| 00:00.01 | Katty | now i have no plans >.< |
| 00:06.26 | raden | awe :( |
| 00:06.28 | raden | poor katty |
| 00:11.48 | SeRi | Katty: Rain? |
| 00:13.31 | SeRi | I will sacrifice 1 goat and 2 chickens for some rain down here in the south... So far the Gods fo not like the idea.. :/ |
| 00:38.00 | Katty | raden: i weeded the garden ...uhh... buckets |
| 00:39.34 | Katty | SeRi: yeah i think we're supposed to get rain here shortly |
| 00:40.19 | Katty | or at least the /radar/ seems to think so. never can tell with missouri |
| 00:41.13 | Katty | raden: and now, laundry! how exciting for a sunday night :/ lol |
| 00:48.52 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
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| 01:12.30 | raden | lol |
| 01:28.18 | apb1963 | off-topic warning... any recommendations as to 1) which API to use for mapping (google, mapquest, yahoo, bing??)... 2) JSON or XML ? |
| 01:28.41 | WIMPy | Where is OSM? |
| 01:29.06 | apb1963 | was that directed at me? |
| 01:29.17 | WIMPy | yes |
| 01:29.19 | apb1963 | googles OSM |
| 01:31.15 | apb1963 | that's news to me... but... kinda plain |
| 01:31.20 | apb1963 | just my opinion |
| 02:00.49 | *** join/#asterisk corretico (~luis@186.96.91.236) |
| 02:06.30 | igcewieling | apb1963: In my limited experience, unless you are good with XML, use JSON |
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| 02:31.25 | Katty | peeks in |
| 02:32.49 | cusco | nothing to see |
| 02:32.51 | cusco | move along |
| 02:33.23 | Katty | moves along to the fridge. |
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| 02:37.07 | Katty | noms chocolate and marshmallows |
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| 02:51.25 | igcewieling | is hunting rogue telephone numbers. |
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| 03:26.20 | absd | Can anyone hit me with a cluebat for this error message? " load_indications: Unable to set the default country (for indication tones)" |
| 03:47.13 | Weezey | igcewieling: done importing? |
| 03:48.01 | Weezey | steals a marshmallow |
| 03:56.58 | absd | if anyone cares, that load indications was simply a typo in a country, and trying to set the default to that country. |
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| 04:21.54 | igcewieling | Weezey: yup! finished overnight. |
| 04:22.21 | igcewieling | we lost a days worth of reports, but that is all. |
| 04:22.42 | igcewieling | more like two days of reports. heh |
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| 08:40.53 | tparcina | Can I turn off verbose on CLI, but still log it in log file? |
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| 08:47.20 | wdoekes | tparcina: yes you can |
| 08:47.21 | tparcina | It seams that logger mute is the command I was looking for. |
| 08:47.28 | tparcina | :) |
| 08:47.46 | wdoekes | you can disable verbose in the console=> line in logger.conf too, if you like |
| 08:47.49 | tparcina | wdoekes: Did you have diferent suggestion? |
| 08:48.40 | tparcina | wdoekes: And if I ever need verbose in console, I have to change that file again? |
| 08:49.36 | wdoekes | you change the file if you expect to not want to see verbose, most of the time |
| 08:49.42 | *** join/#asterisk geeksteve (~geeksteve@emh-nat.poundbury.com) |
| 08:49.51 | wdoekes | if you only want it muted right now, but not normally, you use logger mute |
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| 08:59.46 | danfromuk | Hi, does anyone know whats happened to ipness.com? A client has a DID with them and wants to move but the website is down. |
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| 10:45.35 | eject_ck | what if I see dtmf in log and Blind Transfer #1 Attended Transfer #2, but cnothing happens ? |
| 10:46.00 | eject_ck | I mean I dial #1 but and I see it in console |
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| 11:09.50 | gavimobile | hey folks, what is this dependency? generic_odbc |
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| 11:12.29 | gavimobile | nevermind |
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| 11:28.11 | davlefou | hi, what is the pcmu H263-2000? how can i use it with asterisk? |
| 11:32.49 | kaldemar | davlefou: pcmu is an audio codec, also known as G.711 ulaw. H.263-2000 is a video codec. their names in asterisk are ulaw and h263. |
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| 11:45.01 | davlefou | kaldemar, ok |
| 11:47.25 | *** join/#asterisk kontinuity (~kontinuit@122.178.240.81) |
| 11:47.29 | kontinuity | hi folks |
| 11:47.50 | kontinuity | how do I know if a call has not been answered when its put in a Queue application? |
| 11:48.05 | kontinuity | every CDR record has the disposition as "ANSWERED" |
| 12:16.32 | vlt | Hello. After going from 1.2 to 1.8 last week I miss (at least) one tab completion feature. `meetme list <digit><tab>` doesn’t work anymore. Is this intentional? |
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| 12:50.50 | Tuju | "Looking for tuju-dmz-trunk in default (domain 195.50.204.214)" - how does asterisk map incoming packets to domain? |
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| 13:09.02 | Katty | morning |
| 13:09.08 | Katty | infobot: crittercam |
| 13:09.08 | infobot | it has been said that crittercam is Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
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| 13:17.34 | beardy | checks it out |
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| 13:24.09 | [TK]D-Fender | Tuju: Asterisk doesn't really do domains. |
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| 13:27.29 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:57.37 | SuperNull | hey guys. |
| 13:58.36 | zFleshMissile | Hey guys, does anyone know what a "chan_dahdi.c: Short write: 0/5 (Unknown error 500)" means? Phones have been cutting off and seeing huge amounts of HDLC 6 errors and yellow alarms and I noticed that error in the logs and haven't seen it before. |
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| 13:58.44 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:59.30 | zFleshMissile | Also a "ERROR[29848] chan_dahdi.c: Write to 85 failed: Unknown error 500" in there too |
| 14:00.08 | Tuju | [TK]D-Fender: i'm having authentication problems when it says it Looks for that account in 'default', not really sure what that is, apparently a context. |
| 14:02.40 | *** join/#asterisk boch (~boch@186.182.116.157) |
| 14:02.47 | boch | good morning all |
| 14:03.48 | newtonr | zFleshMissile: those sort of errors are typically not very good. I've seen that caused by both a bad physical layer (wiring, connectors, etc) and also by mis-confguration. I'd contact the support dept for the vendor of your card. |
| 14:04.27 | newtonr | zFleshMissile: I'd try to help troubleshoot further, but I haven't ran into those errors in ages, and I don't work with hardware PSTN interfaces that often anymore. |
| 14:05.11 | boch | i need to record a video call between two phones, i have tried MixMonitor but had no success, hows the proper way to do this?? |
| 14:06.56 | [TK]D-Fender | Tuju: it is... |
| 14:07.15 | [TK]D-Fender | Tuju: It is telling your what EXTENSION it is looking for and in which CONTEXT |
| 14:07.35 | Tuju | so it is context. |
| 14:08.03 | Tuju | things got bit better when i put the hostname into /etc/hosts in client end. (both are asterisk servers) |
| 14:08.19 | [TK]D-Fender | Tuju: No need... I would advise against that |
| 14:08.30 | Tuju | well, it kept complaining. |
| 14:08.33 | [TK]D-Fender | that isn't how things are authed |
| 14:08.46 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 14:09.05 | Tuju | easy to say, when nothing works at this end. |
| 14:09.37 | gavimobile | this is my logger.conf http://pastebin.com/aqsx4JX7 why do I see lines like this in my logs chan_sip.c: Failed to authenticate device 1006<sip:1006@81.218.196.30> I've added security => security and im using asterisk 11? |
| 14:09.52 | gavimobile | that's my public ip address |
| 14:10.19 | Tuju | imo debuggin is hell, why i cannot say a line that i want to follow? |
| 14:10.36 | Tuju | something like 'sip debug extension plops' |
| 14:11.00 | [TK]D-Fender | Tuju: "help sip set debug " |
| 14:11.09 | Tuju | [TK]D-Fender: been there, didn't work. |
| 14:11.17 | [TK]D-Fender | Tuju: But that should not be necessary either... enable FULL SIP debug and show us the call |
| 14:11.31 | [TK]D-Fender | Tuju: Show us. don't just say "didn't work" |
| 14:11.37 | Tuju | [TK]D-Fender: yes, it shows. It's like a grain of sand... in sahara. |
| 14:11.58 | [TK]D-Fender | Tuju: Show us |
| 14:12.31 | Tuju | alca*CLI> sip set debug peer tuju-dmz-trunk |
| 14:12.33 | Tuju | Unable to get IP address of peer 'tuju-dmz-trunk' |
| 14:12.40 | Tuju | why should peer have an ip? |
| 14:13.20 | [TK]D-Fender | Because it'd be nice for that peer to have a place to send calls from... |
| 14:13.37 | [TK]D-Fender | that peer clearly has not registered... or doesn't exist |
| 14:13.47 | Tuju | there you go. So let's go back to that original: "imo debuggin is hell, why i cannot say a line that i want to follow?" |
| 14:14.09 | Tuju | how the hell i'm going to debug registration in the first place? |
| 14:14.25 | Tuju | "lets-debug-then-when-everything-works" |
| 14:14.39 | [TK]D-Fender | You can't follow a peer that you can't IDENTIFY |
| 14:14.46 | igcewieling | Tuju: you can complain or you can try to fix, I recommend the latter |
| 14:14.49 | [TK]D-Fender | if it hasn't registered... how does asterisk know it's THEM>? |
| 14:14.55 | [TK]D-Fender | You are chicken & egging yourself |
| 14:15.04 | Tuju | igcewieling: hold your breath, patch is underway. |
| 14:15.11 | igcewieling | Tuju: wireshark / tcpdump are useful for stubborn SIP problems |
| 14:15.21 | [TK]D-Fender | no need |
| 14:15.24 | [TK]D-Fender | the call is coming in |
| 14:15.30 | Tuju | [TK]D-Fender: i think you said yourself above that authentication is not done via dns |
| 14:15.32 | igcewieling | Tuju: any patches to Asterisk which add new features don't get added to released code for a year or more |
| 14:15.33 | [TK]D-Fender | We just aren't getting it |
| 14:15.44 | Katty | stares at her chart |
| 14:15.55 | Katty | is 2lbs a month drop an acceptable number? it seems low |
| 14:16.11 | [TK]D-Fender | Tuju: DNS is not "authentication" |
| 14:16.12 | igcewieling | Katty: yes |
| 14:16.14 | [TK]D-Fender | Tuju: SHOW US |
| 14:16.22 | [TK]D-Fender | ~pb |
| 14:16.22 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:16.24 | [TK]D-Fender | ^^^ |
| 14:16.44 | Tuju | [TK]D-Fender: indeed, so why not bind registration packets to extension with those crenditals, and not ask fskcing ip then? |
| 14:17.12 | [TK]D-Fender | Tuju: It doesn't mean it's asking for an IP. |
| 14:17.23 | Tuju | [TK]D-Fender: don't flood channel. |
| 14:17.24 | [TK]D-Fender | Tuju: You can't debug something that hasn't IDENTIFIED itself to * yet |
| 14:17.53 | Tuju | [TK]D-Fender: if i send packet with correct line name, it's identified already. |
| 14:18.03 | Katty | igcewieling: yes acceptable, or yes that seems low to you as well? |
| 14:18.05 | Tuju | If i can see it with my own eyeballs, so can software. |
| 14:18.23 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 14:18.26 | [TK]D-Fender | [10:17]Tuju[TK]D-Fender: if i send packet with correct line name, it's identified already. <- who says you got it right and Asterisk accepted it? |
| 14:18.30 | Tuju | but i can already see where the problem is and it's not technical. |
| 14:18.40 | [TK]D-Fender | Tuju: Stop rthis useless whining and SHOW us so we can fix this |
| 14:18.51 | [TK]D-Fender | Tuju: You are going in circles for no good reason. |
| 14:18.52 | igcewieling | Katty: The slower you drop the weight the more healthy. Thankfully, I don't worry about such things as weight. |
| 14:19.04 | igcewieling | [TK]D-Fender: just give up. |
| 14:19.06 | Tuju | [TK]D-Fender: "If i can see it with my own eyeballs, so can software." <------------- did you read that? |
| 14:19.28 | igcewieling | Tuju: I'm sure he did and I'm sure he doesn't care. You work with what you have. |
| 14:19.39 | Tuju | igcewieling: indeed. |
| 14:19.44 | Katty | igcewieling: hrmm. i guess that's true. i'm just impatient :/ |
| 14:19.52 | igcewieling | If you hate Asterisk so much, try FreeSwitch or something. |
| 14:20.21 | Tuju | igcewieling: why do you think Katty hates asterisk? |
| 14:20.23 | [TK]D-Fender | Tuju: You can't debug a peer that * doesn't have an IP for. This is either by virtue of it actually registering to you... or you specifying the host explicitly. |
| 14:20.33 | [TK]D-Fender | Tuju: This ain't Raw-Cat Sigh-Hence. |
| 14:20.37 | igcewieling | Katty: worry less about "weight" and more about general health. |
| 14:20.55 | [TK]D-Fender | Tuju: And the message you came in with... was Aa DIALPLAN one.. not a SIP one. |
| 14:21.39 | Tuju | [TK]D-Fender: yep, that's another thing, i don't get why it refers dialplan context and gives error for OPTIONS packet regardless that it triest o REGISTER. |
| 14:22.33 | [TK]D-Fender | Tuju: And options looks for a specific extension in a specific context as well.... and generally doesn't even MATTER since it's only used as a keep-alive... |
| 14:22.41 | Kobaz | Katty: slower the better |
| 14:23.37 | gavimobile | [TK]D-Fender: could you have a look at my question above? |
| 14:23.38 | Tuju | [TK]D-Fender: so why it doesnt' REGISTER first? it doesn't make sense. |
| 14:23.57 | [TK]D-Fender | Tuju: Why doesn't what register? |
| 14:24.03 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 14:24.13 | Tuju | uuumh, the client? |
| 14:24.59 | [TK]D-Fender | Tuju: IDid I configure it? How are we supposed to know? You've shown us nothing. Maybe it tried and it failed. Maybe the registration attemps never arrived... who know.... |
| 14:25.43 | Tuju | we're talking in general level, we don't *need* to see anything. |
| 14:26.00 | [TK]D-Fender | Then what I've said could be the case |
| 14:26.05 | [TK]D-Fender | You configured the client wrong. |
| 14:26.08 | [TK]D-Fender | Or the packets never arrived |
| 14:26.14 | Tuju | apparently i did. |
| 14:26.17 | [TK]D-Fender | or they did and the auth was wrong and it's rejected |
| 14:26.27 | [TK]D-Fender | Or your peer was set up wrong so it couldn't load at all |
| 14:26.30 | [TK]D-Fender | Or anything |
| 14:26.40 | Tuju | i already tried to go that part, that my context/domain crap is mixed up. |
| 14:26.43 | Tuju | most likely. |
| 14:26.45 | [TK]D-Fender | Since all you're leaving us with is hypotheticals... anything will do apparently. |
| 14:26.49 | [TK]D-Fender | NO |
| 14:26.52 | gavimobile | I think I found my answer |
| 14:27.00 | gavimobile | it was in /var/log/asterisk/security |
| 14:27.01 | [TK]D-Fender | The call arrived and it is LOOKING in the dialplan... so sip.conf is allowing the call |
| 14:27.21 | [TK]D-Fender | Whether it matched a peer isanother matter.... |
| 14:27.47 | Tuju | [TK]D-Fender: I've got two asterisk boxes registering each other and other works, other doens't. problematic tries to register and server-end sees OPTIONS packet and client end says Wrong password. |
| 14:28.15 | Tuju | it should see REGISTER packet if i've got this at all. |
| 14:28.26 | [TK]D-Fender | Asterisk doesn't care about passwords |
| 14:28.29 | [TK]D-Fender | for OPTIONS |
| 14:28.35 | [TK]D-Fender | and options itself is jsut a keep alive |
| 14:28.36 | Tuju | indeed. |
| 14:28.38 | [TK]D-Fender | it is not important |
| 14:30.01 | igcewieling | Tuju: Registration is not required for Asterisk to accept a call from a peer. |
| 14:30.23 | Tuju | igcewieling: I'm not trying to make a call. |
| 14:30.30 | igcewieling | Registration is only required for calls from Asterisk to the client if you don't know the address of the peer ahead of time. |
| 14:30.34 | [TK]D-Fender | The call is arriving, and passing sip.confSo if you're going to neurose about it... feel free to continue wasting your time for nothing. |
| 14:30.34 | [TK]D-Fender | We've seen that much... |
| 14:30.34 | [TK]D-Fender | Actually.. that's all we've seen. |
| 14:30.36 | [TK]D-Fender | But it passes... |
| 14:30.41 | [TK]D-Fender | and what remains... is dialplan... |
| 14:30.42 | igcewieling | Tuju: exactly, so stop talking about registration and move on |
| 14:30.55 | [TK]D-Fender | well registration may be another issue.... |
| 14:31.12 | [TK]D-Fender | Maybe once we see an actual problem we can say something about it.... |
| 14:36.37 | SuperNull | working on an older system here, is it advised to use cdr_odbc over cdr_mysql these days? |
| 14:36.42 | gavimobile | fail2ban has 2 entries in the default jail.conf for my default centos6 install [asterisk-tcp] and [asterisk-udp] |
| 14:36.42 | SuperNull | (1.8) |
| 14:36.45 | gavimobile | which should I enable? |
| 14:37.06 | SuperNull | are you using tcp or udp.. derp |
| 14:37.08 | [TK]D-Fender | gavimobile: YES |
| 14:37.12 | SuperNull | ^^^^ |
| 14:37.53 | gavimobile | SuperNull: well port 5060 uses udp, no? |
| 14:38.27 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 14:38.37 | SuperNull | it can. |
| 14:38.39 | [TK]D-Fender | Ports don't use protocols... |
| 14:38.46 | [TK]D-Fender | your sense of heirachy is off.... |
| 14:38.47 | gavimobile | would it be bad if enabled them both? |
| 14:38.53 | igcewieling | gavimobile: remember Asterisk's format for the messages which fail2ban needs to look at changed between versions. you should confirm the built in rules actually work for your asterisk version |
| 14:38.54 | gavimobile | [TK]D-Fender: cause your not helping me :-p |
| 14:39.07 | [TK]D-Fender | gavimobile: Is there any reason NOT to look at both? |
| 14:39.14 | SuperNull | gavimobile.. most people use 5060 udp so TCP doesn't get clogged with DOS due to half handshakes |
| 14:39.14 | gavimobile | igcewieling: yes, I will confirm |
| 14:39.15 | [TK]D-Fender | gavimobile: How about ... use what ASTERISK uses.... |
| 14:39.24 | [TK]D-Fender | gavimobile: That might make some sort of sense.... |
| 14:39.43 | gavimobile | igcewieling: as long as the log has not been cleared, fail2ban will rescan logs on restart |
| 14:39.53 | igcewieling | gavimobile: the only people who run sip over tcp are those use Microsoft products, which don't support sip over udp |
| 14:40.00 | jmetro | i only use port 47471337 for sip |
| 14:40.12 | gavimobile | igcewieling: like a softphone? |
| 14:40.30 | gavimobile | jmetro: there's less than 65000 ports in total |
| 14:40.32 | gavimobile | :-) |
| 14:40.46 | jmetro | oh i dont use TCP, i use LEETCP |
| 14:40.48 | gavimobile | would it hurt me to enable them both in the jail? |
| 14:40.51 | igcewieling | gavimobile: I didn't know Microsoft had a softphone |
| 14:41.01 | gavimobile | igcewieling: well not microsoft, but windows |
| 14:41.14 | jmetro | igcewieling: Microsoft Communicator i think |
| 14:41.17 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 14:41.21 | igcewieling | gavimobile: Let me be more specific: Microsoft Linc / SIP Server |
| 14:41.24 | gavimobile | zoiper for example or xlite, they may be for windows.. do they use tcp? |
| 14:41.24 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 14:41.31 | gavimobile | igcewieling: ahhh |
| 14:41.38 | igcewieling | gavimobile: I doubt it. They ALL support UDP |
| 14:41.42 | gavimobile | didn't know microsoft had a sip server |
| 14:41.55 | igcewieling | gavimobile: most people don't and that is a good thing for the world |
| 14:42.27 | gavimobile | so once again, can I enable both of them in the jail? |
| 14:42.34 | gavimobile | would it hurt me? |
| 14:42.46 | igcewieling | gavimobile: I doubt it will hurt. |
| 14:42.57 | gavimobile | igcewieling: great! thanks |
| 14:43.01 | igcewieling | Though I normally disable sip/tcp in sip.conf so I'll never get SIP on UDP |
| 14:43.24 | igcewieling | I am SO glad we don't need to use fail2ban anymore |
| 14:43.35 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:46.06 | *** join/#asterisk Draecos (~Draecos@124.150.62.62) |
| 14:49.44 | *** join/#asterisk dr0ck (~dr0ck@c-75-70-61-20.hsd1.co.comcast.net) |
| 14:54.07 | gavimobile | igcewieling: well why not? |
| 14:54.19 | gavimobile | how is that possible? you don't get attacks like the one I posted above? |
| 14:54.34 | igcewieling | gavimobile: because we don't allow any access from off-net using iptables |
| 14:55.11 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:55.44 | igcewieling | We never had any Asterisk/FreePBX server hacked which blocked all access from outside our allocated IP ranges. |
| 14:56.18 | gavimobile | igcewieling: you don't have any remote peers? |
| 14:56.44 | igcewieling | gavimobile: the few remote peers we have have specific exceptions for their IP |
| 14:56.46 | drmessano | TCP is very useful for mobile clients.. Just sayin |
| 14:57.26 | gavimobile | igcewieling: they each have a static ip? |
| 14:57.34 | igcewieling | drmessano: Yes! I'd forgotten about that. Apparently the network "spoofing" and battery saving features work far far better on TCP. |
| 14:57.51 | igcewieling | gavimobile: correct. Static. |
| 14:58.22 | gavimobile | igcewieling: $$$$ |
| 14:58.35 | gavimobile | cha-ching |
| 14:59.09 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:59.10 | igcewieling | gavimobile: They are all on our circuits backhauled to our NOC with QoS and our IPs |
| 15:00.00 | *** join/#asterisk skirge (~skirge@196.15.233.254) |
| 15:00.15 | igcewieling | We have a nickname for BYOB (bring your own broadband) customers: "soon to be former customer" |
| 15:01.06 | igcewieling | customer: *whine* "our call quality is terrible". us: You are not on our internet service, there is nothing we can do. |
| 15:08.43 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 15:10.01 | zFleshMissile | newtonr: thanks for the info, gonna probably replace the hardware for now and see if that changes anything |
| 15:13.16 | *** join/#asterisk dr0ck (~dr0ck@c-75-70-61-20.hsd1.co.comcast.net) |
| 15:16.32 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:19.36 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 15:34.40 | *** join/#asterisk janelleb (~ubunifu@unaffiliated/janelleb) |
| 15:35.39 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:35.39 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:37.29 | janelleb | hey all, I've been using gnokii and a Nokia Phone for sending alerts/pages from a network monitor. Now I'd like to add some dialplan stuff. My question is can Asterisk replace gnokii? are there drivers/modules for using a Nokia Phone with asterisk? |
| 15:38.31 | *** join/#asterisk ryan_turner (Ryan@2600:3c02::f03c:91ff:fe70:c6b0) |
| 15:39.17 | ryan_turner | Hi, I'm curious if I have a Linode VPS can I rent a SIP trunk from some other provider and run my own server for VOIP, just trying to get my old Polycom IP321 up and running. |
| 15:40.15 | ryan_turner | I need some sort of cheap voip phone and I already have a Polycom IP321 from when a previous business I was with bought RingCentral. I dont know the beginning of VOIP so I don't know what infrastructure I need to put in place to get this back to life. |
| 15:40.33 | [TK]D-Fender | ryan_turner: Sure |
| 15:40.52 | ryan_turner | So the general idea I have is sane? |
| 15:41.58 | ryan_turner | Hell could I just setup the SIP trunk directly with the IP321? |
| 15:42.06 | ryan_turner | would I even need to install asterisk? |
| 15:42.09 | [TK]D-Fender | ryan_turner: Using a Linode system for Asterisk? |
| 15:42.38 | ryan_turner | Yes, and the very basic concept that I get phone service from renting a sip trunk and that an IP321 can be configured to connect to Asterisk. |
| 15:42.40 | [TK]D-Fender | [11:41]ryan_turnerwould I even need to install asterisk? <- I;m not sure what you're asking if you AREN'T using Asterisk here.... |
| 15:42.59 | Tuju | it works now. |
| 15:43.06 | *** part/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
| 15:43.31 | [TK]D-Fender | [08:50]Tuju"Looking for tuju-dmz-trunk in default (domain 195.50.204.214)" - how does asterisk map incoming packets to domain? |
| 15:43.45 | [TK]D-Fender | 3 hours after a simple dialplan target notification.... |
| 15:44.09 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 15:44.15 | ryan_turner | [TK]D-Fender, right now my goal is to get this physical phone working where I can dial out and receive calls in. |
| 15:44.49 | [TK]D-Fender | ryan_turner: ok.... |
| 15:45.11 | ryan_turner | I guess Im asking if all I want to do is have the physical phone be able to do that, do I need to even run Asterisk or can I get by with throwing sip trunk info directly on the phone itself |
| 15:45.22 | boch | anyone knows how can i record a video call ? |
| 15:45.56 | [TK]D-Fender | ryan_turner: You can probably use the phone direct if you wanted. |
| 15:46.11 | [TK]D-Fender | ryan_turner: SIP is SIP.... and * is just a B2BUA |
| 15:46.12 | ryan_turner | Ok, I may do that as a very very very basic test. |
| 15:47.39 | ryan_turner | So if I rent a "sip trunk" that should be what's associated with a phone number, right? I mean what Im paying for is literally an interface and some basic server that runs and lets me interact with that via SIP |
| 15:51.16 | *** join/#asterisk russum (~russum@94.139.130.107) |
| 15:52.15 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 15:54.44 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:54.45 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:55.45 | [TK]D-Fender | [11:47]ryan_turnerSo if I rent a "sip trunk" that should be what's associated with a phone number, right? I mean what Im paying for is literally an interface and some basic server that runs and lets me interact with that via SIP <- huh? |
| 15:55.58 | [TK]D-Fender | ryan_turner: You pay SIP PSTN service |
| 15:56.28 | [TK]D-Fender | A rovider who will rent you a DID (phone number) and send you the call via SIP. And to terminate calls to the PSTN via SIP |
| 15:56.37 | [TK]D-Fender | So far "server" doess not exist.... |
| 15:57.07 | ryan_turner | "send you the call via SIP." |
| 15:57.34 | ryan_turner | There has to be some service running that manages that, the encoding, authentication, etc |
| 15:58.00 | [TK]D-Fender | your PROVIDER is that service |
| 15:58.13 | ryan_turner | yes yes, |
| 15:58.20 | [TK]D-Fender | Don't use the term "server" here unless you mean your own. |
| 15:58.36 | [TK]D-Fender | For all we know they are running an appliance soft-switch |
| 15:58.42 | [TK]D-Fender | Does that "count"? |
| 15:58.52 | [TK]D-Fender | Either way... they provide service. |
| 15:59.22 | ryan_turner | Sorry, I dont know voip terminology or concepts. |
| 16:03.31 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 16:03.48 | *** join/#asterisk ibercom (4de568b5@gateway/web/freenode/ip.77.229.104.181) |
| 16:05.15 | *** part/#asterisk russum (~russum@94.139.130.107) |
| 16:05.55 | ibercom | Hello, how can i enable hardware DTMF detection ? |
| 16:07.02 | ibercom | Is it better than software detection ? |
| 16:07.46 | *** join/#asterisk fischli (~fischli@static-31-25-152-181.ewacom.ropa.net) |
| 16:08.09 | *** join/#asterisk dr0ck (~dr0ck@c-75-70-61-20.hsd1.co.comcast.net) |
| 16:08.21 | igcewieling | ibercom: without context, your question has no meaning |
| 16:09.26 | ibercom | Hello, how can i enable hardware DTMF detection ? |
| 16:09.46 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-zabmbzifjrnpyxzb) |
| 16:11.13 | igcewieling | ibercom: first log into the device gui, then pick the page which might have DTMF settings. |
| 16:11.39 | igcewieling | Now, if you give us some CONTEXT we might be able to come up with a better answer. Otherwise stop wasting our time. |
| 16:11.42 | ibercom | igcewieling: With digium card. |
| 16:11.46 | WIMPy | ibercom: What hardware? |
| 16:11.52 | igcewieling | ibercom: there we go! now some context! |
| 16:12.27 | WIMPy | ibercom: By passing a module parameter. See modinfo for the driver you use. |
| 16:13.20 | igcewieling | I don't see any options for hardware DTMF on Digium cards |
| 16:13.32 | igcewieling | Ah, yes, the module itself. |
| 16:14.35 | ibercom | <PROTECTED> |
| 16:15.37 | ibercom | This message: wct4xxp 0000:12:02.0: VPM450: hardware DTMF disabled. |
| 16:15.49 | igcewieling | ibercom: you now now how to enable it |
| 16:16.01 | igcewieling | well, you now know how to get the information on how to enable it. |
| 16:18.36 | sweeper | hey, anyone successfully using jssip w/ asterisk? |
| 16:18.47 | sweeper | I've got a call going through, but no audio |
| 16:18.57 | ibercom | WIMPy: I see "parm: vpmdtmfsupport:int". Need I restart asterisk ? |
| 16:19.52 | dr0ck | you do need to stop asterisk to reload the module with the parameter set |
| 16:20.23 | ibercom | Ok, I try it. |
| 16:20.28 | igcewieling | sweeper: try turning off video |
| 16:21.12 | igcewieling | ibercom: since you are changing modules, you may need to stop asterisk (or at least chan_dahdi) then restart dahdi, then start asterisk or load chan_dahdi |
| 16:21.26 | igcewieling | dr0ck: he is changing kernel module options, not chan_dahdi.conf options |
| 16:22.17 | dr0ck | i know |
| 16:24.05 | ibercom | I'm seening the document "Changing-line-mode-to-T1-or-E1" in kb.digium.com. It is related. |
| 16:24.52 | dr0ck | in that its a kernel module parameter change, its related I guess |
| 16:25.30 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 16:29.34 | sweeper | igcewieling: it's already off |
| 16:30.04 | sweeper | gonna try attaching to the rtc event handlers and see wat's hat... |
| 16:32.34 | *** join/#asterisk Synx|hm (~Synx@unaffiliated/synx-hm/x-1623004) |
| 16:33.35 | Synx|hm | TLS question, if i were to enable TLS on a SIP peer and configure that peer to use a cert/privatekey that asterisk did not have in its repository what would the expected behavior be? (what i am seeing is TLS communication but i was not expecting it to work) |
| 16:44.25 | Katty | peeks in |
| 16:56.52 | [TK]D-Fender | pokes out |
| 16:59.19 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
| 17:01.03 | igcewieling | put that thing away! |
| 17:01.16 | Katty | >.< |
| 17:03.24 | drmessano | Was that an emoticon or an implication of small size? |
| 17:03.47 | drmessano | Maybe something more like ---> . <--- |
| 17:05.33 | Katty | drmessano: no it's that face from southpark |
| 17:05.36 | Katty | also hi! |
| 17:05.38 | Katty | hugs drmessano |
| 17:06.14 | Katty | the . being a nose. |
| 17:06.34 | drmessano | Hi :) |
| 17:07.26 | ryan_turner | Alright, so I've got my polycom IP 321 reset to default and I have asterisk + freepbx installed |
| 17:07.52 | drmessano | I just ate a Tootsie Roll |
| 17:08.13 | ryan_turner | I added an extension in freepbx, so now I should be able to configure the ip 321 with the details for my asterisk install using the appropriate config right? |
| 17:08.34 | ryan_turner | If I could get the damn phone to boot "updating initial configuration"... |
| 17:08.38 | drmessano | I also wrote an emo poem |
| 17:09.33 | ryan_turner | Apparently it found some config, "running ...45-12360-001.sip.ld" |
| 17:09.50 | ryan_turner | damn, and now the display just shows me the IP address, mac, and firmware version... |
| 17:10.02 | drmessano | It's booting |
| 17:10.14 | ryan_turner | Oh ok, so just wait longer. |
| 17:10.28 | drmessano | Yes, do you do this when you boot your desktop too? |
| 17:10.29 | ryan_turner | (Im a total noob and hope to be able to eventually get to configure this via the web gui) |
| 17:10.50 | ryan_turner | If I have no idea that it is actually booting and its stuck at grub> or something like that, yeah. |
| 17:12.14 | igcewieling | We are not here to teach you VoIP, networking, SIP, firewalls, protocols, and telephone setup. |
| 17:12.33 | ryan_turner | igcewieling, Oh ok sorry I |
| 17:13.10 | ryan_turner | didnt really expect that but I was hoping for someone willing to help with just the VoIP broad concepts. |
| 17:13.41 | igcewieling | There are books for that |
| 17:13.49 | ryan_turner | Ok igcewieling, thanks. |
| 17:13.54 | *** part/#asterisk ryan_turner (Ryan@2600:3c02::f03c:91ff:fe70:c6b0) |
| 17:18.28 | drmessano | I used to think that getting someone to the "Hello, World" moment was enough to seed their pursuit of more knowledge.. but in the end, all it does is create another help vampire |
| 17:18.32 | drmessano | Sad, but true |
| 17:19.16 | Katty | i vant to ask you questions. ah ha ha ha. |
| 17:19.31 | Katty | (i hope you read that in The Count's voice, from sesame street.) |
| 17:19.37 | drmessano | lol |
| 17:20.06 | drmessano | I love the example on the Help Vampire Spotters Guide page, but it's not as accurate as it should be |
| 17:20.26 | drmessano | "How do I build a forum?" should be "How do I a forum?" |
| 17:20.48 | drmessano | Because "build" is a verb, which implies some concept of work |
| 17:21.18 | [TK]D-Fender | HOW I CAN VOIP, PLAESE?!?!??! |
| 17:22.22 | drmessano | I am giving an Asterisk presentation at a LUG in a couple weeks and I plan to present a slide on Help Vampires |
| 17:22.47 | drmessano | "Be empowered. This shit is complex and sometimes confusing. Read, Google, Read some more" |
| 17:22.53 | Katty | could you give me a short and sweet description of Help Vampires |
| 17:22.53 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 17:23.05 | drmessano | Katty, http://slash7.com/2006/12/22/vampires/ |
| 17:23.25 | Katty | reads |
| 17:23.26 | igcewieling | sometimes? |
| 17:24.08 | Katty | drmessano: ah right. gotcha |
| 17:24.45 | Katty | drmessano: and the article seems to be onto something. i very much Dislike male Help Vampires |
| 17:24.47 | drmessano | igcewieling, this is a FLUFF PIECE on Asterisk, not a therapy session |
| 17:24.55 | Katty | drmessano: unless they're ripped like crazy, from a JR ward novel or something |
| 17:25.05 | Katty | drmessano: in which case you just gag them so they can't talk and then everything is peachy! |
| 17:25.23 | drmessano | igcewieling, therapy would be "Yes, it's confusing and I can't stop sucking the ear of my teddy bear which I pee on in my sleep over dialplan woes" |
| 17:25.57 | Katty | dialplan woes certainly are dreadful at times :< |
| 17:26.12 | Katty | and it's ALWAYS a typo. *scowlyface* |
| 17:26.15 | drmessano | igcewieling, "Teddy, the context EXISTS, which is it invalid? WHY? Why does your ear taste like tacos? WHY?" |
| 17:26.21 | igcewieling | nah! dialplans are some of the easiest things in Asterisk |
| 17:26.43 | igcewieling | the hard part is SIP. |
| 17:26.46 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 17:26.51 | Katty | trades dialplans with igcewieling |
| 17:27.03 | drmessano | SIP is easy.. you just open up 5060 and do stuff |
| 17:27.17 | Katty | yeah, like sacrafice teddy and a goat |
| 17:27.19 | igcewieling | Katty: dialplan show: -= 46 extensions (439 priorities) in 18 contexts. =- |
| 17:27.35 | igcewieling | and we route over 10,000 telephone numbers on the system |
| 17:27.36 | Katty | you know what's hard? REALLY HARD?! |
| 17:27.42 | Katty | deciding where to go to dinner on a saturday night |
| 17:27.42 | drmessano | Me? |
| 17:27.44 | igcewieling | Katty: diamond? |
| 17:27.44 | drmessano | Wait no |
| 17:27.53 | Katty | covers eyes |
| 17:28.07 | drmessano | X-No-Archive |
| 17:28.07 | drmessano | TYVM |
| 17:28.16 | Katty | igcewieling: are diamonds still the hardest thing? |
| 17:28.23 | Katty | other than drmessano here. |
| 17:28.58 | Katty | google says we've made something harder than diamonds! |
| 17:28.59 | drmessano | No I have OOPS UP stuck in my head |
| 17:29.09 | drmessano | That's one of the funniest, coolest songs ever |
| 17:29.19 | drmessano | "She was soft as bubble bath, I was hard as chinese math" |
| 17:29.32 | Katty | that's very clever. |
| 17:29.38 | Katty | but math is math. |
| 17:29.42 | Katty | but still clever. |
| 17:30.23 | drmessano | I am awesome at Maths |
| 17:30.51 | Katty | all the maths? |
| 17:31.30 | drmessano | I bathe in Math Salts I am so with the Maths |
| 17:31.45 | Katty | sounds like Math Osmosis |
| 17:31.48 | Katty | err Bathmosis |
| 17:31.58 | drmessano | Osmosis does work |
| 17:31.58 | Katty | MATHMOSIS |
| 17:32.40 | drmessano | I slept through my 9th grade science class in 12th grade, with my face in the book, and learned it all |
| 17:32.49 | Katty | http://mathmosis.com/ <- and i thought i was clever :< |
| 17:32.57 | Katty | someone was clearly clever before me. |
| 17:33.50 | drmessano | Katty, great, now those puppets will haunt me in my nightmares |
| 17:34.43 | drmessano | I am filling in their contact form, with your email address, telling them their puppets are scary |
| 17:35.06 | Katty | i have log files. |
| 17:35.15 | Katty | RETHINK YOUR STRATEGY SIR |
| 17:35.16 | drmessano | Email Address (e.g. user@domain.com) <-- SRSLY? |
| 17:35.53 | drmessano | What other form of email address is there? |
| 17:36.19 | drmessano | steve#bandwagon.tirewheel.barrel$HAHA ? |
| 17:36.20 | Katty | distributionlist@domain.com |
| 17:36.44 | drmessano | katty@hermail.com? |
| 17:37.00 | Katty | @yourmom.com |
| 17:37.20 | Katty | infobot: your mom |
| 17:37.20 | infobot | your mom is probably spelt your mum |
| 17:37.31 | drmessano | Hey now, lets get off of moms because.. Oh, I can't even go there |
| 17:37.44 | Katty | drmessano: it's Your Mum |
| 17:37.49 | Katty | drmessano: didn't you hear infobot? |
| 17:38.08 | drmessano | Lets get off of Your Mum because it is time for tea and biscuits? |
| 17:38.33 | Katty | it's because YOU said it that it takes on a whole new meaning |
| 17:39.12 | drmessano | That's going way off topic.. Maybe something like: |
| 17:39.28 | Katty | oh right. we're getting off topic again. |
| 17:39.34 | drmessano | Lets get off of Your Mum because she requires assistance with her dialplan logic? |
| 17:40.09 | Katty | you can do better. |
| 17:40.33 | *** join/#asterisk Bladerunner05 (~Bladerunn@2-229-126-99.ip196.fastwebnet.it) |
| 17:40.34 | drmessano | Lets get off of Your Mum because her SIP stack is inadequate to handle the load |
| 17:40.39 | Katty | there is a whole plethora of asterisk verbage you can use. |
| 17:40.46 | Katty | see. i knew you could do better. |
| 17:41.39 | Katty | drmessano: do you have a basement (wildly off topic) |
| 17:41.44 | drmessano | Your Mum has no problem with RTP because she's always on a public IP with no iptables |
| 17:41.48 | drmessano | I do not |
| 17:41.52 | Katty | mkay. |
| 17:41.59 | Bladerunner05 | hello on chan_dahdi.conf I do faxdetect = incoming and in incoming call on the last line I do exten => fax,1,Goto(fax-rx,receive,1) when a call come from fax it answer like a regular call |
| 17:42.25 | Katty | hi there Bladerunner05 |
| 17:42.42 | Bladerunner05 | hi Katty |
| 17:42.48 | Katty | blade runner was a good movie |
| 17:43.54 | igcewieling | "Touch my Asterisk!" |
| 17:44.41 | igcewieling | Bladerunner05: in order for fax detect to work, you need an Answer and a Wait or Background or something like that so Asterisk can listen to the audio |
| 17:45.02 | Katty | and by audio he means the KRRRRRRRRRRRRRssssssshhhhhKKKKKRRRRR |
| 17:45.10 | Katty | BEEEEEEEE |
| 17:45.19 | Katty | durrrrSshhhhhKRKRRR |
| 17:49.13 | Bladerunner05 | igcewieling: my dialplan play audio and wait keypress in the last line I do exten => fax,1,Goto(fax-rx,receive,1) |
| 17:49.34 | *** join/#asterisk Dovid (~Dovid@74.123.202.211) |
| 17:51.53 | [TK]D-Fender | Bladerunner05: show us the call and your dialplan.... |
| 17:53.33 | Bladerunner05 | [TK]D-Fender: this is dialplan |
| 17:53.36 | Bladerunner05 | exten => s,1,Answer |
| 17:53.37 | Bladerunner05 | exten => s,2,GotoIfTime(09:00-14:00|mon-sun|*|*?daytime,s,1) |
| 17:53.38 | Bladerunner05 | exten => s,3,GotoIfTime(15:00-23:59|mon-sun|*|*?daytime,s,1) |
| 17:53.39 | Bladerunner05 | exten => s,4,Goto(fuoriorario,s,1) |
| 17:53.40 | Bladerunner05 | exten => fax,1,Goto(fax-rx,receive,1) |
| 17:53.42 | [TK]D-Fender | Bladerunner05: PASTEBIN!!!!!!!!!!!!!! |
| 17:53.44 | [TK]D-Fender | Bladerunner05: PASTEBIN!!!!!!!!!!!!!! |
| 17:53.49 | [TK]D-Fender | Bladerunner05: do NOT flood that in here |
| 17:53.58 | Bladerunner05 | [TK]D-Fender: ops sorry |
| 17:54.17 | *** join/#asterisk peetaur2 (~peter@x2f0841e.dyn.telefonica.de) |
| 17:54.21 | boch | anyone knows how can i record a video call ? |
| 17:54.25 | [TK]D-Fender | Bladerunner05: And you are GOTO-ing OUT of that context. You aren't even IN there for that "fax" to be detected |
| 17:55.43 | Bladerunner05 | [TK]D-Fender: what have to do ? |
| 17:56.00 | Katty | drmessano: I SEE VAMPIRES |
| 17:56.12 | [TK]D-Fender | Bladerunner05: You have to actually have it in the context you are LISTENING in. |
| 17:56.23 | [TK]D-Fender | Bladerunner05: You aren't in that context for even 1 seconds before leaving it |
| 17:56.46 | Bladerunner05 | that's the context for incoming calls |
| 17:57.30 | [TK]D-Fender | Yes and you call doesn't STAY there long enough for a fax to be DETECTED |
| 17:57.45 | [TK]D-Fender | You leave it in less than 1 second |
| 17:58.36 | Bladerunner05 | [TK]D-Fender: I have to use wait(5) ? |
| 17:58.58 | [TK]D-Fender | Bladerunner05: You have to make up your own mind and realize where you are when you expect to be able to detect it. |
| 18:01.37 | Bladerunner05 | [TK]D-Fender: thanks :) |
| 18:02.46 | igcewieling | Bladerunner05: You will find fax detection doesn't work as well as you'd like. |
| 18:03.14 | igcewieling | Amazing the number of people who dial the fax and wait to hear a tone before pressing Send. those people won't work with faxdetect |
| 18:10.54 | drmessano | igcewieling, or the 2000's |
| 18:25.19 | *** join/#asterisk OverOnTheRock (~irc@162.r1.ray.transact.bm) |
| 18:29.13 | Synx|hm | Anyone know how i can force Asterisk to verify the peer TLS certificates? |
| 18:29.41 | *** join/#asterisk peetaur2 (~peter@x2f0841e.dyn.telefonica.de) |
| 18:30.59 | Synx|hm | should probably say, 'a SIP peers certificates' |
| 18:33.05 | SuperNull | odd question but is there anything that REQUIRES 'rtcachefriends' to be yes for sip ?? |
| 18:33.51 | SuperNull | oh damn. its required for MWI. |
| 18:33.57 | SuperNull | gay as hell. |
| 18:34.17 | SuperNull | so i need it for MWI but it needs to be off.. otherwise credentials done age quickly enough and cause registration failures. hrm |
| 18:37.35 | *** join/#asterisk Dovid (~Dovid@74.123.202.211) |
| 18:37.50 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 18:38.49 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 18:40.53 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.218) |
| 18:42.30 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 18:42.45 | Bladerunner05 | [TK]D-Fender: I do a dialplan with a long wait and only fax but it seems to avoid detection.. I'm using digium free fax maybe italian protocol not allowed ? |
| 18:47.20 | *** join/#asterisk jagster` (~chatzilla@unaffiliated/jagster/x-9084543) |
| 18:50.28 | [TK]D-Fender | This has nothing to do with "free fax" |
| 18:50.32 | [TK]D-Fender | You are DETECING a fax. |
| 18:50.40 | [TK]D-Fender | This has nothing to do with how you ACT once you do |
| 18:50.53 | [TK]D-Fender | And I have no idea what you are now supposedly doing. You aren't showing anything. |
| 18:52.04 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 18:52.04 | *** join/#asterisk darkdrgn2k (~darkdrgn2@69-165-131-20.dsl.teksavvy.com) |
| 18:52.10 | darkdrgn2k | hey all |
| 18:52.18 | darkdrgn2k | any one have experiance with audiocodes MP-202? |
| 19:01.46 | *** join/#asterisk Rumbles (~Rumbles@31.205.54.123) |
| 19:13.46 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:14.29 | sweeper | how can I find out if my asterisk came with srtp? |
| 19:15.02 | *** join/#asterisk vedic (~V@183.82.84.84) |
| 19:15.40 | vedic | I was curious to know if any body has been able to do Fast EAGI? or sending recorded sound file to remote server where agi is running? |
| 19:15.52 | sweeper | n/m google |
| 19:16.51 | transfinite | SuperNull: take your homophobia somewhere else |
| 19:17.35 | transfinite | is there a way to change the default numbering of dahdi channels when I have 2 PCI cards with the same driver? |
| 19:17.46 | vedic | Is this http://agi-audiotx.sourceforge.net/ advisable solution to get recorded sound file on remote server? |
| 19:17.58 | vedic | I am using * 11 |
| 19:17.59 | transfinite | right now they're in PCI slot order, but I'd prefer to reverse them |
| 19:18.08 | sweeper | transfinite: like to #asstricks? :) |
| 19:18.21 | transfinite | :-) |
| 19:19.26 | sweeper | hmm. unsure what my next step is here in debuggingthis webrtc setup |
| 19:20.11 | sweeper | jssip successfully authenticates and dials in, and gets passed into the confbridge, but I see no evidence of media anywhere |
| 19:24.25 | igcewieling | sweeper: so few people use webrtc you might have better luck with google |
| 19:24.47 | sweeper | plausible I suppose :/ |
| 19:25.02 | igcewieling | same with srtp |
| 19:26.37 | drmessano | You could check if the module is loaded |
| 19:26.55 | drmessano | For SRTP, that is |
| 19:28.33 | sweeper | yep, it's loaded |
| 19:29.08 | drmessano | Then it came with it |
| 19:29.13 | drmessano | lol |
| 19:31.03 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 19:31.28 | *** join/#asterisk ryan_turner (Ryan@2600:3c02::f03c:91ff:fe70:c6b0) |
| 19:31.41 | ryan_turner | Yay, I successfully setup asterisk, freepbx, and configured my polycom ip 321. |
| 19:32.24 | ryan_turner | So, now all I should need to do is add a SIP trunk and I should be able to dial in/out |
| 19:33.06 | ryan_turner | Do any of you have recommend sip trunk providers? |
| 19:33.14 | Qwell | ~itsplist-us |
| 19:33.14 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 19:36.25 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 19:39.11 | vedic | Is this http://agi-audiotx.sourceforge.net/ advisable solution to get recorded sound file on remote server? I am using asterisk 11 and looking for way to copy recorded files from asterisk to remote server. I am using fast agi |
| 19:39.42 | vedic | I see that agi-audiotx has not been updated since 2008 |
| 19:40.44 | drmessano | rsync? |
| 19:40.45 | *** join/#asterisk teff (~teff@client-86-29-231-156.pete.adsl.virginmedia.com) |
| 19:44.22 | vedic | drmessano: Is rsync good for almost real time sending of file on remote server on the internet? |
| 19:45.35 | drmessano | rsync is going to send the file in real time to the destination |
| 19:46.47 | drmessano | What you've cited as an example, this audiotx application, it simply sends audio files back and forth when executed, does it not? |
| 19:46.59 | drmessano | PUT SOUNDFILE <soundfile> <size> - Copy an audio file to the Asterisk server |
| 19:46.59 | drmessano | GET SOUNDFILE <soundfile> - Copy an audio file from the Asterisk server |
| 19:47.19 | drmessano | rsync or even CP can do that.. you just need to call them with System() |
| 19:48.03 | transfinite | ah, found the auto_assign_spans parameter for the dahdi module |
| 19:48.17 | vedic | drmessano: yea, but I am not sure if that is suitable for asterisk 11. I am using Fast AGI and when I issue record command, audio is recorded & stored at asterisk server. I need that audio at remote server almost immediately. |
| 19:49.21 | vedic | drmessano: At asterisk server, how can I run cp (or scp) command when asterisk recording is done? |
| 19:50.10 | vedic | drmessano: I was just thinking what if Curl is using for pushing the file from asterisk to remote server from the dial plan? |
| 19:50.21 | drmessano | I wasn't suggesting using this application.. I was pointing out that it appears to simply invoke a file copy operation. Why can't you do that with rsync or cp? |
| 19:51.07 | drmessano | You can run whatever you want when a recording is done. It's YOUR dialplan |
| 19:51.25 | vedic | drmessano: Because asterisk is running un attended. How would I know when to issue cp ? For rsync, I think it is possible to configure to do it automated. |
| 19:51.45 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.218) |
| 19:52.14 | vedic | drmessano: Asterisk already provides curl function build in. Was think how best to use it to push a recorded file from asterisk dialplan to remote server |
| 19:52.37 | drmessano | Um ok |
| 19:53.35 | drmessano | You do realize you can invoke almost any command using System(), right? |
| 19:54.01 | jagster` | vedic: as a one time thing? |
| 19:54.03 | drmessano | You don't need a 5 yr old AGI or curl to call rsync or cp or whatever else |
| 19:54.04 | jagster` | just use scp |
| 19:54.21 | drmessano | No, he wants to call it when recordings are finished |
| 19:54.33 | drmessano | [15:49:21] <vedic> drmessano: At asterisk server, how can I run cp (or scp) command when asterisk recording is done? |
| 19:54.47 | jagster` | sounds like a backwards solution |
| 19:54.57 | jagster` | i mean you COULD rsync everytime a vm is finished |
| 19:55.24 | *** join/#asterisk btse (~btse@c83-253-253-142.bredband.comhem.se) |
| 19:55.33 | drmessano | He wanted something similar to the aforementioned AGI, which simply invokes a copy operation. |
| 19:55.42 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 19:55.51 | drmessano | You can use System() with any standard copy operation to do the same |
| 19:56.00 | drmessano | Whether or not its sane, ehhhh |
| 19:57.03 | drmessano | You could also set up a VPN tunnel and mount the remote filesystem, recording directly to it |
| 19:57.31 | drmessano | As far as cURL.. well, it's a verb alright |
| 19:58.29 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 20:01.19 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
| 20:03.42 | *** join/#asterisk btse (~btse@c83-253-253-142.bredband.comhem.se) |
| 20:04.19 | vedic | drmessano: One more technique that comes to my mind is using ReadFile. |
| 20:05.19 | vedic | drmessano, jagster:Using ReadFile, reading the audio file content into a variable and then accessing that variable from Fast AGI " get variable" |
| 20:05.31 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 20:07.22 | vedic | drmessano, jagster: Any issue you see with readfile approach? |
| 20:07.43 | vedic | My audio files are not big. Hardly 20 sec audio max. |
| 20:08.54 | *** join/#asterisk jetlag (~jetlag@pool-71-168-253-55.cmdnnj.east.verizon.net) |
| 20:11.05 | *** join/#asterisk btse (~btse@c83-253-253-142.bredband.comhem.se) |
| 20:13.13 | igcewieling | vedic: why are you making it so complicated? What you want to do is trivial |
| 20:16.51 | vedic | igcewieling: Using rsync is an external process as I see. Not part of asterisk by default. In Fast AGI, I will have to check the existance of the file but using readfile which is part of asterisk already, I won't have to do any thing extra but just read the variable from Fast AGI and write the content of the file to get the recorded file on remote server |
| 20:17.19 | igcewieling | I doubt asterisk variables support binary data, but you are welcome to try |
| 20:18.03 | igcewieling | vedic: The ONLY thing which matters when running fastagi is that your script does not have access to the pbx filesystem. |
| 20:18.19 | vedic | igcewieling: yea I am also not sure. Will have to try it out. In documentation, it just says reading a file but not mentioning anything about binary or ascii etc |
| 20:18.43 | vedic | igcewieling: yea, thats why the whole issue is |
| 20:19.14 | vedic | igcewieling: I need to get the audio on remote server where fast agi is hosted |
| 20:19.27 | drmessano | You're assuming that something built into Asterisk is going to be faster at copying a file because ??? |
| 20:20.01 | vedic | drmessano: Not at all |
| 20:20.07 | igcewieling | heh, got a call with a reported RDNIS of '788810796285111212308230' |
| 20:20.13 | vedic | drmessano: I don't want to setup any thing external |
| 20:20.22 | drmessano | You're not using some function of Asterisk to copy in a special, specific way.. you're basically telling Asterisk, which is not an rsync or cp replacement, to copy files. Why wouldn't you make a call to something designed for the task? |
| 20:20.45 | igcewieling | vedic: sometimes we don't always get what we want. |
| 20:21.35 | vedic | drmessano, igcewieling: Because * is already providing functions and command that I think good to explore to max before using external tools and maintaining them |
| 20:22.09 | igcewieling | asterisk was never designed to transfer files using the AGI protocol. |
| 20:22.18 | vedic | If that works, then remote file access from fast agi would be like a charm |
| 20:22.46 | vedic | igcewieling: right, so I am reading that file into a variable and then accessing that variable. Thats the idea |
| 20:22.50 | igcewieling | as drmessano suggested you could give your script local access to the filesystem using NFS or similar |
| 20:23.18 | igcewieling | vedic: you may have to base64encode the data before putting into a variable. |
| 20:23.26 | drmessano | So you want to use the claw of the hammer to put in a screw, because it's "built in", even though it probably sucks for the task |
| 20:23.34 | igcewieling | I suspect you are going into some unknown limitation on max variable size. |
| 20:23.41 | vedic | igcewieling: Lets try first what * is offering. If that doesn't work then NFS or rsync are good options |
| 20:24.11 | drmessano | System("rsync fromhere tothere") |
| 20:24.26 | igcewieling | vedic: Asterisk never offered to manage binary data in the dialplan, so you are NOT using what is "included" |
| 20:24.35 | vedic | igcewieling: possibly. But if that works, it would be very nice to have a feature in next version (sub versions) of asterisk. That will take a lot of pain for people using Fast AGI |
| 20:24.46 | vedic | drmessano: ok |
| 20:24.57 | igcewieling | vedic: most fastagi people are much less crazy then you are. |
| 20:25.00 | drmessano | Asterisk is offering a built-on function as a convenience at a performance hit over tools designed for the task |
| 20:25.19 | igcewieling | I could write a json parser in dialplan, but that doesn't mean I should. |
| 20:25.57 | vedic | drmessano: Well, I am not suggesting writing anything additional but exploring the commands already existing |
| 20:26.20 | drmessano | Asterisk does implement a nifty cURL function. Would I use it for backing up my p0rn collection when a call comes in from the FBI? No, I would call something external |
| 20:26.30 | igcewieling | file ! |
| 20:26.36 | vedic | drmessano: Why shouldn't somebody try (really haven't somebody tried it already ?) to read a binary file into a variable? |
| 20:26.39 | igcewieling | file: ! |
| 20:26.42 | file | hi |
| 20:26.48 | igcewieling | file: if you have a moment |
| 20:27.11 | file | go for it |
| 20:27.24 | vedic | igcewieling: readfile is available with asterisk. |
| 20:27.26 | igcewieling | file: vedic wants to use readfile dialplan function to read a 20 second audio clip into a dialplan variable, then use a FastAGI to get that variable. what do you think? |
| 20:27.35 | file | dear god no |
| 20:27.39 | drmessano | lol |
| 20:27.44 | file | that won't work |
| 20:27.55 | igcewieling | file: that was my reaction, but wanted to check. |
| 20:27.56 | vedic | file: Would be much better if you could say reasons |
| 20:28.01 | drmessano | haha |
| 20:28.20 | igcewieling | vedic: file is a core asterisk developer, he does not need to state reasons. 8-) |
| 20:28.31 | vedic | drmessano: Not sure what is funny here instead of being hacking around |
| 20:29.15 | igcewieling | file: is your batsignal tied to the string "file" or "file:" ? |
| 20:29.54 | vedic | igcewieling: glad to know. Nice to have conversion with him. But still nowing reasons make you more knowlegeble |
| 20:30.00 | file | it's tied to file |
| 20:30.37 | igcewieling | vedic: the reasons are likely related to basic design of stuff and really too complicated to explain. You are always to read the Asterisk source code. |
| 20:30.37 | file | fine, it MAY work but it wouldn't be terribly performant and the larger the file... the more memory you consume |
| 20:30.47 | file | that stuff is designed to be used to store strings |
| 20:31.03 | vedic | file: I see |
| 20:31.14 | igcewieling | Kind of like asking "Why does NAT cause a problem with SIP?" The answer would take 20 mins. |
| 20:31.29 | Qwell | file: the first time it hits a \0, it's dead. |
| 20:31.35 | file | yeah |
| 20:31.40 | Qwell | So, no binary files. |
| 20:31.57 | vedic | file: What is the safe limit (if any) to file size for readfile |
| 20:32.07 | igcewieling | no chance to do something like BASE64(READFILE(.... |
| 20:32.13 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 20:32.15 | Qwell | igcewieling: no chance at all |
| 20:32.16 | igcewieling | ?? |
| 20:32.23 | file | no idea |
| 20:32.26 | Qwell | BASE64 takes a char * |
| 20:32.28 | vedic | file: ok |
| 20:33.02 | drmessano | vedic, Try to imagine all life as you know it stopping instantaneously and every molecule in your body exploding at the speed of light. That's why we don't do it. |
| 20:33.30 | igcewieling | drmessano: ghostbusters? |
| 20:33.35 | drmessano | Also why we don't cross the streams |
| 20:33.37 | drmessano | Yep lol |
| 20:34.26 | vedic | drmessano: I don't understand your all discourging statements instead of having a reasonably logical, technical answers |
| 20:34.26 | drmessano | Is still say the best way to make Asterisk copy a file somewhere is not to have Asterisk do it at all, but hand it off to rsync or something |
| 20:34.38 | file | use the right tool for the job |
| 20:34.45 | drmessano | [16:34:38] <file> use the right tool for the job <--- |
| 20:34.47 | vedic | drmessano: agree |
| 20:34.50 | drmessano | Been saying that for an hour |
| 20:34.58 | vedic | drmessano: really? |
| 20:34.59 | file | and never XML. |
| 20:35.13 | file | (I couldn't pass up dissing XML) |
| 20:35.34 | mjordan | if XML doesn't solve it, you're just not using enough |
| 20:36.02 | drmessano | haha |
| 20:36.43 | Qwell | file: the answer is obviously json |
| 20:36.52 | vedic | igcewieling: How to issue System from Fast AGI? |
| 20:37.08 | igcewieling | vedic: agi exec |
| 20:37.18 | vedic | igcewieling: ok |
| 20:37.36 | igcewieling | Qwell and file: I'd like a JSON encode and decode dialplan functions |
| 20:37.42 | igcewieling | s/like/love/ |
| 20:37.50 | drmessano | In Asterisk 12, no doubt |
| 20:37.57 | Qwell | igcewieling: what would the use-case be? |
| 20:37.57 | mjordan | vedic: they're correct. Let's assume for a second that there was a way for Asterisk to push the entire recording over FastAGI. Since FastAGI doesn't support the concept of streaming, that means we have to buffer the whole thing in memory (bad). What's more, FastAGI is not an asynchronous event driven, protocol, which means you'd have to query for the thing and then block (bad). |
| 20:38.21 | igcewieling | Qwell: passing arrays between from s |
| 20:38.25 | mjordan | vedic: We'd have to encode the whole recording, passing it over TCP. You'd have to reconstruct it (bad). |
| 20:38.27 | Qwell | igcewieling: HASH |
| 20:38.30 | igcewieling | Qwell: passing arrays between PHP AGIs |
| 20:38.30 | mjordan | vedic: re-encoding it (bad) |
| 20:38.39 | mjordan | vedic: then restore the file on your local system. |
| 20:38.47 | mjordan | All of this is much slower than the solutions proposed. |
| 20:38.56 | mjordan | And would be a technical rube-goldberg device |
| 20:39.13 | vedic | mjordan: hmm... I see |
| 20:39.54 | mjordan | FastAGI's purpose is to control the actions taken on Asterisk channel externally. It isn't to manipulate the file system. |
| 20:39.56 | drmessano | I would rather invoke a robust file copy solution for transferring a file to a remote server than hope some built-in function in any app has my best interests in mind |
| 20:40.10 | igcewieling | Qwell: can you think of any reason for Set or Asterisk to have an issue if I already have the data in JSON format? |
| 20:40.33 | Qwell | igcewieling: assuming it's just strings/ints, no |
| 20:40.43 | Qwell | objects will obviously fail |
| 20:40.50 | igcewieling | objects are evil |
| 20:40.59 | drmessano | It's like using Windows Explorer to copy a file because it happens to do that, and assuming its the best solution for copying a file. In reality, Windows Explorer is slower than SneakerNET 2.0 |
| 20:41.09 | file | we're over it, let's move on |
| 20:41.14 | drmessano | I'm not |
| 20:41.21 | igcewieling | drmessano: that was a really horrid example. 8-| |
| 20:41.24 | drmessano | hahaha |
| 20:42.17 | igcewieling | Qwell: also a json encode for the dialplan to pass "arrays" to an agi script |
| 20:42.59 | sweeper | custom_prepare: SQL Prepare failed <-- any idea what could have caused this all of the suddent? |
| 20:43.44 | igcewieling | as it is now I do stuff like this and then re-parse it all back into an array: CELGenUserEvent(SM_HANGUP,account_sid='${SM_ACCOUNT_ID}' route_sid='${SM_ROUTE_ID}' source='dest' status='${SM_DIALSTATUS}' cause='${SM_HANGUPCAUSE}' answeredtime='${ANSWEREDTIME}'); |
| 20:44.36 | igcewieling | Then later a script reads the events and has to parse that back into an array |
| 20:45.49 | *** part/#asterisk vedic (~V@183.82.84.84) |
| 20:46.55 | sweeper | I am gonna stab something |
| 20:47.55 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 20:49.29 | Vann | Hello |
| 20:49.49 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 20:51.43 | Vann | Don't suppose anyone is around to help me with an odd SIP issue? Basically my SIP channels don't terminate after a hangup... :( |
| 20:51.59 | Vann | Using the latest Asterisk 11.4 from source |
| 20:52.44 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.199) |
| 20:54.42 | sweeper | man asterisk/websocket seems pretty flaky. wonder if it's just my internet that's slowing ICE down or what |
| 20:54.51 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 20:58.25 | sweeper | ok this is weird |
| 20:58.44 | sweeper | asterisk will just 'hang up' and not do anything, no even respond to commands in the cli |
| 21:06.20 | Vann | =/ |
| 21:06.21 | sweeper | did it again. wtf |
| 21:06.28 | sweeper | how do I figure out what's doing this? |
| 21:07.21 | Vann | You try looking in the logs? |
| 21:10.42 | igcewieling | sweeper: what network chipset do you have? (use lspci) |
| 21:12.38 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 21:13.13 | davlefou | Hi, since several day, if a use alaw or ulaw, the sound is very bad. I don't undestand why. I don't thing to have done something. |
| 21:14.05 | davlefou | the sound i good beetweem asterisk and me but not with me and asterisk. |
| 21:14.07 | sweeper | igcewieling: AR9485 |
| 21:14.14 | sweeper | Vann: logs are inconclusive |
| 21:14.25 | sweeper | also, some weirdass stuff is happening with realtime |
| 21:14.47 | sweeper | I saw some missiing fields complained in the logs |
| 21:15.05 | sweeper | now I get " __set_address_from_contact: Invalid contact uri yes (missing sip: or sips:), attempting to use anyway" |
| 21:15.22 | sweeper | now I *KNOW* that the uri I'm sending is not 'yes' |
| 21:15.44 | sweeper | so it's almost like it's trying to work with out-of-sequence sql returns? |
| 21:16.56 | igcewieling | do you have peers in realtime which do not register? (i.e. static ips) |
| 21:18.01 | sweeper | no |
| 21:18.19 | sweeper | ok found the "yes" thing, I had made it a default |
| 21:18.59 | sweeper | used the scroll buffer and didn't notice |
| 21:19.02 | igcewieling | don't mess with the schema included n the asterisk source |
| 21:19.15 | sweeper | what schema? >.> |
| 21:20.16 | *** join/#asterisk OverOnTheRock (~irc@162.r1.ray.transact.bm) |
| 21:20.49 | igcewieling | on my system /usr/src/igc/build/asterisk-11.5.0-rc1/contrib/realtime |
| 21:23.23 | sweeper | hmm |
| 21:23.29 | sweeper | that doesn't even have everything I need |
| 21:23.33 | sweeper | but I'll add to it I guess |
| 21:24.13 | igcewieling | what is it missing? |
| 21:26.16 | sweeper | avpf, encryption, iceenable,hassip, hasiax |
| 21:27.54 | sweeper | well I'm looking at the version from trunk |
| 21:30.25 | sweeper | aaand asterisk hung again :/ |
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| 21:36.56 | sweeper | gr8 |
| 21:37.06 | sweeper | now it's just broken all over the place |
| 21:40.02 | sweeper | definitely realtime related |
| 21:40.12 | sweeper | switched to static and it all played nice |
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| 23:04.13 | Get_The_Fish | Hello all, I've got a 1.8.x PBX installed with low call volume in a traditional PBX capacity, and I'm considering upgrading to 11.x. Anything that I should know from those in the know? |
| 23:05.06 | Weezey | just a few libs to install |
| 23:05.06 | ChannelZ | read UPGRADE.txt for any changes that might affect you' |
| 23:05.17 | Weezey | and nat=yes is gone. |
| 23:05.44 | Get_The_Fish | Gotcha. Got that, how about stability issues, anything like that? |
| 23:05.48 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 23:05.56 | ChannelZ | works fine for me |
| 23:06.42 | Get_The_Fish | Great, thats what I needed to know. I've been dealing with Asterisk since 1.2 and some versions were more mature than others on release. |
| 23:06.50 | Weezey | been running 11 in tests since February, Bumped everything up to 11 last week, working great |
| 23:07.09 | Get_The_Fish | Cool, great to hear. Thanks Weezy, ChannelZ |
| 23:07.27 | ChannelZ | Good luck |
| 23:07.35 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 23:07.52 | ChannelZ | If you're using any binary modules (g729, FFA, etc) make sure you grab new versions of those |
| 23:08.11 | Get_The_Fish | Ok, yeah I'm using g729. Thanks for that. |
| 23:08.37 | Get_The_Fish | Lots of new goodies to play with too! |
| 23:09.42 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 23:10.42 | ChannelZ | ConfBridge is pretty full featured now from 1.8 (from what I remember) |
| 23:11.05 | Get_The_Fish | Yeah quite a few new useful options. |
| 23:12.23 | ChannelZ | tries to finish writing his primer |
| 23:28.28 | apb1963 | has google voice finally bitten the ghost? |
| 23:28.51 | apb1963 | was working up until today.. incoming still works, but outgoing... not. |
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| 23:53.04 | Katty | evening |
| 23:55.01 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.238) |
| 23:55.45 | newtonr | Katty: howdy |
| 23:56.01 | Katty | waves to newtonr |
| 23:56.31 | WIMPy | Good morning. |
| 23:57.22 | newtonr | WIMPy: good afternoon! |
| 23:58.24 | Katty | WIMPy: o. |
| 23:58.26 | Katty | WIMPy: o/ |