| 15:57.28 | kchehab | [TK]D-Fender yes it on ot the captures |
| 15:57.50 | kchehab | [TK]D-Fender in all cases of codecs i have such problem, codec is not the reason |
| 15:58.36 | kchehab | [TK]D-Fender this capture did were i restrict all devices to have G711a unique |
| 15:58.50 | [TK]D-Fender | kchehab, I don't care that you think it isn't a problem. It highlights that the pieces don't match and I don't TRUST any of what you've provided |
| 16:04.11 | kchehab | [TK]D-Fender its the same config unless i disallow all and set g711a unique,trust me the codecs were commented but due copy paste the ; didnt appear |
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| 16:12.30 | igcewieling1 | kchehab: hopefully you set alaw in the asterisk config and not g711a as that is not a valid codec name for Asterisk |
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| 16:43.56 | Greenlight | If I use "autocreatepeer=yes", do I then specify a password in [global] ? |
| 16:44.55 | kchehab | [TK]D-Fender i re did the test again and there is my config + asterisk full debug ,kindly check hen specify a password in [global] ? |
| 16:45.12 | kchehab | [TK]D-Fender kindly check at http://pastebin.com/7f3NWdHB |
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| 17:02.09 | igcewieling1 | kchehab: do you have a g729 codec license? if not, don't allow= it |
| 17:02.55 | igcewieling1 | this may not be valid: externip=74.114.101.165 localnet=74.114.101.160/255.255.255.224 Is your server behind NAT? |
| 17:03.05 | [TK]D-Fender | igcewieling1, He's running an unlicensed one for which leifmadsen dropped out immediately upon hearing yesterday |
| 17:03.25 | igcewieling1 | oh. And I shall do the same. sorry, kchehab but I don't assist with unlicensed codecs. |
| 17:04.00 | pabelanger | spend the $10 man |
| 17:04.20 | Greenlight | Ahh yea he's the guy that came back in with "yea but I have 100 real licences" hmmmm |
| 17:05.03 | kchehab | igcewieling1 i have it and allow as UA2 can call UA1 but UA1 cant call UA2 |
| 17:05.19 | igcewieling1 | kchehab: I said I cannot help you further. |
| 17:05.20 | kchehab | igcewieling1 not its real IP |
| 17:05.38 | kchehab | file hi ,...can you please assist me |
| 17:06.06 | igcewieling1 | kchehab: file works for Digium, he would be fired if he helped you. |
| 17:06.25 | pabelanger | doubt fired |
| 17:06.28 | pabelanger | maybe flogged |
| 17:06.59 | igcewieling1 | pabelanger: risking the rights holder revoking Digium's g729 license? I'd fire him. |
| 17:07.28 | igcewieling1 | thought flogging does sound more fun. |
| 17:08.08 | kchehab | igcewieling1 oops i didnt know that |
| 17:08.25 | kchehab | igcewieling1 whom can i ask ,... |
| 17:08.41 | Qwell | Nobody. |
| 17:08.49 | Qwell | You aren't getting help with that here. Period. |
| 17:09.05 | Qwell | Anybody that does help with that, will be swiftly removed from the channel. |
| 17:09.06 | igcewieling1 | kchehab: I doubt anyone will help anyone who has an unlicensed codec. If they work for Digium they risk the entire company and anyone else doesn't want to anger Digium. |
| 17:09.46 | Qwell | If you continue to try, you will also be removed. |
| 17:11.48 | WIMPy | Sometimes I wonder why we support Asterisk and don't demand people use Switchvox. |
| 17:13.06 | dr0ck | why is that? |
| 17:13.57 | WIMPy | To prefer the commecrial solution? |
| 17:14.02 | Qwell | The only legal solution* |
| 17:14.17 | WIMPy | Where? |
| 17:14.28 | Qwell | Where I am. Where FreeNode is hosted. |
| 17:14.39 | Qwell | Where Digium is a legal entity. |
| 17:14.47 | kchehab | file what the benfits i could have if i buy asterisk licenced commercial |
| 17:14.53 | WIMPy | Freenode is hosted about everywhere. |
| 17:14.58 | kchehab | is it the same ,or more stable |
| 17:15.07 | leifmadsen | it's legal |
| 17:15.10 | Qwell | kchehab: Nobody is saying to buy a commercial version of Asterisk. They are saying that in order to use G.729, you need to have valid licenses for it. |
| 17:15.10 | leifmadsen | that's a benefit |
| 17:15.28 | kchehab | Qwell i have ,113 licensce |
| 17:15.46 | WIMPy | Qwell: That's the same. Just because YOU may need a license, doesn;t mean everybody else does. |
| 17:16.00 | Qwell | WIMPy: We cannot help him here. That's it. End of story. |
| 17:16.01 | dpilon | legaly yes they do |
| 17:16.07 | Qwell | I don't care what the laws in his country are. |
| 17:16.31 | WIMPy | And It's only ever about the digium solg G.729 licenses whereas other required liceses are actively ignored. |
| 17:16.48 | Qwell | There are no other legally licensed modules. |
| 17:16.50 | mjordan | huh? |
| 17:17.12 | WIMPy | Qwell: Yu don;t have to support the freee versions. But others might not have issues with that and probably shouldn;t. |
| 17:18.16 | dpilon | this just wasted 5 minutes of everyone's life |
| 17:18.21 | Qwell | Again. |
| 17:19.10 | WIMPy | Same thing each time. |
| 17:19.26 | drmessano | The distro I use is better than yours |
| 17:19.33 | dpilon | hahaha |
| 17:21.46 | Faustov | speaking of gentoo... |
| 17:21.53 | Qwell | Faustov: let's not |
| 17:22.36 | Faustov | ;) |
| 17:24.18 | igcewieling1 | gentoo is just FreeBSD with a Linux kernel. 8-| |
| 17:24.28 | drmessano | There is a real answer to this G729 issue. This channel is owned and moderated by project staff. If they say the topic of the "free" G.729 is off-limits, even if were only due to the legal implications in THEIR country, then they have every right to moderate it out. While I don't ever agree, on principle with "STFU or GTFO", it's valid. |
| 17:24.36 | WIMPy | We could try this one: As far as I can sse it, it's not legal to use Digium phones in the EU. |
| 17:24.48 | WIMPy | Hope that gets corrected before they appear at CeBIT. |
| 17:25.02 | drmessano | Last time I checked, this is IRC, and channel rules apply. |
| 17:25.19 | drmessano | Even if you grossly disagree with that whole principle :) |
| 17:25.57 | igcewieling1 | I disagree with Digium on plenty of issues, but their policy for unlicensed g729 is not one of them |
| 17:25.58 | WIMPy | I disagree with the disrespect for the rules of other people. |
| 17:27.17 | drmessano | You are well within your right to register #freeg729nowthatkevinmitnickisfree and spread the good word |
| 17:27.58 | WIMPy | I'd just prefer if #asterisk was #asterisk and not #digium-support. |
| 17:28.15 | dr0ck | you mean the rule of law of other people |
| 17:28.21 | navaismo | ¿? |
| 17:28.55 | file | go away for a meeting and THIS happens |
| 17:29.05 | WIMPy | dr0ck: In this case. But generally I wouldn't want to restrict it to legal issues. |
| 17:29.05 | mjordan | file: crap, I totally missed the meeting |
| 17:29.09 | Qwell | file: That'll teach you to leave. |
| 17:29.09 | mjordan | I've been watching this |
| 17:29.23 | file | mjordan, don't worry |
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| 17:29.35 | Greenlight | Digium is the project sponsor and maintainer, and it's not unreasonable for them to have such rules. Quite frankly the status quo is perfect. |
| 17:29.42 | dr0ck | wat? law restricts it whether you like it or not |
| 17:29.44 | file | mjordan, I'll just stand in for you in the future if you want :P less meetings |
| 17:29.47 | drmessano | G729 argument? Has it been 3 months already? |
| 17:29.59 | drmessano | checks his watch |
| 17:30.11 | mjordan | file: I probably would have enjoyed the meeting. And I actually had something for David this time too. Nuts |
| 17:30.12 | drmessano | Yep, mickeys third hand is on the 3 |
| 17:30.19 | Greenlight | Not to mention the fact that the non Digium verison is buggy as hell. |
| 17:30.25 | file | mjordan, IM him? |
| 17:30.37 | mjordan | file: oh, you and your technologies |
| 17:30.44 | Qwell | FAX it to him! |
| 17:30.55 | Qwell | (over WebRTC) |
| 17:30.56 | dr0ck | and use digium FFA module |
| 17:31.06 | mjordan | hides his res_fax_spandsp |
| 17:31.11 | drmessano | lol |
| 17:31.47 | Qwell | wonders where malcolmd ran off to |
| 17:32.02 | file | Qwell, his laptop went to sleep |
| 17:32.08 | mjordan | Qwell: I'm going to write that RFC for April first |
| 17:32.16 | Qwell | mjordan: Do it. |
| 17:32.27 | Qwell | You've got a month. |
| 17:32.36 | drmessano | Coffee over WebRTC would be more productive |
| 17:32.47 | mjordan | realizes that would mean having to actually know more about T.38 than he already knows and gets depressed |
| 17:32.53 | Qwell | drmessano: We do coffee over jabber here. |
| 17:32.54 | drmessano | "I'm a teapot!" |
| 17:33.12 | WIMPy | Is that firmware for the Digium phones that was just announced available somewhere? |
| 17:33.41 | newtonr | i missed all the fun in here! |
| 17:33.53 | file | it's a party and it's not even Friday! |
| 17:34.10 | dr0ck | nothin like showing up sober and late to the party |
| 17:34.20 | Qwell | file: The bot seems to think it is. |
| 17:38.00 | igcewieling1 | I didn't notice any info in the upgrade file for asteirsk 10 or 11. Did anything having to do with progress, inband audio, or ringback change in Asteisk 10 or 11 with regards to SIP? I'm having an odd ringback issue |
| 17:38.37 | igcewieling1 | Drat! I was going to write an RFC for ASCII over DTMF for April 1 |
| 17:38.47 | file | don't think so, and no issues spring to mind from people or -users posts |
| 17:39.05 | igcewieling1 | file: thanks. I was afraid it was not that easy. |
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| 18:06.50 | drmessano | When the Asterisk? |
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| 18:56.21 | igcewieling1 | *sigh* We still have customers down from Sandy. Yay Verizon. |
| 18:57.01 | drmessano | wow |
| 18:57.50 | igcewieling1 | drmessano: yeah. Not many though |
| 18:58.01 | [TK]D-Fender | igcewieling1, Yeah, you go give them your .02 cents worth!!! |
| 18:58.13 | igcewieling1 | in many places VZ will not be repairing the copper, they are installing fiber instead. |
| 18:59.00 | igcewieling1 | [TK]D-Fender: I'm not allowed. Upper management are TOTAL wusses about taking a chance on pissing off VZ. Sort of understandable, they could put us out of business with any number of "mistakes" |
| 19:00.46 | igcewieling1 | if it was up to me we'd have already filed multiple lawsuits against VZ |
| 19:08.35 | *** join/#asterisk moos3 (~textual@cpe-72-224-215-87.maine.res.rr.com) |
| 19:08.55 | moos3 | how can i view only T1 calls and completely ignore SIP calls |
| 19:09.13 | ChannelZ | what do you mean "view" |
| 19:09.49 | WIMPy | dahdi show channels? pri debug? |
| 19:10.35 | moos3 | just need to see if all my t1 lines are filled up, a customer claims they aren't getting thought |
| 19:12.28 | edong23 | lawsuits because of a natural disaster? |
| 19:17.14 | igcewieling1 | moos3: asterisk -rx "core show channels" | grep DAHDI |
| 19:17.25 | WIMPy | That's 'dahdi show channels' then. |
| 19:17.37 | igcewieling1 | no, dahdi show channels shows CONFIGURED channels. |
| 19:17.57 | WIMPy | There's the extension column. |
| 19:18.02 | ChannelZ | it'll show what extension they're in which is sort of useful |
| 19:18.05 | igcewieling1 | edong23: no, but they have a bazillion other problems. |
| 19:18.26 | ChannelZ | otherwise yeah you've got to do it externally if you want to sift out certain channels |
| 19:18.29 | igcewieling1 | WIMPy: try it. It shows the callerid of the last call if there is no active call on the channel |
| 19:18.54 | WIMPy | igcewieling1: Not for me. |
| 19:18.59 | WIMPy | It's empty. |
| 19:19.13 | igcewieling1 | WIMPy: maybe the channel has taken no calls? Maybe it changed since 1.4 |
| 19:19.24 | *** join/#asterisk SuPrSluG (~SuPrSluG@50.75.185.122) |
| 19:19.40 | WIMPy | There are definitely more tahn 0 cannels that have been used. |
| 19:19.49 | WIMPy | And it might well have changed since then. |
| 19:21.08 | igcewieling1 | on my 1.8 it does not show anything in the callerid field on a PRI with some active calls |
| 19:21.30 | moos3 | interesting |
| 19:22.08 | igcewieling1 | they are likely outbound calls rather than inbound calls |
| 19:22.13 | WIMPy | the box I've been looking at ist relatively up to date. |
| 19:22.48 | moos3 | that appears to only show every channel but not if in use when i do dadhi show channels |
| 19:23.41 | *** join/#asterisk timahvo1 (~rogue@105.160.24.50) |
| 19:24.49 | WIMPy | Ypu should both upgrade :-) |
| 19:26.21 | igcewieling1 | asterisk -rx "core show channels" | grep DAHDI should work for any version of asterisk using DAHDI |
| 19:26.47 | WIMPy | Yes, but you have to count yourself. |
| 19:27.13 | WIMPy | With dahdi show channels you see which channels are free. |
| 19:27.48 | igcewieling1 | only if you are running Asterisk version higher than 1.8 |
| 19:28.06 | igcewieling1 | WIMPy: make an outbound call and see if it shows up |
| 19:28.16 | WIMPy | It does |
| 19:35.06 | *** join/#asterisk CrashSys (~kumba@rrcs-97-76-33-146.se.biz.rr.com) |
| 19:35.11 | *** join/#asterisk jkroon (~jkroon@41.244.34.26) |
| 19:35.18 | jkroon | leifmadsen, you around? |
| 19:35.20 | CrashSys | Does dahdi or dahdi-tools require libpri to completely build? |
| 19:35.26 | leifmadsen | jkroon: I am |
| 19:35.42 | jkroon | https://issues.asterisk.org/jira/browse/ASTERISK-17848 <-- i can reproduce (i think) with 11.2.1 |
| 19:35.47 | leifmadsen | neat |
| 19:35.54 | leifmadsen | 302 redirects to newtonr |
| 19:35.54 | WIMPy | CrashSys: If you want ISDN support. |
| 19:36.09 | jkroon | don't have the high cpu usage, but I do get the repeated iax2 max retries exceeded. |
| 19:36.16 | jkroon | newtonr, ? interested? |
| 19:36.26 | leifmadsen | ya sorry, I'm not a bug marshal anymore |
| 19:36.42 | jkroon | no problem. just that you closed the ticket with won't/can't fix. |
| 19:36.53 | jkroon | figured you're as good a starting point as any to try and solve the problem. |
| 19:36.54 | newtonr | looks |
| 19:37.06 | CrashSys | Wimpy: I know that libPRI is for ISDN Primary, but does DAHDI and DAHDI Tools require LibPRI in any way? I know that asterisk needs it |
| 19:37.38 | jkroon | not sure what type 6 and subtype 12 means, but I *suspect* the remote peer has gone away, and that causes iax/2 to try and hangup the channel, but that fails repeatedly without ever actually destroying the channel. |
| 19:37.50 | jkroon | looking at the code this seems viable. |
| 19:38.13 | WIMPy | CrashSys: I can;t remember which way round it is, but in the latest version there is a dependency between dahdi and libpri. |
| 19:38.35 | WIMPy | But despite its name ist not only for primary, but for absic ISDN as well. |
| 19:38.36 | jkroon | code in question at channels/chan_iax2.c:3550-3569 |
| 19:39.02 | CrashSys | So, if dahdi depends on libpri, and libpri depends on dahdi, how do you compile them? |
| 19:39.07 | jkroon | WIMPy, dahdi depends on libpri |
| 19:39.08 | WIMPy | s/absic/basic/ |
| 19:39.26 | jkroon | CrashSys, libpri doesn't depend on dahdi. |
| 19:39.37 | CrashSys | OK |
| 19:39.38 | newtonr | jkroon: just go ahead and file a new issue if its not in 1.8 and doesn't have the CPU consumption issue as well. What timing source are you using? |
| 19:39.48 | CrashSys | SO that break the circular dependency :) |
| 19:39.48 | jkroon | timerfd |
| 19:40.01 | jkroon | newtonr, it's not related to timer i don't think. |
| 19:40.14 | CrashSys | so the build order should be: libpri --> dahdi --> asterisk |
| 19:40.14 | jkroon | ok, creating new issue quick |
| 19:41.37 | jkroon | CrashSys, yes, and you're forgetting the os specific kernel modules. |
| 19:41.44 | newtonr | jkroon: kk, add a log with DEBUG,VERBOSE and iax debug turned up as well |
| 19:41.58 | *** join/#asterisk emora (~emora@213.236.11.83) |
| 19:42.13 | WIMPy | Yes, never use the latest kernel. |
| 19:42.21 | jkroon | newtonr, will add what I can, but this is from a rather busy production system so won't be able to get everything you want. |
| 19:42.26 | jkroon | WIMPy, why not?! |
| 19:42.37 | WIMPy | In fact afte updating my test box, I couldn;t get any of the ISDN channels to compile :-( |
| 19:42.57 | newtonr | jkroon: k, i'll look at it after you file and see what else we might need. will respond on JIRA issue |
| 19:43.06 | jkroon | thanks. |
| 19:43.14 | themrrobert | Still having delays in AMI that make it unusable in our environment |
| 19:43.42 | WIMPy | themrrobert: Have you found out where exactely they are happening? |
| 19:45.01 | themrrobert | Box1 is connected to PBX via network cable (direct, outside of the network, this was an attempt to fix, but still same issue). Box 1 sends the AMI action, and Pbx doesn't respond for a while |
| 19:46.20 | WIMPy | Do you have a pcap with timestamps or something? |
| 19:47.47 | themrrobert | No, but we've tried different "Box 1"'s, also tried different cables, it is a monumentous task trying to track down a specific bad packet |
| 19:49.11 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:49.16 | *** part/#asterisk CrashSys (~kumba@rrcs-97-76-33-146.se.biz.rr.com) |
| 19:54.55 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:56.05 | jkroon | newtonr, looks like the call went through the dialplan as the dialplan got reloaded! could that in any way make a call get "wedged" somehow? |
| 19:58.29 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 19:59.18 | newtonr | jkroon: not sure :) maybe |
| 20:01.12 | jkroon | newtonr, ok, let me just file the bug ... not sure if there is any usable info though ... |
| 20:01.17 | jkroon | i'll leave you to judge that |
| 20:04.55 | *** join/#asterisk willryder (~tedryder@nc-184-3-102-84.dhcp.embarqhsd.net) |
| 20:06.22 | file | SO |
| 20:06.46 | file | what happens when you reload a dialplan is that the new dialplan is built, and then merged into the old one - but that operation will prevent calls from progressing until it is complete |
| 20:07.04 | file | so you don't get into some sort of undefined state by the very reload process itself |
| 20:09.08 | newtonr | interesting |
| 20:09.25 | [TK]D-Fender | file, When a channel is created the entire "current" copy is copied over to it. So only new channels get the new dialplan |
| 20:09.54 | file | nope |
| 20:09.56 | [TK]D-Fender | file, So you don't get "hybrid" processing snafu's |
| 20:10.06 | [TK]D-Fender | Supposed toIIRC |
| 20:14.15 | jkroon | file, so whilst a channel is in the dialplan it won't go into the new dialplan - it'll stick in the old one? |
| 20:14.24 | file | no |
| 20:14.43 | jkroon | ok, so atomically replaced? |
| 20:14.47 | file | yes |
| 20:15.02 | carrar | Asterisk is magically delicous |
| 20:15.13 | jkroon | so if channel x sits on foo,123,3 then it can hit the new dialplan on foo,123,4 ? |
| 20:15.19 | file | yes |
| 20:15.26 | jkroon | that seems ... wrong |
| 20:15.53 | jkroon | and would explain quite a number of interesting things I've seen in the past ... |
| 20:16.18 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 20:16.18 | file | to do otherwise would be very very complicated... |
| 20:16.30 | jkroon | probably |
| 20:16.39 | jkroon | but it would be the right thing to do if possible. |
| 20:17.55 | carrar | jkroon, you make all your changes on a dev/test server and then push them to production last at night in a maint window anyways rights? |
| 20:18.02 | carrar | late |
| 20:18.07 | carrar | heh |
| 20:24.20 | ChannelZ | What happens if an exten is on a priority higher than exists in the newly loaded dialplan? Does that old one wind up remaining as a straggler once the extension leaves it? |
| 20:25.10 | file | no |
| 20:26.45 | ChannelZ | That's good |
| 20:27.58 | file | the PBX core is more of a query and execution system... |
| 20:28.14 | *** join/#asterisk bytemaster (~ewrewr@host81-150-217-168.in-addr.btopenworld.com) |
| 20:28.25 | file | so when you are "in the dialplan" you are really in a loop that just queries the PBX core for the next thing to do, retrieves it, executes, and then moves to the next step |
| 20:28.27 | file | rinse and repeat |
| 20:28.55 | file | (until other stuff happens) |
| 20:29.01 | jkroon | carrar, i could - but i quite possibly won't trigger the specific bug at that time. |
| 20:29.03 | ChannelZ | ok so it's not really locked or a pointer to the actual dialplan being used |
| 20:29.15 | file | right |
| 20:29.32 | *** join/#asterisk vastersk (~vinscentp@124.6.136.142) |
| 20:29.51 | jkroon | eesh, ChannelZ that locked pointer thing seems more sane to me ... |
| 20:29.56 | jkroon | sounds anyway |
| 20:29.56 | vastersk | hey guys...need ur expertise here... |
| 20:30.56 | vastersk | having trouble dialing out the extensions of all remote locations connectd via vpn... |
| 20:32.24 | ChannelZ | jkroon: not really, I think it'd actually be more complicated.. both in implementation and operation in the case where you had to reload on a system with active calls. |
| 20:33.03 | jkroon | implementation would be insane, don't even want to think of that. however i respectfully disagree on the operation. |
| 20:33.30 | jkroon | i'd much rather have a call stick in one dialplan (or context at least) during execution. |
| 20:33.32 | jkroon | https://issues.asterisk.org/jira/browse/ASTERISK-21193 |
| 20:33.37 | jkroon | newtonr, ^ |
| 20:34.04 | ChannelZ | but from what I'm understanding, it does |
| 20:34.11 | jkroon | i'm going to have to forcibly restart that instance in the next few minutes - so if you need/want me to query anything now is the time. |
| 20:34.55 | file | the next time it queries it gets the new dialplan, which happens at each priority |
| 20:35.01 | ChannelZ | I mean I guess you are saying you want a given channel to continue using the dialplan as it existed prior to the reload until it's done |
| 20:35.21 | jkroon | no, consider for example this exten => 123,1,NoOP(), same => Dial(SIP/foo); same => Hangup(); now replace that with 1,NoOP(), 2,NoOP, 3,Dial(SIP/foo), 4,Hangup() |
| 20:35.55 | jkroon | now, whilst the prio 2 Dial(SIP/foo) is in progress the dialplan gets reloaded, once SIP/foo hangs up it will simply get called again, immediately. |
| 20:36.21 | file | if you specified the option to allow dialplan execution to continue for the caller, yes |
| 20:36.34 | jkroon | that's the default now isn't it? |
| 20:36.50 | file | pretty sure no |
| 20:37.02 | file | that would be a major behavior change |
| 20:37.05 | *** join/#asterisk Uthark (~Uthark@190.0.58.186) |
| 20:37.06 | jkroon | either way ... i think my example shows what it's intended. |
| 20:37.26 | jkroon | file, interesting ... i'll definitely test that at some point. |
| 20:37.44 | Uthark | Guys, any opinion or recommendation about Yealink IP phones? |
| 20:38.18 | *** join/#asterisk teff (~teff@client-80-1-163-4.bsh-bng-011.adsl.virginmedia.net) |
| 20:39.48 | jkroon | Uthark, good value for money |
| 20:40.06 | jkroon | don't expect them to be on par with high-end phones, but not the crappiest thing money can buy either. |
| 20:42.52 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-cajweneesyrwiksx) |
| 20:43.52 | Uthark | jkroon: Any good/bad experiences while working with them? |
| 20:44.29 | jkroon | yea, they don't honor annexb=no in g729 sdp, you have to explicitly switch it off on the phone. |
| 20:44.40 | jkroon | other than that - haven't had a days trouble. |
| 20:44.52 | jkroon | unlike some other ... often recommended ... brands |
| 20:48.54 | Uthark | jkroon: OK, thank you, you've decided the fate of two coworkers ;) |
| 20:49.12 | jkroon | that's their fate fortunately |
| 20:51.19 | ChannelZ | disclaim! disclaim! |
| 20:54.15 | *** join/#asterisk areski (~areski@80.174.255.57) |
| 20:57.21 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 21:02.16 | jkroon | file, so how does asterisk deal with chan_sip and it's peers? |
| 21:02.26 | file | I don't understand the question |
| 21:02.45 | jkroon | just picked up a case where during a reload a call to SIP/bar failed, even though bar is probably the most used sip endpoint defined in sip.conf ... |
| 21:03.29 | jkroon | Dial("Local/number@blah-0000377b;2", "SIP/bar/number,,") |
| 21:03.52 | jkroon | netsock2.c: getaddrinfo("bar", "(null)", ...): Name or service not known |
| 21:05.20 | file | it replaces the fields in the existing peer |
| 21:05.48 | file | I refuse to think about chan_sip unless I have to |
| 21:06.12 | jkroon | so i just struck it one in a million type of thing? or should I file another bug? |
| 21:06.27 | file | I have never heard of something like that happening |
| 21:06.51 | jkroon | one in a million then ... fortunately I have a backup server through which the particular call just routed and succeeded. |
| 21:09.29 | *** join/#asterisk n3hxs (~ed@pool-108-16-94-10.phlapa.fios.verizon.net) |
| 21:13.05 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-whbgabsjnjtxgpjk) |
| 21:15.04 | vastersk | anybody who got vpn tunnel skills? |
| 21:15.17 | Qwell | ~poll |
| 21:15.17 | infobot | Script for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html |
| 21:16.15 | leifmadsen | lol |
| 21:16.16 | leifmadsen | wow |
| 21:16.22 | leifmadsen | Fidonet? BBS flashbacks! |
| 21:16.41 | newtonr | jkroon: thanks for filing 21193, i'll check it out when i get a chance |
| 21:17.31 | jkroon | newtonr, np, thanks for looking! |
| 21:19.30 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-146-019.pools.arcor-ip.net) |
| 21:26.49 | vastersk | how do you connect pbxtra server to another server via vpn tunnel? is it just by simply specifying the port mactching vpn router's settings, and should work? |
| 21:27.47 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 21:28.26 | Uthark | vastersk: What kind of vpn tunnel are you using? |
| 21:28.46 | vastersk | l2tp |
| 21:53.05 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:55.40 | Uthark | vastersk: I dont have any experience with L2TP, but I've achieved the same with a simple openvpn tunnel with udp transport |
| 21:57.29 | igcewieling1 | for the most part if you have a real VPN then asterisk doesn't have to know anything about it since it is transparent at the network level |
| 22:02.05 | igcewieling1 | want me to point them to the daffys? |
| 22:05.52 | igcewieling1 | heh, the daffys message was not for here. |
| 22:15.52 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 22:19.50 | vastersk | i got... i appreciate your inputs... thanks <Uthark> |
| 22:36.35 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 22:37.21 | *** join/#asterisk navaismo (~navaismo@189.241.118.172) |
| 22:46.11 | *** join/#asterisk ShoreTel (~poppa@siptool.com) |
| 22:46.23 | ShoreTel | I have asterisk interconnected to my shoretel using siptrunk groups |
| 22:46.31 | Qwell | congratulations |
| 22:46.41 | ShoreTel | routed the SIP through a juniper with SIP packet inspection |
| 22:46.44 | ShoreTel | this thing is leet |
| 22:46.51 | ShoreTel | 3g sip on mobile devices :) |
| 22:47.25 | igcewieling1 | with no QoS |
| 22:47.50 | ShoreTel | no QoS on the cellular network, but the calls are generally ok |
| 22:48.12 | ShoreTel | if I call an ip phone on the intranet from outside, once the call hits the asterisk we roll QoS |
| 22:48.22 | ShoreTel | anything between asterisk is QoS'd |
| 22:48.37 | ShoreTel | udp baby |
| 22:48.56 | ShoreTel | it works well, I have shoretel connected to a cs1000 with numerous pri's and other trunking |
| 22:48.58 | igcewieling1 | *nod* We do end-to-end QoS for our customers so I'm a bit of a snob when it comes to that 8-| |
| 22:49.09 | ShoreTel | it is my master dialplan / distant steering database / least cost routing |
| 22:49.21 | ShoreTel | end to end over 3rd world 3g networks? |
| 22:49.26 | ShoreTel | unlikely :) |
| 22:49.40 | ShoreTel | i used bria on an iphone over 3g in thailand and was crystal clear |
| 22:49.43 | ShoreTel | it had a delay |
| 22:49.50 | ShoreTel | 300ms delay probably |
| 22:50.02 | ShoreTel | which is a lot but considering all calls were free and over wifi it worked great |
| 22:50.16 | ShoreTel | my DID could be used to contact me while overseaas |
| 22:50.28 | ShoreTel | my asterisk integration owns the shoretel mobile licensing |
| 22:50.32 | ShoreTel | that stuff is crap in comparison |
| 22:51.00 | malcolmd | maybe you wanna pitch the shortel and go all-asterisk? ;D |
| 22:51.22 | Qwell | ditch the ALG too while you're at it :p |
| 22:53.00 | drmessano | I've done 3G to 3G/4G from Bria on the iPhone to the * box on our mobile operations center, and it's pretty sweet. Lower latency than I expected |
| 22:53.25 | drmessano | Sure, we're using their data.. but screw THE MAN |
| 22:53.29 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 22:53.40 | pabelanger | Bria drains my g2 too fast |
| 22:53.50 | drmessano | TCP, yo |
| 22:53.55 | drmessano | That makes all the difference |
| 22:54.02 | pabelanger | tries |
| 22:54.22 | drmessano | UDP will drain the battery 10x faster.. Maybe that's an understatement |
| 22:55.33 | *** join/#asterisk pbxbrian (~pbxbrian@unaffiliated/brian98) |
| 22:55.41 | igcewieling1 | sounds like a bug |
| 22:55.50 | drmessano | I don't attribute that 10x to "SIP UDP vs TCP" specifically, because the different shouldn't be THAT big |
| 22:56.05 | igcewieling1 | maybe the phone and cell network spoof TCP keepalives? |
| 22:59.30 | pabelanger | okay, changed to TCP |
| 22:59.40 | pabelanger | will full charge tonight and see |
| 23:02.06 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 23:04.05 | ShoreTel | drmessano ya it works good |
| 23:04.27 | ShoreTel | if you pay the in app features for g729 :) |
| 23:04.29 | ShoreTel | its even better |
| 23:04.36 | ShoreTel | 8k a channel vs 64k is quite a bit |
| 23:04.43 | drmessano | Set up SRTP and you feel like you're doing some spy shit |
| 23:04.54 | ShoreTel | hrm |
| 23:05.14 | ShoreTel | i paid for the video conferencing in app too and applied h.264 |
| 23:05.28 | ShoreTel | works great... portgo pro on teh PC does h264... testing on the local network |
| 23:05.28 | drmessano | I've never gotten the video to work from Bria to Bria |
| 23:05.33 | ShoreTel | works great |
| 23:05.52 | ShoreTel | we should check settings to see what you're doign wrong |
| 23:05.54 | ShoreTel | i have it working |
| 23:05.58 | drmessano | I'm glad it works. That tells me i've just missed something |
| 23:06.02 | ShoreTel | install portgo pro on your PC |
| 23:06.04 | drmessano | Baselines are good |
| 23:06.06 | ShoreTel | and test with that |
| 23:06.14 | ShoreTel | baselines are relieving |
| 23:06.15 | drmessano | Ok |
| 23:06.30 | ShoreTel | it pops up a nice little conference box showing all the people you conference in too |
| 23:06.49 | ShoreTel | only the conference originator can see everyone due to the stream not being duplicated of course |
| 23:07.03 | ShoreTel | but i was thinking of something that would produce video relay for conferencing |
| 23:07.16 | ShoreTel | i don't know if there is a plugin for asterisk to make that work yet or not |
| 23:07.25 | ShoreTel | im always thinking about this type of stuff nobody care about |
| 23:07.26 | ShoreTel | :/ |
| 23:07.32 | ShoreTel | i need a raise |
| 23:08.05 | ShoreTel | time for some gang dang thai red curry and mango |
| 23:08.06 | ShoreTel | cheers |
| 23:20.27 | themrrobert | @leifmadsen: WIMPy : so I've done a lot of packet tracing, and it does look like the AMI is processing events and responding to them immediately (via wireshark listening on the PBX). What then could cause such a delay between send and receive? The software on Box 1 that interacts with PBX ami receives a ton of events via tcp ami updateing it, but it only sends out relatively little. Connected |
| 23:20.28 | themrrobert | directly via gigabit. Box 1 uses windows, i was guessing maybe tcp stack delay? |
| 23:20.50 | leifmadsen | shrugs |
| 23:21.05 | leifmadsen | if wireshark shows asterisk processing immediately, I can only thing it must be a networking issue |
| 23:21.11 | leifmadsen | perhaps QOS or something is causing a delay somewhere |
| 23:22.27 | themrrobert | Yea, I thought that before I connected them directly. : in their own subnet. no switch between: CAT6 between two servers. (on em3 on the asterisk pbx) |
| 23:22.44 | themrrobert | and thats the interface wireshark was listening on |
| 23:22.59 | themrrobert | should listen on the other side |
| 23:23.05 | themrrobert | i'll let you know what i find out |
| 23:24.14 | igcewieling1 | bug in yuor AMI client? |
| 23:24.21 | leifmadsen | is the side absorbing (the client) causing the delay itself? |
| 23:43.42 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 23:45.02 | themrrobert | i'm not sure. i'm going to set up another controlled test, but its going to a bit logistically to synchronize the the two data. that should give definitive answers |