| 00:00.08 | jeev | :< |
| 00:02.39 | jeev | this blows chunks |
| 00:03.09 | jeev | jsmith, if it helps, i can't manage to log DTMF keys |
| 00:05.09 | jsmith-dinner | jeev: Did you turn on DTMF logging in logger.conf, then type "logger reload"? |
| 00:05.29 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 00:06.38 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 00:06.44 | jeev | yes i have. |
| 00:06.56 | jeev | debug => debug |
| 00:06.56 | jeev | console => notice,warning,error |
| 00:06.56 | jeev | ;console => notice,warning,error,debug |
| 00:06.56 | jeev | messages => notice,warning,error |
| 00:06.56 | jeev | full => notice,warning,error,debug,verbose,dtmf |
| 00:07.09 | jeev | i've restarted logger and even restarted asterisk gracefully. nothing. |
| 00:07.55 | Jumpie | jeev is it a matter of, you cant send the dtmf tones across ? |
| 00:09.40 | jeev | i can send it.. i have to tap it RWEALLy fast |
| 00:09.43 | jeev | it stil doesn't log |
| 00:09.54 | jsmith-dinner | And you've got RTP debug on? |
| 00:11.30 | jeev | uh hmm |
| 00:11.41 | jeev | now it is on |
| 00:12.50 | Jumpie | man i need a telco simulator/emulator |
| 00:13.13 | Jumpie | if i have a router with a wic-1t connected to a digium t1 card, can i emulate that? |
| 00:13.13 | jsmith-dinner | Jumpie: Build another Asterisk box ;-) |
| 00:13.28 | Jumpie | i need to practice troubleshooting and configuring diff channels, no way im having a t1 at m house lol |
| 00:13.35 | Jumpie | i gotta fe routers layin around... |
| 00:13.54 | jsmith-dinner | Jumpie: That should work then :-) |
| 00:14.40 | jeev | jsmith-dinner, i enabled rtp and the thing went crazy.. still didn't see it |
| 00:15.28 | Jumpie | but can a router pass along did, etc |
| 00:15.29 | Jumpie | :d |
| 00:15.52 | jsmith-dinner | jeev: I don't know what else to tell you then... Is it possible that the RTP is bypassing Asterisk and going directly to your provider? |
| 00:16.30 | Jumpie | heh that'd suck |
| 00:16.54 | jeev | no i dont see how |
| 00:17.00 | jeev | is it possible |
| 00:17.01 | jeev | i mean |
| 00:17.06 | jeev | i literally tap 4 |
| 00:17.07 | jeev | and it works |
| 00:17.11 | jeev | but if i press 4 like i'm dialing a number |
| 00:17.12 | jeev | it doesn't work |
| 00:17.44 | jsmith-dinner | Sounds like your provider doesn't like the duration of the DTMF presses |
| 00:18.57 | jeev | is it possible for me to modify that ? |
| 00:19.57 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:20.42 | jeev | because i'm unable to find any modifiers, since it's asteriskNOW, i dont see how the src would be helpful to me. |
| 00:21.29 | Jumpie | outtolunc, |
| 00:21.44 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-cc8895a56e143e4e) |
| 00:22.49 | *** join/#asterisk ta^3 (n=tacvbo@189.146.172.41) |
| 00:24.37 | *** join/#asterisk JerJer[mobile] (n=jj@m500e36d0.tmodns.net) |
| 00:28.43 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 00:29.20 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@20150239095.user.veloxzone.com.br) |
| 00:31.54 | *** join/#asterisk Defraz (i=t0tal@72.24.26.7) |
| 00:33.00 | korihor | jeev: your provider use a cisco as5300? |
| 00:33.14 | jeev | no idea man |
| 00:34.44 | jeev | i'm really getting tired of this shit |
| 00:37.19 | korihor | jeev: i had a similar problem, and i changed on rtp.c file rtp->send_duration = 160 to rtp->send_duration = 0 and that worked :) |
| 00:37.57 | jeev | my situation is different, i use asterisk now. |
| 00:37.59 | jeev | asteriskNOW |
| 00:38.13 | jeev | i am planning on setting up asterisk on my bsd box at the datacenter |
| 00:38.17 | korihor | jeev: ok :p |
| 00:38.31 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:38.31 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:38.32 | jeev | maybe i'll copy that and try it |
| 00:38.34 | lmadsen | dances |
| 00:38.52 | jeev | lmadsen, you have to answer the question. ASAP |
| 00:38.59 | lmadsen | I don't know what the question is |
| 00:39.05 | lmadsen | and I don't *have* to do anything |
| 00:39.11 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 00:39.18 | [hC] | you have to keep dancing, is what you have to do! |
| 00:39.23 | jeev | :< |
| 00:39.27 | jeev | the question is |
| 00:39.47 | jeev | when i dial out.. i press an option "4", if i tap it lightly and quickly, the remote IVR will pick up.. if i dial it like i'm dialing a number.. it wont realize it |
| 00:39.52 | jeev | know what the problem be? :D |
| 00:40.14 | [hC] | you should probably try pressing digits with your dialing wand |
| 00:40.15 | lmadsen | what version of asterisk? |
| 00:40.23 | jeev | 1.4.18.1, asteriskNOW |
| 00:40.29 | lmadsen | what is the other end? |
| 00:40.40 | [hC] | (ie are you calling out SIP, IAX, or Zap) |
| 00:40.44 | jeev | i guess that's the common question, i will have to email them. |
| 00:40.45 | jeev | SIP |
| 00:40.58 | [hC] | you can do a sip debug and see what their useragent is |
| 00:41.04 | jeev | k sec |
| 00:41.09 | [hC] | ....maybe... :) |
| 00:41.18 | [hC] | i dunno if asterisk advertises version in the useragent |
| 00:41.36 | jeev | doesn't advertise the version. |
| 00:41.52 | *** join/#asterisk media82 (n=DK@chello084113018116.7.12.vie.surfer.at) |
| 00:42.23 | Nivex | how did asterisk-announce get owned? |
| 00:42.36 | jeev | anywayyyyyy |
| 00:42.39 | jeev | yea, it's SIP |
| 00:42.41 | jeev | kind of weird. |
| 00:43.26 | *** join/#asterisk Kumbang (n=kumbang@125.163.83.153) |
| 00:43.37 | jsmith-dinner | Nivex: What do you mean? |
| 00:44.25 | Nivex | I got online casino spam, and it's been through the asterisk-announce list |
| 00:44.40 | media82 | how did asterisk become a fake b2bua? if i send an INFO request with any other mime type than dtmf and such...that it returns 200 no matter what!? |
| 00:45.04 | JT | asterisk is a B2BUA, how is it fake? |
| 00:45.37 | media82 | JT> it will return 200 before even sending the INFO to the other side |
| 00:46.27 | jsmith-dinner | Nivex: Now I see it... yeah, we made some changes to the spam filters that were in front of the mailling lists... first they were too strict, now they're too loose :-( |
| 00:46.36 | Nivex | doh |
| 00:48.00 | jeev | is 1.4.18.1 a recommended version/ i see that on my freebsd ports system... |
| 00:48.05 | jeev | wondering if i should just hack up the ports and set 1.4.19 |
| 00:48.12 | jeev | or email the maintainer |
| 00:53.11 | *** join/#asterisk Trionnis (i=Trionnis@s233-51-251.nap.wideopenwest.com) |
| 00:53.26 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 00:53.47 | media82 | JT> i've looked at the source, asterisk is "SHIT" in the b2bua domain. it doesnt map right in almost any case...its not a softswitch...its a shitswitch |
| 00:54.17 | *** join/#asterisk PaulQ (n=PaulQ@72.29.76.254) |
| 00:55.01 | PaulQ | How come I can't set {CALLERID(name)} in a macro correctly, num works but I am seeing the name on the phone |
| 00:55.31 | PaulQ | Hopping I am doing something stupid :-) |
| 00:56.39 | file | media82: what version of Asterisk? I just checked the source for 1.4 for handling INFO and it sends back a 415 for ones it does not know about, as for mapping... do you mean SIP responses to applicable cause codes? |
| 00:58.55 | file | well, unless you have an INFO with no Content-Length in which case we treat it as a keep alive packet |
| 00:59.34 | PaulQ | Even if I set a variable way ahead of time the macro never sets it |
| 00:59.47 | PaulQ | So weird.. |
| 01:00.06 | media82 | i'm asking...is that the way an INFO message should be treated at all |
| 01:00.18 | file | what would you expect to happen? |
| 01:00.21 | PaulQ | Wonder why my macro cant see variables.. |
| 01:00.29 | media82 | and then im asking...is that the way a b2bua should handle requests |
| 01:00.38 | media82 | at all... |
| 01:00.49 | media82 | if its not known...just forward it... |
| 01:00.53 | media82 | whats the deal!? |
| 01:00.57 | file | but Asterisk doesn't just speak SIP |
| 01:01.08 | media82 | true....is that an argument? |
| 01:01.32 | file | if you want to be |
| 01:01.33 | media82 | im saying b2bua all the time here... |
| 01:01.41 | PaulQ | Ah ha! |
| 01:01.58 | PaulQ | Its a screening macro the $CALLERID applies to the person screening not the other end |
| 01:02.04 | PaulQ | Wow thats so not how you would expect it to be handled! |
| 01:02.30 | media82 | >@file if you find something you cant map...thats ok...but if you do...why restrain it? |
| 01:02.49 | PaulQ | and the variable doesnt set because the macro screen is between the CallED and asterisk so its not the same channel as the variable set |
| 01:02.58 | PaulQ | and there is no way to pass options to a screen macro |
| 01:03.03 | PaulQ | Well this is gheey |
| 01:03.53 | file | media82: because Asterisk was designed with the idea of being protocol agnostic, and the entire idea of passing through stuff you don't understand in that context is... suicide |
| 01:04.32 | file | you *could* turn Asterisk into a pure SIP platform that does exactly what you say... but there are better solutions out there |
| 01:05.16 | PaulQ | Ahh ^ allows you to pass arguements to the screen |
| 01:06.17 | jeev | http://b2bua.org/chrome/site/ |
| 01:06.19 | jeev | anyone use that? |
| 01:06.26 | jeev | http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch |
| 01:06.27 | jeev | i mean that |
| 01:07.02 | PaulQ | M(macro-screen^${CALLERID(name)}^${CALLERID(num)}) that makes me giggle |
| 01:07.30 | jeev | what's that do |
| 01:08.27 | PaulQ | Hmm, it doesnt work in AEL? |
| 01:08.28 | PaulQ | :-/ |
| 01:08.43 | PaulQ | wth |
| 01:09.35 | PaulQ | Maybe AEL lets me do | | in the M() area |
| 01:09.56 | media82 | @file> its about softswitches...and asterisk does a horrible job in it...so for example you are telling me that INFO (X content) is not mappable to -> INFO (X content) |
| 01:10.10 | file | media82: Asterisk is a toolkit originally designed to be a PBX. |
| 01:10.11 | media82 | im not talking about other protocols ... |
| 01:10.22 | file | but no, it's not |
| 01:11.14 | media82 | if you look at the mapping matrix from ...i dont know... ss7 <-> sip .....dont you see some things missing? |
| 01:11.27 | file | like what? |
| 01:11.48 | media82 | like...im saying not...everything maps 1:1 |
| 01:11.58 | media82 | but if it does...you make a problem out of it |
| 01:12.07 | file | like what? examples help... |
| 01:12.16 | media82 | ok... |
| 01:12.45 | media82 | IAM -> INVITE |
| 01:13.02 | media82 | Number complete -> to what!?! |
| 01:13.14 | file | but Asterisk isn't designed like that |
| 01:13.19 | file | it doesn't "map" things |
| 01:13.23 | media82 | oh all the sudden |
| 01:13.40 | media82 | i thought its a b2bua |
| 01:13.51 | file | it is, but not at that level |
| 01:13.59 | media82 | like eben SIP INFO <-> SIP INFO doesnt work...that figures |
| 01:13.59 | Trionnis | argumentative little shit, aren't you? ;-) |
| 01:14.08 | file | it's not mapping protocol to protocol. |
| 01:14.27 | file | each channel driver implements the protocol side of things (for example SIP) and then provides an interface for Asterisk |
| 01:14.38 | file | for dialing numbers, sending DTMF, sending audio, reading audio |
| 01:14.45 | media82 | i know what it does--- |
| 01:14.55 | media82 | im looking at it....and im pooking |
| 01:15.58 | media82 | well obviously you guys have never had to do with softswitched in the real world...kinda senseless to talk about it then |
| 01:16.07 | Trionnis | obviously! |
| 01:16.14 | Trionnis | rolls eyes |
| 01:16.18 | file | and yet Asterisk is working in the real world |
| 01:16.30 | media82 | not for me...it isnt |
| 01:17.03 | media82 | ive done my homework guys...believe me...ive written my own proxy here...if it wasnt for copyrights id use it |
| 01:17.14 | file | proxy is not a softswitch, 'nor a B2BUA |
| 01:17.19 | jeev | well, i built 1.4.19 on my fbsd box at the datacenter.. |
| 01:18.16 | file | a lot of people like OpenSER + Asterisk |
| 01:18.24 | Trionnis | raises hand |
| 01:18.29 | Trionnis | that's what we're running |
| 01:18.47 | Trionnis | doing close to 35mil minutes through it... works for us *shrug* |
| 01:19.03 | media82 | im thinking about OT + audiocodes actually# |
| 01:19.07 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net) |
| 01:19.08 | drmessano | Asterisk = Unicorns |
| 01:19.08 | Trionnis | ugh |
| 01:19.12 | Trionnis | audiocodes |
| 01:19.18 | Trionnis | buggy POS |
| 01:19.33 | hackeron | is anyone able to get MusicOnHold working with the asterisk on ubuntu gutsy? (installed by apt) |
| 01:19.35 | media82 | at least they got the SBC down |
| 01:19.53 | Trionnis | forgive me if I don't believe that |
| 01:20.21 | *** join/#asterisk Darthclue (n=jdale@76-233-19-118.lightspeed.snantx.sbcglobal.net) |
| 01:20.26 | Trionnis | after listening to their techs insist for over 2 months that our mediant 3000 supported TBCT, and then figuring out that it didn't |
| 01:21.02 | *** join/#asterisk frogonwheels (n=michaelg@203.59.141.93) |
| 01:21.05 | media82 | well i got the mediant 2000 running here...and i dont have any softswitch troubles...because its configurable |
| 01:21.07 | Trionnis | that and the whole "oh, it's broken? you want an RMA? ok, ship it to Israel and wait a month and a half for a replacement" |
| 01:21.11 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 01:21.32 | drmessano | Hmm.. Who buys a product that takes 45 days to RMA |
| 01:21.36 | drmessano | Without a spare |
| 01:21.37 | Trionnis | not me |
| 01:21.43 | Trionnis | and we had a couple, thank the gods |
| 01:21.46 | media82 | and OT (australian i think) does ss7<->sip<->isdn |
| 01:21.51 | drmessano | Asterisk has an RMA time of .043 nanoseconds |
| 01:21.56 | media82 | "CONFIGURABLE" |
| 01:22.31 | media82 | -> RESPONSECODE -> ISDN ---- CAUSE VALUE -> SS7 |
| 01:23.40 | media82 | > yea right...if it works at all...i have to put developers on the INFO trouble just because INFO<->INFO doesnt work |
| 01:23.42 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-122-158.PHNX.QWEST.NET) |
| 01:24.00 | file | smacks head against wall |
| 01:25.03 | media82 | or...> 183 Early Media -> 180... |
| 01:25.03 | Darthclue | puts a pillow between the wall and files head |
| 01:25.37 | file | fortunately nobody is forcing you to use Asterisk |
| 01:25.39 | drmessano | file |
| 01:25.53 | media82 | i aint...i didnt say i do... |
| 01:26.01 | media82 | fortunately |
| 01:26.10 | drmessano | So, why are you here? |
| 01:26.32 | PaulQ | Give me a T...Give me a R... Give me a O...Give me a L |
| 01:26.41 | drmessano | No shit |
| 01:26.46 | PaulQ | Give me one more L! |
| 01:26.48 | Nivex | <3 Asterisk |
| 01:26.55 | media82 | cause im interested in solutions...kinda...i dont rely on it..but i would like to see asterisk "at least" go in the right direction |
| 01:26.57 | drmessano | DUZ AKERISK REALITY WORK? |
| 01:27.03 | Nivex | Just listened to my LUG meeting from home |
| 01:27.09 | drmessano | media82: You dont even USE ASTERISK |
| 01:27.14 | drmessano | So how can you see it do anything? |
| 01:27.21 | drmessano | t.r.o.l.l. |
| 01:27.24 | media82 | we have 2 ax running |
| 01:27.26 | media82 | 1.2 |
| 01:27.28 | media82 | and 1.4 |
| 01:27.29 | jeev | asterisk is officially better than sex. |
| 01:27.34 | frogonwheels | :) |
| 01:27.36 | PaulQ | jeev, easy now... |
| 01:27.41 | jeev | i'm serious |
| 01:27.41 | media82 | both suck..the hell out of softswitching |
| 01:27.51 | drmessano | [21:25] <file> fortunately nobody is forcing you to use Asterisk [21:25] <media82> i aint...i didnt say i do... [21:25] <media82> fortunately |
| 01:27.54 | PaulQ | You are having terrible sex |
| 01:27.55 | jeev | i havefn't had it in a while. girlfriend is a virgin and staying one till marriage.. so it'll be better for a while. |
| 01:28.06 | drmessano | Sounds like you use the hell out of it |
| 01:28.16 | PaulQ | "Asterisk is better than my sex life" please! |
| 01:28.20 | jeev | yes |
| 01:28.26 | JT | jeev: lol are you joking? |
| 01:28.29 | jeev | but my sex life is non-existant |
| 01:28.34 | jeev | joking about what |
| 01:28.41 | JT | gf being a virgin |
| 01:28.46 | jeev | yea she is one.. i love it |
| 01:28.47 | drmessano | Asterisk is better than sex because deadlocks are a lot easier to recover from |
| 01:28.49 | JT | hahaha |
| 01:28.53 | jeev | :) |
| 01:28.53 | PaulQ | Tell her she is doing gods work |
| 01:28.58 | PaulQ | The only way to make more virgins is to fuck :P |
| 01:28.59 | JT | marrying someone without having sex first is pure insanity |
| 01:29.04 | file | dances |
| 01:29.05 | Nivex | "Documentation is like sex. When it's good, it's great. When it's bad, it's better than nothing." |
| 01:29.06 | drmessano | JT: Agreed |
| 01:29.12 | [hC] | its like buying a used car without a test drive |
| 01:29.16 | PaulQ | JT: 10/4 |
| 01:29.25 | jeev | i'd rather marry a girl nobody has slept with. :) |
| 01:29.29 | PaulQ | aww |
| 01:29.30 | JT | you're crazy |
| 01:29.32 | PaulQ | That's cute |
| 01:29.32 | jeev | sex is sex |
| 01:29.34 | jeev | cuming is easy |
| 01:29.38 | PaulQ | Cause virgins are great at sex ! |
| 01:29.40 | PaulQ | Fuck that, Give me a whore |
| 01:29.41 | Darthclue | nah, asterisk doesn't work. That's why my answering machine always wants to know what I'm wearing. |
| 01:29.44 | JT | spoken like a true virgin |
| 01:29.44 | jeev | if she's fucked a million people.. she wont be a good mother |
| 01:29.48 | [hC] | i also dont understand people who move in together until they get married |
| 01:29.49 | [hC] | wtf. |
| 01:29.49 | JT | ... |
| 01:29.51 | [hC] | time bomb. |
| 01:29.51 | media82 | i dont really know why you all get that offensive about me talking about the loops in asterisk.....must be a european thing... :) |
| 01:29.51 | Nivex | I gather he's in it for more than just the sex. |
| 01:29.52 | drmessano | HA |
| 01:29.56 | JayTee52 | anyone who thinks Asterisk is better than sex is 1) doing it wrong 2) a sad pathetic excuse for a human being or 3) suffering from some neurological birth defect that makes one insensitive to pleasure. |
| 01:30.12 | file | what if Asterisk gets you sex? |
| 01:30.12 | jeev | :) |
| 01:30.16 | JT | don't worry, divorce will be on the cards soon enough |
| 01:30.18 | jeev | heheh |
| 01:30.23 | jeev | har har har |
| 01:30.26 | media82 | fike> id go with it |
| 01:30.33 | Nivex | phone sex over IP! |
| 01:30.34 | JT | i guess some people have to learn from their mistakes\ |
| 01:30.35 | drmessano | media82: Because you're trolling.. especially the "european" comment.. nice |
| 01:30.39 | media82 | SIP<->SEX interworking. |
| 01:30.47 | media82 | nuff said |
| 01:31.04 | PaulQ | http://shell.vaerchi.com/~paultech/186695367941_0_0.jpg |
| 01:31.06 | PaulQ | Sex is fun kids |
| 01:31.07 | PaulQ | NSFW |
| 01:31.09 | drmessano | Pr0noIP |
| 01:31.09 | PaulQ | NSFW |
| 01:31.26 | jeev | that's a fat girl |
| 01:31.37 | drmessano | Thats a big pimple |
| 01:31.45 | jeev | hhah |
| 01:31.46 | PaulQ | jeev: Skinny |
| 01:31.47 | Trionnis | my wife says the same thing! |
| 01:31.52 | Trionnis | oh wait... different topic |
| 01:32.02 | drmessano | Trionnis: She tells me the same thing |
| 01:32.10 | Trionnis | not likely |
| 01:32.15 | media82 | so you all dont get it...thats fine with me...more money for me then |
| 01:32.20 | Trionnis | she likes strange peenor |
| 01:32.22 | jeev | heh |
| 01:33.33 | Jumpie | well apparently callwithus sucks bigtime |
| 01:33.51 | drmessano | Jumpie: Yes |
| 01:34.28 | drmessano | Asterisk doesn't have enough support for velociraptors |
| 01:34.30 | jeev | korihor, your rtp.c did not work. |
| 01:34.34 | media82 | how about a flash<->asterisk video/audio gateway |
| 01:34.48 | drmessano | media82: Go code it |
| 01:34.55 | media82 | i did |
| 01:34.59 | media82 | but you know |
| 01:35.01 | media82 | the INFO thing |
| 01:35.03 | jeev | i have a patch to play nintendo games over asterisk |
| 01:35.17 | drmessano | Red5? |
| 01:35.22 | media82 | i would contribute...but you all dissed me...so |
| 01:35.30 | media82 | FMS |
| 01:35.37 | drmessano | Never heard of it |
| 01:35.37 | media82 | actually it doesnt matter |
| 01:35.40 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 01:35.54 | media82 | i can do flash<->sip |
| 01:36.02 | media82 | flash<->skype |
| 01:36.07 | media82 | flash<->3g |
| 01:36.11 | Jumpie | cwu doesnt allow good seperation of inbound/outbound |
| 01:36.11 | JT | flash? |
| 01:36.29 | media82 | or if you like |
| 01:36.33 | media82 | asterisk<->flash |
| 01:36.38 | media82 | asterisk<->skype |
| 01:36.43 | Jumpie | next thing is to get a router and bring a t1 into here |
| 01:36.43 | media82 | asterisk<->sip |
| 01:36.44 | JT | as in flash, that web stuff? |
| 01:36.49 | Jumpie | lol |
| 01:36.50 | media82 | asterisk<->flash |
| 01:36.53 | drmessano | Why do you need the asterisk community to kiss your behind? Release it, put up crap Drupal site, and tell the world how great you are... |
| 01:37.09 | media82 | i will! dude |
| 01:37.16 | drmessano | So do it.. |
| 01:37.28 | media82 | just remember i said that asterisk sucks in interworking.. |
| 01:37.46 | media82 | 2008 |
| 01:37.48 | drmessano | Come back when you have a URL |
| 01:37.54 | media82 | i do |
| 01:38.07 | drmessano | Is your code posted? Can I use it NOW? |
| 01:38.11 | media82 | its all working right here...boy |
| 01:38.13 | frogonwheels | I have just finished making a script so that my voicemail from my voip provider can get emailed to my router - unpacked and placed in the asterisk voicemail.. |
| 01:38.19 | frogonwheels | tell me about asterisk sucking in interworking! |
| 01:38.21 | drmessano | Give me a link to download working code |
| 01:38.27 | media82 | i got the rtmp down biatch |
| 01:38.36 | JayTee52 | groans |
| 01:38.42 | drmessano | Link? |
| 01:38.43 | media82 | BUT |
| 01:38.44 | drmessano | Code? |
| 01:38.46 | drmessano | Download? |
| 01:38.51 | media82 | i also got 3g<->flash |
| 01:39.05 | frogonwheels | drmessano: who was that to? |
| 01:39.11 | drmessano | media82 |
| 01:39.22 | Trionnis | I'm gonna have to go get a pair of hip waders... the bullshit is getting deep in here... |
| 01:39.34 | media82 | i can do RTSP<->FLASH right now |
| 01:39.45 | drmessano | media82: Where can I download this great code? |
| 01:39.49 | JayTee52 | I'd go for a full wetsuit with helmet |
| 01:40.05 | media82 | you cant...:) becaue asterisk sucks!!! |
| 01:40.08 | Trionnis | that might be a better idea |
| 01:40.09 | drmessano | media82: You're completely full of crap |
| 01:40.14 | JayTee52 | we're gonna be up to our necks in it in minutes at this rate |
| 01:40.15 | Trionnis | not completely |
| 01:40.17 | media82 | oh really |
| 01:40.18 | drmessano | media82: You have nothing |
| 01:40.25 | Trionnis | every time he opens his mouth, more flows out |
| 01:40.30 | drmessano | media82: You have no link, no apps or code to show |
| 01:40.31 | JayTee52 | trolling gangsta coder wannabee |
| 01:40.32 | media82 | give me a mms stream of any choiuce right now |
| 01:40.34 | Trionnis | he's gonna dry up and fly away on the wind in a bit |
| 01:40.42 | drmessano | media82: Give us a download link |
| 01:40.45 | media82 | i will stream it to flash right now |
| 01:40.48 | drmessano | media82: Give us a download link |
| 01:40.49 | media82 | for you to see |
| 01:40.51 | drmessano | media82: Give us a download link |
| 01:40.56 | Jumpie | what is ht he's claming he can do? |
| 01:41.02 | Jumpie | claiming |
| 01:41.09 | media82 | i CAN do it |
| 01:41.14 | media82 | i reversed RTMP |
| 01:41.20 | JT | your mum said you could do it |
| 01:41.22 | Jumpie | but what....do what |
| 01:41.22 | JT | so you can |
| 01:41.33 | drmessano | He's got all these 1337 mad Flash <> SIP, Flash <> Skype, Flash <> Frigidaire Washer/Dryer Combo skills |
| 01:41.37 | drmessano | But can't show proof |
| 01:41.50 | media82 | if there is an admin...here to verify it...let me speak to him and i will show it |
| 01:42.02 | JT | why an admin? |
| 01:42.03 | drmessano | ROFL |
| 01:42.22 | file | an admin for what? O.o |
| 01:42.29 | media82 | because i only have 1mb up...if i tell you all to connect...it will break |
| 01:42.30 | drmessano | JT: He will only show his code to an admin.. in the basement of the Alamo |
| 01:42.41 | JayTee52 | lol |
| 01:42.54 | JT | with a suitcase of cash |
| 01:42.56 | media82 | guys...im not kidding...but anyways |
| 01:43.04 | drmessano | Donnie, you're out of your element |
| 01:43.06 | JayTee52 | the stars at night are big and bright (clap, clap, clap, clap) .......... |
| 01:43.16 | drmessano | Here in the heart of asterisk! |
| 01:43.19 | file | #asterisk After Hours, providing entertainment for years |
| 01:43.20 | JT | media82: but since you're such a big voip player, surely you have a server in the datacentre with hundreds of megabits |
| 01:43.22 | JayTee52 | ^5 |
| 01:43.36 | lmadsen | promises not to connect |
| 01:43.37 | drmessano | file: I made a mad Asterisk SIP <> Cheese Grater proxy |
| 01:43.42 | media82 | > im a player...but i dont have the bandwidth |
| 01:43.57 | drmessano | file: But I am dialup |
| 01:44.01 | JayTee52 | Cheese grater proxy! hahahahaha |
| 01:44.03 | media82 | if there is an admin who contacts me in private i will show it to him |
| 01:44.18 | outtolunc | file you up for it <G> |
| 01:44.19 | drmessano | media82: What good is your code then? |
| 01:44.19 | file | I'm the admin of my apartment's network, does that count? |
| 01:44.21 | media82 | flash<->rtsp |
| 01:44.26 | media82 | flash<->sip |
| 01:44.28 | JayTee52 | "We're sorry but gorgonzola is not authorized to access the outbound context" |
| 01:44.39 | drmessano | [brie] |
| 01:44.39 | Trionnis | yeah, but I wrote an Asterisk <-> Skype <-> eBay <-> Flickr <-> toaster <-> digg <-> frydaddy proxy... IN FORTRAN! |
| 01:44.51 | drmessano | Trionnis: EGGZACHARY |
| 01:44.55 | Trionnis | lol |
| 01:45.06 | Trionnis | but you can't download it |
| 01:45.12 | Trionnis | I only have 23423 mbits up |
| 01:45.16 | drmessano | You forgot reddit.. there goes your greenlight on Fark |
| 01:45.25 | Trionnis | and it's copyrighted by Steve Ballmer |
| 01:45.29 | lmadsen | yay #asterisk after hours :) |
| 01:45.34 | Trionnis | he threatened me with a chair if I released it |
| 01:45.40 | JayTee52 | my dick is copyrighted by Steve Ballmer |
| 01:45.43 | *** join/#asterisk d3wayne (n=deeewayn@76.29.245.9) |
| 01:45.43 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 01:45.47 | file | developers! developers! developers! |
| 01:45.51 | Jumpie | now we just need someone to write an app that lets you control WOW characters with key presses :) |
| 01:45.58 | outtolunc | chair... i thought you said 'hair' at first <G> |
| 01:46.01 | JayTee52 | never seen a fat man sweat more than that guy |
| 01:46.08 | Trionnis | he doesn't have many of those |
| 01:46.10 | media82 | you dont believe me do you? |
| 01:46.15 | Trionnis | don't think he'd risk losing them |
| 01:46.18 | JT | my asterisk integration with COBOL on Cogs is complete |
| 01:46.20 | drmessano | I was working on a microsoft silverlight <> MGCP gateway using Visual Basic 3.0 on a Gopher daemon |
| 01:46.26 | drmessano | But it's closed source, sorry |
| 01:46.30 | media82 | lol |
| 01:46.31 | Trionnis | AWW |
| 01:46.32 | JayTee52 | media82, are you a Night Elf Mohawk, third level? |
| 01:46.33 | Trionnis | cries |
| 01:46.53 | media82 | ok...first one to message me private..will get the demo |
| 01:47.01 | JT | http://www.coboloncogs.org/INDEX.HTM |
| 01:47.02 | Trionnis | *crickets* |
| 01:47.04 | Qwell | show me flash <> skype |
| 01:47.11 | JayTee52 | don't take the bait! it's just more gay porn |
| 01:47.24 | JayTee52 | show me "paint the fence" |
| 01:47.28 | drmessano | JayTee52: If he gave everyone one the link.. and his bandwidth got hosed... and his dad couldn't check his ebay auctions.... dude, hell. |
| 01:47.32 | media82 | right now...any |
| 01:47.34 | media82 | rtsp |
| 01:47.36 | JT | cobol on cogs is waaay better than ruby on rails |
| 01:47.40 | JayTee52 | show me "wax on, wax off" |
| 01:47.42 | media82 | stream or mms stream you know |
| 01:47.50 | media82 | in flash |
| 01:47.50 | Qwell | rtsp is...kinda useless O.o |
| 01:47.55 | Trionnis | something tells me he's got the "wax off" well studied |
| 01:48.05 | drmessano | media82: Is your webserver in mom's sewing room? |
| 01:48.24 | JayTee52 | Cobol= Completely Obsolete But Obstinately Lingering language |
| 01:48.33 | media82 | ok...well forget it...if youre interested in this then email me at weaponx@inode.at |
| 01:48.35 | Trionnis | I blame IBM |
| 01:48.43 | drmessano | weaponx? That's... uber |
| 01:48.50 | Qwell | media82: again - show me flash <> skype |
| 01:48.58 | Qwell | I'll make you a rich man. |
| 01:49.03 | drmessano | flashes his skype at Qwell |
| 01:49.05 | Trionnis | adds weaponx@inode.at to 300 gay pr0n lists |
| 01:49.17 | JayTee52 | show me "sanda floor" |
| 01:49.19 | media82 | i can show you anything<->anything |
| 01:49.25 | Trionnis | and a couple tranny midget lists for good measure |
| 01:49.30 | media82 | yomomma<->yomomma |
| 01:49.37 | drmessano | Cheeseburger <> Hot Dog, in PHP.. make it happen |
| 01:49.51 | media82 | ok you all got my email...peace out |
| 01:49.51 | Qwell | drmessano: has to be flash |
| 01:49.51 | file | drmessano: no, IA64 assembler! |
| 01:49.58 | drmessano | ROFL |
| 01:50.12 | drmessano | Flash <> Corn Dog |
| 01:50.12 | Trionnis | can someone show him banhammer <-> his ass? |
| 01:50.17 | Trionnis | :) |
| 01:50.36 | JayTee52 | Flash <> Bratwurst |
| 01:50.38 | media82 | before i go...i really dont understand why anyone contacted me...this is a big deal... |
| 01:50.52 | Jumpie | hey JayTee52 |
| 01:50.52 | drmessano | Who contacted you? |
| 01:50.54 | Trionnis | I wouldn't understand why anyone contacted you either |
| 01:50.56 | Jumpie | im all up and running now :D |
| 01:50.59 | media82 | noone |
| 01:50.59 | Jumpie | thanks to outtolunc patience |
| 01:51.00 | drmessano | We'll ban them |
| 01:51.00 | Trionnis | unless they're just a massochist |
| 01:51.03 | drmessano | Oh |
| 01:51.06 | drmessano | Noone did |
| 01:51.15 | drmessano | This is a huge deal |
| 01:51.24 | drmessano | 2008: SIP <> Skype |
| 01:51.30 | drmessano | 2009: Cancer <> AIDS |
| 01:51.34 | drmessano | you're cutting edge |
| 01:51.38 | media82 | well i think so..its not like im saying INFO<->INFO works in asterisk |
| 01:51.57 | media82 | im saying SIP<->FLASH works |
| 01:52.02 | drmessano | media82: Post it on YouTube |
| 01:52.09 | drmessano | media82: Let us all see it work |
| 01:52.18 | Trionnis | he can't... he only has 1mbit up |
| 01:52.26 | drmessano | media82: Surely you can master the art of FLASH <> INTERNET |
| 01:52.27 | Trionnis | it would take all his bandwidth to upload it |
| 01:52.35 | media82 | good idea..since you all seem to be european idiots i will post it |
| 01:52.36 | JayTee52 | media82, post it on YouTube with the Benny Lava audio track |
| 01:52.45 | drmessano | <-- european idiot |
| 01:52.50 | media82 | i mean really...i was saying |
| 01:52.52 | Trionnis | <-- american idiot |
| 01:52.57 | media82 | the first person to contact me |
| 01:52.58 | Trionnis | (green day ftw) |
| 01:53.01 | JayTee52 | < --- plain vanilla American moron |
| 01:53.06 | media82 | and noone contacted me.. |
| 01:53.08 | media82 | funnny |
| 01:53.24 | drmessano | media82: Don't take it personal.. we just hate you |
| 01:53.25 | JT | media82: we're european idiots? but you're the one with the austraian host name |
| 01:53.30 | JT | is austria not in europe? |
| 01:53.33 | JayTee52 | the first person to contact you? what? wins a Winnebago? |
| 01:53.39 | *** join/#asterisk steliosk (n=Stelios@athedsl-105743.home.otenet.gr) |
| 01:53.40 | drmessano | A NEW CAR! |
| 01:53.40 | Trionnis | hey file... what kinda bribes do you accept to ban people? |
| 01:53.43 | media82 | does that make me european? |
| 01:53.45 | Trionnis | I'm going to start a collection |
| 01:53.54 | file | Trionnis: an international airport |
| 01:53.54 | media82 | the ip? |
| 01:53.58 | file | and a private jet. |
| 01:54.06 | Trionnis | hm |
| 01:54.11 | Trionnis | I can probably do the airport |
| 01:54.13 | file | I would settle for just the private jet |
| 01:54.18 | Trionnis | I'm about 1/2 a mile from O'Hare right now |
| 01:54.23 | drmessano | media82 is a 25 yr old 1337 haxor who got a keygen for a same Flash development kit for media.. |
| 01:54.26 | JT | media82: if you are in austria, that makes you european |
| 01:54.30 | media82 | i mean really guys...the first one to contact me is gonna see a live demo...is it that hard to understand |
| 01:54.30 | Trionnis | I'm sure no one would notice if I dug it up and carted it off |
| 01:54.31 | drmessano | some* |
| 01:54.39 | file | Trionnis: I think they might |
| 01:54.45 | Trionnis | hmn |
| 01:54.47 | Trionnis | maybe |
| 01:54.50 | JayTee52 | austria? isn't that where they wear those gay leather shorts with suspenders? |
| 01:54.55 | file | Qwell accepts Skinny phones, might be cheaper to bribe him |
| 01:55.01 | drmessano | $25 bucks says he wants someone to go visit his site and get infected with something |
| 01:55.09 | drmessano | Sounds like hes begging for his code to get in the wild |
| 01:55.13 | drmessano | Like a script kiddie |
| 01:55.28 | media82 | $50 bucks says ...you are full of it |
| 01:55.43 | drmessano | You're begging |
| 01:55.46 | media82 | you all insult me...but i still havent gotten one mail yet...funny |
| 01:55.53 | JT | GUYS |
| 01:55.57 | JT | PLEASE TALK TO ME |
| 01:55.59 | outtolunc | just load the demo onto ustream.tv (as a recording) with password |
| 01:56.01 | JT | MANBEARPIG IS REAL |
| 01:56.04 | JT | IM FULLY SERIAL |
| 01:56.10 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d513db0fd584c160) |
| 01:56.12 | media82 | im on 1mb i cant do shit... |
| 01:56.13 | drmessano | ROFL |
| 01:56.16 | JayTee52 | lol |
| 01:56.21 | drmessano | I can't believe no one has emailed me |
| 01:56.26 | drmessano | Youre so stupid |
| 01:56.32 | drmessano | STOOPED |
| 01:56.36 | drmessano | I have mad code |
| 01:56.40 | drmessano | MAD, MAD, MAD, Code |
| 01:56.42 | media82 | i got a flash video conf right here |
| 01:56.46 | Qwell | media82: for the third time - SHOW ME |
| 01:56.47 | JT | SUPER SERIAL |
| 01:56.48 | outtolunc | i only got 300k up and i can stream to usteam.tv <G> |
| 01:56.51 | media82 | you name the mms or rtsp link |
| 01:56.57 | JT | media82: Qwell is an admin |
| 01:56.58 | JT | media82: Qwell is an admin |
| 01:57.00 | JT | !!! |
| 01:57.01 | JayTee52 | uploads media82's email address to the local Viagra spam server |
| 01:57.03 | Qwell | Show me flash <> skype |
| 01:57.04 | media82 | and i will stream it to the conference |
| 01:57.04 | drmessano | Qwell asked to see it |
| 01:57.06 | drmessano | Show him |
| 01:57.07 | file | the right term is op |
| 01:57.09 | drmessano | He's an @ |
| 01:57.11 | JayTee52 | there, now you'll get some mail |
| 01:57.15 | media82 | skype is a detail... |
| 01:57.20 | Trionnis | ok, well this has been fun and all, but I suppose I should get back to fighting with Level 3 and their absolutely FARKING STUPID interop paperwork |
| 01:57.21 | media82 | its just another cam thingy |
| 01:57.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:57.23 | jeev | what do you guys think about that LINKSYS WIP330 ? |
| 01:57.26 | file | Trionnis: ugh... |
| 01:57.33 | Trionnis | ugh is right |
| 01:57.33 | file | Trionnis: I remember that from years ago, how many pages is it up to? |
| 01:57.36 | Trionnis | 55 |
| 01:57.37 | drmessano | media82: Show Qwell |
| 01:57.40 | media82 | name a link! |
| 01:57.49 | media82 | are you guys retarded? |
| 01:57.49 | Trionnis | and yes, it's a major pain. in. the. ass. |
| 01:57.49 | file | Trionnis: do they still want inband DTMF over G729? |
| 01:57.52 | Trionnis | yep |
| 01:57.54 | *** join/#asterisk Katty (n=Ryan@adsl-68-92-251-223.dsl.stlsmo.swbell.net) |
| 01:57.59 | file | ah how some things never change... |
| 01:58.03 | Qwell | I said, show me skype |
| 01:58.05 | Trionnis | even though we'll never use 729 at all |
| 01:58.15 | Trionnis | doesn't play well with ASR |
| 01:58.16 | Katty | and no, for the last time, i'm not ryan. |
| 01:58.16 | Trionnis | heh |
| 01:58.17 | file | Trionnis: have fun |
| 01:58.20 | Trionnis | oh yes |
| 01:58.22 | Qwell | Katty: liar |
| 01:58.23 | media82 | ok i guess skype works too.... |
| 01:58.23 | Trionnis | sooo much |
| 01:58.29 | Katty | sighs |
| 01:58.30 | Trionnis | later amigos :) |
| 01:58.33 | file | tickle tackles Katty |
| 01:58.37 | Trionnis | try not to torture him too bad |
| 01:58.38 | Katty | hugs file |
| 01:58.39 | media82 | ok os we go... |
| 01:58.43 | drmessano | media82: Give the link to Qwell.. he is our leader |
| 01:58.48 | media82 | how about GIGA TV <-> Skype |
| 01:58.48 | Trionnis | ah hell, never mind, do it as much as you like |
| 01:58.48 | Katty | file: my lappy died :< |
| 01:58.51 | Qwell | If you're *really* Katty, you'd know what I dislike :( |
| 01:58.56 | file | drmessano: did you really just say that? |
| 01:58.59 | Katty | hugs Qwell |
| 01:59.02 | file | Katty: awwwww |
| 01:59.04 | drmessano | FACEBOOK <> FLASH? |
| 01:59.05 | Qwell | ACK |
| 01:59.07 | file | Katty: did you stabbity it? |
| 01:59.07 | drmessano | Thats what I want |
| 01:59.12 | Qwell | drmessano: OMFG, FLASH <> FLASH |
| 01:59.14 | jeev | is willing to paypal $1.00 to whoever fixes my DTMF issue. YES 1 DOLLAR. |
| 01:59.15 | Katty | file: no :/ |
| 01:59.17 | drmessano | LOL |
| 01:59.19 | Katty | file: it's an ole lappy |
| 01:59.27 | file | Katty: :( |
| 01:59.34 | drmessano | AS/400 <> FLASH |
| 01:59.36 | media82 | this is funny...you all dont even know which way to go |
| 01:59.44 | JT | ETHERKILLER <> FLASH |
| 01:59.46 | drmessano | media82: SEND A LINK TO QWELL |
| 01:59.49 | jeev | media82, you wanna make $1 ? |
| 01:59.52 | drmessano | media82: HES AN OP |
| 01:59.55 | drmessano | media82: DO IT |
| 01:59.57 | drmessano | media82: DO IT NOW |
| 02:00.03 | drmessano | media82: OR SHOVE IT |
| 02:00.05 | drmessano | ahem |
| 02:00.05 | file | do the locomotion with me! |
| 02:00.06 | drmessano | Sorry |
| 02:00.09 | Nivex | Mr. Moderator: Motion to kickban media82 |
| 02:00.13 | Qwell | ~locomotion file |
| 02:00.17 | Qwell | pfft |
| 02:00.24 | drmessano | ~troll |
| 02:00.25 | jbot | extra, extra, read all about it, troll is a race on some muds; a guy under a bridge: an annoying robot; a port scanner, or somebody faking being clueless to be shown the One Linux Way so he can argue against it for his psychology thesis on linux advocates, or an employee of trolltech, or at http://www.kuro5hin.org/story/2001/7/27/51233/2979 or http://www.catb.org/~esr/jargon/html/entry/troll.html |
| 02:00.33 | media82 | > seriously guys...ive been coding for 3 days now...just tell me what mms or http link and where to |
| 02:00.34 | Qwell | I've never seen a MUD with a troll. |
| 02:00.36 | file | nobody is going to be kickbanned. |
| 02:00.36 | Katty | i have a troll mage. |
| 02:00.40 | drmessano | 3 days? |
| 02:00.47 | drmessano | How many doritos is that? |
| 02:00.51 | JT | ultra RAD |
| 02:01.03 | media82 | this seems like the spanish inqusition |
| 02:01.05 | jeev | my offer is now raised to $1.10. |
| 02:01.07 | JayTee52 | troll is also a fat balding man in Redmond, WA that is known to throw furniture whenever someone uses the G word. |
| 02:01.07 | JT | Eextreme programming |
| 02:01.19 | drmessano | <media82> I have seen hackers 1337 times, ZOMGROFLCOPTER |
| 02:01.26 | JT | media82: if i said i had a time machine, i'd expect the spanish inquisition |
| 02:01.29 | Katty | wonders what new laptop to get. |
| 02:01.33 | JayTee52 | jeev, relaxdtmf=yes in your sip.conf, now send me a buck ten |
| 02:01.40 | drmessano | Nooooobody expects the spanish inquisition |
| 02:01.44 | media82 | wtf...i havent even domed yet...are you guys reatarded? |
| 02:01.46 | drmessano | Our two weapons.. |
| 02:01.46 | file | jeev: inbound or outbound DTMF, if SIP what mode |
| 02:01.51 | Katty | drmessano: how about the French Banana war? |
| 02:01.52 | jeev | JayTee52, that didn't work! go ask cody.. tell her i wanna fark DONNA |
| 02:01.55 | JT | media82: your condom is not in place? |
| 02:01.57 | Katty | drmessano: you ready for that one? |
| 02:01.58 | Darthclue | wonders what realm katty has her troll mage on |
| 02:02.00 | jeev | outbound, rfc2833 |
| 02:02.07 | drmessano | Katty: Sure |
| 02:02.13 | media82 | why doesnt anyone believe me? |
| 02:02.14 | file | jeev: phone calling a provider? |
| 02:02.20 | jeev | file, if i quickly tap the number, the remote IVR will accept it.. but if i just hit it like a regular call.. the IVR wont take it |
| 02:02.24 | drmessano | media82: We asked you to send a link to qwell |
| 02:02.27 | jeev | i'mc alling a court, everywhere i call, evern my credit card company |
| 02:02.29 | Qwell | wonders why Darthclue wants to know |
| 02:02.31 | drmessano | media82: He's an OP |
| 02:02.33 | jeev | if i quicly tap the option, it'll accept.. if not, it wont notice it. |
| 02:02.36 | JayTee52 | Wax on, Wax off <> Flash |
| 02:02.38 | drmessano | media82: and you refuse |
| 02:02.38 | media82 | i dont know what qwell is |
| 02:02.44 | drmessano | HES TALKING |
| 02:02.45 | file | jeev: what provider? dtmfmode on both the phone AND provider's sip.conf entry? |
| 02:02.45 | Katty | Darthclue: i'm on a pve server. it won't matter. |
| 02:02.46 | Qwell | ... |
| 02:02.48 | drmessano | IN CHANNEL |
| 02:02.53 | JayTee52 | Sweep the leg, Johnny <> Flash |
| 02:02.54 | drmessano | Effin retard |
| 02:02.55 | file | ~qwell |
| 02:02.55 | jbot | [qwell] a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 02:03.05 | jeev | file: i'm using x-lite and have tried other softphones, but yes, rfc2833 to provider. |
| 02:03.11 | drmessano | JayTee52: ROFL.. Awesome reference |
| 02:03.19 | drmessano | SWEEP THE LEG |
| 02:03.22 | drmessano | I am dying |
| 02:03.27 | media82 | ok...well...i offered it to you...i find it funny that noone believes me |
| 02:03.33 | Darthclue | katty: lol |
| 02:03.50 | JayTee52 | back in the early 80's I used a product called Qwell but it had nothing to do with cholesterol. It came with a special comb :-) |
| 02:03.55 | JT | media82: /msg chanserv access #asterisk list |
| 02:03.57 | drmessano | ROFL |
| 02:03.58 | file | jeev: pastebin complete console output with dtmf set in logger.conf to go to console, you'll have to do logger reload... |
| 02:04.03 | JT | media82: shows Qwell is an op |
| 02:04.05 | Qwell | JayTee52: misspelled |
| 02:04.10 | Katty | Qwell: what does Qwell mean anyway? |
| 02:04.12 | jeev | ok, since i compiled on freebsd, lets see if dtmf will now show, sec please. |
| 02:04.15 | Qwell | ~qwell |
| 02:04.15 | jbot | somebody said qwell was a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 02:04.16 | Qwell | ^^ |
| 02:04.18 | JayTee52 | Quell then |
| 02:04.26 | JT | media82: /msg nickserv info qwell, shows that qwell is real |
| 02:04.30 | JT | media82: no more excuses |
| 02:04.32 | JayTee52 | it's been awhile and I've avoided public restrooms ever since |
| 02:04.36 | Katty | Qwell: so your...an... orange? |
| 02:04.47 | drmessano | Qwell is a debugger used in a very early beta of PHP for DOS |
| 02:04.48 | Qwell | ...eww |
| 02:04.49 | Katty | gets straw, eyes Qwell |
| 02:04.57 | Qwell | O.o |
| 02:05.02 | Qwell | Don't s... |
| 02:05.05 | Katty | k |
| 02:05.05 | Qwell | I'm not going to complete that |
| 02:05.13 | *** part/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 02:05.16 | Katty | probably for the best *hee* |
| 02:05.20 | Qwell | agreed |
| 02:05.23 | file | snuggles up to Katty |
| 02:05.35 | JayTee52 | Qwellish was a secret code word to activate swarms of killer frogs and other strange things in Leonard Part 6. |
| 02:05.45 | Qwell | glares at file |
| 02:05.47 | file | all the pop in my fridge has decided to turn into slush or freeze... yet everything else has not |
| 02:05.52 | JayTee52 | and yes, I was one of the two people in America that actually watched that movie |
| 02:05.52 | file | this perplexes me |
| 02:06.11 | media82 | no...i mean seriously...cant a credible person contact me ...are you all just idiots |
| 02:06.14 | Katty | pamples on file |
| 02:06.23 | jeev | file: awkward, my build on freebsd is not properly displaying anything on console. |
| 02:06.28 | JT | media82: Qwell is incredibly credible |
| 02:06.32 | drmessano | OMG |
| 02:06.36 | file | jeev: hrm? what'cha mean? |
| 02:06.36 | lmadsen | media82: we're all idiots most obviously |
| 02:06.36 | drmessano | Leonard Part 6 |
| 02:06.40 | Katty | Qwell don't like hugs. that makes me sad. |
| 02:06.44 | lmadsen | media82: what do you need a contact for? |
| 02:06.49 | Katty | we should get Qwell into the free hug campaign. |
| 02:06.50 | lmadsen | missed the convo |
| 02:06.53 | Katty | file: let's get signs. |
| 02:06.55 | Katty | file: and crayons |
| 02:06.57 | file | I got a hug from Qwell last time I was at the office |
| 02:07.03 | jeev | i was runing asterisknow on the local system but built 1.4.19 on my freebsd box at the datacenter, nothing is truely showing on the screen of asterisk -r |
| 02:07.03 | Qwell | RELUCTANTLY |
| 02:07.08 | Qwell | but <3 |
| 02:07.11 | jeev | i also enabled full logging and full log file was not created. |
| 02:07.16 | lmadsen | I ended up with a couple of hugs from Marko during IT360... |
| 02:07.18 | file | jeev: hrm, restart it? |
| 02:07.25 | Katty | well if i worked with file or Qwell i'd be hugging people every day. |
| 02:07.26 | drmessano | lmadsen: Imagine this.. Flash <> ANYTHING |
| 02:07.30 | Qwell | lmadsen: yeah, that's common :p |
| 02:07.31 | file | lmadsen: that's nothing special |
| 02:07.34 | drmessano | lmadsen: Imagine this.. Flash <> toothbrush, he's done it |
| 02:07.38 | Katty | sadly, southeast missouri has very few i want to come withing 20 feet of, much less hug. eww. |
| 02:07.43 | jeev | i did. |
| 02:07.45 | lmadsen | Qwell / file: yes, this I know |
| 02:08.06 | jeev | shoul i just tar up my asteriskNOW config and move it here? |
| 02:08.15 | Qwell | media82: so, not going to show me then? |
| 02:08.17 | file | jeev: did you install the sample configs? |
| 02:08.40 | jeev | how does asteriskNOW not have an ftp client builtin.. |
| 02:08.43 | jeev | yes i've installed those. |
| 02:09.22 | file | odd. |
| 02:09.44 | jeev | how POS is asterisknow, doesn't even have built in FTP |
| 02:10.11 | JayTee52 | Filezilla FTW! |
| 02:10.29 | jeev | anything with zilla in it is homo |
| 02:10.38 | drmessano | FileZilla is the best client |
| 02:10.57 | JayTee52 | it's nice to do sftp |
| 02:11.07 | Katty | in soviet russia, zilla files you. |
| 02:11.11 | JayTee52 | lol |
| 02:11.31 | *** join/#asterisk ManxPower (n=manxpowe@213.sub-75-203-228.myvzw.com) |
| 02:11.32 | jeev | ahh sftp |
| 02:11.34 | jeev | :> |
| 02:11.37 | JayTee52 | ManxPower, hi |
| 02:12.38 | jeev | file, i've got to run, my friend is btiching, hope you're here when i'm back! |
| 02:12.39 | jeev | thanks |
| 02:12.50 | file | I may be asleep. |
| 02:12.52 | bitzero | is fully willing to view this guys demo. |
| 02:12.58 | file | but I will be around tomorrow |
| 02:13.01 | bitzero | but he wont even tell me what it is he wants to show. |
| 02:13.10 | JayTee52 | media82!!!! we've got a winner here!!!! |
| 02:13.14 | Qwell | bitzero: flash <> anything |
| 02:13.15 | Katty | file: are you going to go nap now? |
| 02:13.20 | file | Katty: alas, no |
| 02:13.27 | file | Katty: I'm thinking of getting a drink |
| 02:13.29 | JayTee52 | flash <> bratwurst |
| 02:13.31 | drmessano | You guys really suck.. I did want to see the FLASH <> Skype gateway.. and all you did was piss him off with your tom foolery mockery monkey shenanigans |
| 02:13.34 | drmessano | Be ashamed |
| 02:13.42 | JayTee52 | lol |
| 02:13.45 | Nugget | heh |
| 02:13.49 | bitzero | Qwell: what does that even MEAN? |
| 02:13.56 | Qwell | bitzero: I don't know! |
| 02:13.56 | Nugget | pours a tequila shot for file |
| 02:14.03 | Qwell | he wouldn't show me |
| 02:14.04 | Katty | well. |
| 02:14.06 | JT | i personally wanted to see his SS7 <> Flash gateway |
| 02:14.07 | Katty | it's naptime for me. |
| 02:14.21 | drmessano | BRI <> Flash ? |
| 02:14.24 | media82 | so no one is interested in flash<->sip ? |
| 02:14.26 | file | I wouldn't mind a martini. |
| 02:14.30 | bitzero | DStar <> flash!!!?!?! |
| 02:14.35 | JT | bitzero: haha |
| 02:14.35 | drmessano | HA |
| 02:14.35 | Qwell | media82: flash<>sip has been done |
| 02:14.37 | outtolunc | i wanna see MONEY >> MYHAND |
| 02:14.38 | Qwell | show me skype |
| 02:14.43 | file | or a strawberry daiquiri |
| 02:14.49 | bitzero | JT: You know what DStar is? |
| 02:14.52 | JT | yes |
| 02:14.54 | drmessano | DStar <> 80m AM <> Flash |
| 02:14.56 | bitzero | They just put up a sat! |
| 02:15.04 | JT | icom did? |
| 02:15.05 | media82 | flash<video,audio><->sip |
| 02:15.07 | bitzero | can't wait to get the DStar module for his IC-V82 |
| 02:15.09 | media82 | ? been done? |
| 02:15.23 | Qwell | yes |
| 02:15.24 | media82 | where biatch? |
| 02:15.25 | *** join/#asterisk Darthclue (n=chatzill@76-233-19-118.lightspeed.snantx.sbcglobal.net) |
| 02:15.28 | drmessano | Red5 does all that, media82 |
| 02:15.38 | bitzero | JT: I dunno who paid for the DStar sat - it's operating on 2M and 440 |
| 02:15.46 | JT | ok |
| 02:15.48 | bitzero | JT: Info is on the ARRL website. |
| 02:15.51 | outtolunc | you really shouldn't call teh guys with @ biatches |
| 02:16.16 | JT | nah mate |
| 02:16.18 | Nivex | *sigh* D-Star's neat and all, but why did they have to go and use a proprietary codec? |
| 02:16.19 | JT | misunderstanding |
| 02:16.19 | media82 | i havent seen an rtmp impl with audio,video to -> sip |
| 02:16.21 | media82 | or 3g |
| 02:16.26 | JT | he's a non-european austrian |
| 02:16.37 | bitzero | Nivex: Thats not the only problem with DStar |
| 02:16.40 | JayTee52 | wearing lederhosen |
| 02:16.43 | bitzero | but it sure is a step in the right direction. |
| 02:16.46 | bitzero | so I'll support it. |
| 02:16.54 | media82 | dude...sorry to disappoint you...but im not austrian |
| 02:17.06 | JayTee52 | Heidi will be crushed |
| 02:17.08 | JT | media82: what are you? |
| 02:17.08 | Qwell | JT: flash<>IRC, obviously |
| 02:17.12 | outtolunc | he didn't dispute the lederhosen <G> |
| 02:17.21 | JayTee52 | hehe |
| 02:17.24 | media82 | thats like calling some shithead american in paris...a france twat |
| 02:17.35 | JayTee52 | or a twatwaffle |
| 02:17.46 | media82 | exactly |
| 02:17.47 | JT | twagette |
| 02:17.47 | Nivex | oh great, we've devolved into the ethnic slurs |
| 02:17.49 | media82 | :) |
| 02:18.15 | JT | media82: well you called the channel stupid europeans |
| 02:18.25 | JT | media82: most of us aren't in europe |
| 02:18.25 | JayTee52 | they had to stop shooting off fireworks at EuroDisney cuz the french kept throwing up their hands to surrender. |
| 02:18.31 | Nivex | "Mom! He started it!" |
| 02:18.34 | Nivex | all y'all shutup! |
| 02:18.51 | JT | Nivex: argh american slang, save me :D |
| 02:18.56 | drmessano | media82: http://osflash.org/red5 <--- |
| 02:19.02 | drmessano | media82: You're late |
| 02:19.05 | Nivex | JT: american south to boot :) |
| 02:19.07 | outtolunc | gets a fly swatter and reaches into the back of the channel |
| 02:19.08 | media82 | i stand on that ground...i mean..i was saying i had something great on my hands...and all you did is being dicks in a "rich" way |
| 02:19.12 | drmessano | media82: Been done |
| 02:19.28 | drmessano | media82: http://osflash.org/red5 |
| 02:19.33 | media82 | ok...so flash<->sip is no big deal |
| 02:19.39 | drmessano | and it actually works with Asterisk |
| 02:19.42 | drmessano | Well too |
| 02:19.55 | JT | media82: definitely an excellent reason to call us europeans, it all makes sense now |
| 02:19.56 | Qwell | that's what I've been trying to tell you |
| 02:20.00 | drmessano | Unlike youre crappy code |
| 02:20.03 | drmessano | your* |
| 02:20.17 | drmessano | Which doesn't work because Asterisk is too hard |
| 02:20.22 | media82 | JT> if you look at the back posts...mostly i was being made fun of |
| 02:20.30 | drmessano | Red5 works with that difficult-ass asterisk crap |
| 02:20.40 | Qwell | and yet you STILL haven't shown us anything to back up your claims |
| 02:20.42 | Qwell | SHOW ME |
| 02:20.49 | Nivex | screenshot or it didn't happen :) |
| 02:20.55 | drmessano | media82: Maybe you should beg the Red5 team to let you make the coffee |
| 02:21.02 | drmessano | media82: They made need a bitch |
| 02:21.04 | media82 | i can do rtsp<->flash right now |
| 02:21.06 | drmessano | may* |
| 02:21.08 | media82 | i said that before |
| 02:21.08 | Qwell | so can red5 |
| 02:21.10 | media82 | i also said |
| 02:21.12 | Qwell | I want to see skype |
| 02:21.13 | tzanger | I love how the americans make fun of the french without knowing any history whatsoever |
| 02:21.18 | drmessano | Red5 does rtsp |
| 02:21.21 | drmessano | and SIP |
| 02:21.33 | media82 | that whoever sends me a personal message first will get to see it |
| 02:21.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:21.42 | media82 | so far...noone did |
| 02:21.56 | Qwell | 'If an admin asks, I'll show them' |
| 02:21.59 | Qwell | I'm an "admin" |
| 02:22.01 | drmessano | Qwell is an admin |
| 02:22.03 | Qwell | show me |
| 02:22.05 | drmessano | ahem, sorry |
| 02:22.15 | media82 | and SIP...actually its -> you can do one...you can do all |
| 02:22.16 | *** join/#asterisk andresmujica (n=andresmu@190.24.108.22) |
| 02:22.28 | drmessano | Red5 does that |
| 02:22.36 | media82 | i doubt it |
| 02:22.50 | drmessano | It sure does |
| 02:23.05 | drmessano | You're late to the table |
| 02:23.05 | media82 | flash<->rfc2429 |
| 02:23.13 | JayTee52 | bbiab, gonna roast some more coffee |
| 02:23.15 | media82 | it doesnt believe me |
| 02:23.30 | drmessano | flash<>rfc2178 too |
| 02:23.39 | Qwell | lots of things do h.263 |
| 02:23.50 | Qwell | h.263 isn't novel by any means |
| 02:23.52 | drmessano | h.263 is... so 2001 |
| 02:23.58 | Qwell | 1999, actually |
| 02:24.03 | Qwell | 98* |
| 02:24.07 | drmessano | Qwell: I was a late bloomer |
| 02:24.11 | Qwell | RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video (H.263+) |
| 02:24.18 | Qwell | ^ RFC2429 |
| 02:24.37 | drmessano | I'd love to see this Flash <> Skype |
| 02:24.40 | lmadsen | tzanger: I think that's funny too |
| 02:24.48 | tzanger | lmadsen: hey, how was it360 |
| 02:24.53 | lmadsen | tzanger: meh |
| 02:24.57 | lmadsen | it was fun with the people I saw :) |
| 02:25.00 | Qwell | flash <> rfc 1149 |
| 02:25.02 | media82 | nono you got it all wrong |
| 02:25.05 | lmadsen | the conference itself was too lightly attended |
| 02:25.11 | media82 | 2190 is 1996 |
| 02:25.13 | lmadsen | me <> myself |
| 02:25.16 | media82 | 2429 is 1998 |
| 02:25.33 | media82 | h263 might be wrong, but that is what adobe uses |
| 02:25.42 | tzanger | lmadsen: hmm, how about taug after? |
| 02:25.45 | media82 | and it is also a h323 standard |
| 02:25.50 | media82 | and sip likes it also |
| 02:25.52 | tzanger | did digium stop by? |
| 02:25.57 | drmessano | That's old school |
| 02:26.03 | tzanger | I have been sick as a dog this week |
| 02:26.06 | plik | oooh, did I miss anything fun,,, is it worth the scroll back? |
| 02:26.07 | craigk | hmmm genzaptelconf is not finding one of the channels on my tdm400p card ... anybody else seen this ? |
| 02:26.10 | tzanger | that and my car engine exploded :-( |
| 02:26.12 | media82 | btw. there are also h263 hd movies out there |
| 02:26.23 | lmadsen | ya, there was 3 peeps from digium, krisk, and a bunch of others |
| 02:26.26 | lmadsen | the taug meeting was decent |
| 02:26.41 | Qwell | plik: no |
| 02:26.44 | lmadsen | has some h.263 hd movies |
| 02:27.01 | [hC] | has some h.264 hd movies |
| 02:27.02 | plik | Qwell: thought as much :0 cheers thoough |
| 02:27.05 | lmadsen | but 54mbps wireless connection isn't enough to stream them to my xbox 360 |
| 02:27.36 | media82 | are there any directshow programmers are there? |
| 02:27.39 | drmessano | media82: Can do any protocols that haven't already been done? |
| 02:27.50 | drmessano | media82: Seems youre a bit late |
| 02:28.07 | media82 | protocol wise? |
| 02:28.09 | media82 | true |
| 02:28.18 | media82 | have you implemented 2190 before? |
| 02:28.23 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 02:28.27 | media82 | have you implemented 2429 before? |
| 02:28.30 | drmessano | Well Flash <> Everything you mentioned has been done |
| 02:28.40 | drmessano | Where is the "wow"? |
| 02:29.07 | media82 | it has..show me... |
| 02:29.08 | lmadsen | it's right there ^^^ |
| 02:29.08 | plik | over there, next to the pow ? |
| 02:29.13 | media82 | i can show you any time |
| 02:29.23 | drmessano | media82: I provided a link to one |
| 02:29.28 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:29.32 | drmessano | media82: Which is more than you have done |
| 02:29.57 | media82 | sorry..i did not see your link... |
| 02:30.07 | drmessano | media82: I posted it 5 times |
| 02:30.23 | media82 | but i have also not seen yours...obviously you dont care anyway...since its already been done |
| 02:30.28 | drmessano | media82: http://osflash.org/red5 |
| 02:30.28 | plik | point out the 'page up' button |
| 02:30.29 | drmessano | media82: http://osflash.org/red5 |
| 02:30.33 | drmessano | Thats 2 more for ya |
| 02:30.49 | drmessano | Red5 does all you've mentioned, and more |
| 02:30.52 | drmessano | So, "wow" me |
| 02:30.58 | outtolunc | says do it again, do it again <G> |
| 02:31.15 | media82 | so where does it say that it does rfc2190<->rtmp |
| 02:31.36 | drmessano | Guess you need to read |
| 02:32.13 | media82 | guess...you need to read---because on2 h263 does not conform to the standard |
| 02:33.05 | media82 | next-< |
| 02:33.13 | drmessano | So far i've provided more links than you.. You've proven nothing more than you can copy/paste links to standards off of some web page |
| 02:33.37 | media82 | i told you all to provide me with a link...but no one showed up |
| 02:33.42 | drmessano | Qwell did |
| 02:33.48 | drmessano | He mentioned it a dozen times |
| 02:33.51 | drmessano | So you fail |
| 02:34.03 | drmessano | ~failburger |
| 02:34.04 | jbot | You fail at life. Have a failburger with fail fries and a large diet fail. |
| 02:34.11 | media82 | in a pm...i only got 1mbit |
| 02:34.12 | media82 | up |
| 02:34.19 | drmessano | So? |
| 02:34.21 | drmessano | He PM'ed you |
| 02:34.21 | media82 | so this is not hte place |
| 02:35.09 | drmessano | Sounds to me like you want someone to e-mail you for malicious purposes.. which is why you've been so insistant on it. I'm still not convinced you're not some script kiddie. |
| 02:35.16 | media82 | <PROTECTED> |
| 02:35.40 | drmessano | But anyway.. I've said my piece. Peace out |
| 02:35.40 | media82 | drmessano> why dont you provide a link and i will show you |
| 02:36.25 | media82 | you know ... im not lying..i worked really hard on this...i find it funny that this is being made a joke of |
| 02:36.57 | djs | A couple of weeks ago, I asked for recommendations on an unlimited * compatible voip provider, and did not understand that they don't work quite like that really. So, now I would appreciate advice on just a good voip provider with reasonable rates. I would like 4 phone numbers, if possible. Hoping I can find something for $50 or less/mo with some minutes included in that. I hope this is a more reasonable request now. |
| 02:39.52 | *** join/#asterisk Faithful (n=Faithful@vg102.vodafone.com.au) |
| 02:40.36 | media82 | and when i say rtsp/sip/3g<->skype i really mean it |
| 02:41.15 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:41.15 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 02:41.17 | lmadsen | daft punk! |
| 02:41.27 | drmessano | daft punk <> Flash? |
| 02:41.42 | media82 | yo momma<>skype |
| 02:41.59 | JayTee52 | I sympathize. I was met with the same skepticism when I tried to raise venture capital for my anti-gravity machine. |
| 02:42.18 | drmessano | I just laughed so hard my wife scoffed at me |
| 02:42.37 | media82 | JT> well then you know...whats funny is that noone even tried to contact me |
| 02:42.55 | media82 | you know if you contact me i will actually show you |
| 02:43.27 | JayTee52 | yeah, but if you show me yours does that mean I have to show you mine? |
| 02:43.36 | media82 | fu |
| 02:43.40 | drmessano | Sweep the leg |
| 02:43.46 | JayTee52 | hehehe |
| 02:44.37 | media82 | its funny...because it does work...noone seems to care.....everyone just seems to care that iam just a liar..... |
| 02:45.07 | drmessano | We don't think you're a liar.. |
| 02:45.23 | media82 | but |
| 02:45.41 | drmessano | We just think you need more attention at home, and mom is too busy with your new stepdad to care. I love that movie. |
| 02:46.00 | media82 | so you think what i say i basically false |
| 02:46.19 | drmessano | Not at all |
| 02:46.23 | drmessano | But see |
| 02:46.23 | JayTee52 | I think it's pathological but then I'm not a psychiatrist |
| 02:46.50 | drmessano | I am doctor, media82.. and I want to help you |
| 02:47.11 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:49.18 | JayTee52 | "Resistance is futile! We are the Flash <> SIP" |
| 02:49.49 | andresmujica | hi all |
| 02:49.56 | lmadsen | howdy |
| 02:49.57 | JayTee52 | hi |
| 02:50.08 | andresmujica | i'm having some issues with bandwidth. |
| 02:50.22 | andresmujica | i'm using g729 but at peaks times i've got quality issues... |
| 02:50.24 | JayTee52 | hums "Always look on the bright side of life" from Monty Python's Life of Brian |
| 02:50.30 | drmessano | He's PM'ed me, and thus far, provided me with nothing |
| 02:50.37 | andresmujica | i wonder if changin to g723.1 or ilbc could do any better... |
| 02:50.50 | lmadsen | sounds like packet loss or jitter possibly |
| 02:50.53 | JayTee52 | I'd avoid ilibc |
| 02:50.57 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 02:50.57 | JayTee52 | it's deprecated |
| 02:51.00 | drmessano | andresmujica: Where did you get G723? |
| 02:51.17 | lmadsen | I doubt changing codec will fix anything |
| 02:51.23 | drmessano | Since you can only legally have/use G723 with a digium transcoder card |
| 02:51.29 | andresmujica | compiled ones. i'm at a no software patents country. |
| 02:51.37 | andresmujica | anyway we bought teh g729 |
| 02:52.24 | *** join/#asterisk s0lid (n=s0lid@210.213.198.61) |
| 02:53.48 | andresmujica | ohh didn't knew about that with the g723! |
| 02:59.13 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 03:00.00 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
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| 03:13.24 | drmessano | sweep the leg |
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| 03:13.26 | drmessano | no mercy |
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| 03:14.51 | jameswf-home | I need to do a realtime app that takes outboung audio and plays it backwards |
| 03:17.26 | JayTee52 | backwards? |
| 03:17.40 | JayTee52 | that'd sound a bit odd, wouldn't it :-) |
| 03:19.16 | drmessano | ?dluow tI |
| 03:19.29 | drmessano | haaaN |
| 03:20.30 | JayTee52 | "llac ruoy tcerid I woh, oCparCmoC ot emocleW" |
| 03:20.58 | drmessano | I'm beginning to think that media82comm is vaporware |
| 03:21.02 | drmessano | I want my weekend back |
| 03:21.26 | JayTee52 | hsalf <> epykS |
| 03:22.41 | drmessano | I think he lost his keygen |
| 03:23.21 | JayTee52 | lost more than that most likely |
| 03:23.36 | JayTee52 | I've gotta get some REM. talk to ya later! |
| 03:23.54 | JayTee52 | nite everyone |
| 03:23.54 | drmessano | Later |
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| 03:39.24 | CpuID2 | hey ppls, anyone here found a decent way to handle calls and sms text messages on a single gsm module with voice calls integrated with asterisk? |
| 03:39.32 | CpuID2 | i dont really wanna have to have 2 separate devices with 2 SIMs if i can avoid it |
| 03:40.02 | CpuID2 | and preferably not like a device with an analog FXS port on it and serial for SMS, id like something so the calls are digital all the way eg. sip based or something |
| 03:42.05 | Kalamansi | CpuID2 what is your distro? |
| 03:49.12 | CpuID2 | gentoo |
| 03:49.25 | CpuID2 | right now just more looking for potential options hw wise |
| 03:50.19 | CpuID2 | as i mentioned, goal being to have one gsm device/module/etc with a single SIM that i can use for calls and sms messages, sms will be coming from a set of scripts somewhere (mainly system/network monitoring) and calls will be to/from asterisk, preferably without going via an analog fxo/fxs channel to the hw |
| 03:50.29 | CpuID2 | ive been playing with the 2n voiceblue lite, but its a bit of a POS |
| 03:58.38 | jeev | file! |
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| 04:02.50 | drmessano | Is that the new version of "fore"? |
| 04:02.55 | drmessano | I didnt get the memo |
| 04:04.46 | jeev | uh, i built this on my fbsd box and i'm not getting ANY logging on the screen other than sip debug . |
| 04:09.14 | jameswf-home | ~bsd |
| 04:09.14 | jbot | BSD is a UNIX operating system. An asterisk port is currently availible if you feel you must, or a way to set your pc back 30 years, progress is overrated |
| 04:11.19 | jeev | heh |
| 04:11.21 | jeev | hater |
| 04:11.24 | CpuID2 | w00t managed to get this voiceblue doing what i wanted :P |
| 04:11.59 | Mavvie | That isn't difficult, it is just a sip gateway. |
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| 04:15.35 | jeev | is the loggnig scheme between a default set up... asteriskNOW and 1.4.19 totally different? i'm not getting console reports anymnore |
| 04:15.57 | Kalamansi | ~linux |
| 04:15.58 | jbot | i heard linux is the cure for cancer, AIDS and slavery to corporations |
| 04:16.12 | Kalamansi | ~gentoo |
| 04:16.13 | jbot | i heard gentoo is foo |
| 04:16.18 | Kalamansi | ~windows |
| 04:16.19 | jbot | well, windows is a 32 bit hack on a 16 bit operating system, originally designed for an 8 bit CPU, with a 4 bit system bus, made by a 2 bit company that can't stand 1 bit of competition... or the World of Warcraft bootloader, or the most important collection of bugs |
| 04:16.29 | Kalamansi | nice |
| 04:16.40 | Kalamansi | ~compatible for asterisk |
| 04:16.52 | Kalamansi | ~os |
| 04:16.53 | jbot | somebody said os was (Operating System) The program that allows you to access the basic functions of your computer. It is the minimum software required to run a program. The best one by far is the MacOS. |
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| 04:43.10 | paulq | Queue penalty dont work as I expect :( |
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| 04:43.23 | paulq | If _MAX_ is set I would expect it to ring _MAX_ and < |
| 04:43.35 | paulq | But it just rings MAX= :-/ |
| 04:43.42 | paulq | So if I have a agent with 1 and the other with 2 |
| 04:43.54 | paulq | and 1 doesnt pickup the queue and it timesout to pen 2 |
| 04:43.58 | paulq | I would expect 1 and 2 to ring |
| 04:44.25 | paulq | (Does that seem right?) |
| 04:48.04 | jameswf-home | neat : http://cb.vu/unixtoolbox.xhtml |
| 04:49.08 | paulq | I'd assume a IT worker knows all that? lol |
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| 04:51.39 | drmessano | That's hardcore |
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| 04:59.33 | jeev | guys, i'm having an issue, i went from asteriskNOW to asterisk 1.4.19 on my freebsd and the console (logging) isn't the same, i've copied logger and it's still the same.. it's not identical to how asteriskNOW was logging. |
| 05:01.41 | jeev | logger.conf has full, but full wont be created. |
| 05:03.53 | jeev | cock, cause it runs in verbose, i'm an idiot |
| 05:08.16 | jeev | any reason why asteriskNOW starts as root ? |
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| 05:21.19 | *** join/#asterisk tinapa (n=niwre@67.159.178.21) |
| 05:21.33 | tinapa | hi, anyone have a solution for this problem? /msg NickServ IDENTIFY |
| 05:21.42 | tinapa | Apr 11 06:15:56 WARNING[29053]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 05:21.50 | tinapa | sorry wrong paste hehe |
| 05:23.43 | jeev | heh |
| 05:23.52 | jeev | maybe you need to create the device |
| 05:25.00 | tinapa | jeev, any ideas how can i do that? |
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| 05:27.07 | paulq | Does the box have a sound card? |
| 05:28.14 | paulq | cd /dev/ |
| 05:28.15 | paulq | ./MAKEDEV -v sound |
| 05:28.15 | paulq | ? |
| 05:28.19 | tinapa | paulq it doesnt have |
| 05:28.24 | tinapa | its a rackmount server |
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| 05:38.32 | paulq | Grr |
| 05:38.40 | paulq | anyway to stop a phone from sending circuit-busy after ringing for X amount of time |
| 05:38.43 | paulq | Its a Cisco |
| 05:38.55 | paulq | I want it to ring forever |
| 05:40.44 | tinapa | paulq: i got it, i just noload chan_oss.so in the modules.conf |
| 05:41.35 | paulq | well yeah :P |
| 05:41.39 | paulq | thats not fixing its ignoring |
| 05:41.45 | paulq | you dont need chan_oss without a sound card |
| 05:41.46 | paulq | sorry |
| 05:41.51 | paulq | Working on my own thing hehe |
| 05:41.54 | tinapa | okay |
| 05:42.06 | paulq | Give me a shout if you need anything |
| 05:42.56 | tinapa | hehe i need to setup an outbound rout to my SIP trunk, cant find a clue |
| 05:43.09 | tinapa | rout/route |
| 05:44.03 | the_5th_wheel | Hi. I use a GRNvoip sip trunk, and it seems that its connection via ulaw, even tho i have a disallow=all allow=G729 |
| 05:44.52 | the_5th_wheel | What i have seen tho is that the actuall sip connection end up being to another server, other than the one i have defined |
| 05:45.07 | the_5th_wheel | Any ideas how i can change this |
| 05:45.15 | the_5th_wheel | Since bandwith is rather expencive down here |
| 05:46.52 | jeev | is MOH not recommended on fbsd? due to mpeg1234 or some shit ? |
| 05:57.44 | Corydon76-dig | MOH no longer requires the use of mpg123 |
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| 05:57.58 | jeev | ahh |
| 05:59.24 | jeev | how can i debug MOH, doesn't seem to be functioning.. i click hold, it starts then stops. |
| 05:59.47 | jeev | [Apr 10 22:59:02] VERBOSE[7973] logger.c: -- Started music on hold, class 'default', on channel 'SIP/trunk_1' |
| 05:59.54 | jeev | [Apr 10 22:59:02] VERBOSE[7973] logger.c: -- Stopped music on hold on SIP/trunk_1 |
| 06:01.35 | jeev | ahh, it's still loading the wrong dir |
| 06:06.49 | jeev | Asterisk Ready. |
| 06:06.49 | jeev | *CLI> Warning, flexibel rate not heavily tested! |
| 06:06.49 | jeev | mpg123: Can't rewind stream by 33 bits! |
| 06:06.49 | jeev | Warning, flexibel rate not heavily tested! |
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| 06:16.38 | Corydon76-dig | jeev: are you using 1.2 or 1.4? |
| 06:17.09 | tinapa | i have a [cbiout] trunk, how can i make a dialplan that if the extensions dial 9131399999 it will go to that trunk? |
| 06:17.55 | Corydon76-dig | jeev: in musiconhold.conf, you want to be using mode=files and transcode your files to 8000Hz, single-channel, uncompressed wav |
| 06:18.14 | tinapa | exten => _9.,1,Dial,SIP/${EXTEN-1}@cbiout,tr doesnt work |
| 06:18.22 | jeev | 1.4 |
| 06:18.48 | jeev | on Corydon76-dig, i will try that now.. exactly how much bandwidth does this take? to play per "leg" |
| 06:18.49 | Corydon76-dig | tinapa: you're missing a ':' |
| 06:19.02 | tinapa | Corydon76-dig in which part? |
| 06:19.03 | Corydon76-dig | jeev: same as voice |
| 06:19.12 | Corydon76-dig | tinapa: ${EXTEN:1} |
| 06:19.23 | tinapa | ok let me try |
| 06:19.31 | Corydon76-dig | tinapa: you're also using deprecated syntax |
| 06:19.32 | jeev | is there any bsd software you know or linux, whatever.. that i could compile for transcoding ? |
| 06:19.45 | Corydon76-dig | jeev: try sox |
| 06:20.09 | jeev | ok,thanks.. so i was using rfc2833 but it wasn't working when i'd keypress.. wasn't working properly.. wholesaler changed me to inband.. but now, dialing out takes at least 10-15 sec |
| 06:20.16 | tinapa | Corydon76-dig what is the new syntax look like? |
| 06:20.26 | Corydon76-dig | tinapa: Dial(...) |
| 06:22.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:22.25 | tinapa | ok thanks Corydon76-dig |
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| 06:23.05 | jeev | Corydon76-dig, you heard of anything like that before? inband causing outgoing lag ? |
| 06:23.37 | Corydon76-dig | Nope |
| 06:23.41 | jeev | damn |
| 06:23.49 | jeev | i'll work it out with the provider i guess, it literally takes 10-15 seconds to dial out. sucks |
| 06:23.57 | jeev | i'm building sox. |
| 06:24.52 | jeev | mpg123 -s audio01.mp3 > audio01.pcm |
| 06:25.10 | jeev | -w <filename> write Output as WAV file |
| 06:27.02 | jeev | bah, /dev/dsp issue. |
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| 06:44.30 | tinapa | Corydon76-dig: i added this line in the context: exten => _XXXXXXXXXXX,1,Dial(SIP/cbiout/${EXTEN:1},20,tr) |
| 06:44.51 | tinapa | the phones said account disabled everytime they dial |
| 06:46.43 | tinapa | or it says service is unavailable |
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| 06:53.24 | jeev | Corydon76-dig |
| 06:53.24 | jeev | [Apr 10 23:52:33] WARNING[30497]: format_wav.c:148 check_header: Not in mono 2 |
| 06:53.24 | jeev | [Apr 10 23:52:33] WARNING[30497]: file.c:322 fn_wrapper: Unable to open format wav |
| 06:53.24 | jeev | [Apr 10 23:52:33] WARNING[30497]: res_musiconhold.c:265 ast_moh_files_next: Unable to open file '/usr/local/asterisk/moh/9th': No such file or directory |
| 06:53.24 | jeev | -- Stopped music on hold on SIP/trunk_1 |
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| 07:31.36 | bougie | hello :) |
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| 07:34.59 | Telemac | Hello |
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| 07:39.25 | Sir0x | good morning |
| 07:40.04 | Telemac | I'm trying to setup B410 with chan_misdn into asterisk, but I don't succeed in getting misdn ports up. I'm using a 2.6.22 kernel, asterisk-1.2.27, zaptel-1.2.25, libpri-1.2.7 and latest install-misdn-mqueue from beronet. I've tried with ports as TE_PTP and TE_PTMP (te_ptmp=1,2,3,4 or te_ptp=1,2,3,4 in /etc/misdn-init.conf) but nothing work. Is there anyone having a idea how can I fix that or at least having more information about what's wrong ? |
| 07:44.17 | *** join/#asterisk mbranca (n=matteo@mi-gw1.voismart.net) |
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| 07:51.20 | rico29 | hi |
| 07:51.30 | rico29 | I need some help again |
| 07:52.01 | rico29 | I want users to be able to connect to my pbx without registration |
| 07:52.15 | rico29 | this first thing works fine, with autocreatepeer |
| 07:52.52 | rico29 | next, I want to redirect people who connect without registration (with autocreatepeer) to a special context |
| 07:53.02 | rico29 | and I don't know how to do |
| 07:53.58 | rico29 | i've tried regexten=context_free_reg but it doesn't seem to work |
| 08:00.47 | mort_gib | Hi, I upgraded from 4.1.17 to 4.1.19 |
| 08:00.53 | mort_gib | Compiled all modules |
| 08:00.56 | mort_gib | Now i get |
| 08:00.57 | mort_gib | <PROTECTED> |
| 08:01.02 | mort_gib | for several modules |
| 08:01.06 | mort_gib | Ideas?? |
| 08:01.56 | rico29 | what did you upgrade ? |
| 08:02.05 | mort_gib | Asterisk |
| 08:02.15 | rico29 | asterisk 4.1 ? |
| 08:02.18 | rico29 | o_O |
| 08:02.23 | rico29 | :) |
| 08:02.26 | mort_gib | from 4.1.17 to 4.1.19 |
| 08:02.42 | rico29 | 1.4.17 |
| 08:02.46 | rico29 | 1.4.19 |
| 08:02.47 | rico29 | no ? |
| 08:03.16 | rico29 | you've install asterisk-addons , |
| 08:03.18 | rico29 | ? |
| 08:03.57 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 08:04.49 | mort_gib | I think so, I'll try again |
| 08:04.49 | mort_gib | but surely voicemail is included in Asterisk |
| 08:04.51 | mort_gib | not addons |
| 08:07.08 | rico29 | ok |
| 08:07.16 | rico29 | mort_gib, > any idea for my problem ? |
| 08:07.22 | *** join/#asterisk bps (n=none@host.250.19.23.62.rev.coltfrance.com) |
| 08:08.03 | mort_gib | well, in sip.conf you would use context=your_fancy_context |
| 08:08.45 | mort_gib | Why would you want to use autocreatepeer |
| 08:09.59 | rico29 | because I want peers to be able to register to my asterisk server from anywhere in my lan |
| 08:10.11 | rico29 | without creating any entry in sip.conf or in realtime |
| 08:10.18 | mort_gib | Yes, but you don't need autocreatepeer for that... |
| 08:10.23 | rico29 | ah ? |
| 08:10.29 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
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| 08:10.45 | mort_gib | You would use that option if you want EVERYBODY to be able to register and you your server |
| 08:11.24 | *** join/#asterisk matt_ (n=matt@2001:770:168:1:20b:cdff:fe04:843a) |
| 08:11.26 | rico29 | mort_gib, > what's the difference with what I said ? |
| 08:11.35 | rico29 | and what's the best way to do this ? |
| 08:12.28 | mort_gib | Uhm, I would create users in sip.conf and map them to extensions in extensions.conf |
| 08:12.49 | mort_gib | They can connect to the server if they can telnet serverip 5060 |
| 08:12.49 | Nugget | telnet is eeeeeeevil! |
| 08:12.57 | mort_gib | Yes, quite |
| 08:13.02 | mort_gib | but usefull |
| 08:13.36 | rico29 | mort_gib, > i said I don't want to create anything in sip.conf |
| 08:13.48 | mort_gib | Why?? |
| 08:13.55 | rico29 | I don't want to create thousands and thousands peers |
| 08:14.09 | rico29 | pecause I want everybody to be able to register on my serv |
| 08:14.27 | mort_gib | But you don't mind creating thousands of entries in extensions.conf |
| 08:14.27 | rico29 | the lan is for testing, but asterisk is on the wan |
| 08:14.53 | rico29 | mort_gib, > for extensions.conf, if I can redirect peers tu a context, it will not be a problem |
| 08:15.03 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 08:15.06 | rico29 | i'll use an AGI or something else |
| 08:15.54 | rico29 | i juste want to redirect the users which go through "autocreatepeer" in a defined context |
| 08:15.57 | mort_gib | I think it will. If you don't have extensions how can the users call each other?? |
| 08:16.21 | rico29 | mocker, > I think that with an AGI + realtime, it's possible |
| 08:16.27 | rico29 | bot that's not my question |
| 08:16.38 | rico29 | mort_gib, sorry, not mocker |
| 08:16.51 | mort_gib | I'm sure it is... |
| 08:17.13 | mort_gib | But outside my expertise |
| 08:18.00 | rico29 | so no idea for context defining with autocreatepeers ? |
| 08:18.14 | mort_gib | I would be tempted to use a database to create users, and populate sip.conf and extensions.conf from that database |
| 08:18.38 | rico29 | mort_gib, > I use realtime, that's better |
| 08:18.46 | mort_gib | I'm sure it is... |
| 08:19.43 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 08:19.53 | mort_gib | Haven't tried out that yet, but it says in the wiki that aoutocreatepeers uses global options, so you would use context=auto_created in the global section I guess |
| 08:20.14 | rico29 | ok, let me try |
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| 08:22.00 | mort_gib | What kind of installation are you trying to create?? Obviously not a standard PBX type?? |
| 08:22.58 | rico29 | mort_gib, > thanks a lot, seems to work |
| 08:23.04 | mort_gib | :-) |
| 08:23.23 | rico29 | mort_gib, > i'm creating a PABX for a university in france |
| 08:23.43 | mort_gib | P"A"BX ?? |
| 08:23.44 | rico29 | not creating it, but configuring asterisk |
| 08:23.56 | rico29 | PBX, or IPBX, or everything you want ;) |
| 08:24.01 | mort_gib | Please explain |
| 08:24.02 | rico29 | :D |
| 08:24.18 | rico29 | I have to |
| 08:24.20 | rico29 | mmh |
| 08:24.35 | rico29 | hard forme to explain everything in english :p |
| 08:24.49 | mort_gib | So you are placing Asterisk in front of a proprierity PBX to be able to use VOIP?? |
| 08:25.05 | rico29 | No, I only Use asterisk |
| 08:25.07 | mort_gib | -Sorry my French is very rusty :-( |
| 08:25.11 | rico29 | :) |
| 08:25.40 | rico29 | i have to make internal users reachable, with localisation |
| 08:25.47 | rico29 | and many services |
| 08:26.11 | mort_gib | So how are you handling extensions |
| 08:26.40 | rico29 | then I have to allow external users, like you for example, to reach teachers by dialing teachername@university in a softphone |
| 08:26.50 | rico29 | what does "handling" means ? |
| 08:27.20 | bps | gérer dans ce cas |
| 08:27.31 | mort_gib | Well, I thought that in Asterisk calling any specific user is done via the extension |
| 08:28.02 | mort_gib | so you map an extension in extensions.conf say TEACHER=SIP/teacher |
| 08:28.17 | rico29 | no |
| 08:28.24 | rico29 | i will have to use an agi i think |
| 08:28.28 | rico29 | for dialing by name |
| 08:28.37 | rico29 | or realtime |
| 08:28.40 | rico29 | it may work |
| 08:28.44 | rico29 | i don't know |
| 08:28.45 | rico29 | :) |
| 08:28.48 | rico29 | i'll see |
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| 08:29.10 | mort_gib | Ah, but then you are out of luck because autocreatepeer uses user part of the Contact: header field's URL. (snip) |
| 08:30.02 | mort_gib | So unless you hand out usernames, or userconvensions, and evaluate these runtime any user would be able to label him/herself as Bruce Wayne |
| 08:30.21 | rico29 | that's not my problem :D |
| 08:30.56 | mort_gib | I like the way you think, but you, or the helpdesk person will be well known on campus :-) |
| 08:31.26 | mort_gib | I hope the helpdesk person is studying to become a accounant! |
| 08:31.39 | rico29 | no, i don't think so, 'cause it's really experimental |
| 08:31.39 | mort_gib | Or better Financial Controller |
| 08:31.46 | rico29 | :) |
| 08:32.18 | mort_gib | Ok, but you still have to come up with a way to make any registration unique |
| 08:32.31 | rico29 | mort_gib, > yes |
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| 08:32.56 | mort_gib | So an automated handing out of user (header) id's |
| 08:33.00 | rico29 | is there a way to use regexps in exten ? like exten => _[a-z],1,... |
| 08:33.04 | mort_gib | ie student numbers |
| 08:33.11 | mort_gib | -If you have them |
| 08:33.21 | rico29 | i have nothing |
| 08:33.40 | mort_gib | But your poor end users?? |
| 08:34.14 | rico29 | hum... |
| 08:34.19 | rico29 | "poor users" |
| 08:34.20 | rico29 | :D |
| 08:34.20 | mort_gib | I don't think so, but there are other ways |
| 08:34.29 | mort_gib | -Fresh outta luck ;-) |
| 08:35.05 | mort_gib | To me it sounds a bit like mayhem, and very interesting |
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| 08:41.45 | vlt | Hello. Why is _0NXXX. more specific than _0[789]00XX.? |
| 08:42.32 | vlt | I expected the second pattern to win for exten 090055555 |
| 08:50.55 | *** join/#asterisk Jocko (i=daemon@c-67-168-165-185.hsd1.wa.comcast.net) |
| 08:51.51 | Jocko | Hello, Is there such a thing as a context wide variable? |
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| 08:53.36 | bps | http://www.voip-info.org/wiki/index.php?page=Asterisk+variables |
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| 08:56.40 | Jocko | bps: I read that, but I'm not clear on inherited scope with something like Set(__VARIABLE=x). Will this carry the variable though the whole context? |
| 09:00.11 | bps | you can declare a variable value for a context, but it won't get to next channel created on this context |
| 09:00.31 | bps | you can use global var like MYCONTEXT_MYVAR |
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| 09:14.06 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 09:14.09 | Chris-NB | hi |
| 09:15.30 | Chris-NB | can someone explain what the parameters nationalprefix and internationalprefix in zapata.conf exactly do? |
| 09:16.02 | Chris-NB | is this correct: if the incoming isdn call hast a ton set to national, then the national prefix is added to the callerid |
| 09:16.29 | Chris-NB | if the incoming isdn call has ton set to internation, then the internationalprefix is added to the callerid. <- correct? |
| 09:18.30 | Chris-NB | or do I miss understand these parameters? |
| 09:19.18 | Jocko | bps:thanks |
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| 09:23.19 | ice_croft | ~book |
| 09:23.20 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
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| 09:34.27 | Jocko | BPS: I'm trying to set a context wide variable that is unique to the current caller. Would something like this "exten => 10,1,Set(MESSAGE${AUTH_MAILBOX}=${EPOCH},g)" allow me to call the variable MESSAGE${AUTH_MAILBOX} in a later statement like "exten => 20,1,GotoIf($[${MESSAGE${AUTH_MAILBOX}}=123456]?30,1)" ? |
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| 09:35.46 | McDouglas | anyone could tell me whats wrong with this line? exten => 08,n,Set(foo=${IF($[ ${CALLERID(num)} = ""]?"Ismeretlen Hivo")}) |
| 09:39.14 | rico29 | 'if' is a function ? |
| 09:39.44 | McDouglas | according to this: http://www.voip-info.org/wiki/view/Asterisk+func+if yes |
| 09:40.09 | rico29 | thankjs |
| 09:41.14 | rico29 | i don't see where the proble ims, sorry |
| 09:41.38 | Chris-NB | can someone explain what the parameters nationalprefix and internationalprefix in zapata.conf exactly do? |
| 09:41.47 | Chris-NB | it's not mentioned in the * book |
| 09:43.15 | bps | Jocko: use for SetGlobalVar, but you need a better value than EPOCH to identify the caller |
| 09:45.08 | Jocko | bps: I'm retaining messages that multiple callers create over multiple calls so I need something unique to the caller and the time they created it. |
| 09:47.32 | bps | Jocko: I just saw setglobalvar isn't necessary anymore, set with g just do fine |
| 09:47.51 | bps | Jocko: long time I didn't do thing inside the dialplan |
| 09:48.04 | Jocko | BPS: all the variables I create during the call will be set to null at the end of the caller's session. I have them going through 10 extensions in the same context and want to use the variable in each extension. |
| 09:48.30 | Jocko | BPS: Cool, I'll try the set +g and null them out at the end of the call. |
| 09:49.25 | bps | if you can use agi or fastagi, especialy if you're integrating things with other systems |
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| 09:50.37 | MDK2MDK | hello everybody :) |
| 09:52.11 | Jocko | bps: I am, but I need to walk the caller through a buch of prompt to get the required variable values and messages before I hand them off to an agi perl script. |
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| 10:01.48 | mvanbaak | hurray. nice netsplit there |
| 10:02.15 | Telemac | I'm trying to use B410 with asterisk and chan_misdn. I can see my 4 TE_PTMP isdn port and channels is asterisk, but I don't succeeded in getting incoming call for test at least to a SIP phone. Ports are defined to be on context from-pstn in misdn.conf and and I got "exten => s,1,Goto(otherctx|103|1)" in that context in extensions.conf... So if anyone could tell me what's wrong ? |
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| 10:19.06 | dworkin | can iax trunk go through 2 NATs (on one end) without problems? yes/no answer is enough, i'll research the rest. |
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| 10:28.02 | jblack | dworkin: One way would be to use openvpn to make two tunnels. |
| 10:28.22 | jblack | which you could agregate into one fatter virtual tunnel. |
| 10:29.35 | dworkin | jblack, of that i am aware but i was hoping for a simpler way. and since iax is known for nat friendliness, i wonder if it can do that. |
| 10:34.24 | Telemac | It seems that my B410 and chan_isdn is ok (Port 1 Type TE Prot. PMP L2Link UP L1Link:UP Blocked:0 Debug:20), but I doesn't get incoming call, but I don't get incoming call for test at least to a SIP phone. Ports are defined to be on context from-pstn in misdn.conf and and I got "exten => s,1,Goto(otherctx|103|1)" in that context in extensions.conf. The SIP phone is in otherctx à 103 extension ... So if anyone could tell me what's wrong ? |
| 10:40.31 | McDouglas | anyone could tell me whats wrong with this line? exten => 08,n,Set(foo=${IF($[ ${CALLERID(num)} = ""]?"Ismeretlen Hivo")}) |
| 10:43.09 | viperdude | . |
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| 10:50.40 | bps | dworkin: for a trunk you probably know servers' addresses, create a port forwarding rule on router or gateway |
| 10:52.33 | dworkin | bps: thank you, i know how to go around it. that's what i'd use with sip. but can iax handle two NATs on its own? (i can't test it right now, i'm at work, but i'd research it now and try it later) |
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| 10:56.54 | bps | dworkin: don't know for two nat but I'm interested when you got the final word. you can also put light tunnels between le nat endpoints |
| 10:57.30 | dworkin | bps. well, if you're here when i try it, i'll let you know |
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| 11:09.34 | aiurea | hi |
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| 11:09.57 | aiurea | is it possible to use the g726-40 codec in asterisk 1.2? |
| 11:11.46 | ice_croft | ppl, how can i send fax to pstn through fxo gw? |
| 11:13.06 | ice_croft | i mean "fax->fxs->asterisk->fxo->pstn" |
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| 11:27.20 | skirmisha | guys |
| 11:27.54 | skirmisha | in realtime mysql for sip_peers, i have field regseconds |
| 11:28.09 | skirmisha | how can i convert that value in readbale format |
| 11:30.39 | skirmisha | ??? |
| 11:37.05 | jer | skirmisha, some basic math |
| 11:37.53 | jer | 60 seconds in a minute, 60 minutes in an hour, 24 hours in a day, 7 days in a week... use a modulo function in your frontend language of choice |
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| 11:45.29 | skirmisha | i use perl |
| 11:45.54 | skirmisha | something like that - $seconds = 94054; |
| 11:45.54 | skirmisha | @parts = gmtime($seconds); |
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| 11:52.43 | jer | well that's fine, you just need to have the code to make it "pretty" in your frontend application which displays this info |
| 11:52.50 | jer | asterisk needs to store it in the format it does |
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| 12:04.27 | *** join/#asterisk Great_Randew (n=Andrew@mail.claired.com) |
| 12:06.42 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 12:07.18 | skirmisha | guys |
| 12:07.36 | skirmisha | this regseconds is not showing correct info |
| 12:08.31 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 12:09.23 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:18.42 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 12:20.11 | Telemac | It seems that my B410 and chan_isdn is ok (Port 1 Type TE Prot. PMP L2Link UP L1Link:UP Blocked:0 Debug:20), but I doesn't get incoming call, but I don't get incoming call for test at least to a SIP phone. Ports are defined to be on context from-pstn in misdn.conf and and I got "exten => s,1,Goto(otherctx|103|1)" in that context in extensions.conf. The SIP phone is in otherctx à 103 extension ... So if anyone could tell me what's wrong ? |
| 12:22.44 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:23.05 | *** part/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 12:28.39 | RoyK | skirmisha: the value holds the time the registration expires |
| 12:31.09 | RoyK | skirmisha: meaning if you set expire time on the client to 300sec, regseconds is set to now+300 |
| 12:32.25 | skirmisha | i understand |
| 12:38.14 | *** join/#asterisk MPWizard (i=cjs@85.195.17.209) |
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| 12:45.45 | *** join/#asterisk machpo (n=mcp@ketone.jawfish.com) |
| 12:48.45 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 12:59.13 | cmantito | is it bad if I saw app_hasnewvoicemail.so and thought appcanhaznewvoicemail?.so |
| 13:00.21 | *** join/#asterisk CVirus (n=GoD@82.201.178.99) |
| 13:00.53 | M1s3ry | this is a common mistake with internet junkies and normal geeks. Through rehabilitation and therapy, this can be fixed. |
| 13:02.53 | cmantito | hehe |
| 13:03.38 | *** join/#asterisk Darthclue (n=Darthclu@fw149.nisd.net) |
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| 13:09.47 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:12.00 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:16.48 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
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| 13:27.42 | jfg | aaarg ! i don't understant how to use iaxclient iaxc_register() function.. can someone help me ? |
| 13:28.14 | jfg | i created a peer in my iax.conf who uses [internal] context |
| 13:28.34 | jfg | in this context i've an exten (888) to call an agi on another server |
| 13:28.50 | jfg | so i use iaxc_register( username, secret, host ) |
| 13:28.59 | jfg | i see in my asterisk cli the peer registration |
| 13:29.13 | jfg | then i try iaxc_call( "888" ) |
| 13:29.17 | jfg | and.. nothing happen |
| 13:29.27 | jfg | no sound, no event in my asterisk cli |
| 13:29.33 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:29.35 | jfg | even with iax2 debug on |
| 13:30.23 | jfg | i don't understand how to use place a call in [internal] context |
| 13:30.26 | jfg | any idea _ |
| 13:30.27 | jfg | ? |
| 13:30.42 | jfg | if i try to call guest@misery.digium.com, it's ok |
| 13:31.06 | jfg | even if in my context i dont authorise calling this exten.. |
| 13:31.15 | jfg | there something i miss, but what ? |
| 13:34.46 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 13:36.17 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:36.17 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 13:42.15 | jfg | please :/ |
| 13:43.18 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:43.54 | *** join/#asterisk plik (i=gorph@phalse.2600.COM) |
| 13:45.02 | *** join/#asterisk ManxPower (n=manxpowe@3.sub-70-221-225.myvzw.com) |
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| 13:52.10 | *** join/#asterisk BeeBuu (n=chatzill@59.38.97.153) |
| 13:52.16 | BeeBuu | hello,all |
| 13:52.38 | BeeBuu | is there any sample of DEADAGI? |
| 13:52.43 | *** join/#asterisk af_ (n=getsmart@88-149-240-185.dynamic.ngi.it) |
| 13:54.48 | *** join/#asterisk Fusoya (i=quality@togi.homeunix.org) |
| 13:55.03 | CallCtr4Sale | hi guys.. is there a 1800 analog line that can handle 24 calls? |
| 13:55.26 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 13:56.29 | *** join/#asterisk boblutz (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 13:56.50 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 13:57.02 | boblutz | I am finally back in town |
| 13:57.13 | Zeeek | we missed you boblutz |
| 13:57.19 | BeeBuu | CallCtr4Sale: 1800 analog line? |
| 13:57.31 | mort_gib | T1 |
| 13:57.34 | boblutz | Zeeek: Conference today? |
| 13:57.37 | Zeeek | ya |
| 13:57.40 | boblutz | sick |
| 13:57.48 | Zeeek | yes |
| 13:57.54 | M1s3ry | ~pb |
| 13:57.55 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
| 14:00.58 | *** join/#asterisk UnixDog (n=UnixDog@ppp-69-238-167-52.dsl.irvnca.pacbell.net) |
| 14:01.12 | ManxPower | CallCtr4Sale: No. An analog line can handle 1 call. A channelized voice T-1 can handle 24 calls, a PRI T-1 can handle 24 calls. All T-1s are digital. |
| 14:01.48 | [TK]D-Fender | CallCtr4Sale: No. |
| 14:02.18 | CallCtr4Sale | ok thanks... |
| 14:02.31 | CallCtr4Sale | so what digium card will i be needing for a t1 |
| 14:02.56 | M1s3ry | any of these cards |
| 14:02.58 | M1s3ry | http://www.digium.com/en/products/digital/ |
| 14:03.24 | M1s3ry | depends on the number of T1's you have, the PCI slot you have on the server, and if you want echo cancellation or not |
| 14:04.42 | CallCtr4Sale | they have 4 t1's |
| 14:04.48 | mort_gib | You might want to look at Sangoma and Media gateways |
| 14:05.24 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 14:05.33 | CallCtr4Sale | why sangoma? is it good? |
| 14:05.36 | ManxPower | CallCtr4Sale: any T-1 Digium card will work. Any Sangoma card that supports voice (some Sangoma cards are data-only) |
| 14:05.54 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:05.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:06.00 | ManxPower | (well any Sangoma card that supports T-1) |
| 14:06.12 | mort_gib | Yes ManxPower |
| 14:06.59 | *** join/#asterisk hacim (n=micah@debian/developer/micah) |
| 14:07.01 | mort_gib | I don't think that Sangoma has that big a range of T1 cards, but I like their support |
| 14:07.56 | [TK]D-Fender | mort_gib: Thats because they don't need half as many as Sangoma are 3.3V & 5V compliant. They also haven't chewed their way through 3 EC solutions. |
| 14:07.59 | mort_gib | ManxPower: The guys I buy VOIP hardware from recommended Sangoma over Digium |
| 14:08.08 | hacim | i have one user registering via SIP from the far-east, some days he cant register at all using a SIP client in linux, but he reboots into windows and he can register with gizmo as his sip client, anyone have any ideas what I can look at? |
| 14:08.38 | mort_gib | 3 EC?? I use their BRI cards in Europe |
| 14:08.49 | mort_gib | I know T1!=BRI |
| 14:08.59 | [TK]D-Fender | hacim: Maybe the client is configured wrong, maybe he has a firewall on the linux side. Maybe the windows client accoutns for NAT & networking automatically and the Linux one doesn't, tec. |
| 14:09.52 | [TK]D-Fender | mort_gib: Echo cancellation. Sangoma has their single Otasic solution since day 1 and its stuck with them. Great stuff. |
| 14:09.59 | hacim | [TK]D-Fender: well he can register right now from the linux client, just 8 hours ago he could not, with no changes other than he slept |
| 14:10.00 | ZPertee | mornin |
| 14:10.12 | mort_gib | Yes it works quite well |
| 14:10.18 | [TK]D-Fender | hacim: Then if you're saying NOTHING changed, then its an act of God. |
| 14:10.42 | plik | or PEBKAC |
| 14:10.47 | hacim | [TK]D-Fender: nothing on his machine, or the asterisk server, so its likely somewhere inbetween, perhaps latency |
| 14:11.07 | hacim | he is in india and the server is in washington state, so its about as far away as you can get |
| 14:11.19 | boblutz | thinks |
| 14:11.25 | [TK]D-Fender | hacim: Keep in mind Indai tends to block SIP as the telcos are monopolies. |
| 14:11.29 | jfg | someone to help me with libiaxclient ? |
| 14:11.35 | [TK]D-Fender | India* |
| 14:11.41 | hacim | [TK]D-Fender: sorry, he's in pakistan not india |
| 14:11.46 | jfg | i have problems to place calls when registered to an IAX server |
| 14:11.53 | [TK]D-Fender | hacim: Not sure if they're much better. |
| 14:12.01 | mort_gib | jfg: could run VPN between sites?? |
| 14:12.16 | jfg | mort_gib: ? |
| 14:12.52 | hacim | [TK]D-Fender: thats interesting, but that wouldn't explain why it works in mac, but not in windows... i'm suspecting a registration timeout set too high causing the nat rule timeout or something |
| 14:12.54 | mort_gib | Well, I have a setup, alas from London to a local site where they run VPN (OpenVPN) so the traffic is on port, eh 80 in this case |
| 14:13.17 | *** join/#asterisk merkurie (n=merkurie@192.153.163.44) |
| 14:13.18 | [TK]D-Fender | hacim: Different sofware as well... who knows. You going to have to try a bunch of stuff... |
| 14:13.22 | Zeeek | Question: would AT&T block SIP if they could get away with it? |
| 14:13.58 | plik | Zeeek: only intermittently - to ensure greater frustration for users |
| 14:14.22 | GrumpyOldMan | they would sell your mother if they could get away with it. |
| 14:14.40 | Zeeek | plik that's exactly what the monopoly did in France when the cable started offering internet |
| 14:15.00 | *** join/#asterisk Defraz (n=T0tal@72.24.26.7) |
| 14:15.21 | plik | yeah, so you don't honestly think AT&T would behave any better do you\/ |
| 14:15.30 | Zeeek | Because they owned the network (remember, taxpayers paid for it!) they kept screwing with it as soon as the cable was available. Outages all the time |
| 14:16.04 | Zeeek | Those days are over, after about 1,000,000 lawsuits they were forced to comply on all counts, including portability of numbers |
| 14:16.35 | merkurie | anyone got any tips for starting a business selling asterisk based pbxs? |
| 14:16.37 | Zeeek | And now they are offereing FTTH themselves - but the law here requires that they have competition |
| 14:17.06 | glaz | how do I reload voicemail.conf |
| 14:17.08 | Zeeek | I have a foolproof plan to make a SMALL FORTUNE selling asterisk based pbxes |
| 14:17.19 | Zeeek | FOOLPROOF I tell you! |
| 14:17.22 | plik | merkurie: check out the competition first, and work out your high value USP |
| 14:17.22 | merkurie | Zeeek, you selling your plan? |
| 14:17.38 | merkurie | plik, not much competition in my area |
| 14:17.49 | Zeeek | The way to make a small fortune selling asterisk pbxes is to start with a BIG FORTUNE |
| 14:17.52 | merkurie | plik, none that i know of |
| 14:18.06 | *** join/#asterisk bougie (n=bougie@APoitiers-256-1-10-5.w90-11.abo.wanadoo.fr) |
| 14:19.09 | merkurie | plik, USP? |
| 14:19.30 | merkurie | plik, Unique selling proposition? |
| 14:19.34 | plik | Unique Selling Point - why should a custmer choose you? |
| 14:19.44 | Zeeek | My IP 500 is still running after the SIP update |
| 14:19.52 | Zeeek | w00t |
| 14:19.57 | plik | being the only supplier that you know of in the area isn't really a agood one |
| 14:20.32 | Zeeek | There's only one way to succeed in this sea of competiton pricing: SERVICE |
| 14:20.56 | plik | unless you can give a uch more personal service AND your clients want / need that |
| 14:21.01 | merkurie | plik, what about pricing? do most people try to bundle a pbx and sell it as a single price? or do they have a pay-by-the-hour consulting type? |
| 14:21.27 | plik | no idea - that's why you need to check out the competition |
| 14:21.59 | plik | some people call it "research" |
| 14:22.20 | merkurie | plik, i think i read about that once =) |
| 14:22.30 | *** join/#asterisk _Krieger_ (n=krieger@193.39.118.158) |
| 14:22.39 | Zeeek | ~research |
| 14:22.40 | jbot | somebody said research was what the internet is a tool for... the more you can use it (search the web or databases, find applicable reading material, reading that material, looking in there for references to others), the more independent you will be. Ask me about sicco and cooperation in asking for help. |
| 14:22.40 | plik | ya, me too... I think |
| 14:22.57 | Zeeek | ~sicco |
| 14:22.58 | jbot | Please ask sicco questions, questions that are Specific, Informative, Concise, Complete, and On-topic. Ask me about research and cooperation in asking for help. |
| 14:22.58 | merkurie | plik, isn't this research? =) |
| 14:23.20 | Zeeek | ~sicco huh? |
| 14:23.40 | plik | it's a vague stab at attempting to research, but not really the kind thats gonna get you sound business advice |
| 14:23.52 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:23.56 | merkurie | plik, oh |
| 14:24.05 | Zeeek | ~sicco how can I get my dialplan working |
| 14:24.35 | jfg | is there an asterisk dev here ? knowing iax.. ? |
| 14:24.40 | plik | "erm... some guy on irc said to try that" |
| 14:25.21 | merkurie | plik, what about trademark and trade name infringement? |
| 14:25.26 | Zeeek | developers are not allowed on this channel. They eat too many cheetos and we have to vacuum too often |
| 14:25.31 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 14:25.32 | merkurie | plik, digium gonna come after me if i'm not a reseller? |
| 14:26.55 | boblutz | merkurie: with extreme predijuce |
| 14:27.12 | Zeeek | or extreme apple juice depending |
| 14:27.14 | merkurie | boblutz, nice |
| 14:27.24 | plik | merkurie: keep coming up with these questions, then when you're done, set about getting definitive answers to them yourself (from sources other than IRC) ... that will be some way towards research |
| 14:27.50 | merkurie | plik, k |
| 14:27.52 | plik | don't forget to keep asking more relevant questions, and checking your facts |
| 14:28.00 | merkurie | plik, k |
| 14:28.19 | Zeeek | facts? We don' need no stinkin' facts |
| 14:28.22 | CallCtr4Sale | is a t1 a zaptel device too? |
| 14:28.49 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 14:29.10 | [TK]D-Fender | CallCtr4Sale: No, T1 is a telephony signalling method. |
| 14:29.12 | plik | Bonus Hot Tip -- you don't have to compile your list of questions in the channel.... try using vim or notepad... you'll find less people get bored with your groundwork that way, and you may even get more sound advice when you need it |
| 14:29.27 | ZPertee | what's the difference in sip.conf between port= and bindport= |
| 14:29.34 | merkurie | plik, is e-mail good for research? |
| 14:29.50 | CallCtr4Sale | but asterisk reads it as a zaptel channel right? |
| 14:30.26 | plik | if you have someone will to engage with you it might be OK... definitely not if you're just gonna spam people asking for stuff though |
| 14:30.36 | merkurie | plik, k |
| 14:30.45 | Zeeek | plik no, you need gcc++ to compile your questions |
| 14:30.55 | [TK]D-Fender | ZPertee: "port=" is for peers to tell * what port to send packets to. "bindport=" tells * what port to receive SIP requests against. This is under [general] only. |
| 14:31.06 | plik | google knows lots of pages that know about setting up business - I'm sure they will help give you ideas |
| 14:31.15 | [TK]D-Fender | CallCtr4Sale: Depends how you get from T1 to Asterisk. |
| 14:31.18 | Zeeek | [TK]D-Fender manning the fort |
| 14:31.28 | *** join/#asterisk ReD-MaN (i=r00t-rox@172-220.static.golden.net) |
| 14:31.38 | Zeeek | This is like the fourth day of my ip500 not rebooting |
| 14:31.44 | Zeeek | I'm soooo impressed |
| 14:31.53 | [TK]D-Fender | Zeeek: lol, lowered expectations! |
| 14:32.02 | Zeeek | yeah, egggs acly |
| 14:32.34 | Zeeek | File in "In all fairness" dept: it didn't reboot spontaneously that often |
| 14:33.02 | Zeeek | Hey, I've been spending 4 hours a day at the shredder preparing for our move. I have to share this |
| 14:33.42 | ZPertee | [TK]D-Fender, can I have multiple bindports? I am using a Linksys ATA in which I have to use different ports when I setup each user. how would I setup asterisk to accept differnt ports (5060, 5061, 5062, and so on) |
| 14:33.44 | merkurie | plik, thanks for your help, i'm gonna go get started |
| 14:34.03 | plik | merkurie: good luck |
| 14:34.09 | Zeeek | Receipt: $200 for USR Sportser 56K (aka Spiral of death) modem in 1997. Man in line behind me asing if we had a special line for it. |
| 14:34.32 | [TK]D-Fender | ZPertee: No, Asterisk will LISTEN against only one port. With your ATA, asterisk will SEND to the port you specify in your PEER setup under "port=" |
| 14:34.47 | Zeeek | ZPertee it'll work, don't worry |
| 14:34.58 | *** join/#asterisk fransena (n=fransena@207.229.0.38) |
| 14:35.21 | ZPertee | Zeeek, its not thats why I'm worrying |
| 14:35.26 | Zeeek | heh |
| 14:35.41 | Zeeek | reaches for beer bottle to drown multipl sorrows |
| 14:35.51 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 14:36.03 | Zeeek | notices additional sorron in that bottle empty |
| 14:36.33 | ZPertee | Zeeek, at least its FRIDAY! |
| 14:36.41 | fransena | Question about the appliance: Is it possible to move audio files to the unit over the network? Reason I ask is I have someone recording all the greetings to high-quality WAVs then I resample them to what will work on whatever system I'm on (Cisco, Asterisk etc.) |
| 14:37.05 | Zeeek | ZPertee there is that, thanks |
| 14:37.07 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:37.47 | ZPertee | Zeeek, thats my trick when in the depths of despair count the days till Friday! |
| 14:37.56 | *** join/#asterisk waKKu (n=komo@unaffiliated/wakku) |
| 14:38.15 | Zeeek | Surely someone must have ideas for asterisk parodies of the Hillary 3AM phone call? |
| 14:38.28 | Zeeek | Let's see. Allison could maybe record something for us? |
| 14:39.21 | Zeeek | Help me write the script. I'll get her to record it! |
| 14:39.35 | Zeeek | It's 3 AM... |
| 14:39.36 | C4away | ZPertee, how long does it take you to count to a max of 5? |
| 14:40.07 | C4away | unless you work 7 days, or need to cheer yourself up on your days off .. then max of 7 |
| 14:40.13 | ZPertee | C4away, somedays longer than others |
| 14:40.17 | C4away | lol |
| 14:41.13 | ZPertee | thinks Zeeek has a great idea! |
| 14:41.21 | Zeeek | some days it doesn't pay to get out of bed though |
| 14:41.28 | boblutz | can you hook a reciever up to an Asterisk box? |
| 14:41.28 | Zeeek | and this is one of those for me |
| 14:41.37 | Zeeek | a receiver of? |
| 14:41.43 | boblutz | radio waves? |
| 14:41.58 | boblutz | I thought I had read something like that before..I dont know what its called |
| 14:42.06 | Zeeek | yeah there is something like that |
| 14:42.59 | boblutz | that is very handy for paranoid people |
| 14:43.56 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 14:44.01 | hacim | myself and another sip user are able to call a meetme and talk, but if we call each other, the sound is there for one second and then its dead |
| 14:44.30 | C4away | app_rpt runs radio repeaters |
| 14:44.49 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 14:45.03 | boblutz | So one could, in theory, call their asterisk box, listen to police frequencies, and then drive home drunk from the bar accordingly? |
| 14:45.32 | boblutz | So if you were to get pulled over, they wouldnt be like "why do you have this reciever in your car? |
| 14:45.42 | C4away | oh that's easy |
| 14:46.01 | C4away | run chan_oss and make sure alsa mixer is set properly |
| 14:46.25 | C4away | then create an extension to connect you through chann_oss to the sound card input |
| 14:46.43 | hacim | how do I enable the SIP method message? |
| 14:46.49 | C4away | then you could put anything that has a line-level output into the mic port |
| 14:46.58 | C4away | sip method? for what? changing channels? |
| 14:47.23 | C4away | oh, sorry, I'm tired, not paying attention to who is talking |
| 14:47.34 | hacim | C4away: no, for IM messages between SIP clients |
| 14:47.42 | C4away | yea, I got you confused with boblutz |
| 14:47.57 | C4away | I don't know how to do that |
| 14:48.39 | C4away | but boblutz: if you want to actually run a radio repeater like a HAM radio repeater or commercial radio or whatever, you can control it through a serial connection to trigger various repeater actions |
| 14:48.48 | C4away | like switching between receive and transmit |
| 14:49.13 | C4away | if you just want to listen you need a DTMF decoder circuit if you want to control the scanner and just put the audio into the line-in jack |
| 14:49.57 | Zeeek | http://VoipUsersConference.org has all the details about how to reach the conference via SIP oir PSTN |
| 14:49.58 | boblutz | C4away: Yea, I was wondering if it would be possibly to change freq |
| 14:50.42 | C4away | you could have preset frequencies set to 1-6 and then 7 scan back, 8 hold freq, 9 scan forward |
| 14:51.20 | C4away | just need a DTMF decoder that outputs a binary switch, then map that to the interface for the scanner using basic logic chips |
| 14:51.38 | C4away | unless the scanner already can be controlled by DTMF |
| 14:52.04 | C4away | but if you just wanted it to auto-scan and stop when it gets a frequency with audio on it then you don't need any control just let it scan and hold while there is audio |
| 14:52.09 | C4away | patch that into the line-in and you are good |
| 14:52.34 | boblutz | well, assume the bar is within 10 miles of my Asterisk box...I would be hearing relevant frequencies, no? |
| 14:53.48 | plik | hmmm,... you could even have it set up so you can press # if you get pulled over, and then asterisk automatically makes a sreies of emergency calls pulling thepolice away from your minor road offence !! ;) |
| 14:54.00 | C4away | haha |
| 14:54.05 | boblutz | dos via asterisk? |
| 14:54.06 | C4away | have it spoof the cid and ani |
| 14:54.49 | boblutz | good idea, but SWATing is highly illegal and uncool |
| 14:54.57 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:55.05 | C4away | I can imagine the flite voice calling in bomb threats and other horrible crimes at random addresses |
| 14:55.07 | boblutz | If I got pulled over, I would rather just throw a smoke bomb down and run |
| 14:55.11 | C4away | they wouldn't catch on to that ... |
| 14:55.20 | boblutz | lol, flite |
| 14:56.05 | ZPertee | boblutz, you could just call a taxi.... |
| 14:56.17 | *** join/#asterisk talntidwrk (n=t@66.208.251.170) |
| 14:56.21 | boblutz | ZPertee: I swear to drunk im not god |
| 14:56.29 | talntidwrk | hehe. |
| 14:56.31 | Zeeek | nice |
| 14:57.13 | ZPertee | hehe |
| 14:57.20 | boblutz | Ill have to play around with this when I get home...itd be fun to listen to while stuck in rush hour traffic |
| 14:57.38 | Zeeek | Calling everyone: In one hour the big Shoe is here: http://VoipUsersConference.org - all about asterisk - what is a trademark of Digium |
| 14:58.59 | hacim | anyone have any ideas about sip bridging problems? Connection is made, half a second of sound is heard, then nothing |
| 14:59.12 | Zeeek | SIP happens |
| 14:59.53 | hacim | in this case, its not :) |
| 15:00.04 | hacim | we both can sip into a meetme and that works, oddly |
| 15:00.05 | Zeeek | that's bad. Very bad |
| 15:00.12 | boblutz | Zeeek: What is the IRC channel again? |
| 15:00.59 | Zeeek | #voip-users-conference |
| 15:01.03 | Zeeek | or |
| 15:01.09 | Zeeek | http://VoipUsersConference.org has all the details about how to reach the conference via SIP oir PSTN |
| 15:01.25 | Zeeek | That's in ONE hour here on the VBC network |
| 15:01.50 | Zeeek | Replace VoIP with SoIP - services over IP |
| 15:02.17 | Zeeek | too bad soip.com is taken |
| 15:02.21 | *** join/#asterisk Skarmeth (n=Skarmeth@201009117121.user.veloxzone.com.br) |
| 15:02.41 | Zeeek | someone has registered ?oip.* |
| 15:03.02 | Zeeek | Thinking "Ya, a lot of stuff will come up on IP so I make money" |
| 15:04.08 | Zeeek | Try it: You'll see [a-z]oip.com are taken |
| 15:04.27 | boblutz | name-parking sucks |
| 15:04.35 | Zeeek | "Heh" over IP: hoip.com |
| 15:04.36 | [TK]D-Fender | Zeeek: soup? soip2. gtfo. |
| 15:04.49 | [TK]D-Fender | :p |
| 15:05.12 | Zeeek | someone called me a few days ago for declic.com |
| 15:05.30 | Zeeek | I asked for 5k and they were shocked. We've been using the name for 10 years + |
| 15:05.34 | denon | Zeeek: boy, that name shark will be ticked when we dump IP and go back to NetBEUI! |
| 15:05.41 | denon | routed NetBEUI, that is |
| 15:05.57 | Zeeek | denon oh, yes, I use it all the time at the orifice :) |
| 15:06.18 | [TK]D-Fender | .... |
| 15:06.24 | Zeeek | what? |
| 15:06.29 | Zeeek | bouche bée ? |
| 15:06.48 | Zeeek | but seriously: http://hoip.com/ |
| 15:07.20 | Zeeek | http://yoip.com/ |
| 15:07.26 | Zeeek | Enough! |
| 15:07.43 | UnixDog | http://wiki.contribs.org/SME_Server:Download |
| 15:08.13 | Zeeek | http://joip.com/NonMember/hp.aspx |
| 15:08.32 | Zeeek | Please someone help with the 3AM call for Allison |
| 15:08.44 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 15:08.46 | hacim | sip no make me happy |
| 15:08.56 | boblutz | I can make a .wav of Allison using the Cepstral TTS |
| 15:09.00 | boblutz | just tell me what u want it to say |
| 15:09.13 | *** join/#asterisk rotozip (n=rotozip@c-68-34-139-139.hsd1.mo.comcast.net) |
| 15:12.25 | Zeeek | oh yeah? |
| 15:12.41 | Zeeek | Well I still want funny stuff so let's all think of the lines |
| 15:14.53 | rotozip | I am runninf Asterisk v 2.4.0.1 and am using a Sipura spa3000... I get a random single dtmf tone while on calls and am wondering if there is anyway to eliminate this. |
| 15:15.14 | UnixDog | there is no asterisk 2.4.0.1 |
| 15:15.21 | UnixDog | your on drugs |
| 15:15.26 | Qwell | ~trixbox |
| 15:15.27 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 15:15.45 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:16.36 | rotozip | sorry asterisk 1.4.18.1 |
| 15:16.52 | ManxPower | rotozip: What make you think it was 2.4.0.1? |
| 15:17.07 | Qwell | freepbx, of course |
| 15:17.26 | rotozip | I was looking at the -v of a module. |
| 15:17.54 | Zeeek | asterisk -rx "stop now" |
| 15:18.18 | hacim | canreinvite=no is the answer to my woes |
| 15:18.49 | ManxPower | rotozip: put a copy of the [general] and the part for the SPA from sip.conf onto pastebin.ca |
| 15:19.10 | rotozip | ok |
| 15:19.41 | ManxPower | If I find out you are running a GUI I'll feed you to the pet gator. |
| 15:19.46 | Zeeek | We be staring soon: http://VoipUsersConference.org |
| 15:20.00 | Qwell | ManxPower: of course he's running a gui |
| 15:20.12 | ManxPower | Qwell: he's not admitted it yet. |
| 15:20.19 | Qwell | yes he has |
| 15:20.39 | ManxPower | Oh. I don't help assholes that look for help on the wrong channel. |
| 15:21.46 | *** join/#asterisk xenonex (n=xenonex@92.47.0.5) |
| 15:22.35 | Zeeek | My name is Zeeek and I am running a GUI. Windows XP (and OS X). Not crazy enough to run asterisk on tose boxes yet |
| 15:23.23 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-165-70.dsl.teksavvy.com) |
| 15:24.01 | ZPertee | way to go zeeek! |
| 15:24.29 | Zeeek | I think I just found a 1997 receipt for WIndows |
| 15:24.55 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 15:25.03 | Zeeek | so we're moving! And I may just go for a hosted pbx after all |
| 15:25.36 | Zeeek | But for my USA service the AA50 with my various accounts at Nufone, VoicePulse and Junction will be very cool |
| 15:25.41 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:25.41 | *** join/#asterisk l2cache (n=l2cache@117.178.101.97.cfl.res.rr.com) |
| 15:25.41 | outtolunc | traitor!! <G> |
| 15:25.56 | Zeeek | My hosted guy uses asterisk!!!! |
| 15:26.12 | Qwell | Zeeek: moving where? |
| 15:26.14 | l2cache | Is anyone aware of a command line tool or enabling sox to convert/mix my .g729 files to wav? |
| 15:26.22 | Zeeek | And he's acharter member of the Club Asterisk de Paris like me |
| 15:26.25 | outtolunc | but you have to have one cloes enough to 'pet' it like it is a faithful hound <G> |
| 15:26.29 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 15:26.35 | Zeeek | Qwell no where in firing range, don't worry :) |
| 15:26.54 | Zeeek | outtolunc yes, there is that. Or at least change the cpu fan |
| 15:27.06 | outtolunc | nods |
| 15:27.23 | Zeeek | I hate the very idea of CPU fans |
| 15:27.36 | Zeeek | but, can't live with 'em, can't live without 'em, eh? |
| 15:27.53 | Trevor_b | If a call comes in on PRI to an asterisk server, and i want to transfer that over to another asterisk server (in mass quantity) would i be best to go with IAX and ulaw or gsm, or what combination of protocol and codec would use the least CPU and memory? |
| 15:27.56 | outtolunc | the reason i am always muted (no mic) is due to the constant turbine noise |
| 15:28.10 | l2cache | Is anyone aware of a command line tool or enabling sox to convert/mix my .g729 files to wav? |
| 15:28.28 | Zeeek | Yes, working in the prison license plate facility has its challenges |
| 15:28.32 | [TK]D-Fender | Trevor_b: Depends where that other server is and how you could physically connect them. |
| 15:28.43 | Trevor_b | Same server room. |
| 15:29.07 | [TK]D-Fender | Trevor_b: You could use SIP / IAX / TDMoE / extra PRI port, etc.... |
| 15:29.09 | outtolunc | yeah, and i am at home <G> |
| 15:29.09 | Zeeek | freenode.net #voip-users-conference |
| 15:29.13 | Trevor_b | contemplating have a single server run like 10-12PRI and then have multiple systems behind it doing more work and using it as the gateway. |
| 15:29.41 | jeev | [TK]D-Fender, wholesaler solved my problem with switching me to inband. |
| 15:29.51 | *** join/#asterisk bl4q (n=Bl@dslb-088-064-146-089.pools.arcor-ip.net) |
| 15:30.43 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 15:31.01 | Trevor_b | Yeah testing the TDMoE as well, just wondering if someone knew better and worse combos. I mean i know VoIP to VoIP what to do, but zap to VoIP, not sure if something in VoIP was more natively like the hardware will hand it to asterisk, or if it doesnt matter, you have to convert to some codec so its all going to cost the same... Although I suppose TDMoE would in theory reduce that since its using PRI based signalling still? hmmmm. |
| 15:32.02 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 15:32.40 | ManxPower | TDMoE does not seem to be well supported since IAX2 Trunking was added to Asterisk |
| 15:32.50 | ZPertee | Zeeek, what was the channel for the voip talk you mentioned, sorry my irc quit |
| 15:33.14 | [TK]D-Fender | Trevor_b: If you can, the most stable and inexpensive way I would do it is to add a NIC to each, wire a cross-over cable, and use that as the media. pass whatever protocol you want over it. since you'll have all the BW you'll even need, go SIP w/ G.711u |
| 15:33.28 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:33.41 | rotozip | ManPower: here is my sip.conf http://pastebin.ca/981102. I have the sip conf in several custom files so I pasted all of it. |
| 15:33.42 | rotozip | nat=no |
| 15:33.42 | rotozip | port=5061 |
| 15:33.42 | rotozip | qualify=yes |
| 15:33.42 | rotozip | secr |
| 15:34.06 | ManxPower | An ulaw or alaw call should use .08Mbps, IIRC. |
| 15:34.07 | rotozip | whoops sorry did not mean to paste those parameters here |
| 15:34.34 | dworkin | (* 1.2.17) in meetme conferencing pressing *1 works for all sip phones. however for pstn phones, it only works if they're in the conference room alone. as soon as somebody else enters the room, pressing star only announces that $NAME is in the conf room. is this a know bug, or my misconfiguration? |
| 15:34.36 | ManxPower | rotozip: you took so long I forgot what your problem is. |
| 15:34.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:35.07 | dworkin | adds that *1 is for muting/unmuting |
| 15:35.11 | ManxPower | rotozip: you need a dtmfmode=rfc2833 in the SPA section of sip.conf |
| 15:35.17 | rotozip | ManxPower: sorry, I get random single dtmf tones while on a call. |
| 15:35.45 | l2cache | Is anyone aware of a linux command line tool to convert between g729 => wav ? |
| 15:36.10 | ManxPower | rotozip: you do not want to enable both alaw and ulaw. Pick one. |
| 15:36.21 | ManxPower | l2cache: since G729 is patented, sox won't have it. |
| 15:36.46 | l2cache | So have you heard of a program to convert it through linux? |
| 15:36.56 | ManxPower | l2cache: if you have Asterisk 1.6beta AND you have a purchased G729 license, you can convert from the Asterisk SLI. |
| 15:37.09 | ManxPower | l2cache: there will be no free program that incluses G729 |
| 15:37.29 | rotozip | ManxPower: cool, I will switch those settings. Thanks. |
| 15:37.39 | l2cache | But the 1.6 ver allows you to convert using the built in codec_g279 module? |
| 15:37.50 | l2cache | if you have it |
| 15:37.51 | ManxPower | l2cache: NO! G729 is PATENTED and LICENSED. |
| 15:38.01 | l2cache | lol, thanks |
| 15:38.09 | Zeeek | we don need no stinkin license... |
| 15:38.12 | ManxPower | If you have a g729 license from Digium, then 1.6beta should allow you to convert to/from G729 to/from anything |
| 15:38.23 | l2cache | perfect |
| 15:38.27 | l2cache | thanks Manx |
| 15:38.29 | denon | Zeeek: not even GPL? |
| 15:38.45 | Zeeek | well, yeah |
| 15:38.48 | denon | ManxPower: Asterisk is LICENSED too .. but I still get it for free |
| 15:38.51 | denon | why not GPL? |
| 15:38.53 | denon | trolls :) |
| 15:38.55 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-177.static.twtelecom.net) |
| 15:39.10 | Zeeek | owns two channels of g729 from Digium |
| 15:40.46 | *** join/#asterisk Zeeek_ (n=IceChat7@86.66.255.138) |
| 15:43.37 | Trevor_b | [TK]D-Fender: Its not the bandwidth I was worried about, but the CPU/Memory requirements for codec conversion, and how many calls the system will be able to push through. |
| 15:44.01 | Zeeek_ | the channels are open: Dial(SIP/123@66.212.134.192) |
| 15:44.02 | [TK]D-Fender | Trevor_b: Yes, and G.711u is the codec used on N/A T1's |
| 15:44.04 | Trevor_b | before memory or CPU get hit, i remember its mainly 1 of the two, just not wichi. |
| 15:44.11 | Trevor_b | AH |
| 15:44.14 | [TK]D-Fender | Trevor_b: So that will be virtually no load |
| 15:44.28 | Trevor_b | ok, so any proto with G.711u and codec conversion is as minimal as I can get. |
| 15:44.31 | [TK]D-Fender | Trevor_b: And on a dedicated comm path. |
| 15:44.51 | [TK]D-Fender | Trevor_b: I'd suggest SIP over that seperate link. |
| 15:45.00 | Trevor_b | k |
| 15:45.18 | Trevor_b | [TK]D-Fender: better luck then with IAX? |
| 15:45.46 | [TK]D-Fender | Trevor_b: Yes |
| 15:45.50 | *** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3) |
| 15:46.26 | Trevor_b | [TK]D-Fender: You ever used TDMoE? The only complaints i can find about it are from like 2004-2006 or so, but since then its like nobody complains, but nobody comments on using it. Only supported in business edition, so not sure if people stopped trying to use it, or if the bugs finally got worked out. |
| 15:47.19 | [TK]D-Fender | Trevor_b: Thats pretty much it... nobody's using it :) |
| 15:47.34 | [TK]D-Fender | Trevor_b: Probably not worth the effort. |
| 15:47.47 | *** join/#asterisk jbeez (i=jbeez@jbeez.net) |
| 15:48.09 | jbeez | can anyone recommend some decent hardware handset manufacturers/models? polycom maybe? |
| 15:48.26 | Trevor_b | [TK]D-Fender: Whats the most PRI's you have ever run to a single asterisk system? |
| 15:49.01 | *** join/#asterisk bougie (n=bougie@APoitiers-256-1-10-5.w90-11.abo.wanadoo.fr) |
| 15:49.51 | Trevor_b | jbeez: Polycom 320/330, 550, 650 series are really good. I am prejudiced against the 501's, but thats mainly quirky hardware issues i had with used models.... |
| 15:49.53 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 15:49.57 | [TK]D-Fender | Trevor_b: In those that I've dealt with, 2. I've heard of 4+ in those I'v chatted with. Without transcoding or EC concerns you do quite a lot. I'd bet 16 via 2 8-port cards easily enough on a "decent" nominal PC with 2GB ram. |
| 15:50.24 | [TK]D-Fender | jbeez: Polycom. Model we'd suggest will depend on your needs/wants |
| 15:50.56 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 15:51.09 | [TK]D-Fender | Trevor_b: But in larger intstall people tend to want to break that out of the * box and go with heavy-duty gateways ($$) |
| 15:51.18 | Trevor_b | Yeah we got callcenter systems running Sangoma A104D's, but a new client wants to build up to a 300 seat system. Currently they have 4PRI's on a TRIXBOX(ugh!) system as the gateway, sorta surprised they were able to get trixbox to push the interoffice and callcenter calls for that many agents without slowing to a crawl. |
| 15:52.52 | [TK]D-Fender | Trevor_b: Sure the configs are crappy, but that shouldn't impact the hardware chosen. |
| 15:52.53 | *** join/#asterisk lst (n=liquid@openwrt/developer/lst) |
| 15:53.07 | jeev | anyone know why asterisk listens on port 2000/tcp ? |
| 15:53.13 | jbeez | ok, here is the deal... |
| 15:53.30 | Zeeek_ | the channels are open: Dial(SIP/123@66.212.134.192) |
| 15:53.56 | Zeeek_ | #voip-users-conference |
| 15:54.06 | lst | i'm interested in putting together a test network for some academic research on ss7/sigtran. Anyone have a moment to chat with me about some of the viable options to do so? |
| 15:54.40 | jbeez | I just started at a new job like 3 weeks ago, they have a 2u hp server with asterisk on it, that they can't get to work with the PRI from cavalier telephone, and they paid some company to put it in and for the hardware like $20k so far, and in the meantime they have a inter-tel pbx with old inter-tel phones and they need to get more phones soon, but dont want to buy intertel phones for $$$$ if we are switching off of it soon, so I need to test some hardware hand |
| 15:54.47 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:54.54 | jbeez | we dont even have a login to this asterisk box atm, |
| 15:54.59 | [TK]D-Fender | jeev: http://www.google.ca/search?hl=en&q=asterisk+port+2000+tcp&btnG=Google+Search&meta= |
| 15:55.07 | [TK]D-Fender | jeev: Google is your friend. |
| 15:55.07 | Qwell | w00t, skinny |
| 15:55.28 | [TK]D-Fender | Qwell: The first module I explicitly disable every time :) |
| 15:55.48 | [TK]D-Fender | Qwell: Followed by pbx_ael, and chan_mgcp :) |
| 15:55.48 | Qwell | but not mgcp? |
| 15:55.48 | Qwell | heh |
| 15:55.54 | Qwell | pbx_ael > you |
| 15:55.59 | [TK]D-Fender | Qwell: Oh yeah, no useless crap on my setups :p |
| 15:56.17 | jbeez | even if they end up just using the asterisk box as a conference bridge they are OK with that, but then they might want to change to a new PBX system like a mitel voip, and hopefully these polycoms will work with it, the 650s look like nice phones, we only need like 30 phones for this office anyway |
| 15:56.20 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:56.27 | DagMoller | AEL is nice |
| 15:57.07 | Zeeek_ | file qwell |
| 15:57.08 | *** part/#asterisk mocker (n=kyle@mocker.org) |
| 15:57.16 | [TK]D-Fender | jbeez: IP 320/330's cover 99% of user needs. |
| 15:57.21 | Qwell | ? |
| 15:57.22 | file | what what what? |
| 15:57.35 | [TK]D-Fender | echo cancels file... |
| 15:57.58 | jbeez | ty |
| 15:58.03 | Zeeek_ | hey |
| 15:58.15 | jeev | file.. |
| 15:58.15 | jbeez | what is the diff between 320/330 ? |
| 15:58.16 | jeev | hi |
| 15:58.19 | file | haylo |
| 15:58.22 | Qwell | jbeez: 10 |
| 15:58.26 | jeev | file: wholesaler changed me to inband yesterday... and it works. |
| 15:58.27 | [TK]D-Fender | think I'm going to go buy a Soekris NET5501..... |
| 15:58.28 | jbeez | sounds good |
| 15:58.32 | Qwell | one has a switch port, I think |
| 15:58.37 | Qwell | (the 330) |
| 15:58.38 | jeev | only issue i've noticed now is that when i dial, it may take up to 10-15 seconsd to get through every time. |
| 15:58.45 | [TK]D-Fender | jbeez: IP 330 has a 10/100 pass-through port. |
| 15:58.50 | jbeez | nice, ty |
| 15:58.58 | ManxPower | jeev: can you be a little more vague? |
| 15:59.02 | jeev | ;) |
| 15:59.18 | Zeeek_ | the channels are open: Dial(SIP/123@66.212.134.192) |
| 15:59.50 | Qwell | Zeeek_: did you need something? |
| 15:59.55 | Zeeek_ | yes |
| 16:00.05 | Zeeek_ | come on over |
| 16:00.17 | Qwell | am I going to dial in again, and not have the question asked? O.o |
| 16:00.56 | Zeeek_ | don't know |
| 16:01.20 | Qwell | I'll be here, poke me if you need something |
| 16:01.52 | Zeeek_ | k |
| 16:05.08 | *** part/#asterisk hacim (n=micah@debian/developer/micah) |
| 16:05.14 | file | jeev: O.o |
| 16:05.20 | jeev | heh |
| 16:05.21 | file | how odd |
| 16:05.24 | jeev | ya |
| 16:06.13 | jeev | i guess it's fast now. |
| 16:06.28 | jeev | i am considering just dropping moh |
| 16:06.42 | jeev | if the client is put on hold, is the no sound no bandwidth or less bandwidth ? |
| 16:07.51 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 16:07.51 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:08.29 | Zeeek_ | Last call Dial(SIP/123@66.212.134.192) - or see http://voipUsersConference.org |
| 16:08.34 | Zeeek_ | bye for now |
| 16:08.46 | *** part/#asterisk Zeeek_ (n=IceChat7@86.66.255.138) |
| 16:08.56 | Qwell | guess he didn't need me afterall |
| 16:09.20 | jbeez | do any of you guys use pf/altq to handle your QoS? |
| 16:12.54 | RobH | Anyone have any information on making voicemail attachments save and email out as mp3? I do not want to do it, but osme of my users cannot download wav files due to size. |
| 16:13.18 | Qwell | RobH: can't write mp3 with asterisk. |
| 16:13.32 | eric2 | use gsm |
| 16:13.37 | RobH | I did not think I could, but wanted to ask. I tried and got the 'cannot write, only read' messages. |
| 16:13.49 | RobH | gsm != blackberry pearl |
| 16:13.58 | RobH | damn things are hard to feed. |
| 16:14.04 | eric2 | or use sox to convert from wav to mp3 |
| 16:14.33 | RobH | that would be outside of * and through some kinda spool address and conversion, too complicated. |
| 16:14.40 | RobH | was hoping there was a way to do it inside * |
| 16:14.55 | RobH | I think I am just going to tell them 'nope, cannot do it for you' |
| 16:14.56 | RobH | heh |
| 16:14.57 | eric2 | not that hard.. but yes, it's outside of * |
| 16:15.20 | RobH | Well, I will add it to my 'maybe' list then |
| 16:15.40 | RobH | I would think do that by emailing a decidated email box, parse the attachment, convert, and send it back out |
| 16:21.11 | clyrrad | Is it possible to have an AGI script write dial plan lines like exten => s,n,GoToIfTIme blah blah blah? I have a bunch of "after hours" settings saved in a PGSQL DB, and I would like to SELECT from that table and have the approperiate GoToIfTime lines written, I am just not sure how to add to the dial plan dynamically like that. How can I go about this? |
| 16:22.52 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 16:23.55 | ManxPower | 560244 |
| 16:24.04 | ManxPower | 4 |
| 16:24.38 | RobH | Anyone know the ubuntu/debian packages to install for ogg support? menuslect says vorbis and ogg, but that doesnt help, since I have the libogg and libvorbis stuff installed |
| 16:25.35 | ZPertee | clyrrad, not sure how to answer your question. but I know that you can run asterisk in realtime and have the configuration stored in a DB. not sure if that would be an option for you or not |
| 16:26.17 | ManxPower | clyrrad: you are making it WAY too complicated. |
| 16:26.29 | clyrrad | ManxPower: what do you suggest? |
| 16:27.10 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 16:27.27 | ManxPower | clyrrad: Have the AGI decide what to do, have it set an dialplan variable named something like TIMEROUTE=context,extension,priority, then as the priority after the AGI, do a Goto(${TIMEROUTE}) |
| 16:27.56 | ManxPower | where context, extension,priority is the actual context, extension, and priority you want to jump to. |
| 16:28.20 | ManxPower | If you jump around the dialplan from inside an AGI -- well let us just say it's not pretty. |
| 16:28.55 | ManxPower | clyrrad: I do something similar, but my AGI sets a variable that is later used by Dial |
| 16:29.15 | clyrrad | ManxPower: the onlything to that is there will be multiple TIMEROUTES, for holidays, closed hours etc... |
| 16:29.35 | clyrrad | ManxPower: kind of like having 5 or 6 GoToIfTime extens (if that makes more sense) ? |
| 16:29.40 | ManxPower | clyrrad: For any single call will there be multiple TIMEROUTES? |
| 16:30.15 | clyrrad | ManxPower: Yes, so a call comes in, and we need to check multipe timeroutes, IE: (is it a weekend, is it a holiday, is it after hours during the week etc....) |
| 16:30.22 | ManxPower | I'm telling you to do all your routing logic INSIDE your AGI, then whatever the result is for that call, set the TIMEROUTE variable to that. |
| 16:30.42 | ManxPower | clyrrad: That is not what I asked. I asked for any SINGLE call will there be more than one final destination? |
| 16:31.05 | *** join/#asterisk _alex_df_ (n=Alex@200.78.229.18) |
| 16:31.22 | clyrrad | ManxPower: no based on the evalutation of one of the GoToIfTIme extens it will end up in one spot or another before the call is terminated |
| 16:31.24 | ManxPower | Trust me, it's easier to do complex if statements in a real language than it is in the dialplan |
| 16:31.34 | ManxPower | clyrrad: I am telling you to NOT USE gotoiftime |
| 16:31.56 | ManxPower | take all the complex logic and put it in your AGI program |
| 16:32.20 | clyrrad | ManxPower: yea I hear ya :) - im just giving the "thought" of what I am after and the GoToIfTIme App is a good analogy of what im trying to do |
| 16:32.34 | ManxPower | clyrrad: you can use gotoiftime if you really like pain. |
| 16:32.40 | clyrrad | hahahaha |
| 16:33.04 | ManxPower | Of if you don' know perl, C, or PHP, but you had better learn one of them fast. |
| 16:33.15 | clyrrad | I agree the AGI would be more practical for this |
| 16:33.23 | clyrrad | yea im more than fine with PHP |
| 16:33.27 | clyrrad | I would write it in PHP |
| 16:33.57 | Juggie | agi is fine, but if you plan to have any volume, i would not recomend php |
| 16:33.59 | ManxPower | exten => _XXXX,1,AGI(build-route.php) |
| 16:34.12 | ManxPower | exten => _XXXX,n,Goto(${DIALROUTE}) |
| 16:34.32 | *** join/#asterisk mandd (n=dache@dsl-135-236.aei.ca) |
| 16:34.41 | clyrrad | ManxPower: Yep makes sense :) thanks |
| 16:34.46 | ManxPower | now you have shrunk 20 dialplan lines into 2 dialplan lines, and all your complex stuff is in the AGI |
| 16:35.17 | clyrrad | Yea, its alot clearner thats forsure |
| 16:35.33 | *** join/#asterisk [T]ank (n=[T]ank@206.71.78.158) |
| 16:35.34 | clyrrad | is it even possible to add to the dial plan dynamically? (im not gonna do it) just curious.......... |
| 16:36.25 | [T]ank | i still have a sangoma a104d for sale if anyone is interested in it. if so pm me. |
| 16:37.04 | ManxPower | clyrrad: in theory yes, in practice, that is what AMI and AGI are for. |
| 16:37.06 | *** part/#asterisk viperdude (n=viperdud@87-127-248-176.no-dns-yet.enta.net) |
| 16:37.18 | *** join/#asterisk JViz (n=JViz@72.242.173.74) |
| 16:37.33 | clyrrad | ManxPower: gotcha |
| 16:37.59 | _alex_df_ | hello, been running 1.4 for about a month now. The system is under heavy sip to zap load. It has a sangoma 4 port E1 card with all 120 channels used most of the day. Specs Quad xeon, 4GB ram, raid0 15k disks, centos 5.1. System will run for days under this load as long as i do not interact with the manager. Anything I do via manager will result in SIP clients loosing registration and not being able to re-register until i have to restart * in order |
| 16:37.59 | _alex_df_ | to restore service. Any ideas where to start debugging? |
| 16:38.21 | l2cache | Hey Manx, you can do the file convert from the CLI in 1.4, just tested it. Thought you'd like to know. |
| 16:38.32 | ManxPower | l2cache: nifty. |
| 16:38.41 | ManxPower | l2cache: I don't use 1.4 |
| 16:44.40 | *** join/#asterisk toyowheelin (n=gbolte@209.90.232.34) |
| 16:44.43 | toyowheelin | hey all |
| 16:45.19 | ManxPower | _alex_df_: upgrade. 1.4.0 had an incredible number of major bugs. |
| 16:46.03 | _alex_df_ | ManxPower, my bad using svn of about a week ago |
| 16:46.14 | ManxPower | _alex_df_: try using a released version |
| 16:46.42 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 16:46.45 | ManxPower | make sure you have the latest zaptel, libpri, and Asterisk (assuming you need PRI and Zap support) |
| 16:47.09 | ManxPower | If this was a general issue with 1.4 many people would be having problems |
| 16:47.40 | toyowheelin | I was wondering if there was a way to make asterisk announce over a PA system when there is a call waiting on an extension |
| 16:47.43 | _alex_df_ | ManxPower, all svn, but ok i'll switch to latest released versions see if that makes a difference |
| 16:49.07 | ManxPower | _alex_df_: if the code was tested it would have been released, not stuck in SVN |
| 16:49.47 | JViz | how much like an NSS is an Asterisk server with multiple GSM gateways? |
| 16:49.52 | _alex_df_ | ManxPower, force of habit, i don't think i've ever run anything but cvs then svn |
| 16:50.57 | lst | anyone around with ss7 experience? |
| 16:51.24 | ManxPower | _alex_df_: you thrill seeker, you. |
| 16:51.36 | JayTee52 | in a normal boot of an * server is ztcfg supposed to run in the init script each time? |
| 16:51.41 | *** join/#asterisk sione (i=sione@ocs.net) |
| 16:51.50 | ManxPower | JViz: Never heard of NSS, but I suspect "not very much" is the answer. |
| 16:52.07 | ManxPower | JayTee52: only if you want zaptel to work |
| 16:53.15 | JViz | ManxPower: NSS is the Network and Switching Subsystem of a GSM network. |
| 16:53.20 | sione | so I unpluged my modem to my sercuity stuff that when it calls home it triggers ghost rings, and i am still getting them ghost rings on my TDM410 :( |
| 16:53.56 | ManxPower | JViz: then the answer is "not at all" |
| 16:54.31 | ManxPower | sione: so I guess the modem is NOT causing the problem |
| 16:55.01 | ManxPower | sione: the thread about this problem on the mailing lists this week was not helpful? |
| 16:55.23 | sione | im not on any mailing thread so i did not catch that |
| 16:55.49 | ManxPower | sione: It sucks to be you. |
| 16:55.53 | sione | blah! |
| 16:56.17 | ManxPower | I don't know if the problem was RESOLVED, as I don't ever use analog so I did not pay attention to the thread |
| 16:57.10 | sione | wonder if SBC tech is probing the line with their buttset cuasing it :) |
| 16:58.24 | sione | if only I can get asterisk not to pick up call on RP and just pick up on ring my problem would be solved :) |
| 16:59.52 | JayTee52 | ok, my server hangs after loading the module for the PRI while trying to run ztcfg. I previously had it up and running though. |
| 17:01.18 | JayTee52 | and the server is locked up at that point |
| 17:01.20 | ManxPower | JayTee52: are you using Sangoma? |
| 17:01.33 | JayTee52 | no, I'm using a Digium TE212P |
| 17:01.55 | ManxPower | I assume that's a hardlock? |
| 17:01.59 | JayTee52 | yes |
| 17:02.06 | ManxPower | Then you know what to do. |
| 17:02.07 | *** join/#asterisk robeph (n=robf@router.asteriasgi.com) |
| 17:02.34 | JayTee52 | I can't even Ctrl-Alt-F1 into a terminal and the eth0 isn't up to ssh into it. |
| 17:02.46 | rotozip | ManxPower: I can call my works conference line and listen to the hold music and I can see that at a certain point in the music is where I get a load DTMF tone. I set relaxdtmf to no in zapata.conf and it looks like it may have fixed it. Do you think that is feasible or a coincidence? |
| 17:02.51 | JayTee52 | ManxPower, start in interactive and not load zaptel? |
| 17:02.53 | robeph | any one know of any rfcs / standards papers I can look at that covers telephony system voltage standards etc? I know the voltages but this requires I have something thats "authoritive" to reference =\ and I can't find any such docs |
| 17:03.01 | ManxPower | JayTee52: no, call digium. |
| 17:03.21 | JayTee52 | we bought it from telephonydepot, will Digium support it? |
| 17:03.28 | robeph | JayTee52: if they made it |
| 17:03.39 | ManxPower | robeph: Bellcore / Telecordia have those standards, but they are hundreds or thousands of dollars to purchase. |
| 17:03.44 | JayTee52 | ok, I'll give them a call |
| 17:03.47 | robeph | ManxPower: :O |
| 17:03.57 | C4away | wikipedia.org ? |
| 17:03.57 | C4away | lol |
| 17:04.04 | ManxPower | JayTee52: You have a HARDWARE COMPAT issue -- that is something Digium should handle |
| 17:04.15 | JayTee52 | ManxPower, thanks |
| 17:04.29 | Corydon76-dig | robeph: yeah, the standards are rather expensive |
| 17:04.32 | C4away | how authoritative do you need this source to be? |
| 17:04.46 | Corydon76-dig | robeph: you might luck out and be able to find it on the itu.int site |
| 17:05.02 | Corydon76-dig | robeph: you can download ITU standards for free |
| 17:05.17 | Corydon76-dig | Just a matter of finding the right standard |
| 17:09.01 | ManxPower | Corydon76-dig: you can download SOME ITU standards for free. |
| 17:09.22 | Corydon76-dig | ManxPower: Strom downloaded ALL of them and gave me a copy |
| 17:09.24 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 17:09.33 | Corydon76-dig | as of about a year ago |
| 17:10.36 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 17:10.48 | ManxPower | Corydon76-dig: weird. |
| 17:10.58 | Qwell | the entire set is *huge* |
| 17:11.02 | Strom_C | yeah |
| 17:11.05 | ManxPower | ALL is quite a large number |
| 17:11.11 | Qwell | it's like, what, 1.7GB? |
| 17:11.13 | ManxPower | Strom_C: how did you get them? |
| 17:11.14 | Strom_C | it's like one or two gigabytes |
| 17:11.28 | Strom_C | ManxPower: they were/are available for free on the itu website |
| 17:11.49 | Qwell | 1.6G /home/qwell/ITU-T_Bookshelf/ |
| 17:12.01 | Corydon76-dig | Heh |
| 17:12.06 | Strom_C | so after a half hour of clicking like mad, I stopped, said "wait, PERL!" and just left it running until it finished |
| 17:13.01 | ManxPower | They must have changed their policy |
| 17:13.20 | Strom_C | yeah, it was advertised as a "limited trial period" |
| 17:14.00 | ManxPower | ah. |
| 17:14.01 | *** join/#asterisk aaawrekng (n=mark@c-76-121-221-213.hsd1.wa.comcast.net) |
| 17:14.11 | ManxPower | Last time I tried, they gave you three free specs |
| 17:14.20 | Qwell | only 3? that's...useful |
| 17:14.23 | Qwell | s/ful/less/ |
| 17:14.37 | Strom_C | yeah, that was the policy some time ago |
| 17:14.51 | Strom_C | i noticed the free specs in...what, March 2007? |
| 17:14.55 | Qwell | Strom_C: You know what we need? |
| 17:15.01 | Qwell | ITU book on tape. |
| 17:15.02 | sione | will be back later |
| 17:15.12 | Qwell | You should get on that. |
| 17:15.25 | Strom_C | hahaha |
| 17:15.32 | Qwell | "SS7, as read by William Shatner" |
| 17:15.37 | Strom_C | ManxPower: looks like they made it permanent |
| 17:15.39 | ManxPower | Strom_C: these days I tend to be about 1 year behind current reality |
| 17:15.39 | Strom_C | http://www.itu.int/publications/default.aspx |
| 17:15.54 | Strom_C | Qwell: have you ever seen Telcordia Notes on the Networks? |
| 17:16.00 | Qwell | no? |
| 17:16.26 | Strom_C | it's basically a brief summary of how everything in telecommunications in north america works |
| 17:16.33 | Qwell | brief? |
| 17:16.35 | Strom_C | and it's a mere 1500 pages long |
| 17:16.38 | Qwell | is that anything like The Brief History of Time? |
| 17:16.40 | Qwell | indeed... |
| 17:16.48 | Qwell | s/The/A/? |
| 17:17.16 | Strom_C | when I was about to quit Ticketmaster, I decided to try printing it because I figured I had been good and not abused the printer for a year |
| 17:17.29 | Corydon76-dig | History of the world, Part II ? |
| 17:17.31 | Strom_C | even double-sided, it's a massive thing |
| 17:17.47 | Qwell | Strom_C: well, publishers use a bit thinner paper... |
| 17:17.51 | Qwell | printer paper is pretty thick |
| 17:17.55 | Strom_C | true |
| 17:18.01 | Qwell | but even still |
| 17:18.25 | Corydon76-dig | If you dropped the manual off the Empire State Building, would it kill someone? |
| 17:18.47 | Corydon76-dig | Or just give them a headache, like a penny? |
| 17:19.02 | Strom_C | Corydon76-dig: that probably depends on whether it was allowed to flap open |
| 17:19.10 | M1s3ry | 15 lb book flying at terminal velocity into someone's head... I think it would be messy |
| 17:19.39 | *** part/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 17:19.41 | Corydon76-dig | M1s3ry: it's already been demonstrated that a penny does nothing but hurt like a bitch |
| 17:20.00 | Corydon76-dig | (No, it doesn't embed itself) |
| 17:20.35 | jeev | Corydon76-dig, well, i had some issues yesterday with MOH, im thinking about just taking it out.. or i'll find a new mplayer. |
| 17:23.40 | jeev | is it possible to put a "beep beep" instead of moh ? |
| 17:26.55 | *** join/#asterisk Darthclue (n=Darthclu@fw149.nisd.net) |
| 17:27.17 | *** join/#asterisk xenonex (n=xenonex@92.47.0.5) |
| 17:29.51 | [TK]D-Fender | jeev: make a huge recording of "beep beep" and use that as a MoH class |
| 17:30.13 | *** join/#asterisk s0lid (n=s0lid@210.213.242.60) |
| 17:31.51 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 17:32.03 | jeev | hahaha fender |
| 17:32.38 | jeev | the girl here said that she was getting some static.. randomly. is that really due to the sound card? i hope it's not the connection |
| 17:32.48 | jeev | i can't seem to transfer a call, i enabled it in features |
| 17:33.52 | RobH | jeev: when you dial the person and the call is connected, you need to ensure you use the t or T option |
| 17:33.56 | RobH | core show application dial |
| 17:35.19 | [TK]D-Fender | jeev: Details would be nice... |
| 17:36.59 | jeev | about the static? tk ? |
| 17:38.45 | [TK]D-Fender | jeev: About exactly what hardware is being used where this happens. |
| 17:39.18 | *** join/#asterisk kurama10 (n=kurama10@host-200-94-16-180.block.alestra.net.mx) |
| 17:39.22 | kurama10 | Hi, anyone know of some sort of monitoring asterisk to send mails when something goes wrong |
| 17:39.50 | jeev | she is using actually it was a dell server with an added sound card. let me go get the details. |
| 17:44.16 | ManxPower | ~ecfo |
| 17:44.17 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly realises its a crappy design based on a half baked 20 year old apps note. |
| 17:45.08 | jeev | so the possible solution is the rxgain |
| 17:45.25 | ManxPower | jeev: I've not heard of anyone having this issue for quite a while. |
| 17:46.51 | jeev | my rxgain is on default. |
| 17:48.16 | *** part/#asterisk [T]ank (n=[T]ank@206.71.78.158) |
| 17:52.35 | [TK]D-Fender | jeev: You still haven't described your situation. How do you expect us to advise you? |
| 17:54.13 | jeev | i'm sorry tk, i got side tracked by not being able to transfer.. appreciate your help, let me get it |
| 17:55.00 | [TK]D-Fender | jeev>i can't seem to transfer a call, i enabled it in features <- again no details. |
| 17:55.35 | jeev | ah |
| 17:55.36 | jeev | sorry |
| 17:55.45 | jeev | [featuremap] |
| 17:55.46 | jeev | blindxfer => #1 ; Blind transfer (default is #) |
| 17:55.46 | *** join/#asterisk DirtyDD (n=DirtyD@ool-18bddaa0.dyn.optonline.net) |
| 17:55.49 | jeev | i pretty much want a blind transfer |
| 17:56.06 | jeev | but RobH told me that i need to add t or T option, correct? |
| 17:56.11 | DirtyDD | SNOM and SLA. |
| 17:56.20 | *** part/#asterisk kurama10 (n=kurama10@host-200-94-16-180.block.alestra.net.mx) |
| 17:57.59 | [TK]D-Fender | jeev: Lets try this again... |
| 17:58.42 | jeev | ok... |
| 17:59.00 | [TK]D-Fender | jeev: HARDWARE! What equipment are yuo using exactly to do what? These details matter. 99% of decent phones & softphones support native SIP transfers and don't need * to do anything for them. |
| 17:59.28 | jeev | X-Lite, it doesn't allow it because i dont have eyeBeam |
| 17:59.36 | [TK]D-Fender | jeev: Its like asking yrou doctor about possible health issues for 10 hours before telling him you're asking about your HAMSTER. |
| 18:00.42 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 18:01.33 | jeev | ok bro, i use X-Lite, i hoping i could pound and transfer elsewhere. |
| 18:02.41 | [TK]D-Fender | jeev: Ok, X-Lite doesn't do transfers.pastebin a complete call attempt at verbose 10 and include your features.conf |
| 18:03.13 | jeev | ok, should i make an outgoing call or incoming |
| 18:03.57 | jeev | http://pastebin.com/m2494c94e |
| 18:04.01 | jeev | that is pastebin for features.conf |
| 18:06.56 | jeev | Fender, i'm having trouble with the placement of the "T", i broke something. |
| 18:07.11 | jfg | i need some support on libiaxclient, could someone help me ? |
| 18:07.44 | robeph | pokes Strom_C |
| 18:07.53 | Strom_C | cocks |
| 18:08.02 | cpm | balls |
| 18:08.05 | cpm | said the Queen, |
| 18:08.10 | robeph | :o |
| 18:08.14 | Strom_C | and everything in between |
| 18:08.14 | cpm | if I had to, I'd be King |
| 18:09.03 | robeph | heh, I didn't need the whole standards , just something "authoritive" on line voltages, not just a wiki page or something... so no real need for me to even consider buying up some 1k$ standards documentation >< |
| 18:09.10 | robeph | cpm: it's good to be king. |
| 18:09.11 | *** join/#asterisk akafurious (n=akafurio@CPE00121713f42b-CM000a73a87e26.cpe.net.cable.rogers.com) |
| 18:10.22 | cpm | indeed |
| 18:10.23 | cpm | ! |
| 18:10.52 | jeev | Fender |
| 18:12.36 | jeev | Fender, you want the DTMF included? |
| 18:13.08 | *** join/#asterisk Brixius (n=Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 18:13.41 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:14.38 | [TK]D-Fender | jeev: Whats your CLI output? |
| 18:14.43 | Brixius | Hello, does anyone know if there is a way to allow saydigits to be interrupted in an ivr? |
| 18:14.47 | ManxPower | robeph: Ring voltage is 90VAC and non-ringing voltage is -48VDC |
| 18:15.16 | ManxPower | Brixius: no, I always used Background to play the individual digits. Even wrote a macro, IIRC |
| 18:15.20 | jeev | debug. |
| 18:15.57 | [TK]D-Fender | jeev: WHERE rather.... |
| 18:16.20 | jeev | i have it going to my main screen and also the log |
| 18:16.22 | cpm | always thought that ringing was 90v pulsed DC, which I guess is the same thing, just not sine |
| 18:16.34 | Brixius | ManxPower: ok, thanks, not exactly the answer I was hoping for, but it's what I needed to know. |
| 18:16.54 | cpm | I do know what it feels like when you stripping wire with your teeth though |
| 18:17.05 | robeph | ManxPower: yeah I thought so, but what I can't find, is off hook, is 4-6 volts, every where I look it doesn't denote negative voltage, which seems odd, but I want to be for sure the actual standard.. |
| 18:17.14 | Strom_C | what kind of idiot strips wire with his teeth? |
| 18:17.20 | cpm | me, |
| 18:17.21 | Qwell | raises his hand |
| 18:17.22 | robeph | Strom_C: *raises hand* |
| 18:17.30 | robeph | Strom_C: usually only if it's live though |
| 18:17.31 | Brixius | raises hand |
| 18:17.35 | robeph | otherwise the thrill isn't there |
| 18:17.37 | cpm | not so much any more, but for decades, it was the handiest wirestripper around |
| 18:17.37 | Strom_C | jesus, you guys...they make tools for that, you know |
| 18:17.51 | jeev | [TK]D-Fender, http://pastebin.com/m6916ac11 |
| 18:17.54 | robeph | heh |
| 18:17.54 | Qwell | 90vac to your teeth hurts :p |
| 18:18.03 | cpm | man, when that line rings, that'll wake ya up |
| 18:18.09 | Brixius | doesn't do it on live wires anymore, kindof painfull at times. |
| 18:18.11 | cpm | it does indeed |
| 18:18.12 | robeph | Qwell: 90vac will cause heart arhythmia ;) |
| 18:18.32 | cpm | may, not will |
| 18:18.40 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 18:18.49 | robeph | let it cross mid point and that may is like betting on winning the lottery |
| 18:18.54 | cpm | yeah, |
| 18:19.12 | cpm | but in yer teeth, it's just 'enlightening' :) |
| 18:19.13 | robeph | that stuff is dangerous, but its more a where ya arc it through rather than a chance based on luck |
| 18:19.14 | *** join/#asterisk hardwire (n=bip@xvm-189-205.ghst.net) |
| 18:19.21 | Qwell | cpm: it's something you don't do twice |
| 18:19.22 | robeph | yeh true |
| 18:19.27 | Strom_C | robeph: I can extract the relevant bits of telcordia sr-2275-4 for you... |
| 18:19.46 | [TK]D-Fender | jeev: -- Executing [s@macro-stdexten:1] Dial("SIP/222-086b8000", "SIP/6000&IAX2/6000|20") in new stack |
| 18:19.52 | cpm | Qwell, yes, a lesson learned iwht zen like clarity |
| 18:19.54 | jeev | i just disabled IAX. |
| 18:19.54 | robeph | Strom_C: that'd be awesome =) |
| 18:19.57 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:19.57 | [TK]D-Fender | jeev: You aren't even ALLOWING DTMF transfers here. |
| 18:20.14 | jeev | with "T" or "t" ? |
| 18:20.26 | [TK]D-Fender | jeev: Yes |
| 18:20.30 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 18:20.32 | jeev | ok |
| 18:20.59 | jeev | exten => s,1,Dial(${ARG2},20) |
| 18:21.02 | jeev | to exten => s,1,Dial(${ARG2},20,Tt) |
| 18:21.03 | jeev | ? |
| 18:21.44 | [TK]D-Fender | jeev: if you want both to be able to be transfered. |
| 18:21.59 | jeev | yea, i'd like incoming and outgoing calls to be transferred. |
| 18:22.17 | jeev | ok, i've restarted. let me test |
| 18:22.19 | [TK]D-Fender | jeev: NO, not "incoming/outgoing". this is CALLER/CALLEE |
| 18:22.28 | jeev | ya |
| 18:23.23 | jeev | -- Executing [222@numberplan-custom-1:1] Macro("SIP/6000-086b4000", "stdexten|222|SIP/222") in new stack |
| 18:23.24 | jeev | -- Executing [s@macro-stdexten:1] Dial("SIP/6000-086b4000", "SIP/222|20|Tt") in new stack |
| 18:23.27 | jeev | something still dysfunctional ? |
| 18:23.32 | *** part/#asterisk toyowheelin (n=gbolte@209.90.232.34) |
| 18:25.03 | [TK]D-Fender | jeev: better |
| 18:25.31 | *** join/#asterisk wonderworld (n=voici@ip-62-143-163-248.1211G-CUD12K-01.ish.de) |
| 18:26.13 | wonderworld | hi, i am trying to record a sound from within an agi script. this is from the agi debug log: |
| 18:26.29 | wonderworld | AGI Rx << RECORD FILE "/tmp/pphonein/833340322" "wav" "*" 30 BEEP |
| 18:26.29 | wonderworld | <PROTECTED> |
| 18:26.29 | wonderworld | AGI Tx >> 200 result=0 (timeout) endpos=480 |
| 18:26.56 | jeev | ok, i attempted, #225 or #1 225, it's not working |
| 18:26.57 | wonderworld | i can't figure out why it's timing out. because i set the timeout to 30 seconds? |
| 18:27.04 | jeev | let me get you dtmf. |
| 18:27.19 | [TK]D-Fender | jeev: And your blind transfer has been remapped, and I don't see an attended transfer being set. NOR do you set your dynamic_featuress variable |
| 18:27.36 | jeev | oh, attended transfer is required? |
| 18:28.00 | [TK]D-Fender | jeev: Go to the WIKI, pull up "config features.conf", and READ. |
| 18:28.09 | jeev | ye, i just noticed it |
| 18:28.51 | wonderworld | i can't record at all, i hear the beep and it stops recording immediately after that, going on in my script |
| 18:30.27 | wonderworld | the file actually is recorded, i get about 50 bytes large files in /tmp/pphonein |
| 18:31.20 | jeev | [TK]D-Fender, i just want US to be in control of transfers, you know what i mean, so let me figure i tout |
| 18:33.02 | wonderworld | never mind, i found out what was wrong. RECORD FILE wants the timeout in milliseconds. 30 must have been damn small..... |
| 18:35.53 | [TK]D-Fender | wonderworld: Yeah, sometimes they hide it in the BIG PRINT :) |
| 18:38.45 | *** join/#asterisk infinity3 (i=brendon@saleen.netcal.com) |
| 18:38.58 | infinity3 | is there a sip provider that gives you like 50 cents free to test with? |
| 18:41.51 | jeev | [TK]D-Fender, this features.conf is either very poorly documented or i'm an idiot.. you'd agree with the latter |
| 18:42.02 | [TK]D-Fender | jeev: possibly both. |
| 18:42.38 | jeev | do you have an example i could steal with you? i'm so confused right now :) |
| 18:43.05 | [TK]D-Fender | jeev: WIKI page + sample features.conf. |
| 18:43.16 | [TK]D-Fender | jeev: use the basics. |
| 18:43.18 | jeev | i have looked, why do you think i was quiet for 10 min |
| 18:43.37 | *** join/#asterisk nicchap (n=nicchap@204.101.222.130) |
| 18:43.38 | jeev | what exactly is this "Set(DYNAMIC_FEATURES=hangup#play#testfeature)" |
| 18:44.23 | jeev | the examples aren't relevent :/ |
| 18:45.21 | [TK]D-Fender | jeev: keep reading... |
| 18:45.47 | jeev | ok bro i will |
| 18:45.53 | jeev | sorry for my impatience |
| 18:46.05 | [TK]D-Fender | jeev: There is a pretty blatant one ont he WIKI page... |
| 18:46.12 | [TK]D-Fender | jeev: http://www.voip-info.org/wiki/view/Asterisk+config+features.conf&view_comment_id=15759 |
| 18:46.56 | nicchap | anyone know if the "facilityenabled" flag in zapata.conf can be used/set on a per channel basis? Trying to enable RLT on some calls and disbale on other calls |
| 18:47.18 | Shotygun | Does anybody here knows how can I determine if the SIP destination has DND enabled upon Dial? For having better error handling than CONGESTION.. |
| 18:47.27 | [TK]D-Fender | nicchap: Everything in zapata.conf is per-channel |
| 18:47.38 | [TK]D-Fender | Shotygun: No, you can't |
| 18:47.55 | nicchap | ok tks [TK]D |
| 18:47.55 | Shotygun | [TK]D-Fender: No way perhaps to use the 480 reply? |
| 18:48.17 | ManxPower | Shotygun: try HANGUPCAUSE |
| 18:48.22 | [TK]D-Fender | Shotygun: DND on your phones isn't a state it tells *, but rather jsut tells you phone tor eject or not. How it answers is up to the phone. |
| 18:48.22 | ManxPower | not DIALSTATUS |
| 18:48.51 | Shotygun | HANGUPCAUSE returns 38, which is pretty much same for CONGESTION, "Network out of order" |
| 18:49.00 | *** join/#asterisk bluregard (n=Bloo@76.29.119.76) |
| 18:49.27 | bluregard | hello |
| 18:50.16 | ManxPower | Shotygun: I've not ever seen any SIP device that returned that code on a DND |
| 18:51.03 | Shotygun | ManxPower: Snom320 with firware v7 does, wait I'll try to show you (not in the office so need to figure out how to turn dnd on remotely on the device) |
| 18:52.03 | Shotygun | brb, vpn |
| 18:53.42 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
| 18:53.46 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 18:56.46 | jeev | [TK]D-Fender, is this what you're talking about? "testfeature => *9,callee,Playback,tt-monkeys ;Play tt-monkes to callee if *9 was pressed - use 'callee' or 'caller' " |
| 18:56.53 | Jumpie | yo guys |
| 18:57.11 | jeev | sup Jumpie |
| 18:57.18 | Jumpie | is it possible to use a digium appliance, but blow away the asterisk now stuff and load a full os/asterisk install on the flash? |
| 18:57.36 | Jumpie | or would that not be feasible |
| 18:57.37 | Jumpie | lol |
| 18:57.45 | Qwell | Jumpie: the AA50? |
| 18:59.34 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 18:59.54 | Jumpie | Qwell, ...well either or |
| 19:00.01 | Qwell | or what? |
| 19:00.10 | Jumpie | im just trying to evaluate my hardware platforms, was hoping for somethin with some fxo and or t1 cards integrated |
| 19:00.16 | Jumpie | qwell the higher end ones |
| 19:00.21 | Jumpie | dunno all the models offhand yet |
| 19:00.26 | Qwell | I don't think we don't sell anything with AsteriskNOW on it.. |
| 19:00.35 | Jumpie | i thought those appliances did? |
| 19:00.36 | Jumpie | my bad |
| 19:00.57 | Jumpie | im just trying to find either one piece of hardwared that has what i need, or one vendor that sells it all |
| 19:01.11 | Jumpie | i was gonna go dell for server and telephonydepot for cards but seeing if an alternative |
| 19:02.19 | Qwell | nothing wrong with Dell |
| 19:02.27 | Jumpie | oh i know |
| 19:02.29 | Qwell | (well, okay, there is a lot wrong with Dell, but that isn't the point) |
| 19:02.32 | Jumpie | lol |
| 19:02.35 | Qwell | I also hear good things about supermicro |
| 19:02.42 | Jumpie | yea, i was lookin at theirs too |
| 19:02.48 | Jumpie | about same price |
| 19:02.56 | Jumpie | but personaly have worked with power edge servers, not super micro |
| 19:03.04 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 19:03.12 | Jumpie | i just know im gonna have a standard bundle, which is 1 t1 card, and 4fxo card |
| 19:03.17 | Jumpie | was hoping to have it all included |
| 19:03.19 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 19:03.38 | ManxPower | I'll never tough another supermicro computer ever again |
| 19:03.42 | ManxPower | ..er..touch |
| 19:05.28 | Jumpie | bad exp eh |
| 19:05.42 | ManxPower | Jumpie: almost made me lose a client |
| 19:05.52 | Jumpie | ouch |
| 19:06.02 | Jumpie | was it their cust service? or hte hardware itself |
| 19:06.10 | ManxPower | I ended up having to buy the system from the client, and buying them a comparable system that worked with a Digium card. |
| 19:06.19 | ManxPower | Jumpie: HDLC Abort |
| 19:06.33 | ManxPower | any time there was any significant disk activity |
| 19:07.21 | ManxPower | Sp basically Supermicro cost me $1,200 |
| 19:07.28 | ManxPower | never again |
| 19:07.50 | Jumpie | ouch |
| 19:07.55 | Jumpie | so it was the card, not the os? |
| 19:08.01 | Jumpie | i think issues like you spawned their disclaimer |
| 19:08.10 | Jumpie | it said to ensure their board works with certain os, and shows an os matrix |
| 19:08.12 | Jumpie | oll |
| 19:08.28 | Jumpie | success with power edg servers though with your clients? |
| 19:11.12 | SwK | someone sent up hunstville the bomb |
| 19:11.43 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 19:11.49 | ManxPower | Jumpie: I think the chipset was just locking interrupts for a long time. I've seen similar problems reported by people |
| 19:12.25 | jeev | i'm so unbelievably confused, features.conf has to be the worst thing i've had to deal with so far. |
| 19:12.35 | Jumpie | hmm ok |
| 19:12.37 | Jumpie | good to know |
| 19:12.38 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:12.38 | jeev | i dont understand how/why it's so difficult to transfer a call |
| 19:12.43 | Jumpie | ill be damned if i eat 1200 |
| 19:12.44 | Jumpie | lol |
| 19:12.56 | jeev | http://www.voip-info.org/wiki/index.php?page_id=1483#editcomments that doesn't even tell you what to do or what to understand, it just talks about tt-monkeys |
| 19:13.10 | ManxPower | jeev: It's not hard to transfer a call. You press the TRANSFER button on your phone and do the transfer. This is not rocket science and you don't need features.conf to do a transfer. |
| 19:13.29 | [TK]D-Fender | ManxPower: In hisX-lite, yes he does |
| 19:13.40 | ManxPower | Now if your phone is too STUPID and BRAIN DEAD to have a transfer feature, then you will need to use the evil features.conf DTMF transfer hack. |
| 19:13.44 | jeev | Manx, i'm using X-Lite. |
| 19:14.04 | ManxPower | jeev: so your phone is to stupid and brain dead to have a transfer button. |
| 19:14.07 | [TK]D-Fender | jeev: You need to enable atxfer & blindxfer |
| 19:14.17 | jeev | [TK]D-Fender, could you give me a hint on something? i've read and read, i dont seem to understand what the point of this is, it just talks about tt-monkeys |
| 19:14.29 | jeev | ok Fender, i have done that. now up there it says #1 then at the bottom, it talks bout #9 |
| 19:14.40 | ManxPower | jeev: most people with decent phones (hardphone or softphone) can transfer just fine with no need for features.conf |
| 19:14.50 | [TK]D-Fender | jeev: Set(DYNAMIC_FEATURES=hangup#play#testfeature) <- I said set the DAMN VARIABLE |
| 19:15.11 | Jumpie | i have the freebie xlite |
| 19:15.12 | ManxPower | You are only having this issue because you don't have a transfer button on your phone. |
| 19:15.14 | *** join/#asterisk CallCtr4Sale (n=hellowor@124.6.168.4) |
| 19:15.15 | Jumpie | xfer not available hehe |
| 19:15.18 | CallCtr4Sale | hi everyone |
| 19:15.38 | CallCtr4Sale | im looking for a toshiba strata consultant |
| 19:15.40 | jeev | ManxPower, what are you trying to say? both of us just told you X-Lite doesn't have transfer enabled. |
| 19:15.56 | jeev | when the time comes, i will get phones with transfer buttons. |
| 19:16.07 | ManxPower | jeev: What I'm trying to say is that you made a very poor choice when you decided what phone to use. |
| 19:16.10 | Jumpie | lol |
| 19:16.17 | *** join/#asterisk RoyK (n=roy@ip-210-6-149-91.dialup.ice.no) |
| 19:16.22 | jeev | Manx, ever heard of testing things? |
| 19:16.43 | ManxPower | jeev: sure, but usually you want to test things as close to reality as you can. |
| 19:17.01 | jeev | yea well i'm sorry i can't do anything right now, i'd prefer to get features working for now.. then worry about the actual phone later |
| 19:17.07 | ManxPower | You are basically trying to evaluate a ford mustang by driving a ford pinto. |
| 19:17.15 | jeev | i would never drive a ford mustang. |
| 19:17.21 | Jumpie | outtalunc recommended a fairly cheap wifi nokia phone, i cant remember which model |
| 19:17.26 | Jumpie | he said it was about $100..anyone remember? |
| 19:17.31 | jeev | WIP300 maybe? |
| 19:17.38 | ManxPower | jeev: and yet, all this time you spent trying to use features.conf will be wasted when you get a real phone. |
| 19:17.39 | jeev | i want the WIP330 though |
| 19:17.45 | Jumpie | when i was a baby my parents had a pinto :) |
| 19:17.54 | jeev | ManxPower, the office which i want to set up will deal with soft phones for now. |
| 19:17.56 | Jumpie | jeev, i just wanted somethin wifi that works |
| 19:18.00 | Jumpie | i dont need 987082760927 bells and whilstles |
| 19:18.06 | jeev | hehe , i think it's cool |
| 19:18.17 | Shotygun | jeev: I'm now testing snom m3, it seems now |
| 19:18.19 | Shotygun | now = nice |
| 19:18.21 | jeev | [TK]D-Fender, sorry i missed your message, i did set the dynamic features. |
| 19:18.25 | jeev | which is that ? |
| 19:18.40 | Shotygun | wifi phone |
| 19:18.46 | Shotygun | wifi voip phone, not a cell. |
| 19:18.50 | *** join/#asterisk spjuden (n=pithen@mail.graphlogic.com) |
| 19:19.15 | jeev | yea, that's what the WIP330 is |
| 19:19.58 | jeev | [TK]D-Fender, it's the stuff after the Dynamic features, like the actual config to testfeature. |
| 19:20.06 | Jumpie | how much Shotygun |
| 19:20.14 | spjuden | What would cause me to occasionally get an error that the callerid checksum failed (and thus not get caller id info for that call)? It happens on all my FXO chans, but all FXO chans do get good data the majority of the time |
| 19:20.15 | Shotygun | I just took the snom m3 out of the box. It comes in a set I find useful but might be annoying to some. There is a base station which is connected to power & ethernet. And comes with another unit which is just the charger. You can purchase additional handsets that come with only a charger and no base |
| 19:20.44 | Shotygun | Jumpie: Can't recall, I got it for trial-out. |
| 19:21.05 | ManxPower | spjuden: the rxgain being set a little too high or too low is the most common cause of that in my experience |
| 19:21.41 | spjuden | ManxPower, interesting, ill take a look at that |
| 19:21.45 | ManxPower | I suppose I should go work on my tan before it starts storming |
| 19:21.59 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 19:22.50 | Jumpie | manx honestly do you recommend digium over sangoma or no? |
| 19:22.55 | Jumpie | i think sangoma is only one that offers pci-e cards right? |
| 19:23.04 | spjuden | ManxPower, east coast? |
| 19:23.13 | ManxPower | Jumpie: I do not recommend one over the other. |
| 19:23.26 | ManxPower | spjuden: Birmingham |
| 19:24.05 | ManxPower | *I* use Sangoma on 4 or so systems, but I have NOT removed the Digium cards from the existing systems |
| 19:24.12 | spjuden | ManxPower, if you're getting the same from we're supposed to get in hartford, its supposed to be pretty crappy weekend |
| 19:24.21 | spjuden | s/from/front |
| 19:24.49 | jeev | would this be it? exten = _1XXXXXXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},${trunk_1_cid}) TO exten = _1XXXXXXXXX!,1,Macro(trunkdial,${trunk_1},Tt/${EXTEN:0},${trunk_1_cid}) |
| 19:24.50 | ManxPower | spjuden: It's supposed to start storming here soon. We are about 3 hr drive almost due west of Atlanta |
| 19:25.02 | jeev | add the Tt after trunk_1 |
| 19:25.35 | [TK]D-Fender | jeev: Look what your macro is doing. |
| 19:25.38 | ManxPower | jeev: Dial(Tech/DEST,TIMEOUT,OPTIONSLIKETANDt) |
| 19:25.53 | ManxPower | jeev: we don't know your macro, we only know what you need to put on your dial line |
| 19:26.12 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 19:26.18 | jeev | exten = s,1,set(CALLERID(all)=${IF($["${LEN(${CALLERID(num)})}" > "6"]?${CALLERID(all)}:${ARG2})}) |
| 19:26.19 | jeev | exten = s,n,Dial(${ARG1}) |
| 19:26.19 | ManxPower | jeev: either you are an Asterisk guru with a sick sense of humor or you are using a GUI version of Asterisk. |
| 19:26.31 | Shotygun | heh |
| 19:26.42 | jeev | i was using the gui... for 2 days, then decided to just build 1.4.19 on my bsd box... so i copied some of the config over. |
| 19:26.50 | ManxPower | jeev: and yet you are making us parse everything to hopefully know what the value of ARG1 is. |
| 19:27.18 | ManxPower | jeev: the primary reason we hate Asterisk guis so much here is because they have such complicated CONFIG files. |
| 19:27.38 | ManxPower | So you basically installed asterisk from source, then copied over all the stuff we have about guis into your system. Way to go. |
| 19:27.44 | [TK]D-Fender | ManxPower: I think I may have been involved with some of that.. |
| 19:27.47 | ManxPower | sorry, have not have |
| 19:28.01 | ManxPower | [TK]D-Fender: pervert 8-) |
| 19:28.15 | jeev | yea, i've got to admit, it was rather stupid (the gui version) |
| 19:28.17 | ManxPower | I shall leave the poor sod to you then. |
| 19:28.21 | [TK]D-Fender | ManxPower: You're turn to be the "pot" ;) |
| 19:28.21 | bitzero | is going to go insane. |
| 19:28.25 | [TK]D-Fender | your* |
| 19:28.34 | Jumpie | hmmm |
| 19:28.42 | [TK]D-Fender | SOD OFF :p |
| 19:28.45 | Jumpie | iis getting pci-e really going to make a huge performance difference over pci? |
| 19:28.47 | [TK]D-Fender | </aussie> |
| 19:28.48 | Jumpie | when it comes to these cards? |
| 19:28.51 | Jumpie | i'd like to stick with the same thing |
| 19:28.56 | jeev | hm |
| 19:28.58 | Shotygun | Any of you used SIP_HEADER() before? |
| 19:29.06 | [TK]D-Fender | Jumpie: not for T1 |
| 19:29.28 | [TK]D-Fender | Jumpie: Its a question of what ports you have an need to keep free,etc. |
| 19:29.43 | jeev | so was $ARG1 $trunk_1 ? |
| 19:30.35 | Jumpie | fender, well basicaly |
| 19:30.47 | Jumpie | my average 'package' is gonna be a 1 t1 card and 4fxo card |
| 19:31.06 | jeev | http://pastebin.com/m2e58f401 |
| 19:31.51 | jbeez | Jumpie: knock it off |
| 19:31.57 | Jumpie | knock what off? |
| 19:32.02 | jbeez | :> |
| 19:32.53 | Jumpie | jbeez |
| 19:32.53 | Jumpie | lol |
| 19:32.53 | Jumpie | sup |
| 19:32.53 | jbeez | n/m |
| 19:32.53 | Jumpie | what choo doing here |
| 19:32.53 | jbeez | had some questions about handsets |
| 19:32.53 | jbeez | my new job has this asterisk box sitting here, they dropped like $20k on it and aren't even using it |
| 19:33.39 | Shotygun | jbeez: And I thought seeing $3k spent for something similar like that is horrible.. |
| 19:34.08 | jbeez | well its a nice 2u hp box with sas drives, and im sure the server itself costs some bucks, and they have some pri cards or something in it |
| 19:34.20 | jeev | [TK]D-Fender, did you see my pastebin post ? |
| 19:34.26 | Qwell | jbeez: why is it just sitting there? |
| 19:34.50 | jbeez | I still have to find out the name of the company that put it in and call them, but |
| 19:34.58 | Jumpie | yup |
| 19:35.03 | jbeez | they can't get the cavalier pri line working with it |
| 19:35.16 | Jumpie | small world |
| 19:35.19 | jbeez | also the intertel pbx they have wont interface with the asterisk box either |
| 19:35.21 | Jumpie | jbeez found me across the world on another network |
| 19:35.25 | Qwell | jbeez: why not? |
| 19:35.37 | Jumpie | do they have the right card types? either that or not configed right because tis definately possible |
| 19:35.49 | jbeez | don't exactly know, I'm new here, we dont even have access to the asterisk server from what I understand |
| 19:37.40 | Jumpie | hehe |
| 19:39.00 | SwK | depending on how big of an install $20K might not be bad |
| 19:39.10 | SwK | its easy to do a 20K install heh |
| 19:41.00 | *** part/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 19:42.10 | Shotygun | <-- SIP read from 10.200.5.156:2054: |
| 19:42.10 | Shotygun | SIP/2.0 480 Do Not Disturb |
| 19:42.26 | Shotygun | Is there a way I can handle this event without patching up the source? |
| 19:46.19 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 19:48.56 | Jumpie | qwell |
| 19:49.04 | Jumpie | so what do you feel about asterisk in an enterprise environment? |
| 19:49.12 | Qwell | what about it? |
| 19:49.13 | Jumpie | i have a guy in another channel saying asterisk sucks with media bridging, and you need sipx all the way |
| 19:49.19 | Jumpie | and nobody would use it enterprise |
| 19:49.21 | Jumpie | thats gotta be a joke |
| 19:49.44 | Jumpie | i think its a matteer of hardware choice and config and it should be fine, i know peopel use native asterisk in large call environments fine |
| 19:49.53 | Qwell | correct |
| 19:50.57 | mvanbaak | yup |
| 19:51.26 | mvanbaak | and if you go to #freeswitch they will tell you sipx sux and you need freeswitch all the way |
| 19:51.46 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 19:52.00 | mvanbaak | and if you go to the secret paid #cisco channel they will tell you all them opensource thingies are playgrounds and you need cisco all the way |
| 19:52.19 | Jumpie | lol |
| 19:52.24 | Jumpie | yep |
| 19:52.26 | magic_hat | hey everyone. several months ago I turned off e-mail notification for voicemails. Now I want to turn it back on, and can't seem to figure out how to do that... help? |
| 19:52.28 | Jumpie | well i hang in #ciscohelp on efnet |
| 19:52.31 | Jumpie | and he's cheerleading cisco solution |
| 19:52.43 | Jumpie | ims orry but most of my clientel is 50 or less employees |
| 19:52.48 | Jumpie | that dont wanna spend 100k on a voip solution |
| 19:53.41 | magic_hat | i have attach set to 'yes' in voicemail.conf... not sure where else to look. |
| 19:53.45 | Jumpie | im also curious, can you forego a fxo / t1 card, and hook right up to a cisco IAD 2431 if you can get them cheap?> |
| 19:53.58 | Jumpie | or would it not be cost effective |
| 19:57.47 | [TK]D-Fender | Jumpie: What would this IAD be doing for you? |
| 19:58.02 | Jumpie | replaces need for card in the server |
| 19:58.08 | Jumpie | acts as pri/t1 gateway for channelized voice |
| 19:58.12 | Jumpie | i mean im TOLD this may work |
| 19:58.18 | Jumpie | just wondering if it was cost effective |
| 19:58.26 | Jumpie | t1 - iad - asterisk |
| 19:58.37 | Jumpie | it has wic-1t and pots ports on it |
| 19:58.57 | [TK]D-Fender | Jumpie: Cost $? |
| 20:00.00 | Jumpie | well you can get them on ebay for about a grand |
| 20:00.27 | Jumpie | 16 fxs and 1 t1/e1 |
| 20:00.44 | Qwell | you can get a T1 card cheaper than that |
| 20:00.45 | Jumpie | someone showed me an article on tindells.com on how to config it..just wondering if anyone tried it |
| 20:00.55 | Jumpie | qwell ya, but this serves as t1 and fxs/fxo provider |
| 20:01.05 | Jumpie | im not tryin to downplay asterisk gear, im just curious |
| 20:01.20 | Qwell | you'd be adding yet another layer into it |
| 20:01.21 | Jumpie | ill end up spending about $1200 on the t1/fxo/fxs cards w/ echo cancelation if not a bit more right? |
| 20:01.22 | Jumpie | good ones |
| 20:01.27 | Jumpie | ah..true..another poitn of failure |
| 20:02.16 | Jumpie | also, talked to an ebay guy, he works for PHONICEQ he says he used to work for digium and their cards are just as good...any news on those? |
| 20:02.35 | Qwell | ~cheap |
| 20:02.35 | jbot | well, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 20:03.52 | denon | Jumpie: good word of advice .. if you're planning to run a mission critical phone system, provided by a vendor you have to ask "anyone heard of those?" .. |
| 20:04.00 | denon | well, I'd think the advice would be obvious |
| 20:04.03 | Jumpie | denon, true |
| 20:04.18 | Jumpie | but if he's a former employee of digium and says they are just as good...can you discount the possibility that that may be true? |
| 20:04.24 | Jumpie | im not saying it is...but im not saying it isnt either |
| 20:04.30 | Qwell | of course you can discount it |
| 20:04.37 | denon | either way, whether or not it works really has no bearing when a drive changes, his card no longer works, and he's long gone |
| 20:04.49 | denon | s/drive/driver/ |
| 20:04.50 | Jumpie | possibly |
| 20:04.56 | Qwell | if somebody worked for Cisco, and said "My stuff is just as good"... |
| 20:05.03 | Jumpie | but what if it was? |
| 20:05.17 | Jumpie | no way to know without testing? i mean somebody new has to get their start somehow? |
| 20:05.22 | Jumpie | lol again im just playin devils advocate |
| 20:05.40 | Jumpie | what im saying qwell is you cant discount the fact that a clone may ac tually be high quality |
| 20:05.41 | denon | Jumpie: ask him to quantify HOW it's "just as good". Ask him specifics about the quality of mfg, the QA work, the failure rates, etc |
| 20:05.47 | denon | don't just ask "if it's good" |
| 20:05.50 | Jumpie | sure..understood |
| 20:06.02 | *** join/#asterisk luke-jr (n=luke-jr@wsip-70-167-147-10.om.om.cox.net) |
| 20:06.03 | Jumpie | he says they sell to businesses all over the world and have a proven track record |
| 20:06.04 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:06.08 | Jumpie | so ill make him put his money where his mouth is |
| 20:06.11 | Qwell | anybody can "say" that. |
| 20:06.16 | luke-jr | I've been having one-way audio issues on incoming calls for a few days now |
| 20:06.19 | denon | seriously, I don't care what you buy .. I'm just saying these are the kinds of questions that digium can answer, but I'd guess your fleabay buddy will say "oh yeah, they're just as good"" |
| 20:06.32 | luke-jr | it's pretty consistant with Google GrandCentral via Gizmo |
| 20:06.34 | denon | make him put SPECS where his mouth is |
| 20:06.45 | luke-jr | and also seems to be a problem via my work extension when using IAX2 |
| 20:07.40 | luke-jr | anyone else seeing this problem? |
| 20:07.57 | Jumpie | denon, oh..no i i would never taked him on his word |
| 20:08.02 | Jumpie | thats why im gonna get some specs |
| 20:08.05 | Jumpie | : |
| 20:08.59 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 20:09.09 | Jumpie | i wonder if anyon sels a t1/fxo card bundle |
| 20:09.42 | denon | Jumpie: just make sure he's going to be around to properly suppor 1.6 and beyond |
| 20:09.47 | denon | penny wise, pound foolish |
| 20:10.27 | *** join/#asterisk [T]ank (n=[T]ank@c-71-195-194-193.hsd1.ut.comcast.net) |
| 20:11.19 | [T]ank | i also have a digium te405p t1 card i would like to unload. anyone interested? |
| 20:11.28 | luke-jr | [T]ank: free? |
| 20:11.35 | [T]ank | lol |
| 20:11.38 | luke-jr | ☺ |
| 20:11.45 | Jumpie | quanto questa |
| 20:11.49 | Jumpie | sp |
| 20:12.02 | Jumpie | te405 = quad t1? |
| 20:12.24 | [T]ank | yeah |
| 20:12.35 | *** join/#asterisk boblutz (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 20:12.40 | *** part/#asterisk boblutz (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 20:13.19 | [T]ank | Jumpie: interested? |
| 20:18.58 | Siya | reads up |
| 20:19.06 | Siya | ooh * 1.6 |
| 20:19.08 | [TK]D-Fender | [T]ank: I got 5$ for it right here... |
| 20:19.12 | Siya | Any experiences? |
| 20:23.00 | *** part/#asterisk [T]ank (n=[T]ank@c-71-195-194-193.hsd1.ut.comcast.net) |
| 20:23.10 | lirakis_work | does "incominglimit" in sip.conf .. allow a peer to make any combination of inbound or outbound calls up to the specified limit ? |
| 20:23.56 | lirakis_work | .. b/c thebook's wording makes it sound like .. it would allow a peer to make max N inbound calls and max N outbound calls.. effectivley setting a channel limit of 2N |
| 20:24.45 | jblack | hmm. the book only defines the value for user and peer. Doesn't say anything about a friend. |
| 20:24.59 | jblack | Looks like whoever wrote that paragraph didn't know either. =) |
| 20:25.22 | lirakis_work | jblack: friend = user+peer |
| 20:29.51 | magic_hat | anyone know what I'm missing to turn on e-mail voicemail notifications? I turned it off a while back and now can't get it going again. |
| 20:32.33 | jsmith | magic_hat: Give us a little more information on your setup. SIP phones? Analog phones? |
| 20:32.47 | *** part/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 20:32.49 | jsmith | magic_hat: Or just having the voicemail *sent* to email? |
| 20:32.55 | jsmith | was thinking MWI until he re-read the question |
| 20:33.18 | magic_hat | yeah, I just want to get an e-mail when someone leaves a VM. I'm using X-lites. |
| 20:34.11 | jsmith | magic_hat: Did you take the email addresses out of voicemail.conf, or maybe set the emailcommand to something different? |
| 20:34.34 | magic_hat | here's a sample voicemail.conf setting: 10 => 6000,Joe User,joe@user.com,,attach=yes |
| 20:35.53 | magic_hat | I haven't done anything with mailcmd |
| 20:36.05 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
| 20:40.52 | jsmith | magic_hat: That looks correct |
| 20:41.09 | magic_hat | hrm... so it's not sending, and I'm not seeing anything in my logs. |
| 20:41.15 | jsmith | magic_hat: I'm not sure why else they wouldn't get mailed out, unless you messed with the mail system under Linux itself |
| 20:41.44 | magic_hat | nah, this is a brand-new server. haven't had time to screw it up yet. |
| 20:41.59 | magic_hat | maybe the e-mail service doesn't run automatically? I'm on Ubuntu. |
| 20:43.10 | luke-jr | magic_hat: Ubuntu Server? |
| 20:44.11 | magic_hat | luke-jr: yup. |
| 20:44.36 | jsmith | magic_hat: Probably not |
| 20:46.52 | tzafrir_home | magic_hat, it really doesn't take *that* long to mess up a server... |
| 20:49.46 | robeph | weird... spent last 3 hours trying to figure out wth was wrong with these queues... |
| 20:50.21 | robeph | I mean the agi we have put agents in the queue, it says the module is sticking them in there, all normal, until you show queue members and there are none... apparently something funky happened when the modules were compiled |
| 20:50.26 | robeph | and that one was borked |
| 20:50.35 | robeph | recompile all works fine, anyone experienced similar in 1.2? |
| 20:50.58 | magic_hat | tzafrir_home: yeah, no lie. What I meant was: I haven't done anything to the server to f*** it up. |
| 20:51.54 | robeph | tzafrir_home: akin to my case here :p... brandnew setup all worked fine, then queues just flame out and stop working, no reason, recompule the module and magically it works, yet I can find zero cause ;p it's bit flipping gnomes... |
| 20:51.58 | robeph | they do crap like that all the time |
| 20:52.35 | *** join/#asterisk Great_Randew (n=Andrew@stjhnbsu84w-156034170250.nb.aliant.net) |
| 20:56.45 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 20:57.13 | Yourname`` | Sometimes the 1/4th of a second of the first words are chopped off when I call Voicemail() .. why so? |
| 20:57.59 | *** join/#asterisk ManxPower (n=manxpowe@135.sub-75-203-95.myvzw.com) |
| 20:58.53 | robeph | Yourname``: the voicemail msg? |
| 20:58.59 | *** part/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 20:59.03 | Yourname`` | Yeah |
| 20:59.05 | robeph | ie, "|||-avid is not here to take your call" |
| 20:59.09 | robeph | instead of david |
| 20:59.19 | robeph | I notice that with a ton of my ivr/recordings |
| 20:59.45 | ManxPower | 2nd major lightening strike of the storm and the damn power goes out. At least I'll know how long this nifty new extended life battery will last |
| 20:59.55 | Yourname`` | robeph: You got it! |
| 21:00.02 | robeph | ManxPower: wher you at? |
| 21:00.06 | ManxPower | robeph: is that garbled or silent? |
| 21:00.14 | ManxPower | robeph: near Birmingham, AL |
| 21:00.15 | robeph | Yourname``: what |||? silence |
| 21:00.19 | robeph | ManxPower: ah huntsville here |
| 21:00.23 | Yourname`` | robeph: That's what happens to me too! Not just Voicemail. Added a Wait(1), still no worky. |
| 21:00.35 | robeph | like we have the agent login thing, and it's like "gent has logged in" |
| 21:00.37 | ManxPower | robeph: canreinvite=no should fix that. |
| 21:00.59 | robeph | oh whats the mechanism causing it? |
| 21:01.19 | robeph | Yourname``: there ya are according to ManxPower... |
| 21:01.56 | Yourname`` | Let's see. |
| 21:02.23 | robeph | unless he meant canreinvite=no will fix my living in huntsville problem >< |
| 21:02.37 | Yourname`` | Well, canreinvite=no on the peer or where? |
| 21:02.41 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 21:03.07 | ManxPower | Yourname``: for all the peers, I don't think it's supported in [general] but you could look it up. |
| 21:03.15 | Yourname`` | ok |
| 21:03.20 | robeph | ManxPower: whats the mechanism behind that? |
| 21:03.33 | robeph | ie. what causes it. |
| 21:03.52 | robeph | the chop off of preceding audio when that is (default? ) set to yes |
| 21:05.25 | ManxPower | There are three possible things that commonly cause lost audio at the beginning of a call 1) phones are doing a reinvite and it is taking long enough for someone to notice 2) some telcos can't pass audio as soon at the line is answered, but need a short wait. .5 second usually seems to work. 3) a horridly large value for echotraining= |
| 21:05.46 | robeph | ManxPower: ok it's not 2, could be 1, and not 3 |
| 21:06.02 | robeph | since this is a sip phone registered at the pbx the ivr is on |
| 21:06.08 | Yourname`` | Nogo actually..doesn't work. |
| 21:06.12 | robeph | hrm |
| 21:06.31 | *** join/#asterisk DrkShadow (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 21:06.34 | robeph | is there a broad, hey wait setting, like 1/2sec on pickup before playing audio? |
| 21:06.37 | ManxPower | robeph: at the start of the IVR do an Answer and a Wait(.5) |
| 21:06.49 | Yourname`` | I'm also doing a Wait(1) |
| 21:07.05 | ManxPower | Yourname``: the wait does no good without the answer |
| 21:07.56 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 21:08.32 | ManxPower | robeph: for regular phone calls, the amount of time it takes for the callEE to pick up the phone and bring the handset to their ear is usually long enough of a wait |
| 21:08.40 | *** join/#asterisk Tako-san (n=jmkiffia@24.108.192.144) |
| 21:08.45 | robeph | ManxPower: yeh |
| 21:08.57 | ManxPower | so you don't need the answer/wait for non-ivr like things |
| 21:09.25 | robeph | ManxPower: we're just hearing this via a softphone so there is no time for handset to ear loss of audio heh, it just actually answers / plays the sound mid word |
| 21:10.00 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:10.01 | ManxPower | robeph: sounds like 2) to me |
| 21:10.07 | robeph | usually its unnoticeable, since you catch a portion of the A in agents, but it's missing enough its ... kinda annoying |
| 21:10.15 | robeph | yeah does |
| 21:10.16 | Yourname`` | ManxPower: Adding the Answer worked! Thanks. |
| 21:10.36 | robeph | except... there is not "telco" tween me and this |
| 21:10.49 | ManxPower | Yourname``: it is a VERY bad idea to Answer calls unless you need to. In this case you need to for that specific situation |
| 21:11.06 | robeph | its no biggy in this case... |
| 21:11.08 | Yourname`` | ManxPower: Oh? How come.. does Answer mess things up? |
| 21:11.10 | ManxPower | robeph: I could have said "endpoint" instead of telco |
| 21:11.15 | robeph | ah |
| 21:11.16 | robeph | <PROTECTED> |
| 21:11.28 | robeph | <--- trouble literalizing things said |
| 21:11.29 | ManxPower | Yourname``: it screws up billing and CDRs and call times, etc |
| 21:11.51 | Yourname`` | ah |
| 21:11.58 | Yourname`` | Thankfully not using all of the above right now :D |
| 21:11.59 | robeph | ManxPower: hahaha yeah, we had a customer using an LCR that was dumping like 1000's of minutes on them that weren't even inuse |
| 21:12.15 | C4away | anyone know why I Read(var,playfile,11) then I check to see if the var is local and starts with a 1 (which would fail) so I strip the one with Set(var = ${var:1:10}) and var still has the same value? |
| 21:12.43 | robeph | cos they were answering calls, even SIT calls and keeping hungup calls in answer. |
| 21:12.46 | robeph | really annoying |
| 21:13.08 | ManxPower | C4away: You actually red docs! Yay! *grin* Try Set(var=${var:1}) |
| 21:13.11 | robeph | especially annoying when money making customer saw his money making was being hindered by bad billing practices... |
| 21:13.21 | C4away | even though the output shows Set("SIP/test00-c4035390", "FWDTO = XXXXXXXXXX") without the 1 |
| 21:13.36 | C4away | I'll try that |
| 21:14.09 | C4away | and yes, I read a lot of docs |
| 21:14.16 | C4away | seems that's about 80% of what I do |
| 21:14.57 | ManxPower | C4away: Look straight up. That's the learning curve for VoIP. |
| 21:15.29 | C4away | heh, yea, I've been doing this 2 years now and re-writing a dialplan from 1.2 -> 1.4 is still a pain |
| 21:15.31 | *** join/#asterisk korihor (n=humberto@190.78.209.202) |
| 21:15.35 | C4away | especially since I didn't write the 1.2 dialplan |
| 21:16.22 | C4away | and the dialplan had errors all through it, I'm surprised their customers weren't screaming that *69, call forwarding, blacklist, etc weren't working |
| 21:16.23 | mvanbaak | C4away: that's easy |
| 21:16.30 | mvanbaak | C4away: I had to do 1.0 = |
| 21:16.45 | mvanbaak | C4away: I had to do 1.0 => 1.6 several times this month |
| 21:16.48 | UnixDog | lol |
| 21:17.07 | UnixDog | 1.6 is not ready for production |
| 21:17.17 | UnixDog | your useing it in production |
| 21:17.19 | mvanbaak | gheh |
| 21:17.29 | mvanbaak | I run trunk in production in several places |
| 21:17.45 | C4away | well, I'm updating it to work on 1.4 AND writing in little things like "to call this number press 1" on *69 and reading back the entries people dial and verifying them before putting the new value in the DB |
| 21:18.20 | C4away | call forwarding and the like just put the number as dialed into the db even if DTMF digits were missed or they misdialed the number, never read it back to the caller |
| 21:18.58 | C4away | also the call forwarding would accept, and actually ask for, the 11 digit number even though their outbound proxy will fail on 1+local |
| 21:19.40 | C4away | some people just don't think about usability when writing code |
| 21:19.50 | mvanbaak | I never do |
| 21:19.52 | RoyK | http://apina.biz/6609.jpg |
| 21:19.57 | mvanbaak | I just code it the way I like it |
| 21:20.00 | C4away | lol |
| 21:20.08 | C4away | is your name matt? |
| 21:20.42 | mvanbaak | rofl RoyK |
| 21:20.54 | mvanbaak | C4away: no, it's michiel |
| 21:21.00 | mvanbaak | do a /whois on me |
| 21:21.05 | RoyK | mvanbaak: :) |
| 21:21.12 | C4away | lol, ok then I don't have anything against you |
| 21:21.21 | mvanbaak | lol C4away |
| 21:21.22 | C4away | would like to talk to the original author of this dialplan though |
| 21:22.09 | robeph | C4away: He's not hard to find |
| 21:22.13 | C4away | ok, so the number isn't changing even though I can see the set command executing in the CLI and it shows it setting the variable to the shorter value |
| 21:22.25 | robeph | C4away: go to walmart, round christmas time, they got this lil red bucket, he's usually near it ringing a bell |
| 21:22.26 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 21:22.39 | C4away | haha |
| 21:23.01 | ManxPower | C4away: as the priority put in a Noop(var is ${var}), watch the cli, is the value correct? If so you are setting it just fine and something else is wrong. |
| 21:24.57 | C4away | Set("SIP/test00-c4035330", "FWDTO = 3035556666") in new stack |
| 21:24.58 | C4away | NoOp("SIP/test00-c4035330", "FWDTO=13035556666") in new stack |
| 21:25.05 | C4away | not setting it even though the set command says it is |
| 21:25.29 | [TK]D-Fender | C4away, do NOT put whitespace in your application & function calls. |
| 21:25.37 | *** join/#asterisk bluregard (n=matt@76.29.119.76) |
| 21:25.51 | ManxPower | Good call, [TK]D-Fender |
| 21:26.20 | C4away | ah, didn't even notice that |
| 21:26.29 | C4away | what about equating variables? I know it used to be needed |
| 21:26.30 | ManxPower | I'll bet you are setting a variable called "FWDTO " |
| 21:26.52 | ManxPower | C4away: you almost never need spaces in the dialplan |
| 21:28.47 | C4away | yep, worked |
| 21:29.12 | [TK]D-Fender | C4away, never with Set |
| 21:31.10 | ManxPower | Joy. We are under TWO tornado warnings, a severe thunderstorm warning, AND a lake wind advisory |
| 21:31.34 | RobH | looks outside at 83 degrees F and sunny. |
| 21:31.35 | [TK]D-Fender | ManxPower, "Would you like fries with that, sir?" |
| 21:31.41 | lmadsen | spaces are only ever required around an operator in an evaluation, e.g. $[${FOO} = 1] |
| 21:31.45 | ManxPower | oh, and the power is out. 8-) |
| 21:31.55 | lmadsen | and you probably don't actually NEED the space, but it's safe to use them for clarity |
| 21:32.24 | ManxPower | lmadsen: before 1.2, the space was required, IIRC |
| 21:32.30 | lmadsen | ya, I think you're right |
| 21:32.32 | mvanbaak | it was |
| 21:32.43 | mvanbaak | we still have a couple of 1.0 boxen |
| 21:32.48 | mvanbaak | there we _NEED_ the space |
| 21:33.24 | lmadsen | ya, because of the way the parser worked |
| 21:34.35 | [TK]D-Fender | or failed to..... |
| 21:35.21 | [TK]D-Fender | Scary to think I've written a better parser than that and a language to go around it... |
| 21:35.43 | mvanbaak | I never knew your name was guido ;) |
| 21:36.08 | [TK]D-Fender | ? |
| 21:36.18 | mvanbaak | guido van rossum |
| 21:36.25 | mvanbaak | the dude who created python |
| 21:36.39 | mvanbaak | my 1.0 dialplans are 1 line |
| 21:36.48 | [TK]D-Fender | mvanbaak, Nothing to do with me, and I don't even know python. |
| 21:36.52 | C4away | shouldn't your nick be Manx[sans]Power |
| 21:37.01 | mvanbaak | exten => _X.,1,Agi(voipserver.py) |
| 21:37.16 | C4away | or just Manx maybe? |
| 21:37.24 | C4away | and how are you online without power? |
| 21:37.46 | ManxPower | There are these incredibly cool things called "batteries", C4away |
| 21:38.06 | C4away | yea, I have them on all my computer equipment but my 3 21" monitors don't do well on the UPS |
| 21:38.22 | ManxPower | I live on a mountian, almost everything is on a battery backup, and everything else is on surge protectors. |
| 21:38.31 | ManxPower | I'm on my laptop |
| 21:38.39 | C4away | I should put a diesel generator in my basement |
| 21:39.15 | ManxPower | I live on a mountian |
| 21:39.34 | C4away | I was actually thinking a) laptop b) at work in datacenter w/generator or c) an LCD screen on UPS |
| 21:39.54 | mvanbaak | my ups runs linux with asterisk |
| 21:39.56 | ManxPower | Power is pretty reliable since the power company replaced most of the power poles on the mountian |
| 21:40.27 | C4away | we were having odd fluctuations in our power the last few days |
| 21:40.29 | *** join/#asterisk NirS (n=NirS@87.68.3.201.cable.012.net.il) |
| 21:40.47 | C4away | burnt out 4 lightbulbs in the house, glad I had UPS and surge protectors on all my valuable equipment |
| 21:40.48 | mvanbaak | all my computers are on regulated powerfeeds |
| 21:41.10 | mvanbaak | so even if the powercompany is feeding me fluctuating power, my gear wont notice it |
| 21:41.33 | mvanbaak | actually, everything in my house runs on the regulated power |
| 21:41.59 | mvanbaak | cost me some money for the initial setup |
| 21:42.07 | mvanbaak | but it saves everything in the house |
| 21:42.30 | C4away | well, I have big upses on all my stuff that claim to be Auto Voltage Regulating |
| 21:42.37 | C4away | seem to work |
| 21:42.42 | ManxPower | I was in the process of putting the lamps and fans on surge protectors when the power went out. |
| 21:43.33 | ManxPower | The UPS on my tivo is freaking now. |
| 21:43.43 | C4away | I have a few old UPSes that only put out 70-90 VAC .. I should put them on lamps and such since an incandescent will run on 80v |
| 21:44.17 | C4away | they put out 120 when the power is live, just that the voltage regulators are about shot |
| 21:45.02 | *** join/#asterisk asdx (n=diego@adsl-159-164.click.com.py) |
| 21:45.04 | asdx | hi |
| 21:45.36 | C4away | one has an alarm loud enough to wake the neighbors, when I opened it up to test it I put a big wad of tape over the piezo buzzer |
| 21:46.18 | asdx | does anyone knows where can i get a list of phone numbers, from all the countries... i need to know how much digits phone numbers have in different countries |
| 21:47.08 | C4away | it is not standardized |
| 21:47.34 | asdx | how do i do this then |
| 21:47.35 | C4away | I just take the international prefix (here it is 011) and make a rule like 011. |
| 21:47.37 | asdx | i mean |
| 21:47.46 | asdx | my boss says he wants to call every phone in the world |
| 21:47.53 | asdx | how do i configure my extensions.conf for that |
| 21:47.59 | asdx | do i have to put every possibly extension for that? |
| 21:48.07 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 21:48.11 | asdx | with the exact X digits? |
| 21:48.11 | C4away | no, just a match like _011. if you are in the USA |
| 21:48.17 | C4away | pattern matching |
| 21:48.21 | asdx | ok |
| 21:48.23 | asdx | cool |
| 21:48.24 | asdx | thanks |
| 21:48.49 | C4away | <PROTECTED> |
| 21:48.59 | asdx | yeah i know that, thx :) |
| 21:49.03 | C4away | ok |
| 21:49.08 | asdx | so |
| 21:49.17 | asdx | let me try |
| 21:49.18 | [TK]D-Fender | asdx, exten => _9.,1,Dial(Zap/g1/${EXTEN:1}) <- there, now he can dial any #. Now make sure your telco thinks its valid. |
| 21:49.42 | C4away | or _X.,1,...etc... |
| 21:49.55 | asdx | [TK]D-Fender: i see, thanks |
| 21:49.57 | C4away | I have abandoned the 9 for outside line, as I don't use KSUs |
| 21:50.15 | C4away | if DBput is deprecated then 9 for an outside line should be for sure |
| 21:50.34 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 21:51.04 | [TK]D-Fender | C4away, I did that so that his boss could call ANY number on the PSTN without the pattern clashing for something "internal" |
| 21:51.42 | [TK]D-Fender | C4away, I accounted for his psycho-worded "EVERY POSSIBLE NUMBER" idea, and thus a prefix was required unless his boss can ONLY call out to the PSTN with that rule and do nothing else. |
| 21:52.06 | [TK]D-Fender | C4away, You need to think a little deeper when looking at my answers. This one was hidden in the "fine print" |
| 21:52.11 | ManxPower | C4away: no, using a selection code simply good dialplan design |
| 21:52.39 | [TK]D-Fender | ManxPower, In my case it was just a little "generosity" :) |
| 21:53.48 | C4away | selection codes are built into the NPNPA number scheme |
| 21:54.19 | C4away | ever wonder why you can't have a 1s in the phone number in certian places NXXNXXXXXX |
| 21:54.42 | C4away | to prevent matching a local number as a long-distance when using 7 and 10 digit dialing |
| 21:55.05 | ManxPower | C4away: no, 1 is a toll code. |
| 21:55.08 | C4away | take that into consideration and don't create internal extensions that would match with external numbering schemes |
| 21:55.09 | ManxPower | or used to be at least. |
| 21:55.41 | C4away | yes, but if I had a local number of 1234444 then that would match the long-distance pattern in the switch |
| 21:55.50 | ManxPower | 9 is used when you can't determine of 3487 is a 4-digit extension of the first 4 digits of a non-LD 7-digit phone number |
| 21:55.51 | C4away | and it would wait for the rest of the number |
| 21:56.26 | ManxPower | and people don't like having to wait for timeouts. |
| 21:56.44 | ManxPower | If timeouts are OK with you then you almost never need an outside line code |
| 21:56.58 | C4away | exactly, so the North American Numbering Plan Administration determined that 1s couldn't be used as the first digit in a local 7 or 10 digit number |
| 21:57.09 | ManxPower | You also don' need that if you make all calls dialed with 1+ac+number |
| 21:57.21 | C4away | but if you do, you dont' have to wait |
| 21:58.19 | *** part/#asterisk RoyK (n=roy@ip-210-6-149-91.dialup.ice.no) |
| 21:58.34 | C4away | the only time I use prefixes is to force a call out a specific trunk where two or more would be valid for the call |
| 21:58.51 | *** join/#asterisk gego (n=gego@host-091-097-123-018.ewe-ip-backbone.de) |
| 21:58.55 | *** join/#asterisk flynux (n=flynux@2a01:38:0:0:0:0:0:1) |
| 21:59.11 | C4away | it is easier to train someone that # dials the number than to require a prefix such as 9 |
| 21:59.25 | ManxPower | I suspect I'll pry 7-digit dialing from my users' cold, dead fingers. |
| 21:59.26 | C4away | because without the # the number will dial, it is usable for everyone who uses the system |
| 22:00.12 | ManxPower | I feel that people that use phones should be able to remember 9. If they can't then they should not be using a phone. |
| 22:00.41 | C4away | well i don't expect people to be that bright |
| 22:00.59 | C4away | I design my dialplans for usablity and assumed ignorance |
| 22:01.50 | C4away | if someone uses the phone a lot to make calls they can hit # or SEND on their phone, if they don't why force them to learn that 9 means dial outside? |
| 22:02.52 | C4away | and 9+7-digits is 8 digit dialing |
| 22:03.17 | C4away | so I'll have to pry 8-digit dialing from their cold dead hands? |
| 22:05.06 | [TK]D-Fender | I always do 7-10-11 digit transparent dialing, 011. , [3469]11 |
| 22:05.18 | C4away | difference in approach, both are valid ... I just strive for standardization and I don't have to dial 9 from home, or payphones, or cell phones, or anything else... and there is no need to physically switch a relay in the KSU to give me a trunk |
| 22:05.36 | C4away | so I say deprecate it |
| 22:05.48 | C4away | send it the way of the phones without # and *, to the museum |
| 22:05.51 | ManxPower | because people know 9 for outside line -- same as other systems |
| 22:06.52 | C4away | you can strip a 9 if it is there as the number goes out the trunk, but there is no need to require it |
| 22:06.56 | C4away | that's all I'm saying |
| 22:07.45 | ManxPower | It's your dialplan |
| 22:07.54 | C4away | and _9. will still wait for the TIMEOUT(digit) |
| 22:08.07 | ManxPower | I would NEVER EVER use _9. |
| 22:08.33 | C4away | it is what you used as an example, and what started this conversation |
| 22:08.43 | [TK]D-Fender | ManxPower, Don't want them dialing the operator I take it.. |
| 22:08.44 | ManxPower | The only time I ever use a . in a pattern is for 011 |
| 22:09.15 | ManxPower | C4away: and on SIP phones the dialplan is in the phone anyway |
| 22:09.29 | ManxPower | [TK]D-Fender: I can't imagine why they would need to call the operator |
| 22:09.43 | C4away | 0 is the front desk in most pbxes I set up |
| 22:09.44 | [TK]D-Fender | ManxPower, load chan_imagination.so |
| 22:10.13 | ManxPower | There IS one reason I consider valid for not using a leading 9 and that is 9911 |
| 22:10.54 | ManxPower | they dial 91, then like a blind hummingbird forget they dialed a 1 and so they dial it again and then they sheriff shows up |
| 22:10.59 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 22:11.57 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 22:12.26 | robeph | ManxPower: also you could use it inside of a context that may service multiple extensions but is preceded by another context that filters out any unwanted _9's ie _9NXXNXXXXXX->[b] _91NXXNXXXXXX->[b] [b] _9. -> [blah] |
| 22:12.43 | robeph | but using it in a default use context does really seem bad I must agree. |
| 22:14.46 | jeev | [TK]D-Fender, im back :/ |
| 22:15.53 | ManxPower | I wanted to Playback(now-connecting-to-911) followed by Wait(2) |
| 22:16.08 | C4away | design your system for idiots and you will not be dissapointed |
| 22:16.24 | C4away | that or they will surprise you with their even greater than expected levels of idiocy |
| 22:16.45 | ManxPower | as it is, the system e-mails a small alias/mailing list of people with the information about who/when/where someone dialed 911 |
| 22:17.18 | C4away | send it to everyone@company.com |
| 22:17.48 | ManxPower | naw, just the PBX admin, the IT director and the office operator |
| 22:18.10 | C4away | that way if it was a legitimate emergency anyone available can come to help, and if it was a stupid mistake they can be humilitated properly |
| 22:19.06 | ManxPower | The IT director is retireing in a few months, he doesn't want to piss anyone off |
| 22:19.24 | *** join/#asterisk JunK-Y (n=junky@modemcable153.55-201-24.mc.videotron.ca) |
| 22:20.21 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 22:20.21 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:20.47 | C4away | I thought when you were retiring you could have some fun |
| 22:21.00 | C4away | what good is retiring if you have to be extra good in your last few months? |
| 22:21.13 | C4away | might as well be just any other few months in your career |
| 22:22.08 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 22:22.35 | nny_1 | whats the best way (i can read up just need a nudge) on getting Zap 1 to ring to SIP 101 and Zap2 to ring to SIP 102 |
| 22:23.20 | jsmith | nny_1: Point the zap channels at two separate contexts, and in each of those contexts dial the SIP phone, like this: |
| 22:23.24 | jsmith | [line1-context] |
| 22:23.25 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 22:23.32 | jsmith | exten => s,1,Dial(SIP/101) |
| 22:23.35 | jsmith | [line2-context] |
| 22:23.41 | jsmith | exten => s,1,Dial(SIP/102) |
| 22:24.03 | nny_1 | jsmith: ty sounds easy enough thanks |
| 22:24.25 | nny_1 | i do this in zapata.conf right? |
| 22:24.32 | nny_1 | right now i have context=default |
| 22:24.46 | nny_1 | and then all the other joyous zapata.conf stuff |
| 22:31.48 | robeph | yeh well as it stands those zap channels go to [default] |
| 22:33.41 | nny_1 | robeph: so if i wanted zap 1 to go to context default, and zap 2 to go to default 2, would i just make an entire separate entry for it like [channels] context=default echo/signalling etc and then context=default2 echo/signalling etc? |
| 22:34.38 | *** join/#asterisk equanimity (n=alex@cpc2-oxfd7-0-0-cust208.oxfd.cable.ntl.com) |
| 22:34.48 | nny_1 | guess what i am asking is *how* to differentiate between the two in zapata.conf |
| 22:35.28 | robeph | yes |
| 22:36.26 | robeph | make different contexts for each |
| 22:36.36 | robeph | s/context/entry/ |
| 22:36.46 | robeph | hey now thats neat |
| 22:37.36 | nny_1 | http://pastebin.com/m16156f17 |
| 22:37.46 | nny_1 | is what i have now |
| 22:37.48 | nny_1 | ha jbot rocks |
| 22:37.52 | *** join/#asterisk voipman (i=ccrites@minibar.rackmount.org) |
| 22:39.34 | *** join/#asterisk mwalling (i=mwalling@you.dontlike.us) |
| 22:46.50 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 22:48.52 | *** join/#asterisk MDK2MDK (n=NanoTec@41.249.116.87) |
| 22:50.54 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:51.22 | *** join/#asterisk atik7 (n=chatzill@122.53.194.0) |
| 23:02.23 | robeph | man I'd sure enjoy regex [retro]filtering from command line ;p |
| 23:02.51 | robeph | so I could filter out stuff already passed to the cli, and unfilter it so I didn't have to keep doing tests with various debugs on / off |
| 23:03.39 | *** join/#asterisk Defraz (i=t0tal@72.24.26.7) |
| 23:05.39 | riddlebox | how would you guys suggest linking two asterisk systems together over the internet? I created an extension on one system and had the other system use that info as an incoming provider, I could register but couldnt make calls or recieve calls? |
| 23:07.42 | *** join/#asterisk ManxPower (n=manxpowe@208.sub-70-222-74.myvzw.com) |
| 23:08.11 | robeph | riddlebox: use iax or sip trunk? |
| 23:08.29 | riddlebox | I can use either |
| 23:09.17 | robeph | then send calls out over IAX2/${TRUNK}/${EXTEN} |
| 23:09.26 | robeph | where trunk is the iax trunk device |
| 23:09.34 | robeph | (not extension) |
| 23:09.51 | robeph | ir SIP if you so choose.. |
| 23:09.59 | riddlebox | ok so I create an iax trunk to both sites? |
| 23:10.02 | robeph | yep |
| 23:10.46 | robeph | and ya gotta stick a register => in the iax.conf that registers the trunk to the other pbx |
| 23:10.54 | robeph | and viseversa |
| 23:11.18 | ManxPower | robeph: you have trunk=yes in iax.conf? |
| 23:11.26 | robeph | oh yeh forgot that part :p |
| 23:11.35 | ManxPower | it's not a trunk until you put that in |
| 23:11.36 | robeph | I got it working |
| 23:11.38 | ManxPower | ~trunk |
| 23:11.39 | jbot | [trunk] is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 23:11.46 | robeph | I just kinda never think bout that lil part ;p |
| 23:11.57 | robeph | sort of included in "make an iax trunk" |
| 23:13.09 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:13.46 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 23:14.22 | *** join/#asterisk profounded (n=pro@c-68-82-34-163.hsd1.nj.comcast.net) |
| 23:15.28 | profounded | does anyone know of any good sip providers for calling the UK and Canada? |
| 23:16.17 | *** part/#asterisk korihor (n=humberto@190.78.209.202) |
| 23:17.05 | lesouvage | I use ipness and they are pretty good and easy to confige with worldwide numbers available. |
| 23:17.56 | drmessano | ipness? |
| 23:18.04 | profounded | Thank you lesouvage, i'll goto the site to check on the rates! |
| 23:18.15 | drmessano | iPenis? |
| 23:18.43 | profounded | lol |
| 23:18.47 | profounded | tks guys! |
| 23:18.54 | JayTee52 | is that anything like vPenis? |
| 23:19.19 | drmessano | WTF.. ipness is pronounced, "I, penis" |
| 23:19.21 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 23:19.25 | drmessano | That's.. bad marketing |
| 23:19.31 | jbeez | lol |
| 23:20.31 | lesouvage | drmessano: I never read it like you, but now it s in my brains. easy to remember btw. |
| 23:20.33 | JayTee52 | I have a script plugin that outputs all kinds of system information like cpu type, memory, hard disk space and utilization and it also shows bogomips but I've seen similar scripts where instead of bogomips it displays as "vPenis" in inches or centimeters. |
| 23:20.53 | drmessano | lol |
| 23:21.27 | drmessano | Im assuming it's pronounced like "hipness"? |
| 23:22.01 | drmessano | I guess they never expect anyone to spell the IP part |
| 23:22.09 | drmessano | Bad, bad oversight |
| 23:22.59 | *** join/#asterisk JesseT77 (n=Jesse@67-210-205-7.generic.webformix.com) |
| 23:23.08 | JesseT77 | Haldo |
| 23:23.13 | drmessano | Lando |
| 23:23.24 | drmessano | R2? |
| 23:23.34 | JesseT77 | Is there an AEL command to play hold music by any chance? |
| 23:23.50 | JesseT77 | I am mocking up a test extension for the sole purpose of testing the hold queue. |
| 23:24.02 | JesseT77 | making sure the music plays properly, etc. |
| 23:24.29 | lesouvage | pronounce is like I (high) P(lower your voice and keep it short Ness (go back to the orinal tone and make it a little bit longer then the second part |
| 23:25.25 | drmessano | Like HIP-ness |
| 23:25.28 | drmessano | IP-ness |
| 23:25.35 | robeph | hip? |
| 23:25.50 | JesseT77 | ayepness |
| 23:25.54 | drmessano | As in "I am a HIP and trendy guy |
| 23:25.57 | drmessano | As in "I am a HIP and trendy guy" |
| 23:25.58 | robeph | I, Penis |
| 23:26.00 | robeph | I prefer this one |
| 23:26.04 | drmessano | Yes |
| 23:26.11 | drmessano | "I, Penis" works |
| 23:26.12 | robeph | it's like I, Robot, |
| 23:26.15 | robeph | but closer to the truth |
| 23:26.16 | drmessano | Yeah |
| 23:26.47 | drmessano | I am going to email them |
| 23:26.51 | drmessano | Perhaps they don't realize |
| 23:27.03 | JesseT77 | Is there a doc online that explains certain ael commands like "Hangup" "Noop" "Background"? I'm thinking there is a command like that for hold music, but it's not "Hold".. |
| 23:27.21 | JesseT77 | expertsexchange.com |
| 23:27.36 | drmessano | AEL is largely undocumented |
| 23:28.43 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 23:28.54 | JesseT77 | Hmm. Since voip-info.org is a popular wiki on the topic, whence I can't find the procedural docs I seek, shall I make up some docs with no research, and wait for someone else to "fix" them? :D |
| 23:29.21 | drmessano | voip-info.org is largely outdated |
| 23:29.56 | JesseT77 | is it deprecated by any newer resource? (I mean, how/why would a wiki go outdated?) |
| 23:30.56 | JayTee52 | largely outdated is being polite. anything that usually refers to * version 1.2 as beta is pretty much a web museum. |
| 23:30.58 | drmessano | Lack of ownership, as would any wiki |
| 23:31.16 | drmessano | A wiki is only as good as the people updating it |
| 23:31.42 | drmessano | Sadly, everyone wants to have a definitive source on asterisk |
| 23:31.49 | drmessano | So Info becomes scattered, at best |
| 23:32.08 | drmessano | Why update a wiki when you can make your own site and extend your ePenis |
| 23:32.16 | JesseT77 | Is * itslef no longer maintained then? |
| 23:32.19 | JayTee52 | JesseT77, have you downloaded the PDF of Asterisk The Future of Telephony? |
| 23:32.29 | JesseT77 | fires up teh google |
| 23:32.29 | ManxPower | Generally people should consider the docs subdir of the Asterisk source as the official docs. voip-info.org is useful for more general stuff |
| 23:32.30 | drmessano | JesseT77: Are you serious? |
| 23:33.17 | JayTee52 | it has a section that covers the commands. |
| 23:33.17 | drmessano | A wiki is outdated, so the project is not updated? |
| 23:33.17 | drmessano | Bad math |
| 23:33.17 | JesseT77 | JayTee: awesome, I shall check that next then. (google doesn't display any search results from a mirror on that folder or anything) |
| 23:33.17 | ManxPower | JesseT77: "the wiki" is not the official docs, it is a 3rd party docs site |
| 23:34.04 | JesseT77 | I didn't come to that conclusion just because of this wiki, but because of drmessano's suggestion that no definitive source of documentation exists.. which would indicate development trouble of some sort. |
| 23:34.20 | drmessano | heh |
| 23:34.29 | JesseT77 | And then Jay mentioned the docs from source. |
| 23:34.33 | drmessano | Yeah.. because SOOO many projects are so well documented |
| 23:34.42 | ManxPower | I just told you the official docs location |
| 23:35.19 | drmessano | The truth is that when a project shifts from a large base of informed users to a large base of end-users, the documentation becomes used and not contributed |
| 23:35.57 | drmessano | and with the proliferation of ISO based Asterisk installs, the sources of sharing info have become more specific to those builds than general asterisk |
| 23:36.08 | drmessano | Although THE BOOK makes a lot of that moot |
| 23:37.11 | JayTee52 | JesseT77, the first weblink in the following is the link to download the book in PDF format |
| 23:37.13 | JayTee52 | ~book |
| 23:37.14 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:38.21 | JesseT77 | JayTee: Awexome thanks, lestwise I are also browing to my version of docs in the tags folder of svn. :D |
| 23:39.07 | JesseT77 | http://tfot.leifmadsen.com/ seems to be down though |
| 23:40.37 | JayTee52 | JesseT77, the PDF is a downloadable file. It'll open in the browser as a PDF and you can save it if you have the Adobe plugin. |
| 23:40.56 | JesseT77 | yes I'm doing that next; I was just reporting the dead link for your benefits. :] |
| 23:42.36 | JayTee52 | This is for everyone here: My short book report of Asterisk Hacking from Syngress Authors: Ben Jackson aka Black Ratchet and Champ Clark aka Da Beave. THIS BOOK IS A WASTE OF MONEY AND TIME. |
| 23:43.25 | JesseT77 | noted |
| 23:43.27 | drmessano | Any book with "Hacking" in it is usually effin lame as hell |
| 23:44.13 | drmessano | That, and any book written by Kevin Mitnick |
| 23:44.17 | drmessano | Mitnick is a moron |
| 23:45.25 | JesseT77 | Bruce Eckell outmorons Kevin Mitnick |
| 23:46.43 | drmessano | I dunno about that |
| 23:46.56 | JesseT77 | "Thinking in C++" |
| 23:47.37 | JesseT77 | Ignoring how terrible C++ is to begin with, that book is the second worst written collection of words I have ever encountered. |
| 23:48.06 | drmessano | Scott Fulton has him beat |
| 23:48.11 | drmessano | Go read his spew on betanews |
| 23:49.10 | JesseT77 | I don't seek out examples for this category however, I just record when I smack into them ;) |
| 23:49.43 | drmessano | lol |
| 23:50.20 | *** join/#asterisk Defraz (i=t0tal@72.24.26.7) |
| 23:50.48 | JesseT77 | my #1 worst hunk of literature of course is JRR Tolkein's novella "Smith of Wooten Major" |
| 23:51.09 | ManxPower | I thought it was "anything by JRR Tolkein" |
| 23:51.38 | JesseT77 | I hold all of his work that I have ever read in high esteem save that black sheep. |
| 23:52.14 | drmessano | ManxPower: You didn't sob whey announced there would be no "Hobbitt" movie? |
| 23:52.23 | drmessano | when* |
| 23:54.53 | JesseT77 | There's not going to be a hobbit movie?! *waterworx* |
| 23:56.05 | drmessano | lol |
| 23:56.16 | MDK2MDK | hello, when installing Asterisk i configured d Asterisk script baut ther some modules that have XXX in the front in state of [*] |
| 23:56.20 | MDK2MDK | what that mean ? |
| 23:56.27 | MDK2MDK | can some one help me plz |
| 23:56.28 | MDK2MDK | ? |
