00:00.10 | drmessano | First off |
00:00.10 | paulius | I'm not using a multimeter. |
00:00.10 | drmessano | Theres this thing |
00:00.11 | [TK]D-Fender | paulius: Echo is more than impedence. |
00:00.14 | drmessano | Called an impedence bridge |
00:00.20 | paulius | [TK]D-Fender: Would you care explaining? |
00:01.08 | [TK]D-Fender | paulius: Not particularly. This answer is long and should be obtained the same way as your previous question ; JFGI |
00:01.18 | paulius | Right. I'm on it. |
00:01.20 | paulius | I'll let you know. |
00:01.29 | paulius | Btw, [TK]D-Fender, this Cisco phone works amazingly. |
00:01.34 | paulius | And the voice quality is amazing. |
00:02.37 | [TK]D-Fender | paulius: Still won't beat a Polycom, even one cost half as much |
00:02.38 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
00:02.52 | paulius | ... |
00:04.06 | [TK]D-Fender | paulius: 7971 doesn't even support G722 |
00:04.20 | Gremlin | is writing an application to read PCM 16-bit linear resolution sound from Madonna CDs and play it as hold music. |
00:04.24 | [TK]D-Fender | Or so many other common codecs... |
00:04.32 | paulius | hmm |
00:04.33 | Qwell | Ciscos use Polycom hardware anyways :p |
00:04.33 | [TK]D-Fender | Silly Crisco... |
00:04.37 | Gremlin | GSM is a great codec |
00:04.37 | paulius | well but it's using something which works |
00:05.11 | [TK]D-Fender | Gremlin: Writing an app? * jsut DOES this already :p |
00:05.35 | Gremlin | From CDs? Or should I rip my Madonna CDs to OGG? |
00:05.47 | [TK]D-Fender | Gremlin: To any format * supports |
00:06.08 | Gremlin | Will Asterisk support a SIP trunk with 16 lines over a cable Internet connection? |
00:06.15 | [TK]D-Fender | Gremlin: I'd suggest doing it to the same format as used by the channels that are on hold so as not to transcode unnecessarily |
00:06.28 | [TK]D-Fender | Gremlin: Depends on your cable internet conenction now doesn't it? |
00:06.33 | Gremlin | Right, saving CPU cycles is good. |
00:06.56 | Gremlin | [TK]D-Fender: I know it does, but Time Warner won't tell me that. |
00:07.37 | Gremlin | ISPs have moved away from stating specific numbers--marketing like the vagueness of adjectives like "blazing fast". |
00:07.38 | [TK]D-Fender | Gremlin: You gave me 1 set of numbers without any for the OTHER half. |
00:07.55 | [TK]D-Fender | Gremlin: Well I'm not asking them, I'm asking YOU |
00:07.57 | Gremlin | I know I did. |
00:08.15 | Gremlin | And my best guess is 768 Kbps. |
00:08.16 | [TK]D-Fender | Gremlin: If you can't come up with a number, don't even ask the question :) |
00:08.48 | [TK]D-Fender | Gremlin: And 16 calls on that? extremely unlikely in the best possible scenario |
00:08.57 | Gremlin | I would do a test, but currently I have no Internet access there at all (I blame the gov't). |
00:09.21 | [TK]D-Fender | 13 *16 |
00:09.32 | Gremlin | Well, 768 Kbps would be the upload speed. |
00:09.42 | [TK]D-Fender | Gremlin: I know.. |
00:10.09 | Gremlin | Teliax told me they have a GSM codec that can do 8 Kbps. |
00:10.17 | [TK]D-Fender | Gremlin: Actually, it might be doable, but only over an IAX2 trunk connection with something like GSM as a codec |
00:10.37 | [TK]D-Fender | Gremlin: no, their & *'s GSM 6.10 = 13kbps |
00:10.57 | [TK]D-Fender | Gremlin: + packet overhead, whih you'll need IAX2 trunking to survive |
00:12.08 | nullable_type | D-Fender >> I set a channel variable via Set(SOURCE_NUMBER=17789602222) function but it doesn't show when i tried core show channels concise. Do you know why. Or did you mean i pull the channel variable some other way? |
00:12.12 | [TK]D-Fender | Gremlin: Correction... seems you CAN survive non-trucked with GSM or G.729 |
00:12.15 | Gremlin | I could always do cable+ADSL |
00:12.24 | [TK]D-Fender | trunked* |
00:12.36 | [TK]D-Fender | nullable_type: Other way for the channel var method. |
00:12.51 | [TK]D-Fender | nullable_type: Which I told you. Then again I told yuo ANOTHER way by changing your CHANNEL line. |
00:13.04 | [TK]D-Fender | nullable_type: I think you dropped the programme on the floor. |
00:13.15 | Gremlin | I' a bit confused with what you mean by trunked. I would assume that I am trunked if I set up Asterisk to use Teliax over Session Initiation Protocol. |
00:13.57 | Gremlin | If trunk means the PBX lines. |
00:14.04 | drmessano | lol |
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00:14.20 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
00:14.24 | Gremlin | Disclaimer: I'm an idiot. |
00:15.05 | leif[mobile] | bash.org +1 |
00:15.22 | Gremlin | You submitted this to bash.org? :o |
00:15.39 | leif[mobile] | bash.org +2 |
00:18.45 | Pan3D | Gremlin: just read up a bit. if I can pick it up, you can :) |
00:19.04 | Gremlin | Okay. (initiating bash.org evasion) |
00:19.34 | Pan3D | haha, nice evasion |
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00:23.44 | ruben23 | hi anyone have idea to increase my asterisk server performance.. |
00:24.28 | ruben23 | i already got E1 connection but when 25 calls at the same time my voice quality still are compromise.. |
00:25.36 | ruben23 | got a dual core 4GBddr2 ram with 500 GB Sata Drive |
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00:32.51 | stope | echo type calls? |
00:32.57 | [TK]D-Fender | ruben23: That description gives no indication of being too weak |
00:33.41 | ruben23 | <PROTECTED> |
00:34.19 | [TK]D-Fender | ruben23: Well you've told us nothing suspicious for your problem |
00:34.26 | stope | what version of * ? are you running g729? |
00:35.00 | alrs | trixbox? music-on-hold? queues? |
00:35.05 | ruben23 | yes g729 codec..i got this error log on call- http://pastebin.com/m2f564a05 |
00:35.20 | ruben23 | im using asterisk as backend for vicidial.. |
00:38.29 | ruben23 | <PROTECTED> |
00:40.31 | alrs | ruben23: using hardhdlc instead of dchan in zaptel.conf helps a lot if you have a two or four-port Digium card |
00:42.09 | alrs | if you are using multiple one-port cards I predict problems |
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01:01.06 | jaytee | alrs, hardhdlc with PRI instead of dchan? |
01:01.14 | alrs | yes |
01:02.27 | jaytee | where'd you hear about that? |
01:02.54 | alrs | found it somewhere in a commit message in Digium's version control |
01:03.46 | jaytee | ah, the joy of well coordinated and consolidated OSS documentation :-) |
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01:04.49 | alrs | jaytee: It was after spending a whole lot of time following a support guy's suggestion that voicebus would fix everything |
01:04.49 | alrs | this was last year when that was just coming out |
01:04.49 | alrs | of course, voicebus is only for analog cards |
01:04.49 | alrs | whatevs |
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01:05.11 | jaytee | well, I've got two servers with TE212P dual port cards in them and I'm using dchan. It's rock solid so I think I'll leave well enough alone. |
01:05.21 | alrs | makes sense to me |
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01:06.26 | jaytee | I have one of my two spans do an HDLC Abort every wednesday at 1:58AM so I suspect the telco is doing a reset of their stuff at that time. It's too precise and regular. |
01:07.12 | *** join/#asterisk Shizuo (i=shizuo@200-171-49-211.dsl.telesp.net.br) |
01:08.28 | drmessano | jaytee |
01:08.35 | drmessano | Its 2am, check your damn clocks |
01:08.37 | jaytee | drmessano, |
01:09.07 | jaytee | well, my clocks are synched to the domain controllers |
01:09.32 | *** join/#asterisk saftsack (n=saftsack@p5792458A.dip.t-dialin.net) |
01:09.34 | jaytee | so yeah, maybe Time Warner is doing it at 2AM and my servers are 2 minutes fast |
01:09.56 | jaytee | but nothing's broke so I don't give a damn. |
01:09.58 | drmessano | Hmm |
01:10.03 | drmessano | I see what you did there |
01:10.36 | jaytee | look, unless it means more money in my pocket then we aren't having this argument! |
01:12.05 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
01:14.53 | paulius | So does anyone know how to eliminate echo on the SPA 3102? |
01:15.12 | Shizuo | SPA? |
01:15.18 | paulius | Linksys adapter thingy. |
01:15.21 | Shizuo | IS this some kind of digium crap? |
01:15.23 | Shizuo | Oh |
01:15.26 | paulius | Gives an FXS/FXO port. |
01:15.29 | rob0 | Unplug it. ;) |
01:15.35 | rob0 | SCNR |
01:15.39 | paulius | Yes it's crap but I didn't feel like paying tons for a Cisco system. |
01:15.51 | Shizuo | Yeah, digium sucks |
01:16.06 | rob0 | oh the Sipuras are not so bad, for analog stuff. |
01:16.28 | rob0 | definitely THE cost effective way to get into this. |
01:16.31 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) [NETSPLIT VICTIM] |
01:16.33 | Shizuo | Sure |
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01:16.39 | Shizuo | As long as it's not Digium (crap) |
01:16.40 | Shizuo | It's ok |
01:16.41 | [TK]D-Fender | Shizuo: Only one shitting all over Digium.. or heck even mentioning their name here... is you... |
01:16.44 | paulius | rob0: Right. exact;y/ |
01:16.56 | paulius | rob0: But do you have any guidance on how to eliminate the echo? |
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01:16.59 | paulius | It's quite annoying |
01:17.23 | rob0 | I buy Digium when appropriate. I certainly support what they've done for the community. |
01:17.33 | [TK]D-Fender | paulius: www.voxilla.com <- go check their forums. Plenty of guides on how to tweak to deal with echo situations. |
01:17.34 | Shizuo | Ripped off? |
01:18.06 | *** join/#asterisk propellerhead (n=yogurt2u@host253.200-82-98.telecom.net.ar) |
01:18.21 | rob0 | Digium support helped a lot when I was getting started, with a TDM card. Can't complain. |
01:18.45 | Shizuo | They're ok |
01:18.58 | Shizuo | The only issue I have with them is paying IRC shills |
01:19.06 | Shizuo | And making Asterisk baitware |
01:19.16 | [TK]D-Fender | Shizuo: "baitware"? |
01:19.27 | Shizuo | Yes |
01:19.33 | [TK]D-Fender | Shizuo: Do explain... |
01:19.40 | Shizuo | Open-source stuff where the real deal is a paid software |
01:19.50 | [TK]D-Fender | Shizuo: And who are "IRC shills"? |
01:19.52 | Shizuo | Just like Alfresco and others |
01:19.58 | [TK]D-Fender | Shizuo: What "real deal"? |
01:20.04 | *** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
01:20.12 | Shizuo | The non-community version |
01:20.23 | [TK]D-Fender | Shizuo: What makes that "the real deal"? |
01:20.27 | Shizuo | Well |
01:20.41 | rob0 | Huh? The free software asterisk is fully functional. |
01:20.41 | Shizuo | As lots of main developers are connected to the sponsor (jusr like Alfresco) |
01:20.44 | JayTee52 | comes with a stick on decal? |
01:20.52 | Shizuo | The community distribution is a pain in the ass (ON PURPOSE) to install |
01:21.00 | Shizuo | While the paid version is easy and trouble-free |
01:21.16 | [TK]D-Fender | Shizuo: BS... 3 stupid compiles. No different than any other version |
01:21.22 | [TK]D-Fender | Shizuo: Software is software |
01:21.26 | rob0 | Mine ... has no pain from installing asterisk. I *can* think of a PITA, however. |
01:21.44 | [TK]D-Fender | Shizuo: and "on puropose"? This is OSS SOURCE. Its no different to compile than jsut about any other project |
01:21.47 | Shizuo | [TK]D-Fender: Only 3 stupid compiles? |
01:21.54 | [TK]D-Fender | Shizuo: Yup |
01:21.57 | Shizuo | [TK]D-Fender: Why is there NO stupid compiles at the commercial version? |
01:22.07 | [TK]D-Fender | Shizuo: * itself, Zaptel/DAHDI, and Libpri |
01:22.22 | Shizuo | [TK]D-Fender: Why not a community version with no stupid compiles? |
01:22.25 | [TK]D-Fender | Shizuo: I dunno, maybe because it comes preconifugred on an entire ISO? |
01:22.35 | [TK]D-Fender | Shizuo: And maybe you're ignrant of PACKAGED *. |
01:22.36 | Shizuo | Because that would drain support money from Digium |
01:22.42 | [TK]D-Fender | Shizuo: Which wouldn't surprise me |
01:22.56 | Shizuo | Alfresco works that way too |
01:22.58 | [TK]D-Fender | Shizuo: Digium has a binary repo, as does every major distro maker. |
01:23.05 | Shizuo | Lulz |
01:23.09 | [TK]D-Fender | Shizuo: Your point is now wholly zoid. Next? |
01:23.10 | rob0 | The PITA I am thinking of is a person who comes into a Digium-sponsored and -run IRC channel to bash Digium. |
01:23.18 | Shizuo | What a paid shill |
01:23.31 | [TK]D-Fender | Shizuo: Ok, Enough of your bullshit FUD |
01:23.33 | rob0 | I wish I was paid. |
01:23.43 | Shizuo | Shiiiil, shiil |
01:23.50 | rob0 | ignored |
01:23.53 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
01:23.54 | Shizuo | This place should be called ##digium |
01:23.57 | Shizuo | Not #asterisk |
01:24.04 | Shizuo | <PROTECTED> |
01:24.36 | [TK]D-Fender | Shizuo: Let me get an electron microscope so I can FIND IT FIRST |
01:24.46 | *** mode/#asterisk [+b *!*@200-171-49-211.dsl.telesp.net.br] by [TK]D-Fender |
01:24.47 | *** kick/#asterisk [Shizuo!n=joe@64.235.218.194] by [TK]D-Fender ([TK]D-Fender) |
01:24.54 | JayTee52 | thank you! |
01:25.00 | rob0 | I guess that was what he wanted. |
01:25.01 | [TK]D-Fender | So long moron... |
01:25.02 | MaliutaLap | was waiting on that one |
01:26.01 | JayTee52 | where is br? brazil? |
01:26.08 | rob0 | yes |
01:26.19 | MaliutaLap | probably a compromised machine |
01:27.04 | MaliutaLap | one of the channels I'm on on another network is suffering from botnets trying to appear like genuine users |
01:27.22 | JayTee52 | seems to me alot of cantankerous assclowns from there come in here lookin to raise a ruckus |
01:27.38 | rob0 | he was/is real, replying |
01:27.52 | rob0 | not a bot |
01:28.04 | MaliutaLap | rob0: yeah, doesn't mean he was the owner of the machine he was on |
01:28.08 | JayTee52 | yeah, way too stupid to be a bot |
01:28.14 | rob0 | haha |
01:28.19 | MaliutaLap | rob0: bots and compromised are 2 different things |
01:28.32 | JayTee52 | not necessarily |
01:28.38 | JayTee52 | could be both at the same time |
01:29.11 | JayTee52 | a computer compromised by a bot is still compromised and has a bot, the jerkoff using it might not be compromised, just stupid |
01:29.12 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
01:29.26 | MaliutaLap | and not all bots are bad |
01:29.33 | MaliutaLap | ~jbot |
01:29.34 | infobot | well, jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
01:29.44 | JayTee52 | certainly not. infobot is a standup bot |
01:29.56 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a846428e71332e56) |
01:29.59 | JayTee52 | ~botsnack |
01:29.59 | infobot | :), JayTee52 |
01:30.08 | JayTee52 | I like feeding em |
01:30.39 | JayTee52 | damn, I'm still on as a ghost with my regular account due to the power blib I had here 10 minutes ago. |
01:31.09 | JayTee52 | but I'm not showing up in the userlist? |
01:31.28 | jaytee | better |
01:31.45 | jaytee | I really need to buy a 10 Gigawatt UPS |
01:31.55 | drmessano | ebay |
01:32.08 | jaytee | of course! why didn't I think of that? :-) |
01:32.31 | jaytee | how've ya been, danny? |
01:33.35 | KyleK | I only need 1.21 gigawatts |
01:33.46 | drmessano | Me? Pretty shitty |
01:34.04 | jaytee | sorry to hear that bud! |
01:34.31 | drmessano | Its all good |
01:34.33 | jaytee | work's been pretty dull lately. back to break/fix and install crap. |
01:34.53 | drmessano | No high drama? |
01:35.04 | jaytee | but at least the landlord had someone fix the damn roof after my kitchen ceiling collapsed from the rain leaking in. |
01:35.13 | drmessano | Good god |
01:35.15 | drmessano | That sucks |
01:35.45 | jaytee | now of course God has decided to "f" with me some more by sending a storm with high winds and heavy rain :-( |
01:36.09 | jaytee | long as my kitchen floor is dry tomorrow I'll be happy |
01:36.53 | jaytee | If not I'll just toilet paper a local church as revenge |
01:37.07 | drmessano | It'll be ok.. One thing I know about god, other than that he doesnt exist, is that it rains |
01:37.23 | jaytee | hahaha!! |
01:37.43 | MaliutaLap | drmessano: that's god pee'ing on us |
01:37.45 | jaytee | or as Woody Allen once said, "Not only is there no God but try and get a plumber on a Sunday" |
01:38.02 | drmessano | JESUS DIES IN THE 7TH BOOK, JUST LIKE DUMBLEDORE <--- SPOILER |
01:38.12 | jaytee | LOL |
01:38.35 | [TK]D-Fender | drmessano: :F |
01:39.01 | drmessano | That would be an awesome bash |
01:39.39 | MaliutaLap | drmessano: we can bash you over the head with it? |
01:39.59 | drmessano | "I just got back from Bible study" "Hey man, Jesus dies about halfway through" "Damnit!" |
01:40.17 | drmessano | I love spoilers |
01:40.25 | KyleK | awww man |
01:40.29 | drmessano | By the way, in the sixth sense, Bruce Willis is really DEAD |
01:40.32 | KyleK | i just got started ;) |
01:41.34 | drmessano | Meatloaf dies at the end of Bat Out Of Hell 3 |
01:41.39 | drmessano | O.o |
01:42.18 | jaytee | whenever I get cornered by a born again and they bring up the point about the bible being "the literal word of God" I like to point out that in one of gospels when Jesus goes before Pilate he's wearing a scarlet robe and in another gospel he's wearing a purple robe. "So, like He didn't proofread His own work? Dude, that's so lame for someone omniscient and omnipotent, don't ya think?" |
01:42.41 | [TK]D-Fender | drmessano: http://tinyurl.com/5g3sq4 |
01:42.45 | drmessano | ROFL |
01:43.46 | drmessano | Oh crap |
01:43.52 | drmessano | I havent seen 3 of those |
01:43.54 | drmessano | :((((( |
01:43.57 | drmessano | BACKFAIL |
01:45.09 | drmessano | Kristin shot JR.. one of my favs |
01:46.06 | drmessano | I watched every episode of Dallas on TNN at the time, one a day for like a year and a half |
01:46.30 | drmessano | Got to watch the Who shot JR? episode.. but only had to wait til Monday to find out |
01:46.31 | drmessano | heh |
01:47.06 | a1fa | anybody noticed increase ammount of spam in asterisk logs? |
01:47.20 | KyleK | i haven't looked at my logs at all yet |
01:47.28 | jaytee | spam? |
01:47.30 | a1fa | [May 13 18:11:18] NOTICE[29301]: chan_sip.c:14654 handle_request_invite: Failed to authenticate user "MeucciSolutions" <sip:MeucciSolutions@93.190.143.10> |
01:47.31 | drmessano | Yes, the world is burning |
01:47.34 | drmessano | Run! |
01:48.00 | a1fa | people constantly dicking with my internet facing asterisk server |
01:48.26 | jaytee | fortunately my * servers don't "talk" to the outside world, just internal peers. If ya gotta get somewhere else it's PRI baby! |
01:48.33 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
01:48.40 | a1fa | i would consantly have to adjust firewall rules |
01:48.52 | a1fa | to block off these idiots |
01:48.56 | a1fa | i need a sip smart firewall |
01:49.16 | a1fa | 3 - 404s or 401s.. and you are out for a timeout |
01:49.17 | bkw_ | I wouldn't trust a SIP ALG |
01:49.29 | KyleK | make/build a firewall that limits access to USA or Canada |
01:49.35 | drmessano | Just use decent usernames and strong passwords |
01:49.40 | KyleK | im sure a lot of people would find that handy |
01:49.44 | a1fa | KyleK: i got people outside of USA |
01:49.48 | KyleK | oh |
01:49.56 | jaytee | when someone actually invents a secure internet I might reconsider using SIP but until then it's a crapshoot and a bug farm. |
01:50.01 | a1fa | i guess i just need to open for /16s |
01:50.12 | a1fa | SIP is not that bad |
01:50.18 | a1fa | just like any other web protocol |
01:50.23 | KyleK | i kinda like insecure internet |
01:50.29 | KyleK | secure yo shit peeps |
01:50.29 | a1fa | internet is fine |
01:50.30 | drmessano | Seriously.. do you run any other daemons on the public internet? That shit gets pounded like prom night |
01:50.35 | a1fa | its the idiots on the internet |
01:50.37 | drmessano | Just use common sense |
01:50.41 | a1fa | yeah |
01:50.47 | drmessano | ZOMG THE SIP IS FALLING <-- bad response |
01:50.49 | KyleK | I run ssh on a port for my local computer |
01:50.57 | a1fa | i run ssh, http, https, 5060... |
01:51.02 | a1fa | they all get pounded |
01:51.03 | a1fa | constantly |
01:51.06 | KyleK | really? |
01:51.10 | a1fa | yes |
01:51.12 | drmessano | Great, welcome to SIP being exposed |
01:51.16 | KyleK | I run just https and didn't seem to get pounded |
01:51.18 | drmessano | Now secure it and move on |
01:51.48 | jaytee | screw common sense. 96% of all attempts to penetrate my security are from friggin China and the bulk of the remainder are from Eastern European shithole countries like Romania. |
01:51.57 | a1fa | haha |
01:51.58 | a1fa | :) |
01:52.03 | a1fa | no internet laws in romaina |
01:52.06 | a1fa | pound away boys |
01:52.12 | jaytee | I don't trust those commie bastards or former commie bastards |
01:52.31 | drmessano | As long as there's stupid people, they will always win. 101/101 is NOT a secure set of credentials |
01:52.36 | drmessano | Nor is 101/fluffy |
01:52.40 | jaytee | hehehe |
01:52.46 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
01:52.55 | KyleK | hmmm a bit more access than i expected |
01:53.00 | a1fa | all my passwords are Fu@#$Ciked-dO2??\Up! |
01:53.39 | drmessano | all my passwords are the names of Asterisk developers spelled backwards |
01:53.43 | drmessano | Now come on, hack me |
01:54.00 | a1fa | :) |
01:54.02 | a1fa | ring ring |
01:54.08 | a1fa | 1-900-BANANA-FONE |
01:54.10 | MaliutaLap | jaytee: romania only went bolshie because of the americans |
01:54.47 | a1fa | dont make excuses |
01:54.47 | a1fa | although |
01:54.47 | MaliutaLap | jaytee: and their smelly deal with stalin |
01:54.47 | a1fa | the only romaian I like is Sandra Romain |
01:54.47 | a1fa | :) |
01:54.48 | rob0 | There are, for all practical purposes, no Internet laws anywhere. Even where(/if) smart laws exist, LEO's are not smart enough to enforce them. If they were, 419'ers everywhere would be stomped out quickly. |
01:54.50 | MaliutaLap | jaytee: trust the US to do a deal with someone who killed more people than hiler |
01:54.55 | MaliutaLap | hitler even |
01:55.19 | a1fa | cmon |
01:55.30 | a1fa | no body commenting on Sandra Romain |
01:55.31 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
01:55.34 | a1fa | !g Sandra Romain |
01:55.34 | *** join/#asterisk joako_ (n=andrew@opensuse/member/joak0) |
01:55.43 | a1fa | you guys suck :) |
01:55.57 | joako_ | Can someone recommend a VoIP route to Cuba as reliable as AT&T? |
01:55.59 | drmessano | Everyone is so paranoid that their precious Asterisk/FreeSWITCH/OpenSIPSomethingoftheweek SIP daemon is being exposed to all these horrible bad guys doing naughty bad awful things.. Well, no shit.. I dont understand the freak out about it |
01:56.38 | *** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) |
01:56.59 | KyleK | were americans not allowed to phone cuba up until now? |
01:57.00 | a1fa | i am not freaking out |
01:57.06 | a1fa | lol |
01:57.07 | KyleK | keep hearing about cuba :) |
01:58.02 | drmessano | a1fa: So you're the one person in the 50 this week that's come in here screaming ZOMG MY LOGS ARE FULL OF HAXORS that isn't freaking out? |
01:58.04 | MaliutaLap | KyleK: I don't think americans are allowed to think about cuba as anything more than the "red threat" without being done for treason |
01:58.05 | drmessano | Noted. |
01:58.31 | a1fa | drmessano: i just noted the increase in the last 24h |
01:58.37 | drmessano | The red threat? What is this the 1960s? |
01:58.37 | a1fa | there must be some kind of exploit |
01:58.50 | a1fa | :) |
01:59.03 | joako_ | KyleK: When I had AT&T/Bellsouth and their long distance service calling worked rather well. Using various VoIP providers I have tried at least 30 calls in the past 2 weeks and only 2 or 3 calls have even completed to the point of carrying on a conversation and there was massive delay. I gave up, bit the bullet and just paid $1.79/min on my mobile phone (AT&T) and the call connected with no satellite delay. Is there any way to get a decent service |
01:59.14 | MaliutaLap | drmessano: you live in the backwards country, not me |
01:59.16 | *** join/#asterisk JenniferAkemi (n=Jennifer@76-10-182-237.dsl.teksavvy.com) |
01:59.17 | drmessano | a1fa: There is.. it's documented in RFC 2543 |
01:59.20 | joako_ | Mind you I have no issues with calls to 1st world countries on VoIP |
01:59.32 | MaliutaLap | drmessano: I'm sure I'm on a list _as_ the red threat |
01:59.43 | MaliutaLap | hard to be a left wing activist and not be |
02:00.01 | MaliutaLap | especially what the rest of the world considers left |
02:00.14 | Pan3D | heh |
02:00.32 | drmessano | MaliutaLap: No one under the age of 40 cares about Cuba |
02:00.45 | drmessano | MaliutaLap: Its really a non-issue anymore |
02:01.40 | MaliutaLap | drmessano: what's the average age of congress? |
02:01.46 | Pan3D | drmessano: indeed, to everyone *except* those with a vested interest in perpetuating the bad vibes. |
02:01.58 | MaliutaLap | drmessano: I think it's still an issue in the halls of government |
02:01.59 | drmessano | MaliutaLap: 94 |
02:02.06 | Pan3D | hahaah |
02:02.08 | drmessano | MaliutaLap: Make that 93, one just died |
02:02.10 | Pan3D | Senator Byrde |
02:02.28 | MaliutaLap | Senator Bryde of Satan? |
02:02.46 | MaliutaLap | or has Cheney been shooting again? |
02:03.09 | drmessano | Besides which, Congress could care less about Cuba too.. They're trying to stop internet pirates, clog the tubes, and make AT&T richer |
02:03.39 | Pan3D | heh, I was flying in and out of Venezuela during the past 7 years. Nothing like being a walking tarket for the hate mongers :) |
02:04.40 | joako_ | drmessano: Oh, so that explains it. VoIP crappy routes to cuba to boost people to switch back to AT&T |
02:05.01 | drmessano | joako_: THATs the ticket |
02:05.22 | jaytee | told you it was a conspiracy! same people that framed Pete Rose! |
02:05.33 | [TK]D-Fender | USA needs an enemy close by they can distract the population with. |
02:05.54 | jaytee | I nominate Canada! Free Maple Syrup for all! |
02:06.04 | phunyguy | hates conspiracy theories. |
02:06.33 | jaytee | phunyguy, yes it's well documented in your NSA file :-) |
02:06.39 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
02:07.13 | phunyguy | hmm? |
02:07.17 | phunyguy | NSA? |
02:07.19 | drmessano | phunyguy: You also hate brussel sprouts |
02:07.26 | jaytee | No Such Agency |
02:07.36 | drmessano | phunyguy: and you are a big fan of Tool |
02:07.42 | drmessano | phunyguy: Its all in your file |
02:07.44 | phunyguy | everyone hates brussel sprouts |
02:07.49 | phunyguy | and everyone loves tool |
02:07.52 | phunyguy | :) |
02:07.52 | drmessano | I dont |
02:07.59 | phunyguy | okokok one. |
02:08.10 | drmessano | Tool sounds like broken powertools having sex |
02:08.11 | phunyguy | (I mentioned Tool yesterday - so there) |
02:08.11 | jaytee | not Bob DeBenedictus. Bob loves the brussel sprouts |
02:08.29 | *** join/#asterisk matsk (n=matkar@c-118ae253.174-6-64736c10.cust.bredbandsbolaget.se) |
02:08.44 | drmessano | Want to spice up a party.. throw some Neil Diamond on |
02:09.04 | *** part/#asterisk matsk (n=matkar@c-118ae253.174-6-64736c10.cust.bredbandsbolaget.se) |
02:09.11 | drmessano | Hot August Night <--- Nuff Said |
02:09.49 | drmessano | http://www.abc.net.au/reslib/200708/r170041_637422.jpg |
02:11.15 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
02:13.07 | Kobaz | i have a standard analog phone on a grandstream sip gateway, dtmf payload type is 97, and dtmf is rfc2883... i dial out on an iax connection... and i have a Read on the other end.... it never sees any digits comming in... even though I can chanspy either leg of the call, and hear the dtmf's go through |
02:13.13 | jaytee | Neil Diamond rocks!!!!! |
02:13.26 | drmessano | Neil is the man |
02:13.32 | phunyguy | o_O |
02:13.56 | drmessano | Hot August Night - Red, Red Wine <--- Even more enuff said |
02:14.09 | drmessano | Gets no better |
02:14.42 | [TK]D-Fender | drmessano: that sounds like a threat ;) |
02:15.11 | Kobaz | i'll paste up some debug in a sec |
02:15.16 | drmessano | lol |
02:16.06 | [TK]D-Fender | Kobaz: And prove on the first leg that * gets the DTMF. |
02:16.27 | drmessano | I still deeply resent he changed up the tempo in his live performances of Red Red Wine after UB40 went #1 with a reggae version |
02:16.30 | Kobaz | [TK]D-Fender: k |
02:16.42 | drmessano | I had more respect than that |
02:16.44 | drmessano | :( |
02:17.48 | *** join/#asterisk jsolis (n=jimmy@201.240.109.106) |
02:17.55 | [TK]D-Fender | drmessano: think about how Sugar Ray re-released "Fly" without the Rap guy from the first vid & single, and Shania Twain undid that duet she did with (other country guy here) |
02:18.14 | Kobaz | [TK]D-Fender: should the dtmf's show up in sip debug? |
02:18.27 | Kobaz | [TK]D-Fender: i've never debugged dtmf stuff before... not sure where to look |
02:18.44 | [TK]D-Fender | Kobaz: the only proff that is acceptable is the hard kind. Using * dialplan apps that depend on it. |
02:19.10 | [TK]D-Fender | Kobaz: Prove it can navigate VoicemailMain, Read, etc |
02:19.50 | drmessano | [TK]D-Fender: The Shania thing was hysterical.. it was a duet on Country Radio and not on mainstream.. then to back up her hypocrisy, she released a second version of her album with all the alternate cuts |
02:19.54 | drmessano | WTFFFF |
02:20.13 | drmessano | County and Sellout version |
02:20.19 | *** join/#asterisk thuddwhirr (n=wolthuis@mimezine.com) |
02:20.52 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
02:21.25 | Kobaz | [TK]D-Fender: yeah, okay... good point.... no dtmf getting to voicemail |
02:22.24 | [TK]D-Fender | Kobaz: Run along now :) |
02:22.41 | Kobaz | no no but |
02:22.45 | Kobaz | how do i fix this :P |
02:23.13 | thuddwhirr | anyone here want to help me troubleshoot a weird SIP issue? :) |
02:24.00 | jaytee | is listening to Neil Diamond - Cherry, Cherry [3:13 (11%)] |
02:24.35 | [TK]D-Fender | Kobaz: Show us something useful |
02:25.16 | Kobaz | [TK]D-Fender: well, i dont get anything in sip debug, no digits recieved on a Read() |
02:25.25 | Kobaz | i can't show something i don't have |
02:25.39 | [TK]D-Fender | Kobaz: How about CONFIGS from both sides <- |
02:25.45 | drmessano | jaytee: FTW |
02:26.03 | jaytee | hehe |
02:26.04 | drmessano | Guess now would be a bad time to mention I have a 4 CD box set of his |
02:26.09 | drmessano | heh |
02:26.27 | jaytee | I've got Hot August Night and Hot August Night II |
02:26.30 | [TK]D-Fender | drmessano: Sounds like a good firing range back-stop ;) |
02:26.34 | Kobaz | gateway: dtmf:rfc2833 payload type:97 asterisk: peer 5505: DTMFmode : rfc2833 |
02:26.55 | [TK]D-Fender | Kobaz: real pastebin please. |
02:27.06 | Kobaz | comming |
02:27.39 | drmessano | jaytee: I used to pretend Neil Diamond was my real Dad.. that maybe mom made a boo boo after a really exciting concert.. But NOOOOOO |
02:27.45 | drmessano | Thanks for nothing, mom |
02:28.07 | drmessano | stomps off in disgust |
02:28.20 | Kobaz | http://pastebin.com/m5f984574 |
02:28.21 | Kobaz | that's the phone |
02:28.24 | Kobaz | er |
02:28.28 | Kobaz | sip peer on * |
02:29.05 | thuddwhirr | ive got my asterisk server hooked up to a sip trunk. I have a DID with that sip provider pointing at my asterisk. i can originate out fine, and i can receive incoming calls fine, but if I try and originate to that DID, asterisk immediatly issues a cancel when it sees the second invite, with no explination. |
02:29.20 | thuddwhirr | anyone have any thoughts on how to debug that? or why that wouldnt be allowed? |
02:29.24 | Kobaz | trying to figure out how to dump the gateway config |
02:29.49 | jaytee | drmessano, regardless of whether Neil was your real father or not, if you weren't here in this world it would be a much darker, colder and less humorous place so thank god for whatever sperm donor did the deed! |
02:30.05 | Kobaz | thuddwhirr: you'll need to elaborate... i'm not following |
02:30.33 | Kobaz | thuddwhirr: are you're trying to call yourself? |
02:30.39 | thuddwhirr | yes |
02:30.51 | thuddwhirr | looped through my sip provider |
02:31.04 | [TK]D-Fender | Kobaz: imagebin.ca |
02:31.06 | Kobaz | are you sending the right digits |
02:31.18 | [TK]D-Fender | Kobaz: but next I want a failed call with SIP DEBUG for your app test |
02:31.24 | Kobaz | ie: does your provider require/notrequire the areacode prefix, etc |
02:32.20 | [TK]D-Fender | thuddwhirr: maybe you should PASTEBIN the call with SIP debug enabeld so we can see what's happening |
02:32.20 | thuddwhirr | i belive so. i send out the invite, get the right awk back, and then I see the invite for the incoming leg. |
02:32.30 | [TK]D-Fender | ~pb |
02:32.31 | infobot | [~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 , or , http://bin.cakephp.org/ |
02:32.40 | thuddwhirr | ok |
02:33.28 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
02:37.53 | Kobaz | http://pastebin.com/m2ba5961a |
02:38.00 | Kobaz | there's the voicemail with no dtmf |
02:38.09 | Kobaz | http://imagebin.ca/view/2qqHrHu.html |
02:38.18 | Kobaz | there's the bit of gateway config |
02:41.23 | Kobaz | i guess i can play around with the other dtmf types |
02:41.28 | Kobaz | in-audio actually doesn't help |
02:41.45 | Kobaz | all the other phones use rfc2883 just fine (ie: polycoms and such) |
02:41.56 | Kobaz | i can turn on sip info |
02:43.55 | *** join/#asterisk chendy (n=chatzill@58.251.102.216) |
02:44.13 | Kobaz | [TK]D-Fender: okay so.... * is getting dtmf now that i turned on sip info |
02:44.36 | Kobaz | [TK]D-Fender: shitty gateway? that rfc2883 doesn't work? |
02:45.19 | *** join/#asterisk metfan2007 (n=jc@189.146.141.79) |
02:45.23 | [TK]D-Fender | Kobaz: Possibe, but I'm also concerned that it trys as "anonymous" |
02:45.30 | [TK]D-Fender | Kobaz: And keeps failing multiple auths |
02:46.13 | Kobaz | ah, i see |
02:48.19 | metfan2007 | hi all! I have a queue for incomming calls for 10 agents, using callbacklogin, the agents need to make outbound calls too, the problem is that when an agent is making an outbound call, asterisk does not know that he is busy, so It tries to send an incomming call |
02:48.51 | metfan2007 | how do I tell asterisk that an agent is busy when making outbound calls withoud logoff agent? |
02:49.29 | Kobaz | http://pastebin.com/m285b35e6 |
02:49.35 | Kobaz | well i turned off the anonymous checkbox |
02:49.42 | Kobaz | it's sending the real username now |
02:50.07 | Kobaz | still get an unauthorized bit on the first exchange |
02:51.07 | [TK]D-Fender | metfan2007: There is no association between chan_agent using a local channel & Exten and a particular SIP device plaing a call |
02:51.36 | [TK]D-Fender | metfan2007: One uses dialplan, the other is a hardware device (Sorry, could be other than SIP, but hopefully you get the point) |
02:51.46 | [TK]D-Fender | metfan2007: there is no way for * to know. |
02:52.02 | [TK]D-Fender | Kobaz: And still bad DTMF |
02:52.05 | [TK]D-Fender | ? |
02:52.17 | [TK]D-Fender | Kobaz: Checked your firmware? newer or better out? |
02:54.07 | Kobaz | possibly |
02:57.12 | jaytee | [TK]D-Fender, have you seen this from Digg's front page tonight? http://www.acesandeighths.com/8ball_6.html |
02:57.40 | jaytee | I can think of at least 3 people they should have listed that aren't there |
02:58.22 | [TK]D-Fender | jaytee: Yup, finished them all about 15 mins ago |
02:58.33 | jaytee | hehe, figures :-) |
02:58.40 | [TK]D-Fender | jaytee: Antoine Dufour, Eric Johnson, plewnty more |
02:59.15 | jaytee | yeah, Antoine for sure. Tuck Andress |
02:59.53 | drmessano | Slash |
02:59.59 | drmessano | Just sayin |
03:00.04 | jaytee | you're the one that turned me on to Antoine |
03:01.36 | Kobaz | there's new fiemware... lets hope it doesn't break anything |
03:02.08 | Kobaz | firmware |
03:06.28 | Kobaz | [TK]D-Fender: nope... no rfc2883 dtmf after reboot... sip info dtmf still works |
03:08.02 | jaytee | [TK]D-Fender, hey in the youtube vid of Spiritual Groove that's a solo of Antoine, not the one with Tommy Gauthier, is he playing a cutaway classical or a cutaway steel string? The neck looks like a classical but the strings look like standard bronze and the treble sounds like steel not nylon. |
03:09.55 | *** part/#asterisk jsolis (n=jimmy@201.240.109.106) |
03:15.12 | MaliutaLap | Yay! off to meet Sexy Geek Girl ... don't wait up for me :P |
03:18.49 | joako | OMFG you can download Linksys SPC tool from cisco.com now without any bullshit |
03:20.04 | drmessano | Yes |
03:20.15 | drmessano | Thats pretty badass |
03:21.24 | jaytee | SPC tool? whazzat? |
03:21.31 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
03:21.49 | joako | jaytee: the util to generate config files for Linksys SPA-phones and ATAs |
03:22.03 | drmessano | Skeleton configs, basically |
03:22.08 | drmessano | Lets you encrypt them as well |
03:22.11 | jaytee | ah! |
03:22.35 | joako | Time to take down http://spc.pifiu.com I suppose... when I first started the site the logs showed plenty of visitors from .cisco.com lol |
03:23.01 | drmessano | Thats your site? |
03:23.14 | joako | drmessano: Yes |
03:24.03 | drmessano | Right on! I got my first SPC stuff from there |
03:26.21 | jaytee | nite all, time for bed |
03:27.30 | *** join/#asterisk propellerhead (n=yogurt2u@host1.190-30-31.telecom.net.ar) |
03:32.34 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc) |
03:34.44 | *** join/#asterisk Braxus (n=braxus@netblock-68-183-230-56.dslextreme.com) |
03:42.46 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
03:49.10 | *** join/#asterisk blkry (n=blkry@97.95.233.232) |
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04:10.50 | VaGoNeTaS | shit, cant make cellphone calls |
04:10.54 | VaGoNeTaS | chanunavail |
04:11.01 | VaGoNeTaS | and is not being blocked by the telco |
04:11.14 | VaGoNeTaS | what could it be |
04:12.11 | [TK]D-Fender | VaGoNeTaS: Absolutely anything |
04:12.21 | VaGoNeTaS | what u mean |
04:12.24 | [TK]D-Fender | perhaps nothing.... |
04:12.29 | [TK]D-Fender | Somwhre in the middle? |
04:12.29 | VaGoNeTaS | i'm in a different place |
04:12.31 | VaGoNeTaS | with 2 e1 lines |
04:12.33 | [TK]D-Fender | No.. clearly to the left |
04:12.47 | VaGoNeTaS | i can receive but not make |
04:13.11 | VaGoNeTaS | im gonna pb the error so u can see |
04:13.44 | VaGoNeTaS | # |
04:14.13 | VaGoNeTaS | pastebin.ca/1422337 |
04:14.30 | VaGoNeTaS | i cant make this shit working properly |
04:15.03 | [TK]D-Fender | ........... |
04:15.07 | [TK]D-Fender | why do I bother... |
04:15.18 | [TK]D-Fender | VaGoNeTaS: You don't even show the friggen DIAL |
04:15.21 | [TK]D-Fender | and NO DEBUG in there |
04:16.00 | [TK]D-Fender | VaGoNeTaS: there is nothing usable in that PB |
04:16.08 | VaGoNeTaS | i did |
04:16.12 | VaGoNeTaS | pri debug span 1 |
04:16.18 | VaGoNeTaS | so im gonna show you the whole message ok |
04:16.24 | [TK]D-Fender | <PROTECTED> |
04:16.29 | [TK]D-Fender | VaGoNeTaS: We don't see the DIAL! |
04:16.37 | VaGoNeTaS | what u want then |
04:16.43 | [TK]D-Fender | VaGoNeTaS: How the hell do we know what you're doing there? |
04:16.44 | VaGoNeTaS | what u want me to pb to |
04:16.49 | VaGoNeTaS | hahahaha |
04:16.52 | VaGoNeTaS | im trying to reach a cellphone |
04:16.57 | [TK]D-Fender | VaGoNeTaS: show the ENTIRE &^#$ing call |
04:17.01 | VaGoNeTaS | k |
04:17.05 | VaGoNeTaS | wait a sec |
04:17.06 | [TK]D-Fender | VaGoNeTaS: Ther is no DIAL in your pastebin. |
04:20.35 | VaGoNeTaS | im dialing with a ftphone |
04:20.45 | VaGoNeTaS | im pretty much sure that the problem is the fucking dahdi |
04:20.56 | VaGoNeTaS | it doesnt well support the fucking Redfone quad box |
04:20.59 | VaGoNeTaS | piece of shit |
04:21.11 | VaGoNeTaS | shitty 3G box |
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04:24.05 | [TK]D-Fender | VaGoNeTaS: It is a device I would never recommend |
04:24.08 | [TK]D-Fender | VaGoNeTaS: http://support.red-fone.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=20 |
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04:24.33 | VaGoNeTaS | so you with me |
04:24.38 | VaGoNeTaS | its a fucking piece of shit |
04:24.50 | VaGoNeTaS | an expensive piece of shit |
04:29.33 | [TK]D-Fender | VaGoNeTaS: Call them for support |
04:30.05 | VaGoNeTaS | dude |
04:30.14 | VaGoNeTaS | if im gonna call redfone is just for one reason |
04:30.18 | VaGoNeTaS | isulting them |
04:30.23 | VaGoNeTaS | xD |
04:30.45 | VaGoNeTaS | i really prefer digium card |
04:31.02 | VaGoNeTaS | T110 maybe would've been better |
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04:42.53 | GeekBoy | when doing "make all" for DAHDI, It is returning, " You do not appear to have the sources for the 2.6.18-92.1.18.el5.028stab060.8 kernel installed." |
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04:49.24 | [T]ank | getting an error on echo cancelation after upgrade to 1.6. Here are the details, can anyone help? http://pastebin.ca/1422357 |
04:51.38 | [TK]D-Fender | [T]ank: your system.conf did not set the ec to use for that channel |
04:51.43 | [TK]D-Fender | [T]ank: go read the samples |
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05:00.49 | [T]ank | How do i know what echo canceller I have? mg2, kb1, sec2, and sec.? |
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05:15.36 | [T]ank | [TK]D-Fender: reading about ec says that the dahdi-linux compiles mg2, kb1, sec2, and sec cancellation types. I just need to specify which one to use. I also see in the samples echocancel=yes. I have tried all of the above on chan 1 and then restart asterisk. am I even close to where I should be? I am getting the same results with everything I try |
05:16.06 | [TK]D-Fender | [T]ank: You set the EC i your system.conf PER CHANNEL.. go read the samples a few more times. |
05:16.22 | [TK]D-Fender | [T]ank: * compiles them ALL, and loads them dynamically as assigned |
05:16.31 | [TK]D-Fender | [T]ank: No assign = none used. |
05:16.40 | [T]ank | right... got that |
05:17.02 | [TK]D-Fender | checkout time, later all |
05:17.05 | [T]ank | its a fxoks port so my understanding is that there is only 1 chan. am i correct in that? |
05:17.14 | [TK]D-Fender | [T]ank: yes |
05:18.24 | [T]ank | so I have tried echocanceller=<whatever>,1 with each of the echocancel types tried in the whatever field. |
05:18.40 | [T]ank | is that not correct? |
05:19.03 | [T]ank | each type gives me the same result of chan_dahdi.c:2010 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) |
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05:20.55 | [T]ank | so if mg2, kb1, sec2, and sec are all installed, then any one of them should have worked, right? |
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05:31.09 | [T]ank | anyone else here know anything about echo cancellation in /etc/dahdi/system.conf |
05:31.20 | Qwell | [T]ank: you are correct |
05:31.57 | [T]ank | Qwell: I am not sure I am doing any of this right... |
05:32.09 | Qwell | Which version of 1.6 are you using? |
05:32.14 | [T]ank | Is there any chance I do not have echo cancellation installed? |
05:32.31 | [T]ank | 1.6.0.9 |
05:32.46 | Qwell | are your channels working otherwise? |
05:33.06 | [T]ank | yep |
05:33.16 | [T]ank | just getting echo on the outside callers side of the call |
05:33.25 | [T]ank | they hear them selves echo when they speak. |
05:33.29 | Qwell | where are you putting the echocanceller line? |
05:33.36 | [T]ank | the asterisk side of the call sounds just fine |
05:33.47 | [T]ank | let me repaste the config... |
05:33.53 | Qwell | echo is handled on the far end. if they hear echo, they need to fix it |
05:34.20 | [T]ank | http://pastebin.ca/1422375 |
05:35.06 | Qwell | 2.1.0.4? |
05:35.38 | [T]ank | its every single person that calls that gets the echo |
05:35.42 | Qwell | err, 2.1.0.2 for tools |
05:35.52 | [T]ank | oh.. let me check the tools version. |
05:36.09 | [T]ank | yes, 2.1.0.2 |
05:36.32 | [T]ank | I also get the error i mentioned above in the cli regarding echo |
05:36.46 | [T]ank | [May 13 23:20:27] WARNING[12419]: chan_dahdi.c:2010 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) |
05:37.23 | [T]ank | in case it is relevant: Wildcard TDM400P REV I Board 5 |
05:37.31 | Qwell | from what I can tell, your config is right.. |
05:38.09 | Qwell | does `modinfo dahdi_echocan_mg2` show anything? |
05:38.51 | [T]ank | at the linux command line, right? |
05:38.57 | Qwell | yeah |
05:39.22 | [T]ank | drops the cursor to the next line like something should happen. Never completes. |
05:39.31 | [T]ank | wait... just copleted. |
05:39.33 | [T]ank | pasting info |
05:39.55 | [T]ank | http://pastebin.ca/1422379 |
05:40.41 | Qwell | it would yell at you if it didn't exist |
05:41.09 | Qwell | yeah... |
05:42.28 | Qwell | Do you have channels=1 anywhere? |
05:42.59 | Qwell | and I assume you've run ztcfg? (I don't know if that's needed..) |
05:44.02 | [T]ank | you mean the line that would read channel => 1? |
05:44.11 | [T]ank | that is in my /etc/asterisk/chan_dahdi.conf |
05:44.13 | Qwell | no |
05:44.23 | Qwell | dunno, the sample system.conf has channels=X |
05:44.36 | [T]ank | lol... dahdi_cfg -vv |
05:44.43 | [T]ank | Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) |
05:44.45 | [T]ank | testing |
05:45.04 | [T]ank | duh |
05:45.06 | [T]ank | that worked |
05:45.10 | Qwell | :D |
05:45.12 | [T]ank | i have been restarting asterisk |
05:45.18 | [T]ank | forgot to run the dahdi_cfg |
05:45.45 | [T]ank | if the school of hard knocks gave deplomas, I would have a doctorate. |
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05:47.32 | Qwell | if you were a wireless AP, what would your IP be? |
05:47.34 | [T]ank | echo is gone also |
05:47.42 | Qwell | woot |
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05:48.16 | [T]ank | Qwell: thank you, good night |
05:48.16 | Qwell | wonders if his AP even has an accessible IP... |
05:50.40 | Qwell | ah hah! found it. |
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05:51.15 | HeMan | Hi! Where can I find information about the current revission "system" with 1.6.0.9, 1.6.1.0, 1.6.2.0-beta and 1.4? |
05:57.37 | HeMan | is 1.6.0.9 considered stable? |
06:02.03 | Qwell | hmm |
06:03.18 | Qwell | androidvnc is...rather awesome. |
06:04.42 | [T]ank | learns new trick from Qwell |
06:04.45 | [T]ank | :-D |
06:04.50 | [T]ank | now I can die happy |
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07:02.40 | ck_28 | good morning all |
07:02.51 | ck_28 | i followed up digium free fax instalion |
07:03.05 | ck_28 | i am trying to send a file to fax |
07:06.34 | ck_28 | kindly find my config and debug at http://pastebin.com/d60a8686a |
07:08.25 | ck_28 | any one can help ? |
07:22.00 | ck_28 | Dear Admins any help |
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07:51.32 | ck_28 | @russellb can you help me |
07:52.21 | tzafrir_laptop | ck_28, don't ask specific people to help you |
07:52.39 | tzafrir_laptop | the guy you asked has other things on his mind, as he hinted you. |
07:52.59 | tzafrir_laptop | Not to mention that it's in the middle of the night at where he is right now |
07:54.07 | ck_28 | :) |
07:54.42 | *** join/#asterisk Perun (n=perun@2001:6f8:1316:1234:216:3eff:fe07:3160) |
07:54.44 | Perun | hi all |
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07:55.13 | Perun | I want to use usb hfc isdn cards for my asterisk... what should I use, zaptel module or misdn? |
07:56.31 | tzafrir_laptop | what version of asterisk do you use? |
08:03.36 | tzafrir_laptop | prefers zaptel/dahdi , but is biased |
08:08.05 | ck_28 | tzafrir_laptop asterisk version question is for me ? |
08:08.27 | tzafrir_laptop | no. For Perun |
08:08.38 | Perun | tzafrir_laptop: :1.4.21.2~dfsg-3 |
08:08.45 | Perun | debian pkg |
08:08.45 | tzafrir_laptop | I can't support non-free software. Others may be able to |
08:09.31 | tzafrir_laptop | Perun, in that case the answer is simple: mISDN (1.x) has been vetoed out of Unstable Debian by its maintained (Simon Richter) |
08:09.43 | tzafrir_laptop | zaphfc is included in the Debian zaptel package |
08:09.50 | tzafrir_laptop | m-a a-i zaptel |
08:10.33 | Perun | so, I should use zaptel? |
08:10.50 | tzafrir_laptop | I would say that it's the path of least resistance |
08:11.07 | tzafrir_laptop | If you want to use mISDN you need to invest much more work |
08:13.10 | Perun | but what is better for the future? I have seen misdn is the new isdn stack in vanilla kernel? |
08:14.08 | ck_28 | Channel 'SIP/add-084f3c00' fax session '25' is complete, result: 'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' |
08:14.21 | ck_28 | what is the resolution |
08:16.59 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
08:18.04 | ck_28 | [44423.655] T38(0/0/0) Tx CC_FAX_IND |
08:18.05 | ck_28 | 675 <Call 100> : FaxIndicated from(64) |
08:18.05 | ck_28 | 676 <SIP 0> : No authentication information available |
08:18.05 | ck_28 | 677 <SIP 100> : Send INVITE Request |
08:18.05 | ck_28 | 678 <SIP 100> : Receive 100 Trying |
08:18.05 | ck_28 | 679 <SIP 100> : Transaction (72 INVITE) proceeding |
08:18.07 | ck_28 | [44424.695] T38(0/0/0) discard T30D from VoPP by no sess |
08:20.45 | tzafrir_laptop | Perun, mISDN2 is not mISDN |
08:21.03 | tzafrir_laptop | the one in the kernel is mISDN2 |
08:21.40 | tzafrir_laptop | Adding chan_lcr support to the Debian package is work in progress, and will likly happen in the squeeze cycle |
08:23.37 | ck_28 | tzafrir_laptop in which i can post my question ? |
08:23.44 | ck_28 | channnel* |
08:31.00 | Perun | tzafrir_laptop: aa and can I use misdn2 with asterisk and hfc usb cards? (want the NT mode to) |
08:31.22 | tzafrir_laptop | hfc-usb is indeed not supported by zaptel |
08:31.49 | tzafrir_laptop | I suggest you ask on #debian-voip in irc.oftc.net |
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08:35.18 | KyleK | hmm can germany dial a north american toll free number? i know the caller would pay for it, but can a person over there call it? |
08:35.46 | Zeeek | KyleK: I know we can in France |
08:36.11 | Zeeek | a message is read saying "you are calling a tollfree USA number, the bad news is it ain't free for you. |
08:36.30 | KyleK | k |
08:36.34 | Zeeek | so it says if you stay on the line, you will be charged |
08:37.33 | KyleK | ah, I might get a random USA did as well for that then, setting up like 1800grandma kinda thing |
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09:16.36 | *** join/#asterisk Get_The_Fish (n=IceChat7@173-14-4-113-Colorado.hfc.comcastbusiness.net) |
09:17.11 | Get_The_Fish | hello all... can anyone point me to a number that will work as a DTMF tester, like an IVR that will read back the digits that you entered? |
09:17.34 | Get_The_Fish | I have users that complain about DTMF not working intermittently |
09:18.51 | ck_28 | any one can help me to solve the fax t38 capability --T38(0/0/0) discard T30D from VoPP by no previous V21_Flag |
09:20.12 | Zeeek | Get_The_Fish: someone published a dialplan that does that. Possible voicepulse connect or onsip.com or teliax ? |
09:21.01 | *** join/#asterisk Ast001 (n=uros@cable-89-216-155-28.dynamic.sbb.rs) |
09:21.26 | Get_The_Fish | well, I was really looking for a toll free or other number that does that- in other words, I use the provider that I have to connect to the pstn. |
09:21.32 | Ast001 | Hi is there any method for turning off remote unix connection in asterisk cli ? |
09:22.02 | tzafrir_laptop | Get_The_Fish, exten => _X.,1,SayDigits(${EXTEN}); or some variation |
09:22.06 | Get_The_Fish | I have found some that will let me register with them and test, but that doesnt test my configuration with my provider. Am I making sense? |
09:22.44 | Ast001 | I just don't want to see remote unix connection every time myscript connects with asterisk with asterisk -rx |
09:23.18 | Get_The_Fish | I have that, and it's helpful, but I want to know what the "other end" of the call is receiving. Because asterisk is saying everything is peachy, but my users are saying otherwise. Suspecting my provider. |
09:23.54 | tzafrir_laptop | Ast001, this message is a verbose message at verbosity level 3 |
09:24.12 | Ast001 | so I need higher level ? |
09:24.17 | Get_The_Fish | lower level |
09:24.22 | Ast001 | or lower ? ok thanks |
09:24.23 | tzafrir_laptop | You asked Asterisk to give you plenty of noisy messages. Why do you complain when it does so? |
09:24.32 | Get_The_Fish | try "core set verbose 2" |
09:24.34 | Get_The_Fish | or lower |
09:24.41 | tzafrir_laptop | core set verbose 0 |
09:24.43 | Ast001 | ok thank you. |
09:24.58 | tzafrir_laptop | Unless you actually want to troubleshoot / trace something |
09:25.31 | Ast001 | it works thank you tzafrir |
09:25.40 | Get_The_Fish | troubleshooting is overrated. Just keep restarting until it works or windows update tells you they have a a fix.... oh wait... |
09:26.45 | tzafrir_laptop | svn update |
09:26.48 | tzafrir_laptop | make install |
09:26.52 | Get_The_Fish | lol |
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09:43.50 | ck_28 | tzafrir_laptop you are the only active person in this channel :) |
09:43.59 | ck_28 | tzafrir_laptop can you guide me or give me a key |
09:44.40 | tzafrir_laptop | I have already told you that I don't know this digium fax thing |
09:44.54 | ck_28 | thanks |
09:45.17 | ck_28 | what is the gigium channel name ? |
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10:36.39 | Get_The_Fish | tzafrir, you still up and around in here? Any ideas on why a SIP trunk's register packet would have the receiving host's IP in the from: header coming from asterisk? |
10:44.04 | *** join/#asterisk bmg505 (n=leon@196-209-78-114-rndf-esr-5.dynamic.isadsl.co.za) |
10:46.39 | Get_The_Fish | doesnt look like the fromdomain setting in the peer details is working properly, the from: field in the SIP message header is showing the address of the host that is receiving the invite. Any help on this? |
10:54.23 | kaldemar | Get_The_Fish: is it a register or an invite or a reply to one of those? show something concrete. |
10:55.01 | Get_The_Fish | register |
10:55.09 | Get_The_Fish | what do you want to see |
10:56.22 | Get_The_Fish | From: <sip:john@68.164.111.XX>;tag=as3fcc5278 |
10:56.34 | Get_The_Fish | To: <sip:john@68.164.111.XX> |
10:57.16 | kaldemar | the whole message dialog. |
10:57.36 | Get_The_Fish | one sec, lemme pastebin it |
10:58.13 | kaldemar | and relevant sip configurations from both ends, secrets masked. |
10:58.22 | *** join/#asterisk ceeriael (n=ceeriael@93.167.108.90) |
10:59.16 | Get_The_Fish | k one sec |
11:00.13 | miloux | if i do sip show channels, i have one thats: x.x.x.x 3172 3c26701a8220-q7 0x0 (nothing) No Tx: NOTIFY - And that sip dev. is not recieving any calls. How do i clear it? its not listed under soft hangup |
11:00.36 | ck_28 | hi any one can help me to solve the fax t38 capability --T38(0/0/0) discard T30D from VoPP by no previous V21_Flag |
11:03.28 | Get_The_Fish | kaldemar: http://www.pastebin.ca/1422536 |
11:03.52 | Get_The_Fish | ceeriael is in here, we've been working together on this. |
11:04.00 | ceeriael | =) |
11:04.43 | ceeriael | it seems the nomater what is put in "fromdomain" it uses the target IP in the FROM field. |
11:05.12 | Get_The_Fish | I found some guy that submitted a bug to asterisk on this, then said that he solved it without mentioning how. This is a configuration issue somewhere, just not sure where |
11:06.08 | *** join/#asterisk ThoMe (i=tm@tm.muc.de) |
11:06.10 | ThoMe | hello. |
11:06.16 | ThoMe | how i can use/install the codec 729a? |
11:06.28 | kaldemar | nothing wrong with that single message. do you have a problem of some kind? |
11:06.34 | Get_The_Fish | ThoMe, you need to purchase it from digium. |
11:06.41 | ThoMe | Get_The_Fish: oh ok |
11:06.49 | ThoMe | Get_The_Fish: and a alternate? |
11:06.58 | kaldemar | and peer definitions don't affect register statements you have in the sending end. |
11:07.19 | Get_The_Fish | ThoMe: http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC |
11:07.33 | Get_The_Fish | ThoMe: no alternates will be discussed by me. |
11:08.00 | Get_The_Fish | kaldemar, ahhhhh good to know |
11:08.11 | ck_28 | no one have an idea using digium fax? |
11:08.35 | Get_The_Fish | so kaldemar, no matter what you have in the peer details for that host, it doesnt really matter in a registration attempt, is that correct? |
11:09.21 | kaldemar | in the sending end it doesn't. register messages are build based on register statements only (register => ...). |
11:09.29 | *** join/#asterisk saftsack (n=saftsack@p5792458A.dip.t-dialin.net) |
11:09.42 | Get_The_Fish | ok, well, that explains quite a bit then. |
11:09.50 | kaldemar | -build |
11:10.04 | Get_The_Fish | 1.4.22 |
11:10.34 | kaldemar | i meant i had an extra word in my sentence. :) so do you have a real issue with that? |
11:11.40 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
11:12.03 | Get_The_Fish | oh, gotcha. Well, I think that the register statement isnt configured correctly is what it is. Now that I understand the way that the registration packet is done it makes more sense, will look at the registration string. |
11:16.25 | Get_The_Fish | kaldemar, from my reading, the register string is correct here.... john:password@68.164.111.XX/john |
11:16.50 | Get_The_Fish | I cant figure out why it would use 68.164.111.xx as the from: field as well. |
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11:22.07 | kaldemar | Get_The_Fish: see rfc 3261 section 10.2 |
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11:23.03 | Phurl | hi all, i am an experienced hacker learning asterisk and vicidial for the first time. Check out my work twitter http://twitter.com/VcDlAstrsk . Currently looking for some troubleshooting help on the vicidial |
11:23.37 | phunyguy | 1337 h@x0r! |
11:23.39 | kaldemar | Get_The_Fish: it is not a misbehavior. |
11:24.56 | Phurl | phr3ak, w0r7 |
11:25.34 | phunyguy | 3xcus3 m3? w@t j00 ca11 m3? |
11:26.22 | Phurl | hahah |
11:26.57 | *** join/#asterisk DarkRift (n=dark@65.92.171.162) |
11:27.29 | Get_The_Fish | damn, you are correct....thanks kaldemar, I missed that.... |
11:28.35 | kaldemar | no problem. |
11:47.03 | *** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83) |
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12:05.23 | *** join/#asterisk bmg505 (n=leon@196-209-78-114-rndf-esr-5.dynamic.isadsl.co.za) |
12:06.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
12:08.16 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
12:10.44 | ck_28 | <PROTECTED> |
12:11.08 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
12:21.42 | Great_Anta_Baka | how do you check how long a call has been going on for from the * CLI? |
12:22.43 | *** join/#asterisk stope (n=nobody@chelmsford-cable-69-60-242-213.unitz.ca) |
12:24.55 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
12:25.34 | ck_28 | any one tried to install opt |
12:27.22 | tzafrir_laptop | ck_28, what is 'opt'? |
12:28.03 | tzafrir_laptop | Great_Anta_Baka, for starters 'channel show NAME_OF_CHANNEL' |
12:28.18 | Great_Anta_Baka | ah ty |
12:28.20 | tzafrir_laptop | which would be the specific channel and not the call, but would be good enough, I guess |
12:28.33 | *** join/#asterisk [netman] (n=netman@175.Red-79-145-182.dynamicIP.rima-tde.net) |
12:28.36 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
12:32.06 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-5346077f9fd4ff6f) |
12:32.52 | ck_28 | sorry opal |
12:33.12 | ck_28 | tzafrir_laptop http://www.voip-info.org/wiki/view/T38modem+configuration+with+Asterisk |
12:33.31 | ck_28 | requested to enable t38 modem |
12:33.43 | tzafrir_laptop | ck_28, let's start with the simple things: what version of Asterisk do you use? |
12:34.04 | ck_28 | 1.6.0.1 |
12:34.08 | ck_28 | 1.6.1.0 |
12:35.49 | ck_28 | tzafrir_laptop i installed 1.6.0.9 first then update it to 1.6.1.0 |
12:38.59 | *** join/#asterisk tobias (n=tobias@24.225.71.33) |
12:39.04 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
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12:45.51 | *** join/#asterisk thomasrr (n=scroogey@195-240-213-212.ip.telfort.nl) |
12:45.53 | thomasrr | hello |
12:48.55 | thomasrr | does anyone here have experience with voipbuaster voip-in numbers? |
12:51.38 | ck_28 | tzafrir_laptop should i continue or you are busy ? |
12:56.19 | *** join/#asterisk martyn-job (i=be18869a@gateway/web/ajax/mibbit.com/x-4504d6a8c96f778c) |
12:56.25 | martyn-job | Hi |
12:56.25 | martyn-job | every1 :D |
12:57.49 | martyn-job | I need to know if you know some application or way to do an IVR with connection to authorizing like ( VIsta, Master Card) to pay some products or service that my IVR companny want to sale across my IVR .. |
12:57.53 | martyn-job | What do you know about it ? |
12:58.20 | thomasrr | if you are getting a "number is not in service" that's a problem on voipbuster side right (got the number frmo them [voip-in]) |
12:58.46 | thomasrr | its not lke a config issue on my side in asterisk when i am calling from my mobile phone |
12:58.51 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
12:59.00 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
12:59.33 | Phurl | Unable to request channel SIP/1234 what to look for? please one tip. |
12:59.51 | thomasrr | are you sure the extension exists? |
12:59.54 | thomasrr | in extensions.* ? |
13:01.54 | SuPrSluG | Phurl: what's the output of sip show peers @ the cli? |
13:02.15 | Phurl | 1003/1003 192.168.111.67 D 5061 OK (3 ms) |
13:02.19 | Phurl | i can call them,? |
13:02.34 | Phurl | Subdolus, so the peer has to be seen? |
13:03.07 | SuPrSluG | so you don't have and extension made for 1234. you need to make one |
13:03.07 | SuPrSluG | s/and/an |
13:04.09 | Phurl | thats it, great. that is what i wanted to know. |
13:04.13 | Phurl | thanks |
13:04.29 | thomasrr | Phurl: do you know something about voupbuster voip-in numbers? |
13:04.49 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:04.54 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
13:05.11 | Phurl | thomasrr, no, but I can ask someone. what do you what to know? I am in kosovo atm |
13:05.42 | thomasrr | if the message "This number is not in service" is a problem on voipbuster end |
13:05.43 | thomasrr | or mine :) |
13:05.55 | Phurl | nopw |
13:05.55 | thomasrr | i think voipbuster because nothing in the log on my side |
13:06.03 | Phurl | cannot help with that. sorry dude. |
13:06.14 | Phurl | i assumed you googled? |
13:06.19 | thomasrr | yes |
13:06.45 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
13:08.07 | Get_The_Fish | thomasrr, make sure that the sip invite is reaching your * box. go to the asterisk CLI and type "sip set debug" |
13:08.16 | Get_The_Fish | see if you see an invite from them |
13:08.20 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
13:10.59 | thomasrr | cool our sat is getting launched now :) |
13:11.03 | thomasrr | go europe! |
13:12.22 | thomasrr | no such command 'sip set debug' |
13:12.58 | plundra | When you attach to the asterisk cli, can you enable the pretty colors you get while running asterisk in the forground? |
13:13.47 | *** join/#asterisk Gremlin (i=826f3cf2@gateway/web/ajax/mibbit.com/x-05ff357c04f676f2) |
13:13.47 | plundra | The opposite of -n would be the obvious option but I don't see anything like that. |
13:14.17 | Get_The_Fish | thomasrr, you are in the asterisk cli are you not? |
13:14.20 | Phurl | Subdolus, Unable to request channel SIP/cc10 /chan_sip.c: No such host: cc101 |
13:14.22 | [TK]D-Fender | thomasrr: "sip set debug on" |
13:14.30 | Phurl | so i need to setup the extension cc101 |
13:14.37 | Phurl | that would be in extensions.conf? |
13:15.01 | Get_The_Fish | is that a 1.6 thing? sip set debug worked for me in 1.4.23.1 |
13:15.09 | *** join/#asterisk HenrikBe (n=zapphir@h204n4fls32o954.telia.com) |
13:15.29 | ck_28 | any one can help me in my asterisk fax t38 compatability |
13:15.49 | Gremlin | Does Asterisk support DNIS reliably? |
13:15.56 | plundra | Get_The_Fish: I believe it warned about it being deprecated in 1.4, did it not? |
13:16.01 | thomasrr | now i am getting a different error hehe |
13:16.24 | HenrikBe | is there any tutorial or examples on using ajam (ajax asterisk manager) except from the short text on voip-info? |
13:16.47 | Get_The_Fish | it didnt for me |
13:16.59 | Get_The_Fish | Gremlin- yes it does |
13:17.48 | Get_The_Fish | HenrikBe, yeah there is.... google around for it, I remember seeing a several pages on it |
13:17.55 | [TK]D-Fender | Phurl: Stop with these little 1-line wonders and PASTEBIN the entire failed call. |
13:17.57 | [TK]D-Fender | ~pb |
13:17.57 | infobot | [~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 , or , http://bin.cakephp.org/ |
13:17.58 | [TK]D-Fender | ^^^^ |
13:18.17 | Phurl | [TK]D-Fender, thanks i know pb np |
13:18.50 | Get_The_Fish | henrikbe: http://www.the-asterisk-book.com/unstable/manager-interface-ajam.html |
13:19.27 | thomasrr | hmm no notify |
13:19.27 | HenrikBe | gtf: thanks! |
13:19.38 | Get_The_Fish | np |
13:19.41 | thomasrr | maybe its the bloody hipath pbx who steals it :) |
13:20.23 | Gremlin | Is it possible to get Asterisk to "patch" the call through to a certain line/whatever based on the number dialed? |
13:20.31 | Get_The_Fish | henrikbe, here's the doxygen from source on it: http://www.asterisk.org/doxygen/1.4/AstHTTP.html |
13:21.04 | [TK]D-Fender | Gremlin: You can do whatever the hell you want. Its your dialplan <- |
13:21.15 | Get_The_Fish | Gremlin, yes, it is- Freepbx will allow you to do this from the gui. No idea on the dialplan code, but you can use theirs as an example. |
13:21.56 | Phurl | [TK]D-Fender http://www.pastebin.ca/1422632 |
13:22.03 | Gremlin | Okay. |
13:23.54 | stope | I have an SLA problem, http://pastebin.ca/1422633 the phones both ring and the line lights light up and flash but the phones don't actually ring |
13:23.59 | stope | am I missing something? |
13:24.14 | stope | the caller can hear the ringing... |
13:24.31 | Gremlin | Any hope of running Asterisk on a 486 or Pentium 75 MHz, or should I look at something decent like Pentium 166 or higher? |
13:25.14 | Gremlin | I'll have up to ten simultaneous "lines" coming in from SIP with a fairly CPU intensive codec. |
13:25.25 | stope | get a more powerful machine |
13:25.26 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
13:25.29 | [TK]D-Fender | Phurl: I take it you're trying to call out via AMI Originate, correct? |
13:25.43 | [TK]D-Fender | Gremlin: NO CHANCE |
13:25.47 | Get_The_Fish | Gremlin, no |
13:25.54 | Phurl | i think so, I am using the vicidialer, let me see the command to call. |
13:25.55 | Gremlin | Celeron 1.3GHz? |
13:26.09 | [TK]D-Fender | Gremlin: Your details are still vague |
13:26.18 | Get_The_Fish | damn man, you have a 75 MHz pentium that still runs? That thing should be in a museum :) |
13:26.21 | Gremlin | Yeah, they are. |
13:26.35 | Get_The_Fish | celeron 1.3 should work. |
13:26.38 | [TK]D-Fender | Phurl: It is trying to call a DEVICE, not an "extension". |
13:26.42 | Gremlin | Okay, cool. |
13:26.58 | Phurl | good, so the call command is wrong. |
13:27.06 | Phurl | let me see what it is sending |
13:27.07 | Gremlin | Now I just need 10 USB to RJ11 ATAs all running off of two USB ports. |
13:27.11 | *** join/#asterisk Psychobilly (n=moi@adsl72-48.kln.forthnet.gr) |
13:27.12 | [TK]D-Fender | Phurl: so taht would be "SIP/1003", etc |
13:27.48 | Psychobilly | hello, how do i convert this diaplan line in ael 2 syntax: exten => 402,hint,SIP/402 |
13:30.22 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
13:32.05 | *** join/#asterisk nauticalthinker (n=mratliff@c-76-122-200-95.hsd1.tn.comcast.net) |
13:32.45 | nautical | anyone have much experience with doing an Active Directory integration for unified communications? |
13:33.21 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
13:33.43 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
13:35.01 | [TK]D-Fender | nautical: I'm sorry, could you be a little more vague please... |
13:35.49 | Get_The_Fish | nautical, yes. Be prepared to fight with Microsucks kludge of a LDAP implementation |
13:36.23 | Get_The_Fish | might want to try likewise as a client for AD, and go from there. |
13:36.44 | Psychobilly | any hints about hint syntax in ael? :> |
13:37.34 | [TK]D-Fender | Psychobilly: http://www.voip-info.org/wiki/view/Asterisk+AEL2 |
13:37.47 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
13:38.33 | SuPrSluG | <PROTECTED> |
13:39.06 | Phurl | [TK]D-Fender, thanks it looks like a configuration problem in the vicidial, the dial manger table contains 1 Channel: SIP/cc100 |
13:39.27 | [TK]D-Fender | Phurl: GUI's are not our problem. |
13:39.34 | jaytee | I have over 200 phones on my * system. My boss wants me to determine if it's feasible to have something setup where we can dial all the phones at once and broadcast a message. I don't see how without using MeetMe and that many simultaneous calls into a conference bridge on the server I think would cripple it or at least not work as hoped. Am I wrong? |
13:39.40 | Phurl | of course. thanks for your help. |
13:40.13 | Psychobilly | Subdolus i ve tried this, aelpasereturns error |
13:40.15 | Get_The_Fish | Phurl which version of asterisk did you say that you were using? |
13:40.17 | Phurl | [TK]D-Fender, thanks |
13:40.32 | Psychobilly | SuPrSluG i mean |
13:40.39 | mort_gib | jaytee: Sending text to the phones is not an option?? |
13:40.40 | [TK]D-Fender | jaytee: that's kinda rough. What I might do is add an extran conferencing server into the mix and split them up and bridge the meetmes |
13:41.01 | Phurl | Get_The_Fish, i think ubunut 1.4 |
13:41.18 | jaytee | mort_gib, nope not all of them are Polycoms, some are analog off of Linksys ATAs. |
13:41.20 | Get_The_Fish | ok, nevermind then. |
13:41.46 | Get_The_Fish | jaytee, you can do this out of the box with Freepbx, so you can look at their code to get an idea of what you need to do to get that working |
13:41.47 | jaytee | [TK]D-Fender, that's a thought. I'd use Page() anyways since it's a one-way audio setup and all the callees are muted. |
13:42.03 | Phurl | Get_The_Fish, Asterisk 1.4.21.2~dfsg-3ubuntu2 |
13:42.03 | *** join/#asterisk elred (i=sauron@fucksheep.org) |
13:42.04 | mort_gib | jaytee: Then TK's advice might be better |
13:42.15 | jaytee | mort_gib, thanks for the suggestion though! |
13:42.23 | ck_28 | any one can check this http://pastebin.com/m79a3c265 |
13:42.42 | Get_The_Fish | Phurl, a buddy of mine tried this with 1.6 and was getting something similar is why I was asking |
13:42.46 | [TK]D-Fender | jaytee: master spawn process looping out to a page-pool server. 1 real issue you'll run into is call setup lag. |
13:43.10 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
13:43.10 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) [NETSPLIT VICTIM] |
13:43.13 | [TK]D-Fender | jaytee: remembering that you'r ack-ing a LOT of calls. Sometimes can kill a few sec's to get everyone in which makes the speaker feel awkward |
13:43.38 | [TK]D-Fender | jaytee: (beep) .. uneasy pause ... "start speaking" |
13:44.13 | jaytee | we had a major power failure and right now the network engineer is in the boss's office getting reamed out because the UPS systems failed to keep up with the load and the Microsoft Operation Manager monitoring failed to send pages for the power outage |
13:44.38 | *** join/#asterisk Jacke (i=jacke@85.128.98.27) |
13:44.38 | *** join/#asterisk DaveCanoe (n=Dave@66.96.16.50) [NETSPLIT VICTIM] |
13:44.38 | *** join/#asterisk carrar (i=tim@198.136.194.10) [NETSPLIT VICTIM] |
13:44.41 | jaytee | both my IVR server and my primary * server "bounced" but came right back up online. |
13:45.23 | Get_The_Fish | MOM? Seriously???? |
13:45.24 | ck_28 | checking for gcc option to accept ANSI C... none needed which package is missing |
13:45.30 | mort_gib | So MoM didn't work... I'm shanek |
13:45.40 | Get_The_Fish | shocked |
13:45.49 | mort_gib | s/shanek/shaken |
13:46.31 | Phurl | Get_The_Fish, here is a similar problem. no answer. http://code.google.com/p/outcall/issues/detail?id=17 |
13:46.38 | mort_gib | I have trusted MS with my important data for so may year, and with only a lot of problems |
13:46.44 | jaytee | I had Nagios working great for stuff like that but all the linux bigots that run the show here decided to axe it in favor of MOM. It took me about a month to get Nagios working for all the server monitoring and stuff while doing other tasks. The Network Nitwit has taken over 6 months and still can't get it right. |
13:46.54 | [TK]D-Fender | Phurl: No... the channel you chose is not VALID |
13:47.33 | [TK]D-Fender | Phurl: Phurl there is no peer named "cc101" in your SIP SHOW PEERS |
13:47.38 | Phurl | [TK]D-Fender, yes, here is an example of a valid call that I was able to place :http://pastebin.ca/1422645 |
13:47.40 | mort_gib | jaytee: I had two "Pro IT guys" recommending Juniper Firewalls |
13:47.58 | jaytee | lol |
13:48.00 | Phurl | the problem is with the vicidial call generated |
13:48.06 | Phurl | i am now looking into it, thanks |
13:48.08 | mort_gib | jaytee: they came up with a £50.000 project to "secure" the infrastructure |
13:48.20 | [TK]D-Fender | Phurl: That code means nothing to me. |
13:48.27 | mort_gib | jaytee: I left 8 months later, they were still at it |
13:48.33 | jaytee | mort_gib, hahahaa |
13:48.34 | [TK]D-Fender | Phurl: Sho an actual failed call from *'s point of view. |
13:48.54 | mort_gib | jaytee: I did a similar setup for another client using openBSD (CARP rulez) in two weeks |
13:49.02 | [TK]D-Fender | Phurl: We do not care how broken your dialer is, only enough to prove thats it is broken |
13:49.16 | Get_The_Fish | mort and jaytee... same here, cant tell you how many times I've seen that. |
13:49.39 | jaytee | [TK]D-Fender, on the idea of a page-pool, is there some kind of wildcard for dialing all the sip peers? or would I have to code something to pull the list of registered peers and loop through them in a macro? |
13:50.30 | [TK]D-Fender | jaytee: No wildcard... you'll have to script something up or hard-code |
13:51.08 | [TK]D-Fender | jaytee: And you'll get hit with the dialplan line length limit as well. You'll probably ahve to cascade this a few levels deep. Hence the nasty setup delay |
13:51.27 | jaytee | [TK]D-Fender, that's what I thought. I'm going to look into doing a script that will pull the sip peers. |
13:52.01 | mort_gib | Get_The_Fish: In this case client wanted to aggregate ADSL lines, and have redundant Firewalls doing so, along with IPSec 4 sites infrastructure |
13:52.20 | mort_gib | Get_The_Fish: I had issues with online banking, that was it, eh sort of |
13:53.09 | Get_The_Fish | jaytee, this might help you: http://www.pastebin.ca/1422658 this is the freepbx implementation of intercom |
13:53.54 | Get_The_Fish | mort, did you use a distro for pf like pfsense or anything? I had issues with CARP, especially with SIP |
13:54.18 | [TK]D-Fender | Get_The_Fish: LOL... NO chance :) |
13:54.31 | Get_The_Fish | TK, how so? |
13:54.35 | [TK]D-Fender | Get_The_Fish: jaytee knows what he's doing, and FreePBX wasn't made to scal for this |
13:54.53 | HeMan | I'm trying to do call pickup and I've got it to work when I call internally but not when I have an incomming call |
13:54.54 | Get_The_Fish | tk, I can understand that, but it's a start :) |
13:54.59 | mort_gib | Get_The_Fish: No, I used OpenBSD vanilla, but in all honesty they don't run any "realtime" protos |
13:55.04 | Phurl | [TK]D-Fender, thank you for your advice. I think I found the problem. in the vicidial has that c101 in the "phone extension" field. I am going to change it to a know peer. |
13:55.05 | Get_The_Fish | gotcha |
13:58.35 | SuPrSluG | Phurl: are you using vicidialnow? |
13:58.43 | Phurl | Subdolus, yes |
13:59.18 | rue_mohr | I have a good one, the people at the office dont want to attend their transfers, and they want the calls tey transfer off to come back to them if nobody answers... suppose I can set a call source variable |
13:59.51 | *** join/#asterisk spck (n=spck@unioncab.com) |
14:00.04 | *** part/#asterisk spck (n=spck@unioncab.com) |
14:00.11 | *** join/#asterisk spck (n=spck@unioncab.com) |
14:00.19 | [TK]D-Fender | rue_mohr: Go read the CHANNELVARIABLES doc that came with * |
14:00.20 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
14:00.33 | rue_mohr | hmmm |
14:02.20 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
14:04.39 | *** join/#asterisk youngproguru (n=quassel@74.10.229.45) |
14:06.06 | youngproguru | neato |
14:06.25 | Phurl | [TK]D-Fender, thanks it is working now |
14:07.03 | Phurl | i changed the Phone extension: to a known peer and the astguiclient.php can now dial. twinkle gets the call. |
14:07.05 | Phurl | !!! |
14:07.06 | Phurl | thanks |
14:08.21 | *** join/#asterisk coppice (n=chatzill@119.82.66.88) |
14:12.41 | Subdolus | twitches |
14:16.39 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
14:17.43 | HeMan | I got call pickup to work with incomming calls now but transfered calls are strange to pickup |
14:24.03 | HeMan | the call has to be picked up with original destination |
14:24.04 | *** join/#asterisk trentcreek (n=kvirc@200.94.227.117) |
14:24.10 | HeMan | can that be changed in any way? |
14:29.52 | trentcreek | I am attempting to compile DAHDI, It is wanting the kernel source for 2.6.18-92.1.18.el5.028stab060.8, but the Kernel on the system is 2.6.18-128.1.10.el5-i686 |
14:30.04 | trentcreek | How can I get around this? |
14:32.05 | tzafrir_laptop | are you sure? uname -r |
14:32.15 | pmhaddad-work | does anyone have a good e911 tutorial or dialplan example? I'm still stuck on this :/ |
14:33.43 | trentcreek | tzafrir_laptop: the version it needs came up |
14:36.35 | *** join/#asterisk machoman48 (n=machoman@89.203.164.69) |
14:39.18 | *** join/#asterisk my007ms (i=master@botmaster.x86.be) |
14:39.19 | trentcreek | tzafrir_laptop: but, yum list kernel\* does not return the correct sources listed |
14:39.34 | my007ms | what is windows software i can use to convert sound file to gsm files |
14:39.50 | [TK]D-Fender | my007ms: Audacity |
14:40.19 | my007ms | http://audacity.sourceforge.net/ ? |
14:40.40 | my007ms | thanks :) [TK]D-Fender :) |
14:41.03 | [TK]D-Fender | my007ms: How many do you have to convert? |
14:41.23 | my007ms | 4 file in wav format need them to be gsm |
14:41.40 | [TK]D-Fender | my007ms: Just use * CLI for this |
14:41.46 | my007ms | i used to use sox but i am in windows and have no access to any linux box |
14:41.47 | tzafrir_laptop | trentcreek, which version came up where? What about uname -r ? |
14:42.08 | trentcreek | using yum list kernel\* |
14:42.11 | [TK]D-Fender | my007ms: You're in * and you say you have no access to any *NIX box? |
14:42.15 | [TK]D-Fender | #asterisk |
14:42.20 | trentcreek | uname -r returns correct |
14:42.25 | tzafrir_laptop | install Linux under virtualbox |
14:42.37 | my007ms | yes i mean ssh access : |
14:42.44 | my007ms | with root to install sox :) |
14:42.50 | tzafrir_laptop | problem solved |
14:43.51 | [TK]D-Fender | my007ms: Just use * to convert if * supports the base format at all |
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14:45.51 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
14:47.02 | trentcreek | [TK]D-Fender: uname -r returns the vesion it wants |
14:47.27 | *** join/#asterisk BobPierce (n=BobPierc@216.36.132.162) |
14:47.49 | *** join/#asterisk Black_L (n=chatzill@wsip-98-175-64-147.ga.at.cox.net) |
14:47.52 | Black_L | HEllo? |
14:48.48 | Black_L | Allo? |
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14:50.42 | jaytee | [TK]D-Fender, my servers restarted this morning due to a power failure. Since the restart the Polycom 550 callerid display shows the phone number and an IP address like SIP: 5146@XXX.XX.XXX.XXX. The Polycom 330's aren't experiencing this issue. I did a sip reload but that didn't change anything and rebooting the 550's doesn't either. |
14:51.12 | jaytee | I'm thinking I need to reload chan_sip.so |
14:51.19 | [TK]D-Fender | jaytee: Usually you see an IP when a call is sent from a different IP than the one its reg'd to |
14:51.52 | [TK]D-Fender | jaytee: Actually.. jsut on a different subnet <- |
14:52.07 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
14:52.37 | jaytee | [TK]D-Fender, you mean the phone has a different IP now than what is in the sip registry for that device? I'll check that, thanks. |
14:52.38 | [TK]D-Fender | jaytee: So if the phone is 192.168.10.123 and *'s interface is 192.168.11.1 then I believe you end up seeing the IP as well |
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14:58.16 | Katty | hai! |
14:58.28 | *** join/#asterisk LT (n=lt@unaffiliated/lt) |
14:58.43 | [TK]D-Fender | Katty: Mew. |
14:58.47 | [TK]D-Fender | Katty: OH HAI |
14:58.48 | Katty | hugs on [TK]D-Fender |
15:00.52 | *** part/#asterisk JenniferAkemi (n=Jennifer@76-10-182-237.dsl.teksavvy.com) |
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15:01.45 | jaytee | hugs Katty |
15:02.03 | eppigy | ^________________________^ |
15:03.06 | Katty | :>>>> |
15:03.09 | Katty | huggles jaytee |
15:03.11 | Katty | pamples eppigy |
15:03.50 | jaytee | [TK]D-Fender, figured it out. You were correct. My server has dual nics and the secondary interface is in the same VLAN as the phones but not the interface the phones register to. When the server rebooted after the power failure the eth1 interface came up automatically. I've since disabled it from coming up at startup. |
15:04.05 | [TK]D-Fender | jaytee: You're welcome |
15:04.25 | jaytee | [TK]D-Fender, oh, yeah, thank you!!! :-) |
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15:05.46 | eppigy | :D |
15:07.21 | Black_L | Hey guys |
15:07.32 | Black_L | How do i configure a dial plan to run as soon as a phone is picked up? |
15:07.47 | Black_L | I need to pick up the phone, and have it automatically dial 2 more numbers into a conference with it. |
15:08.29 | Black_L | At the same time |
15:08.41 | Black_L | And to not stop ringing those numbers for any reason other then them picking up. |
15:09.06 | sulex | I'd like to limit to 10 mins the amount of time an agent can spend with a caller entered in a queue. not the time a user spends in a queue waiting for an agent to pickup the call, but the amount of time spent talking to the agent himself... ideas? |
15:09.11 | *** join/#asterisk spck (n=spck@unioncab.com) |
15:09.42 | Black_L | If you can just program in a timer that sets off a user-event to end the call after 10 minutes. |
15:09.59 | [TK]D-Fender | sulex: "core show application dial" <- |
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15:10.57 | *** part/#asterisk davood (n=Davood@86.109.41.134) |
15:11.37 | sulex | [TK]D-Fender: I'm not using dial, I'm using queue |
15:11.40 | mmlj4 | Black_L: Answer() |
15:11.59 | Black_L | mmlj4: That doesn't tell me anything... |
15:12.06 | [TK]D-Fender | sulex: You might very well be using dial, go lookat precisely what calls your "agent" |
15:12.15 | Katty | dinged. |
15:13.11 | sulex | [TK]D-Fender: sorry but I think I did not get what you mean, can you expand it? |
15:13.32 | [TK]D-Fender | sulex: I mean go look at how the queue calls your agents |
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15:14.29 | mmlj4 | that's the first part of your project... the line will be answered immediately |
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15:14.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
15:15.00 | kaldemar | Black_L: what kind of a phone? |
15:15.27 | Black_L | Just a typical phone that plugs into the back of my x86 switch box |
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15:15.35 | sulex | [TK]D-Fender: and again... the queue command is calling the agent at the moment... not a dial application. the only timeout I can set on Queue() is the time the user will be let waiting for an agent to pickup... are you suggesting me to leave the Queue() usage for Dial() ? |
15:15.51 | kaldemar | Black_L: was it you asking the same thing yesterday too? |
15:15.56 | Black_L | Yes |
15:15.59 | Black_L | Never got it working |
15:16.09 | kaldemar | well, did you define the channel as immediate? |
15:16.31 | Black_L | Yes |
15:16.42 | kaldemar | how did it not work? |
15:16.53 | Black_L | I have no idea how to program a dial plan |
15:17.05 | Black_L | And i can't figure out where the other coder put the conf file now... |
15:17.42 | Black_L | I think i dial numbers using ext => priority, number, dial() right? |
15:17.52 | kaldemar | what exactly do you want the phone to do when someone picks it up? just join a conference? |
15:18.36 | kaldemar | "ext => priority, number, dial()" is a no. |
15:18.37 | Black_L | Call two other numbers and link them into a 3 way |
15:18.44 | Black_L | Do not stop dialing for any reason |
15:18.54 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
15:18.55 | Black_L | Ring until someone picks up, DO NOT STOP |
15:19.23 | Black_L | Even if one of the other line picks up don't stop ringing on the one that hasn't until they pick up |
15:20.52 | kaldemar | you can't do it with a simple dial then. |
15:21.16 | kaldemar | i've done that using a meetme conference and callfiles. |
15:21.32 | Black_L | Ok then |
15:21.35 | Black_L | How do i do that? |
15:21.47 | [TK]D-Fender | sulex: Again you are not looking at wht the queue is DOING. |
15:21.47 | kaldemar | make the exten put a call file for each callee to the spool directory and then put the caller in a conference. |
15:22.06 | [TK]D-Fender | sulex: pastebin a call going into the queue and look at PRECISELY how this "agent" is getting called |
15:22.08 | [TK]D-Fender | ~pb |
15:22.09 | infobot | [~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 , or , http://bin.cakephp.org/ |
15:22.11 | [TK]D-Fender | ^^^^^^^ |
15:22.18 | kaldemar | make the callfiles connect the callees to the same conference upon answer. |
15:22.34 | [TK]D-Fender | kaldemar: Yup |
15:22.38 | Black_L | kaldemar: Ok so.... any idea how i would do that? |
15:23.09 | [TK]D-Fender | Black_L: [11:21]<kaldemar>make the exten put a call file for each callee to the spool directory and then put the caller in a conference. <--- he just told you |
15:23.25 | Black_L | You don't get it |
15:23.31 | Black_L | I know NOTHING about this system... |
15:23.37 | sulex | [TK]D-Fender: I think I got you now... eureka! the point is, the queue command is using dial to call the agent... maybe if I set an absolute time out on the agent extension I get what I need? :) |
15:23.50 | Black_L | I'm assigned jobs and i'm new so i'm expected to figure it out on the go |
15:23.53 | kaldemar | Black_L: you have all the keywords now. do you have a copy of the book? |
15:23.55 | Black_L | Which is a pain in the ass |
15:24.06 | Black_L | I have Trixbox made easy now |
15:24.11 | [TK]D-Fender | Black_L: Go lookup "call files" on the WIKI, and go read the instructions for MeetMe |
15:24.25 | Black_L | Aye |
15:24.27 | [TK]D-Fender | Black_L: Thats a book I take it.. |
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15:24.37 | kaldemar | if you read up on the dialplan on the simplest level, callfiles and meetme, you should be able to figure it out pretty fast. |
15:24.41 | Black_L | Yes |
15:24.43 | Qwell | [TK]D-Fender: a book by...Kerry |
15:24.48 | mort_gib | Black_L: Hey, do you need cleenex?? |
15:24.56 | [TK]D-Fender | Black_L: Your GUI is not going to do ANYTHING for you, this is entirely custom. |
15:24.56 | Black_L | Why would i need cleenex? |
15:25.03 | Black_L | Yeah i know... |
15:25.07 | Black_L | I have Putty open |
15:25.09 | mort_gib | Black_L: To wipe your eyes |
15:25.12 | Black_L | Doing it through Putty, not the GUI |
15:25.38 | Black_L | What's the address for the Asterisk wiki? |
15:25.49 | [TK]D-Fender | Black_L: If you understand nothing about * configuration itself, you're in for plenty of extra pain just fighting your way around your GUI's crap |
15:25.53 | [TK]D-Fender | ~wikis |
15:25.54 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
15:26.10 | [TK]D-Fender | Qwell: LOL! |
15:26.23 | [TK]D-Fender | Qwell: the blank leading the blind! |
15:26.50 | Black_L | D-Fender: Don't need to tell me man |
15:27.05 | Black_L | Been working on this all yesterday. Not getting anything else done so i'm getting a little agitated. |
15:28.38 | Qwell | Black_L: why not reinstall from scratch and ignore any of the GUI stuff? Why bother working around it? |
15:29.22 | tzafrir_laptop | or rather, "freepbx", and not "gui in general. FreePBX makes a very complicated dialplan |
15:29.25 | kaldemar | Black_L: familiarize yourself with the things mentioned earlier and you'll get it. unless if you've shot yourself in the leg with a GUI, maybe. |
15:31.18 | [TK]D-Fender | Black_L: You'll need to make a custom extension/application, in there you'll need to drop in 2 call files (or issue AMI/CLI Originate commands) to have 8 call out to the 2 other parties. You'll then need dialplan to dump them into a MeetMe room upon some sort of confirmation that they accept. You'll then have to dump that call (yours) into the same room and wait for the others to join. |
15:32.06 | [TK]D-Fender | Black_L: **AND** all of the depends on *'s ability to track call-progress, etc. More work required to auto-kick people from the conference when you're done so you don't accidentally hang channels, etc. |
15:32.17 | [TK]D-Fender | fun fun fun... |
15:33.12 | Black_L | Sorry |
15:33.13 | Black_L | Had to pee |
15:33.17 | kaldemar | let him start simple, that's just plain depressing. :) |
15:33.45 | Black_L | My boss is determined to use Asterisk |
15:33.48 | Black_L | I mean Trixbox |
15:33.50 | Black_L | I have no idea why |
15:33.59 | Black_L | So i can't ditch it |
15:34.07 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
15:35.02 | [TK]D-Fender | Black_L: Well you've got plenty of * to learn for this, or go hire a consultant |
15:35.30 | [TK]D-Fender | Black_L: So why is it he needs a miracle bat-phone to initiate a 3way forced call? |
15:35.40 | Black_L | Local airport |
15:35.47 | Black_L | If a plane crashes, the bat phone shall be maned |
15:36.03 | Black_L | And god so help us if the fire and emergency departments don't pick up |
15:36.06 | [TK]D-Fender | Black_L: Holy shit, you're putting LIVES on the line with Trixbox? |
15:36.17 | [TK]D-Fender | SILLY RABBIT TRIXBOX IS FOR KIDS! |
15:36.18 | kaldemar | skims 3 seconds off of coffee latency. |
15:36.28 | Black_L | It's a backup |
15:36.31 | Black_L | It's not the main line. |
15:36.48 | Qwell | wait, what? |
15:36.51 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
15:37.07 | Black_L | ANYWAYS |
15:37.11 | Black_L | Need to get this working |
15:37.15 | Black_L | And flawlessly obviously |
15:37.20 | Qwell | Not going to happen. |
15:37.23 | Black_L | Time to burn up the search box |
15:37.36 | mort_gib | Black_L: I would opt for * rather than TrixBox |
15:37.46 | mort_gib | Or FreePBX or, or |
15:37.52 | [TK]D-Fender | Black_L: Well I jsut told you what you'll have to do and account for |
15:37.55 | Black_L | I don't know why he wants to use Trixbox so badly |
15:38.03 | Black_L | But he's the boss |
15:38.03 | Qwell | Black_L: because he's dumb. tell him no. |
15:38.08 | Black_L | So we are using Trixbox |
15:38.09 | [TK]D-Fender | trixbox isn't really so much of a problem. |
15:38.15 | [TK]D-Fender | this is jsut costom code ont he side |
15:38.21 | [TK]D-Fender | For a single task |
15:38.23 | Black_L | I can code |
15:38.27 | mort_gib | Black_L: Many bosses are like that, ask Katty |
15:38.29 | kaldemar | make your point well and you can ditch trixbox in no time. |
15:38.30 | Black_L | I just don't know how to use this stuff. |
15:38.39 | [TK]D-Fender | Black_L: You've been told the 3-4 pieces required. get to work |
15:38.44 | [TK]D-Fender | TRABAJO |
15:38.46 | Black_L | Alright alright |
15:38.52 | [TK]D-Fender | (c) eppigy |
15:38.56 | Black_L | I'm already searching documentation |
15:42.30 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
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15:47.03 | mort_gib | Is there any plugins available for monitoring asterisk from Nagios?? |
15:49.14 | *** part/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
15:49.23 | fiddur | http://www.voip-info.org/wiki/view/Asterisk+monitoring |
15:49.24 | spck | tshark -w output.cap "port 5060 or port 5061" |
15:49.29 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
15:49.30 | spck | erps |
15:56.33 | Katty | yes. |
15:56.35 | Katty | bosses dumb. |
15:56.38 | Katty | very, very dumb. |
15:57.48 | Katty | also! |
15:57.49 | Katty | lunch. |
15:57.50 | Katty | afks |
15:58.19 | spck | i ignore my phb's |
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16:10.03 | Vec | Filipe |
16:10.04 | Vec | u there |
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16:11.17 | spck | has anyone been to the opensips bootcamp? |
16:11.49 | *** join/#asterisk spck (n=spck@unioncab.com) |
16:11.53 | spck | srry wrong rom |
16:14.01 | therealcircut | hey all |
16:16.03 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
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16:19.40 | therealcircut | dead today |
16:19.49 | therealcircut | [TK]D-Fender must have meetings |
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16:27.50 | therealcircut | is it possible to playback a recording over a page without using call parking? |
16:28.40 | bmoraca | therealcircut: er...huh? what are you trying to accomplish? |
16:29.22 | [TK]D-Fender | therealcircut: sure. make sure the channel that is calling the page is doing a playback. guess what kind of channel this would be |
16:29.50 | therealcircut | basically i dial extension 779, it records a message, then it pages my phone on the desk and plays back that message |
16:30.22 | [TK]D-Fender | therealcircut: Easily doable so far |
16:30.24 | therealcircut | then it will start ringing the phone the original caller to the extension that was dialed. |
16:30.41 | therealcircut | sorry that was jibberish |
16:31.46 | therealcircut | basically, call comes in. receptionist answers and finds out who it is. Then she dials 7+persons extension which makes a recording |
16:32.13 | therealcircut | then asterisk should call that persons extension, playback the recording over a page |
16:32.26 | therealcircut | and then send the original caller to that persons extension. |
16:32.33 | therealcircut | allowing it to ring |
16:33.13 | therealcircut | and going to vm if theres no answer |
16:33.16 | therealcircut | sounds doable right? |
16:34.29 | [TK]D-Fender | therealcircut: So the callee's phone gets 1 call as a Page" over the speaker. When that ends the intended actual call starts to ring in? |
16:34.43 | therealcircut | yup |
16:35.15 | [TK]D-Fender | therealcircut: Tricky but possible |
16:36.00 | therealcircut | do i need to setup another extension for it somehow |
16:36.02 | [TK]D-Fender | therealcircut: What is the real point of this? |
16:36.13 | therealcircut | announced callerid? |
16:36.17 | therealcircut | i dunno the sheep want it |
16:36.34 | [TK]D-Fender | therealcircut: Have they considered looking at the CID on the stupid phone? |
16:37.14 | therealcircut | and risk wiplash for having to turn their heads?? |
16:37.48 | therealcircut | right now it does the playback fine, parks the call |
16:38.03 | therealcircut | but they dont like having to press the parked call button to pickup the call |
16:38.06 | [TK]D-Fender | therealcircut: When you say "recording", you mean the receptionist says "Joe from Acme Inc" for instance? |
16:38.07 | Katty | pokes head in |
16:38.14 | therealcircut | yea |
16:38.40 | stope | when the system is up and running, if we're using BLF and SLA on a hosted pbx for many clients, is that kept in memory or is it constantly read from the disk when required? |
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16:39.48 | [TK]D-Fender | therealcircut: Make your exten do teh record, then launch a call file or AMI originate to do the page with a local channel as the "Channel:" to do the playback" and end this call. On the exten that triggered the script, allow X seconds to pass (guesstimate) and then process the rest of the normal dialplan. |
16:40.11 | [TK]D-Fender | lunch BBIAB |
16:41.23 | therealcircut | dcc me some burritos |
16:41.26 | therealcircut | pls kthnx |
16:42.40 | Katty | has been watching star trek episodes on youtube |
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16:57.22 | rbd | hi guys, I have an ast 1.4 call acting as a B2BUA for G729/G711 calls....I do a loadtest where I shove 50 calls through it (1 call starting every other second, and after all 50 are established, lasting about an hour)... the CPU goes quite high (>100%) when the calls are starting, but then during the test it is at about 20% ... is this normal? |
16:57.32 | *** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83) |
16:57.50 | rbd | the dial plan has only a simple Dial() command in it to initiate the bridging |
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17:30.44 | stope | rbd: I would assume that its the call setup and teardown that causes the spike in cpu usage |
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17:33.53 | seidren | hi everyone |
17:34.31 | Katty | hi |
17:36.14 | seidren | noob here!!! could someone please enlighten me on how an outbound call center works. I need to know how many phone lines are needed to setup |
17:36.33 | jasonwoot | 42 |
17:36.42 | seidren | ha ha |
17:36.49 | seidren | for real though |
17:37.04 | seidren | is one enough.. i feel stupid to ask |
17:37.23 | Flyser | Is it correct, that I can remove all Hangups from the end of the priority chain in my extensions.conf if autofallthrough is enabled? |
17:37.54 | Flyser | Or is there a slight difference in the behaviour of these options |
17:38.03 | Katty | i think a better question is how many epople are ina call center, and will they all require a channel at the same time. |
17:39.26 | seidren | say i have 10 people making outbound calls simultaneously. do i need 10 phone numbers to make this happen or |
17:39.43 | seidren | i was rephrasing that... |
17:39.56 | Katty | stop thinking about physical lines for a moment |
17:39.59 | [TK]D-Fender | seidren: How many lines depends on your needs. |
17:40.15 | [TK]D-Fender | seidren: And separate NUMBERS from LINES |
17:40.35 | seidren | ok good.. |
17:40.35 | Katty | 10 people on the phone at the same time, means 10 channels |
17:40.39 | [TK]D-Fender | seidren: If you want 10 agents calling out, thats 10 channels right there. |
17:40.52 | seidren | ok 10 channels |
17:41.36 | Katty | how many people calling in? |
17:42.13 | seidren | calling in is unknown |
17:42.20 | seidren | but the main purpose is to call out |
17:42.23 | Katty | then you need to find that out |
17:42.30 | [TK]D-Fender | seidren: Come up with an answer for how many channels you will need |
17:42.32 | seidren | say 2 coming in |
17:42.38 | Katty | 12 channels. |
17:42.41 | seidren | 2 channels coming in |
17:42.46 | seidren | so ya total 12 |
17:42.47 | Katty | 2 phone numbers? |
17:42.49 | Katty | or 1 phone number? |
17:42.58 | seidren | 1 is good |
17:43.02 | Katty | well there you go |
17:43.10 | Katty | that's what you need |
17:44.04 | [TK]D-Fender | seidren: Call up your telco and have them quote you a T1 PRI (full 23 channel, or partial for 13 |
17:44.04 | seidren | ok.. so now to make this happen.. do i call my local telephone company and subscribe for 1 phone line ? and asterisk takes care of the rest ? |
17:44.13 | [TK]D-Fender | seidren: No |
17:44.17 | seidren | ah ya |
17:44.24 | Katty | sip 'trunks' might be an option as well |
17:44.27 | [TK]D-Fender | seidren: Asterisk does not let some boring analog line carry 500 calls at once |
17:44.56 | [TK]D-Fender | seidren: * lets you use a variety of ocnnectivity options you already posess |
17:45.15 | seidren | ok so I need a T1 PRI |
17:46.00 | [TK]D-Fender | seidren: That is one option. Using an ITSP if you have the bandwidth is another |
17:46.02 | [TK]D-Fender | ~itsp |
17:46.03 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
17:46.06 | [TK]D-Fender | ~pri |
17:46.07 | infobot | [pri] [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
17:46.09 | *** join/#asterisk michaely (n=Mike@207.114.199.107) |
17:46.54 | seidren | ok good.. thank you so much |
17:47.18 | *** part/#asterisk machoman48 (n=machoman@89.203.164.69) |
17:47.22 | seidren | i think i understand the scene. |
17:47.38 | Katty | cheers |
17:47.41 | Katty | time for margaritas then |
17:47.42 | jasonwoot | Fender is in league with the POTS providers |
17:48.01 | jasonwoot | SIP trunks or death |
17:48.12 | Katty | little red cookbook, little red cookbook |
17:48.24 | Qwell | jasonwoot: SIP "trunks" cause deaths. |
17:48.30 | coppice | Fender is in league with the devil.... oh, sorry, you said that |
17:48.32 | seidren | the T1 option seems like it needs special cards.. |
17:48.44 | jasonwoot | SIP trunks cured my lyme disease |
17:48.46 | Katty | everything needs cards. |
17:48.50 | Katty | obviously |
17:48.51 | seidren | ITSP only needs as much bandwidth as possible |
17:49.36 | seidren | because I should be able to connect computers to the asterisk box via ethernet and call through |
17:49.53 | *** part/#asterisk michaely (n=Mike@207.114.199.107) |
17:50.05 | seidren | am i correct ? |
17:50.44 | Qwell | seidren: you are. |
17:50.52 | Qwell | but then what happens when your DSL line goes down? |
17:51.04 | Katty | looks like you're not calling the police department |
17:51.10 | Katty | or the fire department |
17:51.14 | Qwell | or the ISP |
17:51.25 | jasonwoot | answer= dont get DSL |
17:51.26 | Katty | or me! |
17:51.47 | seidren | T1 wont fail then ? right ? |
17:51.56 | jasonwoot | you can have a local provider pull fiber for the cost of a T1 install/lease |
17:51.59 | Qwell | seidren: It will fail. |
17:52.07 | Qwell | But there are SLAs on them. |
17:52.12 | Katty | everything fails, at some point or another |
17:52.13 | Katty | even you. |
17:52.22 | Qwell | I don't fail. |
17:52.25 | Qwell | ever. |
17:52.32 | jasonwoot | T1's will bounce the signaling and drop the calls, with SIP trunks, there's a good chance it will recover |
17:52.51 | [TK]D-Fender | Asterisk : When failure is NOT an option (it comes bundled with the software) |
17:52.51 | *** join/#asterisk nephfl (n=none@wsip-98-175-64-147.ga.at.cox.net) |
17:52.57 | nephfl | hello, i need some dialplan help |
17:53.03 | Qwell | [TK]D-Fender: ... |
17:53.04 | Katty | i need a therapist |
17:53.12 | Katty | and maybe a visit to the spa |
17:53.35 | nephfl | I have 2 zap extenions set up for immediate=yes and they dial to a meetme conference |
17:53.40 | [TK]D-Fender | seidren: T1's are monitored circuits where the telco offers SLA for their service. It is probably the most reliable phone link you're likely to find |
17:53.48 | [TK]D-Fender | Qwell: <3 |
17:53.53 | nephfl | but i need the other extensions to ring in order to know when someone has joined the conference |
17:53.53 | jaytee | did you ever notice that the word therapist contains two separate words? the AND rapist |
17:54.21 | jasonwoot | I will venture a guess: who is Jaleel White? |
17:54.31 | seidren | thanks a lot people... |
17:55.04 | seidren | i'll probably come back here when i have more questions.. but for the moment.. i guess i have my options layed out... |
17:55.15 | nephfl | how do i get the remote extension to ring? |
17:55.20 | seidren | money will decide the rest |
17:55.22 | seidren | bye |
17:55.30 | nephfl | extensions |
17:56.20 | jasonwoot | mo money, mo problems |
17:56.20 | jaytee | hums "money changes everything..." |
17:56.37 | nephfl | could i have the first extension dial the other two and transfer them to meetme? or is there a simply command to do it? |
17:58.19 | seidren | ~itsp |
17:58.20 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
17:58.34 | seidren | ~itsplist-ca |
17:58.35 | infobot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
17:58.41 | nephfl | what i have is an emergency phone system with 2 zap extensions and an analog connection to pots... i need it so that if i pick up either extension the other rings and dials out to 911 on the pots line |
17:58.56 | *** join/#asterisk IBC_Jkenney (n=Jkenney@99.23.50.73) |
17:59.30 | IBC_Jkenney | Hello I have a problem i need for asterisk to write partial CDR record to postgres then complete record after call hangup |
18:00.01 | [TK]D-Fender | nephfl: What does the other do BESIDES jsut ringing? |
18:00.11 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
18:00.17 | bmoraca | nephfl: you should probably investigate just using a POTS line directly. asterisk + emergency = too many pieces in the link and too many points of failure |
18:00.27 | nephfl | when you pick up all 3 extensions must be in conference |
18:00.59 | nephfl | if i pick up line one 2 and pots should dial then both need to ring until someone answers and all 3 are on conference |
18:01.09 | funxion | anyone know of a reason that the asterisk app would lock up and drop its sip peers becuase of trunk deadlocks? |
18:01.45 | *** join/#asterisk Justnulling2 (n=Justnull@ool-4b7fd02a.static.optonline.net) |
18:02.46 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
18:04.20 | nephfl | any ideas |
18:04.47 | *** part/#asterisk seidren (n=chatzill@38.111.96.113) |
18:08.49 | jaytee | pony rides on commercial airplane trips |
18:10.36 | jasonwoot | funxion: using ZAP too? |
18:10.45 | funxion | yup |
18:10.48 | funxion | E1 pri |
18:10.51 | jasonwoot | PCI card? |
18:11.38 | *** join/#asterisk mweichert (n=mweicher@216.13.154.21) |
18:12.18 | jaytee | just got in two new Dell Mini laptops. we're going to use them with bar code readers at points of entry to scan tickets and membership cards. I'm gonna name them Frodo and Bilbo |
18:13.23 | mweichert | hello, I have trixbox installed, setup some extensions, and would like to customize one of the extensions by playing back an mp3. To do this, I think I would add some dialplan rules in extensions_custom.conf, correct, under the [ext-local-custom] |
18:13.27 | mweichert | context |
18:14.01 | jaytee | mweichert, wrong channel. try #trixbox |
18:15.14 | eppigy | 8[] |
18:15.24 | eppigy | TRABAJO |
18:15.26 | mweichert | jaytee, okay, thanks |
18:15.27 | eppigy | NO NECESSITO |
18:16.09 | jaytee | ¡usted ahora trabaja! |
18:16.54 | [TK]D-Fender | nephfl: You'll need to script 2 call files to have your outbound calls get dumped into a mettme with you |
18:18.16 | jplank | for a polycom, does this look off to anyone? 9,011x.t |
18:18.32 | jplank | (digit map for a polycom) |
18:22.23 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
18:25.32 | Get_The_Fish | can anyone suggest a good way to see what channels are transcoding? I am using more g729 than I think that I should, and I need to find a good way to troubleshoot this |
18:26.07 | Get_The_Fish | I'm grepping the logs for recordings, playing audio, and out of license warnings |
18:27.24 | trentcreek | What can be used to stress test the combination of the box and trunks? |
18:27.45 | trentcreek | For example I want to simulate 100 calls |
18:28.23 | Get_The_Fish | trent, look into sipp and see if that will work |
18:28.32 | *** join/#asterisk BobPierce (n=BobPierc@216.36.132.162) |
18:31.46 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
18:31.53 | trentcreek | okay, thanks |
18:32.07 | Get_The_Fish | jplank, this is my polycom digit map, it's the default if it helps: "[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" |
18:32.52 | ayeso | If I have modified the source code for app_voicmail.c, (this will sound stupid im sure) how can i compile it and install it into a current working asterisk platform that already has voicemail running? |
18:33.41 | [TK]D-Fender | jplank: Dial 9 prefix... how 1980 |
18:34.21 | [TK]D-Fender | ayeso: Just recompile & install everything, then connect to * and specifically reload that module |
18:35.47 | Get_The_Fish | I hate to ask again, but I could use help on this...can anyone suggest a good way to see what channels are transcoding? I am using more g729 than I think that I should, and I need to find a good way to troubleshoot this |
18:39.03 | *** join/#asterisk matsk (n=matkar@c-198ae253.174-6-64736c10.cust.bredbandsbolaget.se) |
18:39.15 | [TK]D-Fender | Get_The_Fish: "sip show channels" |
18:39.31 | [TK]D-Fender | Get_The_Fish: Map that to "show channels concise" |
18:40.28 | Get_The_Fish | sweet TK, thanks man... |
18:41.03 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
18:41.18 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
18:41.53 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
18:44.33 | *** join/#asterisk mmlj4 (n=jkelly@70.171.94.246) |
18:44.33 | *** join/#asterisk carrar (i=tim@198.136.194.10) |
18:50.24 | Get_The_Fish | ok, so would it makes sense that when a call is sent from the queue it would be using a transcoder? |
18:50.59 | *** join/#asterisk ingenius (n=alektro@netsolution.com.ar) |
18:51.01 | [TK]D-Fender | Get_The_Fish: Depends if all the sounds are in G729 or not |
18:51.07 | [TK]D-Fender | Get_The_Fish: Or if you're recording. |
18:52.13 | Get_The_Fish | all the sounds (that I know of) are in g729, as is the only codec accepted on the trunk, and same with all the ua's |
18:52.33 | Get_The_Fish | not recording all calls. |
18:53.26 | ruben23 | hi how am i going to setup my asterisk server to be fault tolerant or for failover.. |
18:54.32 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-79303cae626cd56f) |
18:55.01 | *** join/#asterisk stupidnic (n=foo@cpe-70-94-229-122.sw.res.rr.com) |
18:56.25 | stupidnic | Can somebody tell me how to fix "Firmware version 0 not supported by this driver contact Voicetronix to have it updated"? The card is an older TDM400 in a server that I had to replace the OS on. I used Debian Lenny, should I revert back to Etch? |
18:56.58 | *** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
18:58.28 | alrs | stupidnic: I have never used anything Voicetronix. Perhaps you could just get rid of the libvpb0 and vpb-utils packages? |
18:59.13 | stupidnic | hmm |
18:59.25 | stupidnic | if I try and remove libvpb0 it tries to remove asterisk as well |
19:00.26 | alrs | stupidnic: Does that voicetronix error actually keep anything from working? |
19:00.35 | stupidnic | Not sure |
19:00.58 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
19:01.06 | stupidnic | I do have a problem that has only occured after I installed the new system (HD failed earleri today on an otherwise perfectly working setup) |
19:01.36 | stupidnic | granted I installed asterisk from apt-get, but the main version is the same as what I was running |
19:01.55 | alrs | I'm running Lenny-packaged Asterisk on production machines and loving it |
19:02.03 | stupidnic | 1.4.18 in etch versus 1.4.22 in lenny |
19:02.13 | stupidnic | Yeah, I am not having trouble with VoIP stuff |
19:02.19 | stupidnic | just TDM/Zaptel |
19:02.23 | [TK]D-Fender | What on earth does Voicetronix have to do with a TDM400 ? |
19:02.24 | alrs | have you built the zaptel drivers? |
19:02.33 | alrs | in module-assistant? |
19:02.38 | stupidnic | alrs: yes m-a etc |
19:03.00 | stupidnic | I can see the Wildcard detecting the channels properly in dmesg |
19:03.27 | stupidnic | but for some reason when I ring the FXO line, asterisk never sees it as ringing |
19:03.47 | stupidnic | [TK]D-Fender: it was an error in the dmesg that I saw right before the wildcard loading |
19:03.53 | stupidnic | so I don't really know... |
19:04.04 | alrs | card shows up OK in zttool? |
19:04.25 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
19:04.27 | stupidnic | Yes |
19:04.30 | stupidnic | I can see it there |
19:04.31 | keith4 | my asterisk box is NAT'd. I have nat=yes for all external peers, and have never had a problem with them. currently, there is one client, also NAT'd, that can hear me fine, but I can't hear her |
19:05.01 | keith4 | sip debug shows: <--- SIP read from 195.113.65.8:9664 --->, followed by two blank lines, and <-------------------> |
19:07.09 | [TK]D-Fender | keith4: PASTEBIN |
19:07.12 | [TK]D-Fender | ^^ |
19:09.29 | keith4 | k, hold on |
19:10.40 | *** join/#asterisk thomasrr (n=scroogey@195.240.213.212) |
19:10.51 | thomasrr | hello |
19:11.06 | thomasrr | does anyone here have experience with using vopibuster voup-in numbers together with asterisk? |
19:14.22 | [TK]D-Fender | thomasrr: Is specific experience required for your actual question? |
19:14.57 | jameswf | no free lunch today :( |
19:15.06 | hardwire | anybody know of a place to send really old and outdated phones for old phone systems. |
19:15.44 | keith4 | [TK]D-Fender: http://pastebin.com/d38130527 |
19:16.17 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
19:17.24 | keith4 | [TK]D-Fender: I'm calling from 6107583228 to 6107171796, which goes Voicepulse->asterisk->sabrina (105) |
19:18.58 | [TK]D-Fender | keith4: PB your sip.conf |
19:19.17 | eppigy | hi |
19:19.24 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
19:19.47 | [TK]D-Fender | eppigy: YOU ARE DAVE! |
19:20.30 | Katty | oh |
19:20.30 | Katty | dave |
19:20.31 | eppigy | YES |
19:20.32 | Katty | hi dave |
19:20.35 | eppigy | hi Katty |
19:20.38 | eppigy | hi [TK]D-Fender |
19:20.41 | Katty | i'mw atching star trek |
19:20.44 | Katty | season 1 |
19:20.46 | eppigy | the new one? |
19:20.47 | Katty | on episode 5 now |
19:20.47 | eppigy | o |
19:20.49 | eppigy | haha |
19:20.50 | Katty | TOS |
19:21.01 | eppigy | I am at work :[ |
19:21.05 | Katty | THE ENEMY WITHIN |
19:21.06 | Katty | so am i |
19:21.09 | eppigy | :D |
19:21.15 | eppigy | I AM THE ENEMY |
19:21.25 | Katty | i don't even what to know what you're within |
19:21.28 | Katty | that's just.. eww. |
19:21.55 | eppigy | o I did not think of that |
19:21.57 | eppigy | haha |
19:22.08 | keith4 | [TK]D-Fender: http://pastebin.com/d5436f254 |
19:22.12 | eppigy | it is an older electro song |
19:22.44 | keith4 | I just had someone off-site/NAT'd set up 104 for a test call, and there were no audio problems. |
19:23.53 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
19:24.24 | nephfl | my call file is just sitting there and not processing |
19:26.57 | Katty | OH NOES the captain beamed up TWICE! |
19:27.43 | Pan3D | Katty: that's a great episode |
19:27.57 | Pan3D | the use of shadows is great |
19:28.09 | Pan3D | makes him look eeeeeeviiiil |
19:28.54 | *** join/#asterisk saftsack (n=saftsack@p5792476A.dip.t-dialin.net) |
19:43.30 | Katty | i like the use of eyeliner |
19:47.57 | *** join/#asterisk mweichert (n=mweicher@216.13.154.21) |
19:48.13 | Katty | the use of stunt doubles is awful |
19:48.39 | mweichert | is it possible to specific a SOCKS proxy in sip.conf for sip channels? |
19:48.50 | mweichert | *specify |
19:50.48 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
19:59.12 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
20:02.02 | *** join/#asterisk voxter (n=voxter@190.241.16.138) |
20:06.13 | stupidnic | On the followme app if I just provide the context that followme should use (with no |a s or n) it is still prompting me to record my name |
20:06.43 | mweichert | in a sip channel, can you specify which rtp port to use? |
20:06.58 | Qwell | stupidnic: how are you specifying the context? |
20:07.39 | stupidnic | Qwell just followme( 2001 ), but I have also tried followme( 2001| ) with the same results (this use to work on my previous version) |
20:08.00 | Qwell | that didn't answer my question |
20:08.15 | *** join/#asterisk goupil (n=goupil@2a01:e35:2f3d:7900:240:63ff:fedc:10e) |
20:08.17 | stupidnic | then your question was vague |
20:08.28 | Qwell | you said you're providing the context |
20:08.29 | Qwell | how? |
20:08.35 | *** join/#asterisk ziram19 (n=chatzill@41.226.184.105) |
20:08.47 | stupidnic | in the extensions.conf I have it specified |
20:09.02 | stupidnic | [followme] 5. Followme(2001|) |
20:09.30 | stupidnic | 2001 matches a context specifed in the followme.conf |
20:09.34 | thomasrr | i am trying to get the voipbuster voip-in number working |
20:09.34 | *** join/#asterisk Micho123 (n=mcho123@77.42.186.182) |
20:09.45 | [TK]D-Fender | keith4: Your REGISTERS break the rest of [general]. |
20:09.49 | thomasrr | only keep getting a message like "this user is currently not online try it later" |
20:09.51 | *** part/#asterisk spck (n=spck@unioncab.com) |
20:09.56 | *** join/#asterisk spck (n=spck@unioncab.com) |
20:09.58 | [TK]D-Fender | keithey have to come AFTER everything else |
20:10.01 | spck | wish i'd quit closing the channel |
20:10.03 | thomasrr | only outgoing calls work just fine the voipbuster trunk |
20:10.08 | thomasrr | what could i be doing wrong? |
20:10.13 | Micho123 | Hi all, Can a peer in sip.conf be a part of several contexts |
20:10.13 | Micho123 | ? |
20:10.22 | [TK]D-Fender | Micho123: No |
20:10.33 | spck | you could include the context of that peer in other contexts |
20:10.41 | spck | i think |
20:10.52 | ziram19 | i have a voip provider connected to asterisk 1.4 that works fine with an ivr |
20:11.16 | [TK]D-Fender | spck: Wrong scope, but I'm sure you are thinking the right goal |
20:11.47 | ziram19 | when i migrate to asterisk 1.6.1 dtmf semms not understand correctly the extension that someone entred |
20:11.47 | spck | ahh k |
20:11.49 | Micho123 | [TK]D-Fender, I'm using asteris realtime...I need to change context based on dialed number...Could this be done with goto? |
20:12.05 | [TK]D-Fender | Micho123: HUH!? |
20:12.22 | Kobaz | Micho123: you answered your own question |
20:12.28 | [TK]D-Fender | Micho123: Your context is FIXED. It does not change based on what you dial. What you dial goes to THAT context period. |
20:13.01 | Kobaz | Micho123: you need to use goto, to jump to the desired context |
20:13.11 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
20:13.21 | [TK]D-Fender | Kobaz: I wouldn't go saying that outright |
20:13.33 | Kobaz | okay well, you don't need... but you can use |
20:13.39 | Micho123 | Kobaz, and the CDRs will be saved on cdr table with the new context or with the peer context? |
20:13.55 | Kobaz | Micho123: the last context will be saved in the cdr |
20:14.06 | Micho123 | Kobaz, let's see then |
20:14.10 | Micho123 | Kobaz, thanks |
20:14.22 | Qwell | stupidnic: show me the call trace when it happens |
20:14.25 | keith4 | [TK]D-Fender: oh, crap |
20:14.26 | Qwell | ~pb |
20:14.27 | infobot | [~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 , or , http://bin.cakephp.org/ |
20:15.23 | ziram19 | there is someone that can response me? |
20:15.27 | stupidnic | Qwell: http://pastebin.com/m63b65539 |
20:15.42 | stupidnic | that is a short version but it illustrates the issue I am having |
20:15.48 | jplank | no one else is seeing that IP attempting to make calls through your PBX that I was talking about yesterday? Between yesterday and today, it has hit every one of my clients at least 5 times |
20:16.00 | stupidnic | I hung up when it prompted me to record my name |
20:16.11 | keith4 | [TK]D-Fender: after *everything* else? or after everything else in the [general] section? |
20:16.16 | [TK]D-Fender | ziram19: You haven't shown anything and we have no reason to suspect some generalized problem against 2 non-specific * versions |
20:16.21 | jcape | jplank: What IP? |
20:16.36 | Qwell | stupidnic: what version of Asterisk? |
20:16.50 | stupidnic | Asterisk 1.4.21.2~dfsg-3 |
20:16.51 | [TK]D-Fender | keith4: REGISTER's always after everything else under [general] and before any peer entry |
20:16.59 | stupidnic | Qwell: debian lenny |
20:17.02 | keith4 | [TK]D-Fender: okay, fixing |
20:17.10 | jplank | 93.190.143.10 |
20:17.19 | jplank | check out http://trixbox.org/forums/trixbox-forums/open-discussion/sip-external-hack |
20:17.22 | jplank | I'm not the only one |
20:17.24 | stupidnic | I tried finding something in the changelog |
20:17.30 | stupidnic | but nothing has jumped out at me yet |
20:17.34 | ziram19 | fender: what can i show? |
20:18.15 | jcape | Does it try to login as sip user 123? |
20:18.16 | [TK]D-Fender | jplank: That post is nonesense |
20:18.28 | jplank | more or less right |
20:18.34 | jplank | but I'm seeing the same thing |
20:18.40 | jplank | 15 different PBXs |
20:18.49 | ziram19 | when i tape 100 for example * understand XX and when i reconnect my astersik 1.4 all works fine |
20:19.03 | jplank | same IP, same originating DID, same terminating number |
20:19.05 | [TK]D-Fender | jplank: zirmProvide CONFIGS and call debug. |
20:19.19 | [TK]D-Fender | ziram19: Provide CONFIGS and call debug. |
20:20.20 | [TK]D-Fender | jplank: there is no backup provided by that post and it proves nothing. a dialplan context doesn't let people in. Its just a place calls can land if sip.conf lets them in and was configured to point there |
20:20.28 | keith4 | [TK]D-Fender: like so? http://pastebin.com/d3b6263bd |
20:20.35 | [TK]D-Fender | jplank: A dialplan error has nothing to do with that. |
20:20.47 | jplank | no, I'm not saying hes hacking anything, if you look at the sip traces he provided, they guy isn't getting through |
20:20.47 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
20:20.51 | [TK]D-Fender | keith4: Yes |
20:20.54 | iratik | Is there a way to clear the originate queue for AMI ? |
20:21.16 | jplank | I'm just finding it interesting that this same person is getting around the Internet so fast |
20:21.18 | jplank | from the same IP |
20:21.23 | [TK]D-Fender | jplank: [May 12 05:30:51] VERBOSE[30399] logger.c: -- Executing [4312297134@from-sip-external:1] NoOp("SIP/93.190.143.10-09635110", "Received incoming SIP connection from unknown pee |
20:21.25 | [TK]D-Fender | r to 4312297134") in new stack |
20:21.43 | [TK]D-Fender | jplank: this seems to say he IS allowing anonymous calls and let it slide right on through. |
20:21.57 | jplank | nah, thats how trixbox works |
20:22.07 | jplank | when its gets rejected, thats the first line in the context |
20:22.23 | jplank | very stupid, I agree, shouldn't setup the call |
20:22.27 | [TK]D-Fender | jplank: "says he doesn't allow anonymous calls", "show an anonymous call coming in" = user error |
20:22.39 | ziram19 | fender do you want an ssh access? |
20:22.41 | jplank | not user error, trixbox error |
20:23.05 | jplank | I'm just saying crazy how I seen the same attempt from on 15 different of my pbxs, from the same IP |
20:23.15 | keith4 | [TK]D-Fender: thanks, i'll have to wait to test it until tomorrow, though. that peer is 6 hours ahead of me, and probably in bed by now ;-) |
20:23.22 | Qwell | stupidnic: report a bug. bugs.digium.com |
20:23.23 | [TK]D-Fender | jplank: Which if its a bug in their interface for not doing what it was conceieved for falls flatly under the realm of "fuck the fucking GUI" :) |
20:23.30 | thomasrr | [TK]D-Fender: I am using voipbuster and requested a voip-in number |
20:23.31 | jplank | lol |
20:23.35 | jplank | agreed |
20:23.40 | [TK]D-Fender | keith4: Yes, better |
20:23.43 | jplank | thats why I stopped using trixbox a while ago |
20:23.49 | jplank | everything was ass backwards |
20:23.52 | thomasrr | but somehow i can call out with the account but not receive calls (nos ip notify msgs) from the voip-in number |
20:24.20 | jplank | I must say though, a lot of it was trixbox trying to take over freepbx (which is kind of backwards in its own right) |
20:24.33 | stupidnic | Qwell: Okay. I will build from source just in case (unless you think it has been addressed) |
20:25.14 | keith4 | [TK]D-Fender: I just noticed that I have listed "localnet=192.168.0.0/255.255.0.0" explicitly. Is this necessary? If so, should I list the other private IP ranges? (like... what if that client is NAT'd 10.x.x.x?) |
20:25.22 | Micho123 | Kobaz, Getting the following warning ...http://pastebin.com/d1dac3ea9 |
20:25.27 | Micho123 | can you check plz? |
20:25.46 | ziram19 | fender :you wan't an ssh access? |
20:25.51 | [TK]D-Fender | keith4: these are YOUR ranges, not THEIRS |
20:25.52 | Kobaz | you're sending the call to an extension/context that doesn't exist |
20:25.59 | jplank | haaaaa last two posts on that thread is great, those people allowed anonymous SIP calls |
20:26.05 | Kobaz | ziram19: no he doesn't want ssh access |
20:26.10 | jplank | one person logged 2000 calls in over an hour |
20:26.18 | keith4 | [TK]D-Fender: okay... so what if a NAT'd client is in that same range? is that a problem? |
20:26.32 | [TK]D-Fender | keith4: completely unrelated |
20:26.39 | keith4 | ok, just making sure |
20:27.14 | [TK]D-Fender | Micho123: Telling you exactly what context & exten its looking for, what more is there to say? |
20:27.43 | Micho123 | [TK]D-Fender, I think i have a priotity issue...1 sec |
20:28.28 | iratik | [TK]D-Fender: ? |
20:29.12 | [TK]D-Fender | iratik: ? |
20:29.45 | iratik | Is there a way to view the AMI originate queue? or clear it? |
20:30.18 | iratik | When you issue an Originate command, you get back "succesfully queued" . How do i manage, look at that queue? |
20:30.57 | [TK]D-Fender | iratik: No clue, and if I did I might have answered you |
20:31.08 | iratik | This must be a tough problem |
20:31.12 | iratik | thanks for your consideration though |
20:31.20 | [TK]D-Fender | iratik: Please don't hold out on the assumption I'm going to answer everyone's questions |
20:31.20 | *** join/#asterisk haryv (n=lanny@S010600a0c93f6f7e.vs.shawcable.net) |
20:31.22 | Qwell | stupidnic: no, it's a bug |
20:31.48 | [TK]D-Fender | checkout time, later all |
20:31.53 | iratik | [TK]D-Fender: I do assume you are the god of asterisk ... |
20:32.07 | Qwell | iratik: Gods are supposed to be omnipresent. |
20:32.11 | Qwell | iratik: He leaves occasionally. |
20:32.21 | iratik | You are good too i think |
20:32.49 | Kobaz | heh |
20:33.12 | iratik | Any clue why my AJAM integration seems to be launch thousands of originate requests ... e.g. , i execute a single originate command , then the next time i issue originate, its like they are doubled up ... until 256 originate commands are being launched. |
20:33.30 | iratik | I'm not launching thousands of originates, but its like asterisk is receiving them |
20:34.27 | *** join/#asterisk paulius (n=paulius@unaffiliated/paulius) |
20:35.05 | Micho123 | Can someone take a look to my extensions table and see why the GoTo is not working well? http://pastebin.com/d62a48b3b |
20:35.07 | haryv | Hiya everyone. Got a new phone well, not new but anyway, having a heck of a time for it to except the aastra.cfg image file, or its UI changes on the 9133i. Getting a display message of Network disconnected with a default time of jan1 12:00. Cli says the phone is registered as soon as it logges in then, this display info pops up. I have tried it with dhcp and static ip address. I can log into the phones UI, make the changes to the networ |
20:35.53 | haryv | Not sure if these phones have a log output that can be put on pastebin.ca |
20:36.00 | Micho123 | I'm getting Channel 'SIP/gw-in.ergatel.net-b7814c98' sent into invalid extension '028945551' in context 'On-net', but no invalid handler |
20:36.00 | Micho123 | Scheduling destruction of SIP dialog '1817550057304200033721@80.169.210.180' in 32000 ms |
20:38.00 | iratik | I figured it out... turns out firebug had reached its max log limit |
20:38.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
20:41.29 | iratik | I didn't see the 1,000s of originate requests from my client side interface |
20:41.51 | iratik | turns out i was rebinding the same element over and over again , exponentially increasing the number of requests |
20:42.34 | Katty | eppigy: episode 6!!! |
20:42.46 | Katty | eppigy: 'Mudd's Women' |
20:43.52 | eppigy | MUDDS |
20:44.33 | eppigy | I am doing bitch work :[ |
20:44.53 | Katty | :< |
20:45.41 | eppigy | yesh |
20:45.57 | eppigy | thats why I am getting out of the pbx game |
20:46.03 | eppigy | too much end user hand holding |
20:46.49 | Katty | :< |
20:47.23 | iratik | By the way guys... not sure how relevant this is.. it has to do with guitar effects and asterisk |
20:47.41 | Kobaz | hah |
20:47.47 | Kobaz | that sounds pretty relevant |
20:47.50 | iratik | You can an amazing delay effect by doing 255 chanspy requests on a channel with a guitar and conferencing the result |
20:48.02 | iratik | it sounds like hell delay distortion |
20:48.59 | iratik | Something like for (i=0, i<255) Originate( Local/23646, 61388316250 ..) where 23646 is the conference you are listening in on, and 6138... stuff is the channel to monitor thet softphone which has guitar patched through the mic |
20:48.59 | mmlj4 | have you tried adding an effects pedal to the mix? |
20:49.14 | *** join/#asterisk jrhunt (n=jrhunt@69-223-16-247.ded.ameritech.net) |
20:49.49 | iratik | no.. i just ran across it accidentally when an ajax interface rebinded the same element on each request leading to an exponential number of bindings.. .which subsequently caused an exponential number of chanspy channels to be created.. maxing out the conference |
20:51.16 | *** join/#asterisk ks3 (n=ks3@74.203.195.1) |
20:51.57 | Flyser | Hi, I have a normal consumer ISDN connection with 3 MSNs. Is it possible to set up a mailbox at ${MSN}01 and something else at ${MSN}02? If yes, can you please point me to some documentation? |
20:52.15 | ks3 | Is there a way to force ReceiveFAX to send a T.38 re-invite? Our provider is T.38 capable, but they expect us to reinvite with T.38 when fax tones are detected, and ReceiveFAX doesn't seem to be doing this. |
20:52.18 | mmlj4 | make menuslelect... what's "module embedding"? |
20:52.37 | Flyser | I think the right word for this is direct dialing in |
20:52.46 | Qwell | mmlj4: it embeds the modules into the asterisk binary, rather than making them shared objects |
20:53.05 | Qwell | mmlj4: it's not useful to most people |
20:53.33 | mmlj4 | fair enough, thanks |
20:54.16 | Katty | eppigy: cargo safely aboard. |
20:58.12 | eppigy | Katty: :>~ |
21:03.00 | Deeewayne | ks3, enable faxdetect |
21:04.33 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
21:05.29 | Katty | eppigy: the definition of attractive has changed a bit, me thinks. |
21:06.36 | ks3 | Deeewayne: I see reference to that in zaptel.conf for Zap channels... can it be used for SIP as well? Our only connection to PSTN is through a SIP provider. |
21:06.38 | *** join/#asterisk sikanrong (n=sikanron@20.104.219.87.dynamic.jazztel.es) |
21:09.38 | sikanrong | hey all, I've been reading the asterisk book and playing around with dialplans on my local net, so I'm familiar with a lot of these concepts, but I have a question: |
21:09.43 | sikanrong | <PROTECTED> |
21:09.45 | eppigy | Katty: how so? |
21:09.48 | sikanrong | exten => 1,1,Answer() |
21:09.49 | sikanrong | exten => 1,n,Dial(SIP/0034xxxxxxxxx) |
21:09.56 | Katty | eppigy: they're not very attractive, to me. |
21:10.09 | sikanrong | how do you use the ITSP for routing calls to the PSTN? |
21:10.30 | sikanrong | i guess that's my question, or do you actually use zap and route stuff to a physical FXO? |
21:10.36 | sikanrong | not sure how this stuff works quite yet |
21:11.30 | haryv | was astralink sold to another company? |
21:12.06 | haryv | got a message from my bank that another company aquired astralink andit was trying to use my credit card :) |
21:12.46 | *** part/#asterisk sikanrong (n=sikanron@20.104.219.87.dynamic.jazztel.es) |
21:12.52 | *** join/#asterisk sikanrong (n=sikanron@20.104.219.87.dynamic.jazztel.es) |
21:14.12 | Qwell | astralink? |
21:14.26 | Qwell | haryv: you mean Asterlink? |
21:14.31 | Qwell | yes, whois them. |
21:15.26 | haryv | I did |
21:16.31 | haryv | Another one bites the dust. I cancelled my acount with this new company. Now need to find another company with a 1800 service. |
21:17.08 | paulius | So I've asked this millions of times, but what providers do experts like YOU use? |
21:18.08 | stupidnic | haryv: I use voicepulse and like them, not affiliated with them in any way |
21:18.14 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
21:18.36 | haryv | voicepulse has a pay as you go 1800 service? |
21:18.47 | mmlj4 | haryv: teliax has that |
21:19.05 | stupidnic | haryv: not sure what exactly qualifies as pay as you go in your eyes though |
21:19.26 | stupidnic | they charge you for the number monthly and then any incoming calls are just billed to your account |
21:19.43 | stupidnic | although they make you keep a balance on hand to deduct from |
21:19.50 | stupidnic | like an escrow account |
21:20.06 | haryv | I really liked the idea of a 1800 service. Dialing into my asterisk box from a pay phone then dialing out really saved me a bunch of coin. |
21:20.55 | stupidnic | I will say this about voicepulse though... I was a bit annoyed when they yanked my local DIDs out from under me a few months back |
21:21.22 | [TK]D-Fender | Toll-free origination often costs a lot more than termination does. |
21:21.37 | haryv | seems asterlinks site is still up. the sales and support extentions are not ringing. |
21:21.42 | stupidnic | apparenttly the CLEC that was providing them the numbers abruptly terminated the agreement they had so all DIDs in my area were lost |
21:23.07 | haryv | who owns asterlink again? |
21:24.10 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
21:25.23 | bmoraca | haryv: probably more expensive than it's worth. lots of providers will charge you a surcharge for toll-free calls placed from payphones |
21:25.29 | haryv | kinda funny, the registered service is moved to hollywood california. Perhaps they layed off the techsupport and sales staff ;) |
21:25.44 | haryv | probebly true. |
21:26.48 | bmoraca | PacWest, for instance, charges $0.80 for each payphone call, in addition to their normal rates...not really worth it |
21:27.00 | haryv | wow |
21:27.11 | eppigy | Katty: who is not? I am so lost :< |
21:27.42 | haryv | better to have a prepaid cell phone |
21:28.03 | *** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled) |
21:28.17 | Katty | eppigy: mudd's women. |
21:28.31 | haryv | I need to knock out this aastra 9133i phone no service issue before going on to something else. |
21:33.07 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
21:33.42 | Katty | eppigy: they're blonde, and stuff |
21:33.56 | Katty | eppigy: too many sequins |
21:36.01 | haryv | katty, are you a admin? |
21:36.29 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
21:36.52 | haryv | hi jaytee, been a while ;) |
21:37.13 | jaytee | hi haryv |
21:38.03 | Katty | haryv: buwha? |
21:38.05 | Katty | hugs jaytee |
21:38.13 | jaytee | man, those new Dell Mini netbooks are so sweet. they're like Hobbit laptops |
21:38.21 | jaytee | hugs Katty |
21:38.34 | Katty | sweet like chocolate? |
21:38.47 | jaytee | sweet like real small and real light |
21:38.50 | haryv | ur still working at the zoo right ? |
21:38.54 | jaytee | yep |
21:39.32 | haryv | okay |
21:39.38 | haryv | :) |
21:39.40 | Katty | haryv: you do not parse. please try again. |
21:39.42 | jaytee | just setup one of the Minis as an admissions scanning station with a USB bar code reader and our POS (Point of Sale, Piece of Shit, same difference) software |
21:39.49 | *** join/#asterisk jdblack (n=jblack@pool-71-181-243-204.sctnpa.east.verizon.net) |
21:40.26 | haryv | I have put some of my IT work on hold. Market sucks :) |
21:40.32 | haryv | some of it at least |
21:41.31 | haryv | jaytee, so what has been happening? |
21:41.53 | jaytee | the big money maker in today's IT market is server consolidation and "virtualization" ( a word I hold in contempt due to it's constant misuse and abuse) |
21:42.05 | bmoraca | haryv: aastra phones are garbage. avoid at all costs |
21:42.06 | jaytee | haryv, same ole, same ole |
21:42.12 | Katty | jaytee: i'll virtualize you in a minute |
21:42.14 | drmessano | jaytee: Im working on virtualizing at home |
21:42.33 | drmessano | jaytee: I moved my girlfriend in, and she sleeps next to my wife now.. in our slightly larger bed |
21:42.44 | drmessano | Ive saved 30% on energy costs |
21:42.49 | jaytee | lol |
21:43.00 | stupidnic | heh |
21:43.08 | haryv | ohh brother |
21:43.34 | bmoraca | we've actually noticed an increase in sales over the last two months |
21:44.05 | Katty | woah |
21:44.05 | Katty | hey now |
21:44.44 | haryv | bmoraca, sales in what? |
21:45.00 | Katty | apparently larger beds. |
21:45.07 | bmoraca | haryv: general IT...computers, monitors, servers |
21:45.11 | *** join/#asterisk goupil (n=goupil@2a01:e35:2f3d:7900:240:63ff:fedc:10e) [NETSPLIT VICTIM] |
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21:45.30 | bmoraca | haryv: for about 6 months, everyone was in repair mode...but we're seeing a lot more sales the last two months |
21:46.12 | *** join/#asterisk Ziaeon (n=ziaeon@75-149-177-2-Miami.hfc.comcastbusiness.net) |
21:46.42 | Ziaeon | Is there a way to limit what extensions an incoming call can dial from an ivr? |
21:46.57 | Katty | just don't include it |
21:47.34 | Ziaeon | My IVR has direct dialing enabled, you dial the extension you want, as long as that extension exists on the server, it dials it. Perhaps I am going about this the wrong way? |
21:47.34 | bmoraca | Ziaeon: don't have a matching extension and use 'i' to inform user of such...like Katty said |
21:47.50 | bmoraca | Ziaeon: what is it you don't want them to dial? |
21:48.07 | Ziaeon | Lets say I only want them to reach the 1000 series extensions from the IVR direct dial, not the 2000 series. |
21:48.12 | bmoraca | Ziaeon: if you only want them to be able to dial specific extensions, you'll probably need more specific pattern matching |
21:48.28 | bmoraca | Ziaeon: then your pattern should be _1XXX instead of _XXXX |
21:48.31 | Ziaeon | So instead of direct dial, I need strict pattern matching |
21:48.43 | Ziaeon | Remember im talking about incoming calls through an ivr |
21:48.46 | Ziaeon | From, say, my cell phone. |
21:49.07 | Katty | wonders what part of don't include them isn't sinking in ;) |
21:49.12 | jaytee | Ziaeon, are you having them dial the digits at a prompt using Read() or WaitExten()? |
21:49.24 | bmoraca | Ziaeon: let's start by having you pastebin your IVR context... |
21:49.25 | Katty | applies asterisk book to Ziaeon's noggin in hopes of osmosis |
21:49.27 | bmoraca | ~pb |
21:49.28 | infobot | [~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 , or , http://bin.cakephp.org/ |
21:49.32 | bmoraca | ^^^^^ |
21:49.58 | Ziaeon | I'm inferring that this is more of a problem with the way FreePBX generates IVRs. |
21:50.04 | jaytee | yeah, pastebin that sucker and let's see what's what |
21:50.16 | jaytee | FreePBX? OMG!!! |
21:50.20 | Ziaeon | Dont whip me! |
21:50.35 | jaytee | I won't whip you, I'll just ignore you :-) |
21:50.50 | bmoraca | Ziaeon: if you're going to use a GUI, you can't complain when said GUI doesn't let you out of the box it creates |
21:50.59 | Ziaeon | Roger |
21:51.07 | Ziaeon | Ok, time to port this IVR to my strpped asterisk |
21:51.11 | bmoraca | Ziaeon: nor can you complain when you can't find your own way out of that box :) |
21:51.12 | Ziaeon | thanks guys :| |
21:51.22 | VaGoNeTaS | buddys |
21:51.29 | VaGoNeTaS | i got this error during the zaptel "make" |
21:51.30 | mmlj4 | can I concatenate variables in extensions.conf? |
21:51.31 | VaGoNeTaS | http://pastebin.ca/1423097 |
21:51.41 | VaGoNeTaS | does anybody knows why i got that error? |
21:51.49 | VaGoNeTaS | its ubuntu 9.04 server |
21:52.03 | jaytee | mmlj4, yes |
21:52.55 | VaGoNeTaS | i've installed the linux-headers-2.6.28-11-server which is my kernel version |
21:53.03 | mmlj4 | I don't see what I want in the book :-/ |
21:53.23 | jaytee | mmlj4, what are you trying to do? |
21:53.50 | [TK]D-Fender | mmlj4: Set <- |
21:53.55 | *** part/#asterisk Ziaeon (n=ziaeon@75-149-177-2-Miami.hfc.comcastbusiness.net) |
21:53.59 | VaGoNeTaS | that's the version that supports this shitty Redfone quad |
21:54.04 | VaGoNeTaS | (zaptel) |
21:54.21 | mmlj4 | I want to put several numbers into a string, separated by colons |
21:54.42 | VaGoNeTaS | http://pastebin.ca/1423097 |
21:54.50 | mmlj4 | say, "$extension:$callerid" |
21:54.57 | [TK]D-Fender | mmlj4: Set <- |
21:55.12 | mmlj4 | because AGI is choking on multiple variables |
21:55.12 | [TK]D-Fender | mmlj4: just DO IT |
21:55.25 | mmlj4 | lemme |
21:55.39 | VaGoNeTaS | tk any suggestion? |
21:56.48 | VaGoNeTaS | ls |
21:56.59 | Qwell | . .. goats/ |
21:57.12 | jaytee | lol |
21:57.22 | jaytee | pygmy goats |
21:57.42 | [TK]D-Fender | Honey Bunches of Goats |
21:57.44 | [TK]D-Fender | YUM! |
21:59.55 | jaytee | perfect example of where product testing and development suffers a communcations breakdown in a large company: the new Tums Smoothies are awesome as an antacid. The packaging they come in sucks |
21:59.56 | *** join/#asterisk phl4kx (n=supervis@webmailserver.nisira.com.pe) |
22:00.43 | VaGoNeTaS | well, THANK YOU SO MUCH guys |
22:00.44 | VaGoNeTaS | damn |
22:01.14 | jaytee | VaGoNeTaS, sorry dude but my espanol is a bit rusty for reading all that make output in your pastebin |
22:01.24 | VaGoNeTaS | crap |
22:01.26 | VaGoNeTaS | damn |
22:01.36 | VaGoNeTaS | im gonna format this shit |
22:01.53 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
22:01.55 | Qwell | how is that going to help? |
22:03.03 | jaytee | VaGoNeTaS, running such a new kernel is begging for compile issues. Jackass Jackoffalope is too damn new. Try Hardy Heron, at least it's an LTS release. Running bleeding edge distros just gives you roids, man. |
22:05.01 | VaGoNeTaS | that's the kernel tha cames with the distro |
22:05.17 | VaGoNeTaS | so your suggestion is install an older version like 8.04-server' |
22:05.20 | jaytee | no shit Sherlock but why are you running something that was released LAST MONTH? |
22:05.25 | Qwell | or install DAHDI |
22:05.48 | jaytee | has Redfone updated their crap for DAHDI yet? |
22:05.51 | VaGoNeTaS | ok im installing 8.04 server |
22:05.59 | VaGoNeTaS | Qwell |
22:06.04 | eppigy | VIRTUALIZATION |
22:06.09 | VaGoNeTaS | dahdi 2.0.0 its the only version that supports Redfone |
22:06.24 | Qwell | and? |
22:06.27 | VaGoNeTaS | the dahdi_dynamic_ethmf module |
22:06.42 | [TK]D-Fender | VaGoNeTaS: Use a more stable distro |
22:06.53 | VaGoNeTaS | i belive that is not supported by my kernel version so im installing ubuntu 8.04 server right now |
22:08.24 | jaytee | I don't care if Mark Shuttleworth went into space or not. I'm still not running a production system on anything other than RHEL or CentOS. |
22:10.08 | eppigy | YEAH SON |
22:10.17 | jaytee | Dad? is that you? |
22:10.21 | Qwell | jaytee: Has Bob Young ever been in space? |
22:10.23 | Qwell | I didn't think so. |
22:10.39 | jaytee | Bob Young? nope, don't think so. |
22:10.39 | eppigy | When he takes a good dmt hit maybe |
22:10.47 | [TK]D-Fender | VaGoNeTaS: And be caeful because Ubuntu updates kernels and tons of shit behind your back which tends to break zaptel/dahdi. |
22:10.48 | eppigy | YOU KNOW WHAT I MEAN |
22:10.56 | [TK]D-Fender | VaGoNeTaS: You are ASKING for trouble using it.... |
22:11.00 | eppigy | yes |
22:11.01 | VaGoNeTaS | hmmm |
22:11.03 | jaytee | Ubuntu, the "kernel of the week" distro |
22:11.06 | VaGoNeTaS | you say ubuntu sucks? |
22:11.08 | eppigy | use yum and exclude kernel updates |
22:11.19 | bmoraca | ubuntu's fine as a desktop OS for tinkering around |
22:11.21 | jaytee | Ubuntu uses apt not yum |
22:11.28 | eppigy | OH RLY? |
22:11.30 | jaytee | bmoraca, agreed |
22:11.32 | [TK]D-Fender | VaGoNeTaS: I'm saying having your kernel change behind your back every week will leave you fucked every week. |
22:11.40 | bmoraca | gentoo ftw |
22:11.40 | eppigy | im sayin use centos |
22:11.44 | eppigy | with yum |
22:11.56 | jaytee | eppigy, ah, that would be yum then |
22:12.00 | [TK]D-Fender | Ok, martial arts time... |
22:12.00 | eppigy | dont you dare crontradict me son |
22:12.01 | [TK]D-Fender | BBL |
22:12.04 | [TK]D-Fender | (very) |
22:12.06 | jaytee | have fun |
22:12.32 | bmoraca | i fuckin hate linux...i wonder if asterisk would compile and run under cygwin... |
22:12.40 | eppigy | wow dog |
22:12.44 | eppigy | have you lost your mind |
22:12.48 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
22:12.48 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
22:12.54 | eppigy | linux can be fun |
22:12.59 | eppigy | if you dont mind learnign something |
22:13.06 | bmoraca | only if you're a masochist |
22:13.07 | leifmadsen | Asterisk 1.4.25 Release Candidate 1 (1.4.25-rc1) is now available for testing! |
22:13.11 | eppigy | negative |
22:13.12 | mmlj4 | ok... Set(ZEXTEN={$XEXTEN}:{$YEXTEN}) seems to fail, or at least my AGI isn't getting it |
22:13.16 | eppigy | once you actually know what youre doing |
22:13.21 | eppigy | it involves little pain |
22:13.28 | leifmadsen | mmlj4: you're using variables incorrectly |
22:13.37 | mmlj4 | leifmadsen: enlighten me? |
22:13.41 | leifmadsen | ${VARIABLE} |
22:13.46 | mmlj4 | hrm |
22:13.47 | leifmadsen | not {$VARIABLE} |
22:13.47 | jaytee | bmoraca, yeah but the Windows ntoskrnl.exe in W2K3 or earlier is such a latency pig it wouldn't be worth the pain |
22:14.17 | bmoraca | jaytee: i wasn't being serious (about cygwin) |
22:14.24 | jaytee | besides, the asteriskwin32 port was originally an April Fools joke that someone got carried away with and it won't die a proper death. |
22:14.40 | bmoraca | lol |
22:14.56 | mmlj4 | that works better... but AGI still ain't being nice to me |
22:14.58 | mmlj4 | :-( |
22:15.30 | bmoraca | i learned a long time ago that timing sensitive applications + windows = fail...maybe with 2k8 Core mode (no GUI) would be acceptible |
22:15.56 | bmoraca | i mean, if I can run asterisk well enough in ESXi, it should work on top of windoe |
22:15.58 | bmoraca | ze |
22:16.07 | jaytee | bmoraca, it might, haven't played with it myself. too busy |
22:16.55 | bmoraca | same here...who has time to test things out anymore anyway? |
22:17.49 | VaGoNeTaS | its about to finish |
22:17.50 | jaytee | on the other hand, I have no issues with linux, it's a fine OS when properly configured regardless of what distro you choose. I just prefer the RHEL or CentOS releases for server applications to Debian or Ubuntu. |
22:17.56 | VaGoNeTaS | (ubuntu 8.04server) |
22:18.05 | VaGoNeTaS | its the company requirement |
22:18.09 | haryv | im4centos |
22:19.07 | eppigy | the only thing that intrigues me about ubuntu currently |
22:19.15 | eppigy | is its desktop release |
22:19.24 | eppigy | and that is comes with ext4 |
22:19.33 | eppigy | *it |
22:20.05 | haryv | Ih8flakey customers |
22:20.20 | *** join/#asterisk h3x (n=Hex@64.192.116.17) |
22:20.20 | jaytee | Canonical does a good job on pushing the envelope on the desktop. They've managed to move the desktop forward quite a bit over the last 4 years or so. |
22:20.36 | jaytee | Gnome is still a major resource pig though. |
22:20.50 | eppigy | how adre you |
22:20.51 | eppigy | dare |
22:21.09 | haryv | guy sets appointment, then calls back a few hours says he cannot make it for that time period, sets it up for the next day, then he calls back and says he cannot make it saying he is going on a road trip. |
22:21.33 | eppigy | tell him you will be in his living room until he gets back |
22:21.40 | eppigy | making yourself at home |
22:21.53 | haryv | I am going to start asking for credit card deposits of customers do that to me. |
22:22.11 | haryv | Its a loss of several hours income of no work. |
22:22.49 | haryv | I call, leave a message..and another one does not call back. Whats the point? |
22:22.49 | stupidnic | Qwell: latest trunk for 1.4 fixed the issue just in case you were wondering |
22:23.16 | jaytee | he probably wasn't, he's got enough to keep himself occupied |
22:23.19 | tzafrir_laptop | hmm... that "solution" from the list didn't work well here. they forgot to escape the ! |
22:23.40 | haryv | jaytee, ever work for your self before? |
22:24.35 | jaytee | haryv, yes I did, once upon a time but I like to eat so I had to choose between working for someone else and having food and money or working for myself and getting stiffed by customers. |
22:25.28 | haryv | thats true. Im wondering it it makes more sence just to work for a established company ;) |
22:25.55 | h3x | established companies are overrated |
22:26.07 | haryv | A well known vancouver communications company wants me to so sales for them. Will see what the meeting this saturday will unfold. |
22:26.17 | h3x | telus? lol |
22:26.22 | haryv | nooooo |
22:26.35 | jaytee | well, I'm looking at doing it again soon as I'd rather work for myself than some of the idiots I've had to work for or with lately and I've gained business experience and learned more about making a contract that a customer can't weasel their way out of. |
22:27.41 | *** join/#asterisk SaiSoma (n=SaiSoma@74.167.136.30) |
22:28.03 | h3x | FLACs are awesome. I'm going to start all over with my music collection. screw mp3 |
22:28.10 | jaytee | lol |
22:28.30 | h3x | i wish i could find a car amp that has S/PDIF inputs |
22:28.53 | jaytee | or a portable music device that plays FLAC files :-) |
22:28.59 | h3x | I think my archos does |
22:29.13 | h3x | but, its pissing me off. I'm assuming I will build a carpc |
22:29.22 | jaytee | pretty sure my SanDisk doesn't |
22:29.25 | h3x | hahahah |
22:29.28 | haryv | h3x, a telus contractor wanted me to sign a 13 page contract. and, work on sunday. I said no. |
22:29.30 | stupidnic | I seem to recall that there was a patch to the ipod replacement software that enabled FLAC play back |
22:29.31 | h3x | you should be lucky your sandisk will support mp3 |
22:29.35 | stupidnic | I forget the name of it though |
22:29.54 | h3x | haryv, telus is a work of the devil |
22:30.00 | h3x | ok fine its better than bell but |
22:30.10 | h3x | they make devil kind of money |
22:30.12 | stupidnic | yeah... RockBox supports FLAC |
22:30.31 | h3x | woah |
22:31.23 | h3x | ahah |
22:31.29 | h3x | well i have a archos 604 i think |
22:31.35 | *** join/#asterisk Alborracho (n=chatzill@190.25.135.1) |
22:31.43 | stupidnic | Yeah not supported by RockBox I don't think |
22:32.05 | h3x | argh |
22:32.16 | Alborracho | hi everyone, can someone help me with a SIP issue? |
22:32.21 | jaytee | drmessano, you still around? |
22:33.13 | h3x | MFKR.... Now they got a 3.5G archos |
22:33.25 | h3x | damn it |
22:33.28 | h3x | thats it |
22:33.48 | h3x | you know what, im not going to build a carpc. first im going to set up a openwrt router in my car |
22:34.04 | h3x | so i can go wifi from the archos and bluetooth from my tomtom, to hsdpa |
22:34.09 | drmessano | I am |
22:34.11 | h3x | its sad that i have to do it that way |
22:34.11 | stupidnic | hacks into h3x's car |
22:34.14 | jaytee | don't forget the Flux Capacitor |
22:34.16 | h3x | hahaha! |
22:34.23 | h3x | its a murano it came with a flux capacitor |
22:34.30 | stupidnic | haha |
22:34.40 | h3x | its in the shop right now |
22:34.43 | h3x | apparently the fluxing died |
22:34.46 | jaytee | drmessano, know of any mini-itx boards that are using the Intel Atom? |
22:34.53 | h3x | ran out of the 1.2 Gigawatts |
22:35.03 | drmessano | Off the top of my head, no |
22:35.16 | bmoraca | jaytee: intel makes one |
22:35.18 | stupidnic | jaytee: there aren't any in that form factor to my knowledge |
22:35.20 | jaytee | I just messed with a Dell Mini today and I was impresed |
22:35.29 | h3x | the computer died and was not reporting check engine issues |
22:35.34 | h3x | so my CVT tranny went to hell |
22:35.35 | stupidnic | plenty of micro-atx |
22:35.43 | h3x | among other things |
22:35.52 | jaytee | micro-atx is too big for what I want |
22:35.57 | Alborracho | does someone know how to modify the "From" header in sip? i need to send it like this "From:<sip:c8oqz84zk7z@privacy.org>;tag=hyh8" |
22:36.07 | h3x | albor: Edit the source code. |
22:36.14 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
22:36.21 | h3x | welcome to asterisk. hahaha. |
22:36.30 | Alborracho | any function in special? |
22:36.35 | h3x | actually isnt there some append sip header thing |
22:36.39 | jaytee | I want to build an Asterisk Appliance and use the Atom processor. I'm going to put it in a shiny lavender colored case and put Trixbox out of business. :-) |
22:36.48 | stupidnic | hmmm I wonder... it isn't clear from this bug report if this is in the 1.4 trunk or not |
22:37.11 | h3x | jaytee: screen print some strippers on the lid |
22:37.16 | Alborracho | h3x: ok thanks |
22:38.44 | bmoraca | jaytee: http://www.newegg.com/Product/Product.aspx?Item=N82E16813121383 |
22:38.53 | drmessano | jaytee: I want to come up with a PBX in a brown case... call it the "Shitty PBX"... so when the boss says "Not spending that kind of money, just get some shitty PBX" |
22:39.00 | drmessano | "YAY, they asked for it by name!" |
22:39.08 | jaytee | bmoraca, thanks dude! |
22:39.08 | leifmadsen | drmessano: lol |
22:39.09 | h3x | drmessano: Now you sound like nortel's marketing with the DMS |
22:39.13 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-5cc7e106559fd860) |
22:39.16 | h3x | ShittySwitch |
22:39.21 | jaytee | LOL |
22:39.28 | stupidnic | bmoraca: thanks for that... I wasn't even aware of that |
22:39.50 | bmoraca | there's a fair number of them: http://www.newegg.com/Product/ProductList.aspx?Submit=ENE&N=+50001157&QksAutoSuggestion=&Configurator=&Subcategory=-1&description=intel+atom&Ntk=&CFG=&SpeTabStoreType=&srchInDesc= |
22:39.55 | tzafrir_laptop | jaytee, why not use a Via board? |
22:40.00 | VaGoNeTaS | you guys were right i guess |
22:40.06 | bmoraca | more to come when nVidia's ION gets around, but those will be much more expensive |
22:40.07 | VaGoNeTaS | coz now is compiling as it should |
22:40.09 | VaGoNeTaS | the zaptel |
22:40.11 | jaytee | it's got a PCI slot so it'll take a Digium or Sangoma card! |
22:40.30 | VaGoNeTaS | i've installed libpri, libfb, fonulator |
22:40.34 | VaGoNeTaS | and now fonulator |
22:40.39 | VaGoNeTaS | i mean zaptel |
22:40.57 | bmoraca | jaytee: yeah, but the trick is finding a case that'll fit one and a PCI riser card that works properly and points the right direction. Sangoma cards are smaller than Digium cards, so I'd recommend them, personally |
22:41.09 | drmessano | jaytee: or come out with a line of cheap routers and PBXs based on open source and call them the "So, ho?" series.. because they answer the question "Aint that a cheap ass piece of crap?" |
22:41.12 | drmessano | "So, ho? |
22:41.45 | jaytee | VaGoNeTaS, glad it's compiling for ya now. |
22:41.56 | VaGoNeTaS | yeah, actually it just finished |
22:41.59 | VaGoNeTaS | now im doing the make install |
22:42.01 | VaGoNeTaS | im just testing |
22:42.13 | VaGoNeTaS | coz with dahdi-linux-2.0.0 |
22:42.14 | jaytee | bmoraca, I haven't read much on nVidia's ION. |
22:42.22 | VaGoNeTaS | i wasnt able to make calls to cellphones |
22:42.24 | stupidnic | patches the source to include a feature I he wants |
22:42.46 | bmoraca | jaytee: it uses Atom, but has a more powerful GPU component...marketed toward HTPC crowd, but way too expensive |
22:43.11 | stupidnic | Yeah I was going to say it still uses the Atom |
22:43.25 | stupidnic | we use Atom's for low power servers in the datacenter |
22:43.33 | stupidnic | damn things are miserly on power |
22:43.47 | stupidnic | .3 amps @ 120V full load |
22:44.07 | stupidnic | I have some dual quad cores that pull 2+amps @120V full load |
22:44.10 | bmoraca | stupidnic: what kind of servers? Atom is an in-order only processor...would suck dick on pretty much everything... |
22:44.54 | stupidnic | bmoraca: we mainly use it for low impact things... tftp servers, dhcp servers, dns caching servers, etc |
22:45.05 | stupidnic | things like that |
22:45.19 | drmessano | I think moving to the Google Cloud is the best thing we can do to conserve power and save the planet... Since if we screw this planet up any worse, Google has enough money to buy a new one |
22:45.20 | jaytee | the Dell Mini with an Atom runs Win XP pretty fast. |
22:45.30 | bmoraca | ESXi virtualize that shit |
22:45.41 | stupidnic | do you have any idea how much power google's DCs use? |
22:45.48 | stupidnic | I drive past one daily |
22:46.01 | jaytee | as fast as a 2.8ghz P4 just by rough "feel" |
22:46.14 | stupidnic | there are 65 1Megawatt what generators surrounding the place |
22:46.31 | stupidnic | that gives some idea of how much power they are using in the place |
22:46.46 | drmessano | "what" generators? Must be what powers the search indexes |
22:46.58 | stupidnic | heh |
22:47.23 | stupidnic | that is one DC alone sadly, there is another one across the street with even more generators |
22:47.26 | drmessano | Yahoo still uses "huh" generators |
22:47.46 | pmhaddad | lol |
22:47.50 | jaytee | and MS uses "WTF?" generators |
22:48.10 | stupidnic | google tries to hide how much their DC use resources wise by puttting them in shell companies so they don't have to list them on their SEC filings |
22:48.18 | drmessano | http://search.yahoo.com/search?p=search&fr=yfp-t-501&toggle=1&cop=mss&ei=UTF-8 |
22:48.39 | drmessano | Result #1 for the term "search" is yahoo.. 2nd is google |
22:48.40 | drmessano | Lies |
22:49.06 | bmoraca | which is interesting, because Yahoo uses google's engine now :P |
22:49.12 | stupidnic | google is ranked the number one most visted site in the world |
22:49.35 | stupidnic | I wish I had their page rank :) |
22:49.40 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
22:49.54 | drmessano | I wonder if Google IT sets google.com as everyones home page |
22:50.00 | bmoraca | stupidnic: and Al Gore has the single largest electricity bill of any one person in the US (or pretty close to it). hypocrisy ftw! |
22:50.14 | stupidnic | bmoraca: hah |
22:50.19 | pmhaddad | drmessano, google doesnt have internal it :P |
22:50.26 | KavanS | drmessano, not sure...pretty sure they maintain their own equipment |
22:50.37 | KavanS | they have a repair department, but I heard it was pretty loose |
22:50.59 | jaytee | I still love Google. If I search for a recipe for some kind of dish, say gazpacho or something I'll get like 4000 hits that are usually relevant. If I search on MSN I get, "Here's your 3000 porn sites!" "Um, I didn't want porn! I wanted a recipe!" "Sure you do! Everyone wants porn!" "No!!!!" "Connection timed out" |
22:51.13 | drmessano | How would google be any different than any other large corporation.. Sure they have an IT department.. |
22:51.42 | KavanS | drmessano, I don't think that they have standard limitations of access placed on the local machines |
22:52.01 | KavanS | but I could be wrong... |
22:52.06 | drmessano | So they don't need basic desktop support? |
22:52.12 | drmessano | Gimme a break |
22:52.23 | pmhaddad | drmessano, i was jokin |
22:52.23 | stupidnic | they use linux on the desktop |
22:52.49 | stupidnic | and no.. that wasn't a joke |
22:52.51 | stupidnic | ;) |
22:52.52 | pmhaddad | drmessano, i did have a friend that worked there last summer, and i dont think he ever mentioned IT |
22:53.05 | KavanS | yeah, I don't think they have the "standard" it dept |
22:53.08 | stupidnic | that's because nobody that works there actually does any work :) |
22:53.08 | pmhaddad | yeah |
22:53.10 | drmessano | ROFL |
22:53.12 | pmhaddad | hahaha |
22:53.13 | KavanS | haha |
22:53.13 | KavanS | yeah |
22:53.21 | KavanS | they make money I guess |
22:53.22 | pmhaddad | just 20 percent project |
22:53.40 | KavanS | yeah, the inherent problem will come down to... |
22:53.43 | KavanS | "everyone is not a genius" |
22:53.48 | stupidnic | a buddy of mine visited the campus a year or so ago... and was like "do you people ever do anything work related?" :) |
22:53.57 | KavanS | sooner or later it will happen...then you will have this excess flow of people fucking off, who really don't produce results |
22:54.16 | drmessano | So no one maintains antivirus on desktops, patching, switches, routers, cabling, basic desktop issues? |
22:54.20 | drmessano | Come on now, people |
22:54.40 | stupidnic | drmessano: they use llinux... they don't get viruses! :) |
22:54.43 | KavanS | drmessano, how many "unsavvy" people do you think google is really hiring? |
22:54.53 | rob327 | i'm sure they have network guys |
22:54.54 | KavanS | I mean you get the accountants and paper pushers... |
22:55.00 | drmessano | KavanS: They dont hire people to work on computers all day, #1 |
22:55.15 | KavanS | who knows lol |
22:55.26 | KavanS | I'll ask the guy I went to highschool with....I guess he does some shit for the gtalk dept. |
22:55.27 | stupidnic | my cousin is interning there this summer... I will find out and let you all know :) |
22:55.38 | drmessano | wow |
22:56.28 | drmessano | The thought that a company even 1/100th of their size could get by without basic IT support is even mind boggling.. Not sure why this doesnt make sense to you guys |
22:56.38 | drmessano | Of course Google has an IT dept |
22:56.46 | drmessano | As does Microsoft, and yahoo, and even Digg |
22:56.49 | KavanS | because IT support is like "check out" staff at the supermarket |
22:57.00 | KavanS | it's going to be less in demand as time goes on |
22:57.21 | KavanS | and I work in IT support |
22:57.23 | drmessano | Just because google is a tech company doesnt mean everyone works on their own fucking computers |
22:57.29 | stupidnic | KavanS: you poor bastard |
22:57.30 | bmoraca | KavanS: just because someone is a programmer does not mean they can maintain a global IP network |
22:57.40 | drmessano | bmoraca: Exactly |
22:57.43 | KavanS | bmoraca, indeed, I'm not saying they entirely maintain it |
22:57.53 | KavanS | but it's not like you are going to need the same size of IT support as you'd need 10 years ago |
22:58.03 | KavanS | at least that's my opinion |
22:58.04 | bmoraca | KavanS: in addition, the vast majority of Google's employees are sales staff and other operational people. not programmers. |
22:58.43 | KavanS | right |
22:58.54 | KavanS | I just doubt it is as large as you'd expect |
22:58.56 | bmoraca | KavanS: Google is the same as any other large company. they have a dedicated inhouse department for IT and probably another one for IS. too many hands in the pot, and all that |
22:59.00 | KavanS | I'm not saying it doesn't exist |
22:59.11 | stupidnic | I remember a job posting for google from many years ago... it specifically stated that support staff needed a solid background in desktop linux, as well as openoffice |
22:59.11 | drmessano | KavanS: I doubt its any smaller than any other company their size would have |
22:59.45 | KavanS | yeah I saw some documentary on g4tv and it showed people using their choice of OS |
23:00.22 | drmessano | Again, they have the same IT needs as everyone else and the non-IT staff, which includes the programmers and guys that make the "product" |
23:00.49 | stupidnic | man |
23:00.53 | KavanS | lol I just don't think it's the same size |
23:00.57 | stupidnic | that pico-itx ion board is TINY |
23:01.08 | KavanS | but I'll ask and find out....who knows I could be wrong |
23:01.23 | drmessano | Why are they gonna pay $75 an hour for a programmer who could be writing code, to work on his own PC when a $20 an hour Workstation specialist can be doing it? |
23:01.28 | drmessano | That makes ZERO sense |
23:01.44 | stupidnic | drmessano: we have conceeded the point... you won... move on :) |
23:01.54 | KavanS | I don't think he's 100% correct... |
23:01.57 | KavanS | lol |
23:01.58 | bmoraca | i want a job maintaining Google's network...i think that'd be all kind of fun |
23:01.59 | KavanS | sorry. |
23:01.59 | stupidnic | shhh |
23:02.06 | stupidnic | don't feed him any more :) |
23:02.09 | drmessano | KavanS: I dont think youve worked in a large company before |
23:02.17 | drmessano | KavanS: So move on |
23:02.30 | KavanS | drmessano, right...what is your definition of large? |
23:02.41 | stupidnic | bmoraca: they make their own servers and switches |
23:02.54 | drmessano | KavanS: Greater than 35,000 employees? |
23:03.15 | KavanS | nope |
23:03.34 | drmessano | What about 10,000? |
23:03.51 | bmoraca | stupidnic: they use commodity parts for their servers, but I HIGHLY doubt they make their own switches and routers. |
23:04.05 | stupidnic | bmoraca: then you would be incorrect |
23:04.09 | KavanS | drmessano, yep |
23:04.26 | bmoraca | stupidnic: source. doesn't make sense for them to rewrite what other companies have already largely perfected. |
23:04.32 | stupidnic | cost |
23:04.37 | stupidnic | 10Ge is stupid expensive |
23:04.48 | stupidnic | even from comapnies like forece10 and cisco |
23:05.23 | drmessano | bmoraca: They make their own LARGE switches.. Datacenter size stuff.. Because in the quantity they need, and demands they have, they can do that.. But not smaller stuff it would make more sense to just buy |
23:05.40 | drmessano | I dont think its even component level |
23:05.45 | KavanS | drmessano, what large firms do you work at? |
23:05.57 | alrs | Google designs their own motherboards and has Gigabyte build them |
23:06.09 | drmessano | I worked for 10 years doing IT a company with about 50,000 employees |
23:06.13 | stupidnic | their servers are CRAZY smart |
23:06.14 | drmessano | in a* |
23:06.38 | KavanS | 50,000 at an IT company? |
23:06.43 | KavanS | or IT dept. at a non-IT focused org? |
23:06.44 | stupidnic | they incorporate a 12V battery backup on the server itself with the 5v pull down built into the server's motherboard |
23:06.44 | drmessano | No |
23:06.48 | KavanS | right... |
23:06.52 | stupidnic | its brilliant |
23:06.53 | drmessano | I didnt say "AN IT COMPANY" |
23:07.03 | alrs | BITCH YOU AIN'T NO NERD |
23:07.14 | stupidnic | especially when you consider how much it would cost to install a Powerware UPS large enough to support the load of a typical DC |
23:07.35 | bmoraca | KavanS: you seem to think that Google is an IT company...it's not. It's an IS company...very, very, very different. |
23:08.01 | KavanS | lol |
23:08.15 | stupidnic | bmoraca: not "concrete" proof but pretty close even if the info is dated |
23:08.17 | stupidnic | http://gigaom.com/2007/11/18/google-making-its-own-10gig-switches/ |
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23:10.21 | bmoraca | stupidnic: that's a little different. they're not building their own switches...they're buying the components and assembling them themselves. "building" implies designing the ASIC and switching engine |
23:10.33 | stupidnic | bmoraca: symantics then |
23:10.43 | bmoraca | stupidnic: i like semantics :) |
23:11.01 | stupidnic | I don't buy components to connect my servers together... I buy Cisco :) |
23:11.06 | stupidnic | see my point? :) |
23:11.08 | drmessano | You poor bastard |
23:11.25 | stupidnic | my two 6509s disagree |
23:11.53 | drmessano | So? |
23:12.05 | bmoraca | if it wasn't so damn loud, i'd use my 7204 as my DSL router at home... |
23:12.38 | stupidnic | I could have easily have said Black Diamonds, its the same as the OS jihad, or any other comparison between differing factions |
23:12.49 | stupidnic | people like what they know |
23:13.00 | stupidnic | I prefer Cisco because I know IOS well |
23:14.31 | drmessano | Sure, but making a comment to the effect of "Look at me, I have two 6509s" doesn't offer anything to the conversation |
23:14.45 | bmoraca | i've been wanting to get my hands on some NetScreens in to see how they compare with the Cisco ASA, but haven't had time, really |
23:15.00 | stupidnic | Neither did your "you poor bastard" comment which was out of context and really pointless |
23:15.00 | drmessano | Like if you told me Linksys phone sucked and I said "My 100 SPA-941s are blinky" |
23:15.51 | drmessano | stupidnic: It wasnt out of context.. maybe you dislikes it, or didnt agree with it, but I urge you to look at the context of something being "out of context" before offering that |
23:15.58 | drmessano | disliked* |
23:16.06 | stupidnic | unless I misconctrude the intent or direction of your comment |
23:16.59 | mmlj4 | os mispselled it |
23:17.22 | stupidnic | Or I could have wasted more bandwidth with the grammer/spelling police |
23:17.27 | jaytee | I've misconstrued before but never miscontruded. I'll have to try that sometime |
23:17.31 | drmessano | grammar? |
23:18.00 | mmlj4 | stupidnic: just joshin' ya |
23:18.15 | stupidnic | mmlj4: must have missed the smiley |
23:18.35 | bmoraca | i saw no smiley |
23:18.50 | stupidnic | sarcasm is a dying art apparently |
23:19.00 | bmoraca | sarcasm doesn't work over the interwebs |
23:19.03 | bmoraca | it's incompatible |
23:19.12 | stupidnic | but its a series of tubes! it must |
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23:21.18 | KavanS | drmessano, my only point is that I think programmers (in my humble opinion) and IT personnel in general...take better care of their equipment than someone who is not as familiar with the inner workings |
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23:22.45 | drmessano | KavanS: Google is sales and programmers for the most part.. and programmers are NOT IT people and no, they don't take better care of their PCs.. Any "IT personnel" google would have are support personnel.. Their business is not IT.. |
23:23.04 | *** join/#asterisk smeegs (n=smeegs@2a01:348:153:17:211:43ff:fea3:521d) |
23:23.13 | drmessano | Unless you're 75 years old and think that all "computer people" are the same |
23:23.18 | KavanS | lol |
23:23.22 | KavanS | programmers do take better care... |
23:23.27 | KavanS | and if not, they know better |
23:23.32 | KavanS | at least the programmers I've worked with |
23:23.36 | KavanS | maybe you've spent some time at MS though... |
23:23.37 | drmessano | I can tell you for a fact they dont |
23:23.43 | drmessano | Nope |
23:25.04 | alrs | I'm with drmessano |
23:25.16 | KavanS | ok, I'm the exception then...lol |
23:25.24 | KavanS | cheers :-) |
23:25.44 | tanner | does asterisk support sccp? |
23:25.46 | alrs | "ok, to ssh to the server we're going to use putty." |
23:25.51 | Alborracho | how can i compile just one module? i dont want to recompile all asterisk |
23:26.07 | alrs | oh wait, no it was scp |
23:26.26 | alrs | moral of the story was that I had to tell the programmer five times, "no you can't double-click that, it is a commandline application" |
23:26.34 | KavanS | wtf? |
23:26.41 | KavanS | jesus christ... |
23:26.47 | KavanS | I must be the only one who works with intelligent people |
23:26.55 | KavanS | or I'm a complete idiot myself... |
23:26.56 | drmessano | They not stupid |
23:26.58 | KavanS | must be the latter |
23:27.00 | drmessano | GRRR |
23:27.03 | drmessano | They're not stupid |
23:27.12 | drmessano | They're very intelligent |
23:27.16 | drmessano | Write good code |
23:27.38 | drmessano | But CAN'T fix workstation issues, or program a switch, or a router, or make a patch cable, or trace a drop |
23:27.43 | drmessano | etc etc |
23:27.57 | KavanS | I don't think that's absolute by any means |
23:28.27 | alrs | a lot of programmers are tool-users |
23:28.41 | alrs | they melt down when you take away their IDE or GUI db client of choice |
23:29.20 | tanner | cat + redirection FTW |
23:29.32 | stupidnic | Alborracho: did you compile from source? |
23:30.38 | Alborracho | i compiled and installed asterisk from sources |
23:30.48 | Alborracho | but i made a change in cha_sip.c |
23:30.49 | paulius | Bad programmers are tool users. |
23:30.55 | paulius | And some programmers are just plain tools. |
23:30.55 | Alborracho | i dont want to recompile everything |
23:30.56 | stupidnic | Okay... then just run make again |
23:31.11 | stupidnic | as long as you don't do a make clean it will only recompile that one app |
23:31.13 | Alborracho | just the module and the do a cp |
23:31.21 | Alborracho | ohh |
23:31.23 | Alborracho | nice |
23:31.26 | Alborracho | thanks |
23:31.33 | stupidnic | I just did this 5 minutes ago |
23:31.55 | stupidnic | I did do a make install though because my module as compiled into asterisk (followme) |
23:32.04 | stupidnic | s/as/is |
23:32.24 | stupidnic | looks at mmlj4 |
23:44.17 | *** join/#asterisk propellerhead (n=yogurt2u@host1.190-30-31.telecom.net.ar) |
23:48.28 | mmlj4 | ? |
23:48.58 | *** join/#asterisk aces1up (n=signup@ip70-173-52-152.lv.lv.cox.net) |
23:49.04 | mmlj4 | stupidnic: joke over, time to get some work done |
23:49.08 | aces1up | what a good brand name for ata's? |
23:49.26 | mmlj4 | depends |
23:49.34 | mmlj4 | you want to provide dialtone, or accept it? |
23:49.58 | stupidnic | sweet... the patch works |
23:50.10 | aces1up | just need something that gets good voice :) |
23:50.14 | aces1up | just using for a home net. |
23:50.26 | aces1up | but want it to be reliable, want to plug a cordless phone into it. |
23:50.40 | aces1up | unless you know any cordless voip phones that are good for less than 100.00 |
23:50.50 | mmlj4 | aces1up: are you plugging phones into the ATA? or are you pluggin the ATA into the wall to get AT&T's phone service? |
23:51.01 | aces1up | pluggin phones into ata. |
23:51.02 | mmlj4 | ok, fine... linksys |
23:51.19 | aces1up | linksys? |
23:51.31 | aces1up | hrmm. ok but haven't had much luck with their routers :) |