| 00:00.12 | zigman | do you want to talk about that.. i'd love to listen |
| 00:00.40 | Nethab | she's not here, that was for effect |
| 00:00.47 | sudoer | hermie, so can polycom phones not use the consultative transfer method? |
| 00:00.56 | Nethab | assisted transfer? |
| 00:01.00 | eKo1 | Has anybody done a 'select distinct(disposition) from cdr' on their cdr table and noticed some funky values in there? |
| 00:01.04 | hermie | sudoer: no |
| 00:01.21 | *** join/#asterisk LUTOR_ASI (~newbie@icable-docsis2-108.ibw.com.ni) |
| 00:01.24 | sudoer | while im talking to someone, i just want to dial 101(extension) and press transfer and thats it |
| 00:01.27 | drumkilla | eKo1: there was a change regarding that recently |
| 00:01.42 | drumkilla | docs indicated it was supposed to be a string, but it was actually using ints |
| 00:01.47 | drumkilla | so it was fixed to use strings |
| 00:01.49 | Nethab | i don't know if that's possible |
| 00:01.55 | sudoer | the old people are having a real hard time pressing 'trasnfer,101,dial,transfer' to send a call |
| 00:01.56 | drumkilla | in stable, it still uses ints |
| 00:02.11 | Juggie | when i download a cvs tarball from sf all the files have ,v at the end, theres a command to fix that, anyone know off hand, i forget... |
| 00:02.12 | drumkilla | eKo1: this was all for cdr_odbc by the way |
| 00:02.16 | *** join/#asterisk Corydon-w (~tilghman@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 00:02.17 | redder86 | hrmmph, this is very good http://paulgraham.com/hs.html |
| 00:02.20 | sudoer | which is kind of a pain in the ass |
| 00:02.22 | zigman | eKo1 NO ANWSER, BUSY, FAILED and ANSWERED |
| 00:02.31 | eKo1 | drumkilla: So you're saying that these weird values I'm seeing is because fo the that? |
| 00:02.37 | LUTOR_ASI | Somebody knows how to choose a specific codec when dialing an extension in the dialplan..? |
| 00:02.39 | eKo1 | drumkilla: I posted that bug by the way. |
| 00:02.54 | drumkilla | eKo1: ha. so you knew all that. sorry :) |
| 00:03.09 | drumkilla | I don't know if that's it or not, I was just making sure you knew about the change |
| 00:03.18 | eKo1 | So if I change the column to an INT, it should be fine right? |
| 00:03.22 | sudoer | hermia, thats how transfers are on your phones also? |
| 00:03.36 | *** join/#asterisk HjemmeRoyK (~roy@110.80-203-29.nextgentel.com) |
| 00:04.10 | Nethab | yes my phones you hit transfer, then dial the number, then transfer again |
| 00:04.26 | xkev | dk, do you comprehend translate.c? |
| 00:04.33 | sudoer | Nethab, you dont find that as horrible? |
| 00:04.47 | ManxPower | LUTOR_ASI, "show applications" and README.variables |
| 00:04.50 | Nethab | that way if the other party doesn't answer you don't push transfer and don't lose then |
| 00:05.11 | Nethab | no that's the way my nortel phones have worked for years |
| 00:05.15 | eKo1 | The values I have in there are 1,2,4,1092636494,1465077313,1498633538. I don't see an 8 anywhere so I guess those large values correspond to an 8? |
| 00:05.35 | Grooby | grrrrrrr |
| 00:05.44 | Grooby | i want FIOS to my house |
| 00:05.47 | Grooby | hehehe |
| 00:05.47 | *** join/#asterisk Lethol (~lethol@201.128.129.125) |
| 00:05.54 | ardor | whats FIOS? |
| 00:06.03 | Grooby | verizon's fiber on something something |
| 00:06.05 | Nethab | sudoer: it's the same with conference, you push it, then dial, then when the other person answers you push conference again to join all three parties |
| 00:06.06 | Grooby | basically fiber to your house |
| 00:06.08 | tzanger | Fiber In Oriface Service |
| 00:06.09 | ardor | First IN out Second |
| 00:06.10 | Grooby | 15mb down and 2mb up |
| 00:06.21 | xkev | drumkilla, I'm having some issues redirecting calls, and it seems ast_translator_build_path() is doing something silly |
| 00:06.22 | Grooby | then i can get all my family onto my * |
| 00:06.24 | sudoer | they are having a very hard time with conferences |
| 00:06.26 | Grooby | no need for long distance |
| 00:06.28 | Nethab | sudoer: otherwise you conference in a ringing tone, and that's annoying |
| 00:06.28 | Grooby | tee hee hee |
| 00:06.30 | Killohurtz | how come I keep getting "Unable to request echo training on channel 1"???? |
| 00:06.31 | eKo1 | Isn't fiber to your house called FTTC |
| 00:06.48 | Nethab | fttp fiber to the premises |
| 00:06.55 | Grooby | no clue |
| 00:07.00 | eKo1 | fiber to the curb |
| 00:07.05 | Grooby | i am just quoting verizon |
| 00:07.11 | Nethab | if it's to the curb then it's not to your house |
| 00:07.12 | xkev | fttc ftth fttp, all the same thing basically |
| 00:07.12 | Lethol | i know i shouldnt be asking here.. but can someone set me up with a 7905 sip firmware image? |
| 00:07.18 | Nethab | hehe |
| 00:07.52 | Grooby | hehehe |
| 00:08.00 | Grooby | i just pull/tug it a little |
| 00:08.04 | Grooby | and it'll be in my house |
| 00:08.05 | Grooby | :-D |
| 00:08.43 | *** join/#asterisk pimpwell (~pimpwell@ool-44c6aaba.dyn.optonline.net) |
| 00:08.54 | Nethab | some of us only have FTTN fiber to the neighbors |
| 00:09.14 | greg_work | Grooby: this is supposed to be a family channel |
| 00:09.25 | Grooby | blink blink |
| 00:09.33 | Nethab | pull it tug it, get it? |
| 00:09.35 | Grooby | Greg, get your head out of the gutter because i have no clue what you are thinking |
| 00:09.36 | Grooby | :-D |
| 00:09.56 | greg_work | hey, i just looked in here and saw that line :p |
| 00:09.57 | sudoer | some of us wish we lived in japan where 100 mb fiber to your house is a option for everyone and costs less than cable |
| 00:10.01 | Grooby | hahahaha |
| 00:10.20 | Grooby | and all the hentais you can get? |
| 00:10.25 | Grooby | i mean gadgets |
| 00:10.27 | Grooby | not hentais |
| 00:11.52 | eKo1 | hentai!? where? |
| 00:12.11 | eKo1 | I get fiber to my office. |
| 00:12.31 | Grooby | roflmao |
| 00:12.36 | eKo1 | It's under-used though (only 2 Mbps). |
| 00:12.39 | Grooby | fiber, it does the body good |
| 00:12.43 | freat | I've got fiber at home for networking... cause I need it |
| 00:13.15 | Nethab | i've got fiber in my stereo system, but that's kinda a waste of bandwidth |
| 00:13.17 | eKo1 | I checked out the cable. The actually fiber makes up only 2 percent of the whole damn cable. |
| 00:13.25 | eKo1 | s/actually/actual |
| 00:13.27 | freat | problem is my drive speed ends up being a bottlenexk |
| 00:13.39 | sudoer | scsi |
| 00:13.48 | Grooby | the only fiber I have are the oranges I eat in the morning |
| 00:13.52 | wankel | scsi rarely provides better performance than modern ATA drives |
| 00:14.03 | freat | yeah but unless you've got fast drives on both ends of the connection it doesn't do you any good |
| 00:14.10 | Nethab | or raid |
| 00:14.22 | Nethab | not to be confused with black flag |
| 00:14.24 | wankel | ide raid is definitely the way to go. cheap, fast as hell. |
| 00:14.25 | firestrm | <PROTECTED> |
| 00:14.26 | freat | I run RAID-0 on two WD Raptors and they are still too slow |
| 00:14.35 | ||Smuggs | yo ardor u said to use fc2 because fc3 has troubles. i just start asterisk and i'm sitting steady at a CLI> prompt? "Asterisk Ready" Im sure that means no problems |
| 00:14.37 | wankel | freat: what kind of controller? |
| 00:14.44 | freat | onboard |
| 00:14.47 | wankel | promise? |
| 00:14.48 | Nethab | the ports are working, but your client sent a bad request |
| 00:14.50 | freat | no way |
| 00:14.51 | eKo1 | Are you talking about fibre now? |
| 00:14.54 | wankel | highpoint? |
| 00:15.03 | LUTOR_ASI | ManxPower: can you explain me what i have to do with show applications, excuse me for my ignorance but i'm new to asterisk. |
| 00:15.04 | freat | it's the intel chipset that is off the southbridge |
| 00:15.04 | Grooby | check your registry string |
| 00:15.10 | Grooby | and your proxy= |
| 00:15.14 | freat | so you're not bottlenecked by PCI bus |
| 00:15.23 | wankel | bottlenecked by the PCI bus? |
| 00:15.27 | freat | heck yeah |
| 00:15.28 | LUTOR_ASI | ManxPower: could you please give me an example.. |
| 00:16.03 | freat | wankel: 32 bit pci bus will max you out at around 130 Megs / second |
| 00:16.20 | wankel | 32-bit, sure. wtf uses 32-bit pci for storage? |
| 00:16.41 | eKo1 | Aha! Asterisk is storing a BFN instead of 8. I knew it. |
| 00:16.55 | freat | well, my home box ain't a 64 bit pci bus... nor have I updated to PCIX |
| 00:16.58 | tzanger | a BFN? Big Fuckin Number? |
| 00:17.02 | eKo1 | Yes. |
| 00:17.23 | firestrm | Grooby, the proxy= is a new one to me.. Ive check rechecked, and checked again the registry string.. its verbatum what terracall says it should be. |
| 00:17.26 | tzanger | I had a call number of -5834191425 once... it crashed out shortly thereafter :-) |
| 00:17.38 | wankel | freat: that sucks. be better off with hardware raid off 64-bit pci :) |
| 00:17.39 | eKo1 | Looks like I'll be up late tonight. |
| 00:17.42 | tzanger | well a 64 bit NULL is pretty fuckin big :-) |
| 00:18.11 | Grooby | oops |
| 00:18.12 | Grooby | i mean host= |
| 00:18.13 | Grooby | sorry |
| 00:18.17 | Grooby | i am retarded |
| 00:18.17 | Juggie | heh, theres a contract comming up soon for 175,000 voip phones from within the CDN gov, i wonder if anyone with an asterisk soulution has the balls to apply. |
| 00:18.19 | freat | wankel: sure, gimme some $$ and I'll build it in a heartbeat. 3 ware controller... whole bunch of WD Raptors... mmm |
| 00:18.34 | Mother_ | and why not? |
| 00:18.57 | firestrm | Grooby, thats ok.. i feel the same way right now.. |
| 00:19.14 | Juggie | does any company exist that can support asterisk & sip at that lvl |
| 00:19.57 | eKo1 | 1v1? |
| 00:19.58 | firestrm | Grooby, that one is also exactly as terracall states it should be. |
| 00:20.00 | buddah | that is a shit load |
| 00:20.06 | *** join/#asterisk exonic (~exonic@209.172.7.134) |
| 00:20.07 | Torq | need IAX termination in manila,phillipines. anybody?? |
| 00:20.13 | wankel | unfortunately, storage performance is getting hard to improve. the only thing that's caused storage performance to improve lately is huge increases in bit density. that can't continue for long, though. spindle performance hasn't been improved much at all lately. |
| 00:20.24 | ardor | i know VOCAL scales pretty good. I like asterisk. |
| 00:20.25 | Juggie | eKo1, not 1v1, lvl aka level |
| 00:20.26 | sudoer | skype |
| 00:20.30 | wankel | so we'll just have to keep stacking on more heads and spindles 'til something better comes along. |
| 00:21.04 | eKo1 | HD technology is bound to change soon with all the flash memory everywhere. |
| 00:21.18 | wankel | flash is slow as balls, though. |
| 00:21.25 | *** join/#asterisk johngalt (1000@dsl081-088-238.lax1.dsl.speakeasy.net) |
| 00:21.30 | eKo1 | For writing only. |
| 00:21.35 | eKo1 | Reading is snappy. |
| 00:21.47 | wankel | and for reading, too, unless you're doing sequential reads. |
| 00:21.57 | firestrm | eKo1, nand flash backed ram drives.. at least thats what is being pushed lately in embedded circles |
| 00:22.28 | eKo1 | Actually the future of HD technology is in AFM. |
| 00:22.43 | sudoer | Nethab, can i see a sample of one of your dialplans for polycom phones? |
| 00:22.48 | wankel | firestrm: yow... well, that'll work for stuff that tends to rewrite the same stuff a lot, where the write-back will really help. |
| 00:23.06 | wankel | three-d holographic opto-magneto-tronic memory! |
| 00:23.08 | exonic | Hey guys |
| 00:23.18 | johngalt | could someone please call my fwd nbr - 577298, I seem to be having problems getting this setup |
| 00:23.18 | BoRiS | Juggie: Where? |
| 00:23.32 | eKo1 | AFM is more promising than holography. |
| 00:23.43 | wankel | 'twas just joking. |
| 00:23.44 | Nethab | sudoer: you mean to dial one? |
| 00:23.51 | wankel | i've been waiting for holographic memory for decades :) |
| 00:24.00 | Grooby | firestrm you see my msg? |
| 00:24.08 | exonic | Anyone in here familiar with festival? I would like to have asterisk stream festival data on the fly? Is there a way to exec Festival() and have it sent to the open channel? |
| 00:24.14 | wankel | the latest goodies from the UK will probably have expired patents by the time they commercialize it |
| 00:24.15 | firestrm | Grooby, just noticed |
| 00:24.54 | *** join/#asterisk rodmaez (~rodmaez@netblock-66-245-225-106.dslextreme.com) |
| 00:25.02 | ||Smuggs | can some1 give me their FWD so I can test this |
| 00:25.33 | sudoer | Nethab, on some of the phones here i cant transfer to 101, but others i can, im not sure which part of config that is |
| 00:25.51 | Mother_ | is the recommended kernel still 2.4 for production? |
| 00:25.53 | eKo1 | Yes. |
| 00:26.12 | LUTOR_ASI | Somebody has dial out from X-PRO to PSTN succesfully? Please i'm desesperate.. i don't know what to to..! |
| 00:26.15 | Mother_ | thanks - I got * running on SuSE 9.2 |
| 00:26.18 | eKo1 | But I'm using 2.6 in a production environment. |
| 00:26.30 | Mother_ | are you finding it stable? |
| 00:26.44 | sudoer | the phones are all using the default configs they had |
| 00:26.47 | Mother_ | I've been running tests and so far it works quite good |
| 00:26.52 | Nethab | sudoer: you mean the dialplan inside the polycom phone? |
| 00:27.05 | wankel | eko: hmm. is there any hope for reasonable write speeds with AFM? |
| 00:27.12 | sudoer | Nethad yes |
| 00:27.15 | sudoer | Nethab yes |
| 00:27.16 | Chuji | Hmm, what would make bridging a Sip to Zap channel work, but a Zap to Zap not? |
| 00:27.23 | Nethab | i've never had to change mine |
| 00:27.30 | Nethab | mine are the default too |
| 00:27.32 | Juggie | Chuji, it should |
| 00:27.49 | Chuji | Juggie : Yeah, indeed |
| 00:28.06 | Juggie | whats the msg? |
| 00:28.27 | Chuji | Juggie : I just get an immediate hangup |
| 00:28.40 | Juggie | what card are you using for your zap lines? |
| 00:28.49 | Chuji | T400 |
| 00:28.55 | Chuji | on the same span |
| 00:29.07 | *** part/#asterisk Torq (~Torq@cpc3-cmbg9-5-0-cust203.cmbg.cable.ntl.com) |
| 00:29.09 | Chuji | Channels work fine, It's a T1 |
| 00:29.13 | Nethab | PBX power monitoring cards are giving errors, need to clear alarms. |
| 00:29.16 | Juggie | yah i have rthe same card |
| 00:29.32 | Nethab | what does that mean |
| 00:30.06 | Juggie | hrmm |
| 00:31.18 | *** join/#asterisk marlowe (~marlowe@bmw.princetonhost.com) |
| 00:31.37 | Chuji | -- Attempting native bridge of Zap/24-1 and Zap/1-1 |
| 00:31.38 | Chuji | <PROTECTED> |
| 00:31.40 | sudoer | Nethab, I have one more question about conference calls |
| 00:31.43 | Chuji | Then boom, done |
| 00:31.51 | Juggie | whats your dial look like |
| 00:31.54 | Juggie | Dial(? |
| 00:32.05 | sudoer | what is the procedure you go through if you are on the phoen with someone and you want to have a 3 way call |
| 00:32.06 | Chuji | It's out of an AGI |
| 00:32.21 | *** join/#asterisk lattice (~lattice@dsl017-054-176.sfo1.dsl.speakeasy.net) |
| 00:32.22 | sudoer | the conference call seems overly complicated also on polycom phones |
| 00:32.41 | Chuji | But the same Dial line works fine if the originating call is Sip |
| 00:32.55 | Chuji | which makes me believe it's somehting going on with the Zap channels |
| 00:33.14 | *** join/#asterisk HjemmeRoyK (~roy@110.80-203-29.nextgentel.com) |
| 00:33.25 | Juggie | whats the agi doing |
| 00:34.54 | Chuji | Juggie : It's just a quazi calling card app. The dial string looks like so |
| 00:35.03 | Chuji | $AGI->exec("DIAL Zap/g1/801$phonenum"); |
| 00:35.33 | Chuji | 801 is the prefix on my switch that pulls a a local trunk group |
| 00:36.09 | Juggie | well, the switch rejecting the call is one thing that could cause what is happening |
| 00:36.33 | Chuji | Yeah, but that doesn't explain why Sip works |
| 00:36.53 | Juggie | true |
| 00:36.58 | Juggie | try echoing the dial string |
| 00:37.06 | Juggie | see if it changes between sip to zap or zap to zap |
| 00:37.14 | Juggie | shoudnt but you never know |
| 00:37.16 | Chuji | I wonder if the "Native Transfer" thing has anything to do with it |
| 00:37.40 | Juggie | native transfer? |
| 00:37.40 | Chuji | native bridge |
| 00:37.42 | Chuji | I meant |
| 00:37.44 | ||Smuggs | call me? 576021 i promise u good times |
| 00:38.07 | Juggie | should be fine |
| 00:38.21 | *** join/#asterisk inezk (z293388@atos.wmid.amu.edu.pl) |
| 00:38.22 | Chuji | Is it trying to bridge the two zap channels natively? |
| 00:38.25 | inezk | hello |
| 00:38.27 | Chuji | without * in the middle? |
| 00:38.36 | Juggie | are you running 1.0.4 of everything? |
| 00:38.47 | Juggie | libpri/asterisk & zaptel |
| 00:38.49 | inezk | anybody implemtn h.323 or other voip protocol in own aplication (in c# maybe) ? |
| 00:38.50 | Chuji | 1.0.2 |
| 00:39.15 | Chuji | Guess I could upgrade, but this seems pretty elementary |
| 00:39.29 | sudoer | Nethab: the conference call seems overly complicated also on polycom phones |
| 00:40.32 | sudoer | Nethab: what is the procedure you go through if you are on the phoen with someone and you want to have a 3 way call |
| 00:41.04 | sudoer | is that considered a conference call? I've having the users do conference calls for 3 people |
| 00:41.17 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 00:41.24 | facek | i am new in asterisk |
| 00:41.34 | Chuji | facek |
| 00:41.35 | Chuji | ~docs |
| 00:41.36 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 00:41.36 | facek | can somebody give me an accoutn. i want to test x-lite. |
| 00:42.06 | facek | Chuji: yes, of course. I have already read it, and now i am waiting for hardware. |
| 00:46.14 | *** part/#asterisk beto75 (~hav@201.133.230.247) |
| 00:47.18 | Chuji | -- Starting simple switch on 'Zap/24-1' |
| 00:47.54 | *** join/#asterisk doushanes (~Owner@c-67-184-189-220.client.comcast.net) |
| 00:48.16 | hermie | damn shame that they are actually the best wireless carrier |
| 00:48.24 | hermie | or i'd drop em like a bad habit |
| 00:50.34 | sudoer | is there a wa yto make or emulate 3 way calls with asterisk? |
| 00:51.01 | Juggie | bridge two lines? |
| 00:51.10 | Juggie | u can do a flashhook if you are on analog lines |
| 00:51.36 | hermie | sudoer: a meetme()? |
| 00:51.44 | modulus_ | nigga wuuuuuuT!? |
| 00:51.48 | modulus_ | jbot g-g-g-g? |
| 00:51.49 | jbot | G-UNIIT!! |
| 00:52.12 | hermie | jbot forget g-g-g-g |
| 00:52.12 | jbot | hermie: i forgot g-g-g-g |
| 00:52.25 | hermie | jbot g-g-g-g is what crackas say |
| 00:52.26 | jbot | hermie: okay |
| 00:52.27 | hermie | :) |
| 00:52.44 | modulus_ | jbot g-unit? |
| 00:52.45 | jbot | g-unit stands for "Guerilla Unit". It's members are Tony Yayo, Lloyd Banks, Young Buck, and the leader 50 Cent. Their official DJ is DJ Whoo Kid. Also see http://www.g-unitsoldier.com/ |
| 00:53.46 | hermie | the whole 50 cent/ja beef was just too much |
| 00:53.48 | hermie | or hip hop |
| 00:53.53 | *** part/#asterisk forrestc (~fwc@206.127.78.199) |
| 00:54.05 | hermie | whatever you want to call the random arrangement of clicks and whistles |
| 00:54.13 | *** join/#asterisk aspworld (~richard@northbay-pppoe-195.vianet.ca) |
| 00:54.14 | ||Smuggs | someone get 50 cents phone number so we can talk tuff to him over the internet |
| 00:54.29 | ||Smuggs | i can use VOIP |
| 00:54.31 | ||Smuggs | i am sooo cool |
| 00:54.36 | *** join/#asterisk laloo (~laloo@042.142-60-66.FTTH-SWI.surewest.net) |
| 00:55.04 | *** join/#asterisk subx (~cstraley@24-148-45-8.stk-bsr1.chi-stk.il.cable.rcn.com) |
| 00:55.41 | laloo | Guys. Can someone please help me? We installed a new TDM400P card a few days ago. Things were going fine for a while. Today, when we rebooted the system, we get an error "No ISA Tormenta card found at d0000" What gives? |
| 00:57.11 | subx | anyone wiling to help with CentOS issue and the chan_sip.c? |
| 00:57.52 | subx | a make command is blowing errors in regards to chan_sip.c |
| 00:58.06 | subx | ideas? |
| 00:58.13 | eKo1 | I'm trying to write a logrotate entry for *. Will a reload cause * to reopen the log files for writing? |
| 00:58.15 | wankel | what errors? |
| 00:58.32 | firestrm | has anyone ever gotten terracall to work? |
| 00:58.36 | subx | chan_sip.c: In function `handle_response': |
| 00:58.36 | subx | chan_sip.c:6784: duplicate case value |
| 00:58.36 | subx | chan_sip.c:6769: previously used here |
| 00:58.41 | *** join/#asterisk FryGuy- (fryguy@c-67-174-57-164.client.comcast.net) |
| 00:58.44 | DrPete | is dtmfmode=rfc2833 inband or out of band? |
| 00:58.48 | wankel | subx: cvs head? |
| 00:58.53 | subx | yes sir |
| 00:58.54 | *** join/#asterisk SimonR (~SimonR@static-1M-b1-14.highspeed.eol.ca) |
| 00:59.01 | Chuji | yeah, same thing |
| 00:59.26 | wankel | hold on, lemme update |
| 00:59.50 | subx | is this common, I made sure I used the stable cvs |
| 01:00.05 | wankel | i've had stable break a few times |
| 01:00.08 | *** join/#asterisk r0d3nt|m (RatMan@209-58-249-211.cust.telepacific.net) |
| 01:00.17 | wankel | usually just trivial stuff |
| 01:00.25 | *** join/#asterisk Smuggs (~Smuggs@S0106000c41a28668.cg.shawcable.net) |
| 01:00.51 | tessier_ | Man lawyers suck. They think they can threaten and scare people into doing stuff. |
| 01:01.04 | wankel | tessier: generally, they're right :) |
| 01:01.17 | *** join/#asterisk mm29955 (~test@host97-216.pool82184.interbusiness.it) |
| 01:01.19 | tessier_ | Not in my case. First thing I do upon hearing from a lawyer is call my lawyer and let those two hash it out. |
| 01:01.23 | tessier_ | Not my problem. |
| 01:01.32 | tessier_ | The only lawyer who isn't an asshole is your own lawyer. Funny eh? |
| 01:01.44 | wankel | oh, he's an asshole, too. just not to you. :) |
| 01:01.44 | tessier_ | He's really an asshole too, it's just that they do anything for money. |
| 01:01.48 | tessier_ | Exactly. |
| 01:01.57 | hermie | tessier_ Of course, your lawyer must drag the negotiations out... |
| 01:01.59 | tzanger | tessier_: so what's that lawyer want you to do? |
| 01:02.11 | *** join/#asterisk santiago (~santiago@63.245.86.101) |
| 01:02.18 | tessier_ | tzanger: eh, best not to discuss legal issues on IRC. :) |
| 01:02.29 | tzanger | tessier_: heh well no need for specifics |
| 01:02.32 | tessier_ | I think he really wants me to suck his cock. |
| 01:02.47 | wankel | "Sir, it's about the incident with the dog" |
| 01:02.50 | *** join/#asterisk mm29955 (~test@host97-216.pool82184.interbusiness.it) |
| 01:02.51 | tessier_ | hehe |
| 01:03.04 | Grooby | firestrm: good luck. I am headin gout |
| 01:03.06 | laloo | can someone please point me in the right direction? Is it possible that the card just went dead on us? |
| 01:03.34 | wankel | subx: hmm. it just built clean for me. |
| 01:03.57 | subx | interesting... lemme check |
| 01:04.07 | wankel | -rw-rw-r-- 1 petro petro 428948 Jan 21 21:05 channels/chan_sip.o |
| 01:05.02 | wankel | evil clowns |
| 01:05.03 | subx | ok I just tried to remake, nada |
| 01:05.11 | subx | same error. |
| 01:05.13 | wankel | what kind of a freak wants to put on makeup and play with children? |
| 01:05.23 | subx | got a chan_sip that I have? |
| 01:05.26 | hermie | wankel: is there another kind/ |
| 01:05.29 | subx | ;-) |
| 01:05.30 | hermie | ? |
| 01:05.50 | subx | that I can have excuse me... |
| 01:05.53 | wankel | subx: you checked out -r v1-0? |
| 01:05.54 | eKo1 | wankel: That's what happens with overpopulation. |
| 01:06.38 | hermie | "Stay all night to save the population" |
| 01:07.48 | hermie | or is it "Stay all night, we'll save the population"? I think so... |
| 01:08.26 | *** join/#asterisk DaviD (sinemm_@81.212.13.57) |
| 01:09.10 | sudoer | is there a way to simulate or do 3 way calling with * |
| 01:09.16 | *** join/#asterisk florz_ (nobody@p508A6EB9.dip.t-dialin.net) |
| 01:09.22 | sudoer | meetme isnt the same |
| 01:10.38 | sudoer | for the polycom to have a '3 way calling', i have to call user, press hold,dial 4000(for conf), press dial,press transfer, make another call to someone, press hold, dial 4000, press dial,press transfer,dial 4000,press dial |
| 01:11.26 | paulc | can't you do 3 way calling on the Polycom itself? |
| 01:11.48 | wankel | is 1.0 still really considered the latest stable? |
| 01:11.48 | freat | I have a polycom ip 500... it has a conference button that works |
| 01:11.49 | eKo1 | Question: Does are reload on the CLI reopen log files? |
| 01:12.01 | sudoer | paulc, i dont know if you can, i dont think so |
| 01:12.14 | freat | sudoer: yes you can |
| 01:12.15 | wankel | subx: okay, head builds clean here, too. hmmm... what version of gcc do you have? |
| 01:12.29 | freat | sudoer: we just bought 14 of them |
| 01:12.41 | sudoer | freat, can you tell me how to iniate one? |
| 01:12.48 | wankel | that doesn't sound like a C99 problem but you never know what kind of trip the parser will get on when it gets lost |
| 01:12.49 | sudoer | im using the 300 here |
| 01:12.55 | *** join/#asterisk file (~symlink@mctn1-142166192101.nb.aliant.net) |
| 01:13.11 | freat | dial a number. once in the call, one of the soft button labels is "conference" |
| 01:13.23 | freat | hit that, dial the second number, then hit conference again |
| 01:13.36 | paulc | sudoer: Dial a number.. press Conf.. dial the second number.. hit conf.. done! :-) |
| 01:13.49 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 01:14.09 | sudoer | oh ,that is the way i am doing it, but the conference call button started acting up last week, i have to use transfer right now for some reason |
| 01:14.30 | freat | acting up? |
| 01:15.21 | freat | sudoer: sounds like you have another issue... |
| 01:15.28 | sudoer | i dont know how its acting up either since i have never changed their config files via ftp, etc |
| 01:16.04 | sudoer | <PROTECTED> |
| 01:16.13 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 01:16.14 | hermie | sudoer: you have polycom phones, right? they can do 3-way |
| 01:16.27 | letherglov | where can I get a three way? |
| 01:16.33 | *** part/#asterisk santiago (~santiago@63.245.86.101) |
| 01:16.38 | wankel | try #hottub |
| 01:17.19 | Primer | ztd-eth.c:90: warning: implicit declaration of function `eth_hdr' |
| 01:17.21 | Primer | hrmm |
| 01:17.38 | wankel | it's c. you're supposed to ignore warnings. ;) |
| 01:17.59 | Primer | my bad, wrong line |
| 01:18.02 | Primer | ztd-eth.c:90: error: invalid type argument of `->' |
| 01:18.09 | *** join/#asterisk dontmsgme (~none@adsl-68-124-160-220.dsl.irvnca.pacbell.net) |
| 01:18.19 | dontmsgme | <PROTECTED> |
| 01:18.19 | dontmsgme | [17:17] <dontmsgme> 69-175-230-122*CLI> == Everyone is busy/congested at this time |
| 01:18.28 | dontmsgme | What does this mean, everytime it tries to dial my SIP phone it says this |
| 01:19.12 | Mavvie | I could msg you the answer, but then I would offend you. |
| 01:19.24 | dontmsgme | Im a grown up, try me |
| 01:19.28 | wankel | yeah, i was gonna tell him, but he clearly didn't want me to |
| 01:19.39 | dontmsgme | =(( |
| 01:19.41 | wankel | primer: what release is that? |
| 01:19.41 | freat | do 'sip show peers' at the console to see if it is actually registered |
| 01:19.56 | freat | I doubt the phone is actually registered with the server correctly |
| 01:20.05 | Mavvie | anyway, the error is in dial_exec.c |
| 01:20.11 | Mavvie | it says it can't dial the sip-phone. |
| 01:20.13 | Mavvie | does it exist? |
| 01:20.15 | dontmsgme | sip show peers |
| 01:20.15 | dontmsgme | Name/username Host Dyn Nat ACL Mask Port Status |
| 01:20.15 | dontmsgme | 12345 (Unspecified) D N 255.255.255.255 0 UNKNOWN |
| 01:20.21 | dontmsgme | It shows reigstered and it will make phone calls |
| 01:20.32 | freat | yep, the server is not seeing it |
| 01:20.36 | wankel | doesn't look very registered to me |
| 01:20.38 | freat | Host should show the IP |
| 01:20.40 | wankel | since it has no ip or port |
| 01:20.43 | dontmsgme | Why would that change over night? |
| 01:20.48 | Mavvie | edwin/edwin 192.168.1.247 D N 255.255.255.255 5060 Unmonitored |
| 01:20.51 | Mavvie | that's what you need. |
| 01:20.52 | Primer | wankel: 1.0.3 |
| 01:20.52 | freat | uhh.. sun spots |
| 01:20.57 | wankel | cosmic rays |
| 01:21.00 | freat | yes |
| 01:21.06 | dontmsgme | Turbulence =( |
| 01:21.11 | dontmsgme | Air pockets =( |
| 01:21.21 | Mavvie | gravity fluctuations. |
| 01:21.24 | freat | can you ping the phone? |
| 01:21.26 | dontmsgme | What am I supposed to do |
| 01:21.39 | *** part/#asterisk eKo1 (~bernd@63.245.57.70) |
| 01:21.57 | wankel | sit the phone down, point your finger at the asterisk server and say very firmly, "Go register." |
| 01:22.08 | hermie | http://tinyurl.com/6akto |
| 01:22.10 | dontmsgme | Ialready did that the phone just told me to come inhere |
| 01:22.16 | freat | damn phones need to learn how to behave |
| 01:22.17 | paulc | sudoer: I'd be half inclined to clear the phone config and set it up from scratch again |
| 01:22.23 | wankel | sounds like your phone needs some discipline |
| 01:22.35 | freat | bad, bad phone |
| 01:22.45 | dontmsgme | I havent changed sip.conf at all |
| 01:22.49 | Mavvie | dontmsgme: stop/start asterisk. |
| 01:22.50 | dontmsgme | Nor has my PBX IP changed |
| 01:22.51 | Umaro | tell it if it doesn't behave, you'll reflash it to a earlier, less functional firmware version. |
| 01:22.56 | hermie | I think the word for this map (http://tinyurl.com/6akto) is circuitous |
| 01:22.59 | wankel | reboot the phone and see what the console says when it tries to register |
| 01:22.59 | freat | dontmsgme? can you ping the phone?? |
| 01:23.13 | dontmsgme | I started and stopped, show peers, and it is the same |
| 01:23.18 | dontmsgme | How do I ping? |
| 01:23.22 | freat | OMG |
| 01:23.23 | Mavvie | "ping" |
| 01:23.25 | Mavvie | ping(1) |
| 01:23.35 | Mavvie | actually, ping(8) here. |
| 01:23.36 | sudoer | paulc, can i set these configs up without using ftp? |
| 01:23.46 | wankel | don't confuse him with the (1). he doesn't have the manuals on the shelf and the shell will bitch about it :) |
| 01:23.51 | sudoer | unfortunately i dont have ftpd installed on any boxes here |
| 01:23.59 | freat | ping 123.123.123.123 |
| 01:24.12 | freat | substitute your phone's IP address where the 123.123.123.123 is |
| 01:24.13 | sudoer | paulc freat: can i see a sample of one of your configs |
| 01:24.15 | jontow | wee, got my script working :) |
| 01:24.22 | jontow | turns out.. there is an easy way to make it work.. |
| 01:24.23 | paulc | sudoer: Yeah - You can do a lot of the config through the phone and/or web interface, but the real power with the polycoms comes from editing the xml files |
| 01:24.29 | freat | hey sure what's your email? |
| 01:24.48 | paulc | sudoer: give me a few minutes cos I'm just setting up a new IP600.. but I can probably send you a template file |
| 01:24.52 | jontow | make an extension to dial that you can use as Local/### |
| 01:25.08 | jontow | the extension can't Answer as priority one, it has to Dial() the outbound number aspriority 1 |
| 01:25.14 | dontmsgme | Yes it is pinging |
| 01:25.20 | dontmsgme | Why isntit registered? |
| 01:25.27 | jontow | .. as, if it answers, it'll bridge to the internal extension and then hangup when the gsm file finishes |
| 01:25.29 | Mavvie | try sip debug and reload the phone. |
| 01:25.30 | wankel | dontmsgme: how often does it register? |
| 01:25.30 | jontow | :) |
| 01:25.40 | dontmsgme | I duno |
| 01:25.47 | sudoer | paulc, the otherproblem i have is i cant pick up the handset and dial 9areacodephonenumber, i have to keep handset down, then dial and pickup, i guess that is part of the xml config too |
| 01:25.48 | dontmsgme | How do I stp this ping? |
| 01:25.51 | wankel | if you just restarted asterisk and you're not saving your presence data in a database and the phone hasn't re-registered... |
| 01:25.52 | cbachman | has all the SunOS pages on his shelf |
| 01:26.01 | *** join/#asterisk imagmo (~imagmo@c-67-160-156-219.client.comcast.net) |
| 01:26.11 | freat | dontmsgme: do you see messages on the server console that it is trying to register? |
| 01:26.23 | Mavvie | hmmm... can't turn off the ping if I'm not mistaken, you have to reboot your system. |
| 01:26.24 | jontow | so now, when httpd drops on my webservers, i get a call with a prerecorded message saying what dropped and on what server :) |
| 01:26.48 | freat | dontmsgme: if you do not see messages that it is trying to register, then the phone is not seeing the server. check the IP of the server and what your phone says |
| 01:26.48 | jontow | grandstreams once an hour too |
| 01:27.06 | wankel | jontow: and it's probably more intelligent than a noc monkey |
| 01:27.07 | freat | dontmsgme: make sure to reboot the phone so you force it to attempt registering |
| 01:27.21 | freat | dontmsgme: is your server on a static IP? |
| 01:27.29 | dontmsgme | -- Registered SIP '12345' |
| 01:27.31 | dontmsgme | Yes |
| 01:27.36 | dontmsgme | I can make calls |
| 01:27.39 | wankel | does it show in peers now? |
| 01:27.39 | dontmsgme | It just doesnt take SIP calls |
| 01:27.56 | dontmsgme | lemme check |
| 01:27.57 | sudoer | ok, thanks paulc |
| 01:28.01 | dontmsgme | Yea it does now.. |
| 01:28.10 | freat | then you need to set up sip.conf correctly if it can't receive calls |
| 01:28.10 | jontow | wankel; sure as hell is more intelligent than ours ;) which also happen to be our 'network techs' |
| 01:28.13 | wankel | okay, so you probably restard asterisk and the phone hadn't re-registered yet. |
| 01:28.17 | freat | look on the wiki for documentation on your phone |
| 01:28.19 | dontmsgme | Wow now its working... |
| 01:28.29 | dontmsgme | I got the magic touch I guess |
| 01:28.46 | jontow | here, they take a reactive monitoring approach.. so they only know about trouble when a customer has noticed and is already pissed off and talking to someone at our call center |
| 01:28.52 | wankel | hehe |
| 01:29.00 | wankel | jontow: watching mrtg graphs is hard! |
| 01:29.04 | freat | well, that ping command is really powerful. probably did the trick |
| 01:29.09 | jontow | where the people in the call center have no ability to do anything, except say "we'll get ahold of someone" or lieing and saying "yep! we're working on that right now" |
| 01:29.15 | jontow | i'd rather know before the TSRs |
| 01:29.21 | jontow | .. and more importantly, the volatile customers |
| 01:29.47 | freat | and if ping doesn't work, use 'zuperping' |
| 01:29.47 | jontow | i'm actually building tools for people to use to do their jobs |
| 01:29.54 | dontmsgme | HOw do I stop this ping flood? |
| 01:29.56 | freat | it is really powerful |
| 01:30.01 | jontow | vs. just giving people a script that says "hold on while i escalate this issue" |
| 01:30.04 | freat | turn off your computer |
| 01:30.14 | freat | try ctrl-c |
| 01:30.21 | jontow | and bothering network techs who have no time as is with simple issues like "you've gone over your quota.. remove some files" |
| 01:30.22 | dontmsgme | That worked |
| 01:30.27 | jontow | .. its retarded, all around. |
| 01:30.35 | wankel | jontow: of course. |
| 01:30.58 | wankel | you don't expect to have clued people in the noc when they get paid $25k/yr, do you? :) |
| 01:32.37 | imagmo | hello, has anybody received a TDM04B card with RJ45's instead of RJ11's? |
| 01:32.49 | freat | yeah just plug the rj11's into it |
| 01:32.57 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 01:32.59 | Veto | is there a 1.0.4 of zaptel? |
| 01:33.13 | freat | imago: I had the same question a while back |
| 01:33.29 | imagmo | freat, so the pins match? |
| 01:33.35 | freat | something about making with rj45s was cheaper |
| 01:33.35 | freat | yes |
| 01:33.43 | imagmo | thanx!! |
| 01:33.45 | freat | np |
| 01:34.02 | imagmo | i'm doing this install on the wire, so I can't look at it myself. |
| 01:34.08 | imagmo | cheers |
| 01:35.16 | laloo | Can someone please help me? I just need to be pointed in the right direction |
| 01:35.35 | jontow | i don't get paid $25k/yr |
| 01:35.37 | jontow | and i'm clued |
| 01:35.38 | jontow | wtf :) |
| 01:35.48 | wankel | you get paid less than that? |
| 01:36.38 | Veto | jbot who is martin |
| 01:36.40 | jbot | I think you lost me on that one, Veto |
| 01:36.54 | jontow | yes, yes i do |
| 01:36.55 | Veto | jbot why is martin always busy? |
| 01:36.57 | jbot | Veto: what are you talking about? |
| 01:36.57 | jontow | $12/hr |
| 01:37.02 | jontow | and its decent pay for around here |
| 01:37.03 | wankel | jontow: damn. you need a new job :) |
| 01:37.17 | jontow | yeah, yeah i do.. know anyone that'll hire a 20yo with no college degree and lots of skill? |
| 01:37.18 | wankel | where? rural india? |
| 01:37.22 | jontow | upstate NY |
| 01:37.34 | wankel | hmm. we might have some SOC openings. not sure. |
| 01:37.58 | *** join/#asterisk Marlow (~marlow@217.67.139.197) |
| 01:38.07 | jontow | plenty of UNIX experience (5+ yrs senior admin with many flavors, less with others), quite a bit of networking, little cisco but enough to get around, quite a bit of asterisk experience |
| 01:38.17 | jontow | some C, lots of shell scripting |
| 01:38.51 | jontow | the only reason i make $12/hr vs. the $8.25/hr from before is because i could script my way out of HP-UX even :P |
| 01:39.21 | Nukemizer | with a basic install of * is there any special requirements to having my * behind the Firewall ? can I just forward ports to * ? |
| 01:39.43 | wankel | depends on what's trying to talk to the * box |
| 01:39.53 | Marlow | :) |
| 01:40.07 | jontow | if its SIP .. good luck |
| 01:40.08 | jontow | :( |
| 01:40.22 | Nukemizer | yep SIP |
| 01:40.25 | Nukemizer | :( |
| 01:40.27 | jontow | yikes :( |
| 01:40.38 | jontow | any chance you can get SER on the outward-lying machine? |
| 01:41.11 | Marlow | jontow : if he could get ser on there, he could just install asterisk there instead :) |
| 01:41.16 | wankel | jontow: hrm. no SOC openings. security engineer openings, but you said not much cisco knowledge? do you know any firewall or ids products? |
| 01:41.33 | wankel | soc will take smart geeks and train them on the products, but the engineers have to come in knowing one or two |
| 01:42.02 | jontow | i've played with snort, ipfw, a bit with ipfilter |
| 01:42.26 | jontow | nothing too serious because i spend too much time putting other people's fires out |
| 01:42.41 | wankel | oh, wait. there's a soc analyst position |
| 01:42.54 | laloo | What does this message in Asterisk log mean? " Requested 8000 Hz, got 48000 Hz -- sound may be choppy" |
| 01:43.04 | *** join/#asterisk MrEntropy (~entropy@ppp243-36.lns2.adl2.internode.on.net) |
| 01:43.05 | MrEntropy | yo |
| 01:43.19 | MrEntropy | does anyone know a good program for encoding IVR's in g729? |
| 01:43.46 | wankel | jontow: http://sh.webhire.com/servlet/av/jd?ai=544&ji=1502430&sn=I |
| 01:44.01 | wankel | i think i get some piddly $500 award if i refer you, too :) |
| 01:44.25 | jontow | i should kill both of my copies of mozilla and firefox and restart them |
| 01:44.32 | wankel | haha |
| 01:44.40 | jontow | methinks 20+ tabs in each window is a bit much with only 256MB of RAM and another 190 processes also :( |
| 01:44.43 | Umaro | Anyone know how to do a 2 way outbound call? I want both sides of the channel to be outbound |
| 01:45.14 | wankel | that's just working in the SOC but you can get to know the products and we have lots of engineering openings you can apply to transfer to once you have the expertise. |
| 01:45.43 | wankel | the SOC actually has a cool manager now. total asshole for a while but he's long gone. |
| 01:45.48 | Marlow | Umaro : with a .call script ? |
| 01:46.38 | Marlow | Umaro : that should do it ... just throw a .call file and let it do the two calls |
| 01:47.09 | PTG123 | anyone in here know how to make it so you can listen to calls live? |
| 01:47.22 | Umaro | Marlow: well yeah, but sorry.. I didn't phrase my question correctly. |
| 01:47.28 | PTG123 | like easedrop |
| 01:47.30 | jontow | wankel; i am not qualified for that job. |
| 01:47.36 | Marlow | Umaro : you get what you ask for :o) |
| 01:47.38 | Umaro | Marlow: Basically, I have a * in EU and a * in NY |
| 01:47.40 | wankel | jontow: you are :) |
| 01:47.54 | Druken | PTG123: that is illegeal |
| 01:47.57 | wankel | ptg123: zapscan/zapbarge |
| 01:48.02 | jontow | not per the requirements |
| 01:48.06 | Umaro | Marlow: my calls from NY to EU are less than my calls from EU to EU |
| 01:48.08 | jontow | i also don't know how i feel about more tech support |
| 01:48.11 | jontow | i got out of that on purpose :) |
| 01:48.41 | Marlow | Umaro : so you want to do callback .. |
| 01:48.53 | Umaro | Marlow: yes, but an automated callback |
| 01:48.55 | wankel | yeah, it sucks. definitely advancement opportunities here, but you're not qualified for the engineering jobs without expertise in at least one security platform. |
| 01:49.07 | jontow | but what do you consider a security platform? |
| 01:49.07 | wankel | what are you missing? the cisco cert? |
| 01:49.11 | Umaro | Marlow: I don't want the caller to know it's doing a callback underneath |
| 01:49.14 | jontow | most definitely the cisco stuff.. all of it |
| 01:49.21 | Marlow | Umaro : that won't work .. |
| 01:49.26 | Umaro | Marlow: why not? :) |
| 01:49.33 | cbachman | has a similar issue. Is in a good job position but no advancement opportunities, unless the boss quits. |
| 01:49.44 | jontow | my cisco experience exists in a few illegal entries into piddly routers in 1999, and a 2511 that i stole from the spares rack here a week ago to fuck with |
| 01:49.47 | Marlow | Umaro : you can't revert a call, that allready is inbound to outbound without hanging up |
| 01:50.02 | Umaro | Marlow: Except if I redirected it via the manager interface, right? |
| 01:50.22 | Marlow | Umaro : no .. it's still inbound .... |
| 01:50.31 | jontow | sounds like another one where they sit you in a call center at a windows pc with a couple webforms and a lot of books |
| 01:50.36 | Marlow | Umaro : it can only get outbound, when the initiator hangs up and is called back |
| 01:50.46 | jontow | which .. btw, happens to be "a lot of books" more than the job i got out of here :D |
| 01:50.57 | *** join/#asterisk pauldy (~pauldy@24-155-82-32.ip.grandenetworks.net) |
| 01:50.58 | jontow | i think i would not mind it, but i don't know :/ |
| 01:51.08 | Umaro | Marlow: well no, there wouldn't be any hanging up |
| 01:51.23 | Umaro | Marlow: they would just kind of... wait until the callback call came and got transferred to them |
| 01:51.24 | Marlow | Umaro : exactly .. so not possible .. |
| 01:51.35 | subx | anyone got braodvoice incoming call issues not out going that works just fine... |
| 01:51.44 | jontow | im not horribly proficient with firewalling, though i get my own thing done, and a few other people's |
| 01:51.46 | Marlow | Umaro : if somebody calls you .. |
| 01:51.50 | pauldy | ok newbie question trying to find a service provider like vonage or lingo for use with an asterisk server any suggestions |
| 01:51.51 | Marlow | Umaro : from a landline .. |
| 01:51.52 | wankel | yeah, like i said support always sucks, but if you're smart and want to get out of the soc we're desparate for third-level engineers. |
| 01:51.56 | jontow | my main skills are sysadminning and odd problem solving in the unix environment |
| 01:52.01 | wankel | and existing employees get first crack and all opennings |
| 01:52.06 | Marlow | Umaro : it's not in your hands to change the billing on that call ... |
| 01:52.08 | pimpwell | my main skills are paying someone to do it |
| 01:52.12 | wankel | you know IP well, though? |
| 01:52.20 | pimpwell | PI and MP |
| 01:52.21 | Marlow | Umaro : only if you call them back .. |
| 01:52.51 | jontow | i'd say so.. im not proficient with the OSI model, i can't explain every protocol up, down, and through, and i couldn't explain the command to apply a cert to an apache/linux server |
| 01:52.55 | JerJer | damn, that is a wise concept... calling people back |
| 01:52.55 | Umaro | Marlow: no, no.. not from a landline |
| 01:53.06 | wankel | if you know *nix really well and IP fairly well i think you'd have a good shot at it unless we're suddenly seeing a glut of applicants. last i knew they were desparate for smart people. |
| 01:53.08 | Umaro | Marlow: from a sip phone on my EU * server |
| 01:53.25 | jontow | but .. i've looked at the OSI model, played with an awful lot of protocols, and applied a cert to a freebsd/jailed apache2 server, with documentation from your site ;) |
| 01:53.29 | wankel | anyway, if you want to talk to someone i can have the recruiter for our office call you. if not, oh well :) |
| 01:53.29 | pimpwell | you need to be able to hand code a TCP/IP packet |
| 01:53.30 | *** join/#asterisk file (~symlink@mctn1-3365.nb.aliant.net) |
| 01:53.30 | Marlow | Umaro : eh ? .. then i don't get the concept of two outbound calls .. |
| 01:53.38 | Marlow | Umaro : i mean your conecpt ? |
| 01:53.43 | Umaro | Marlow: I don't want it to go VoIP to NY, I want it to go TDM from NY to EU |
| 01:53.48 | jontow | i am curious, thats for sure |
| 01:53.53 | Umaro | er |
| 01:53.55 | jontow | i know that i can't make it to college anytime soon, here.. |
| 01:54.11 | Umaro | Marlow: I don't want it to go VoIP to NY and then VoIP to EU, I want it to go TDM from NY to EU |
| 01:54.14 | jontow | i don't get paid enough to save the money i need to do that.. so i need to find something that pays more |
| 01:54.26 | Marlow | Umaro : ah .. |
| 01:54.30 | pimpwell | jontow: start your own business |
| 01:54.31 | wankel | we've got typical edu reimbursement and such. $3500/yr won't pay for brown.edu, but you can get through URI or something local that way. |
| 01:54.32 | jontow | my biggest problem is.. im not willing to leave my gf, and she's in a similar boat with less unix experience than me.. but more cisco experience |
| 01:54.33 | Marlow | Umaro : that's a different story ... |
| 01:54.49 | jontow | pimpwell; that doesn't make you more money to go to college, it puts you in debt, just like college |
| 01:54.52 | wankel | hmm. yeah, hard to move with two people. |
| 01:54.58 | Marlow | Umaro : that would be a background callback from your US server to your EU server .. |
| 01:55.07 | Umaro | Marlow: yes! |
| 01:55.09 | Marlow | Umaro : should be possible .. |
| 01:55.14 | jontow | however.. she also has more call center experience than me, and could very well apply and be a better candidate for the job than me. |
| 01:55.24 | wankel | heh. |
| 01:55.26 | jontow | .. we work together now ;) |
| 01:55.41 | Marlow | Umaro : hang on for a sec |
| 01:55.46 | subx | Anyone want to take on an incoming broadvoice issue? |
| 01:55.49 | wankel | guess she could apply for security engineer ;) |
| 01:55.53 | jontow | :P |
| 01:55.54 | jontow | hehehe |
| 01:55.59 | wankel | hmm. those requirements are silly. |
| 01:56.15 | pimpwell | jontow: write a bunch or essays on why republicans are always right, and send them out to scholarship agencies |
| 01:56.16 | jontow | you know.. im curious.. i really am |
| 01:56.20 | pimpwell | or = of |
| 01:56.20 | wankel | i know most of the SOC staff couldn't come close to them. operate a protocol analyzer? yeah, right. |
| 01:56.36 | wankel | they could probably figure out how to RUN tcpdump with enough googling, but wouldn't have a clue what it means. |
| 01:56.36 | jontow | im gonna bookmark the page and related, and show it to her |
| 01:56.39 | jontow | maybe we'll head to RI |
| 01:56.40 | jontow | heheh :) |
| 01:56.56 | wankel | RI isn't super-exciting, but hell, you're in upstate :) |
| 01:57.13 | jontow | don't know what the company policies are regarding dating a coworker though.. but as long as they knew up front and were ok with it, i'd deal |
| 01:57.54 | wankel | we have a few married couples. i don't know what the official policy is. at most place it's as long as one of you isn't the boss of the other |
| 01:58.12 | jontow | thats what it is here.. and we work on separate halves of the building |
| 01:58.41 | Umaro | Anyone had experience with large amounts of VoIP from US to EU? |
| 01:58.58 | *** join/#asterisk czero (~me@CPE000f6690f84b-CM001225704b6e.cpe.net.cable.rogers.com) |
| 01:59.10 | subx | ok I am giong to pull every last hair on my head |
| 01:59.24 | subx | maybe I'll pull i tout too |
| 01:59.48 | pimpwell | cam it |
| 02:00.05 | subx | ok can I have a comment about braodvoice, |
| 02:00.14 | PTG123 | hey czero |
| 02:00.27 | subx | pimpwell: thanks, I'll hook that up after I am done throwing the server across the room. |
| 02:00.41 | pimpwell | im trying to win my auction on ebay for an IDE to USB cable |
| 02:01.41 | czero | hey PTG |
| 02:01.48 | greg_work | pimpwell: you can get complete usb drive enclosures for $40cdn new |
| 02:02.07 | pimpwell | enclosure? |
| 02:02.33 | MrEntropy | pimpwell: IDE to USB? that's abominable |
| 02:02.45 | Veto | * seems to keep some state information between runs, I assume it's in /var/spool/asterisk. If so, can I clean it out via a careful rm ? |
| 02:02.57 | greg_work | ... a case that fits a 3.5" (and they make them for 2.5" as well) ide drive and has a power supply and plugs into usb |
| 02:03.15 | Umaro | Marlow: whadya think? :) |
| 02:03.21 | greg_work | and you can get them for 5 1/4" drives as well. in fact, i have a usb 2.0 cdwriter sitting beside me that cost like $80 |
| 02:03.38 | pimpwell | ya but I have like 14 120gig seagates right here |
| 02:03.43 | pimpwell | and a laptop |
| 02:04.08 | pimpwell | (that is almost full) |
| 02:04.09 | greg_work | .. and you're going to connect them all with usb ?! |
| 02:04.15 | pimpwell | just transfer |
| 02:04.22 | pimpwell | as I accumulate |
| 02:04.27 | greg_work | why dont you just build a NAS system |
| 02:04.42 | *** join/#asterisk [1]NormAst (~NormAst@207.245.0.131) |
| 02:04.50 | pimpwell | it's all weird movies about the government etc |
| 02:04.53 | pimpwell | just garbage |
| 02:04.57 | Marlow | Umaro : i have seen a script for that somewhere .. |
| 02:05.04 | Marlow | Umaro : but can't seem to find it right now .. |
| 02:05.11 | Umaro | Marlow: awesome, i'd love it |
| 02:08.21 | Damin | Morning. |
| 02:09.59 | *** join/#asterisk cripito (~cripito@c-65-34-156-173.se.client2.attbi.com) |
| 02:10.13 | cripito | hello |
| 02:10.20 | dontmsgme | Hi |
| 02:15.51 | subx | hello has anyone had luck with broadvoice incoing configs? |
| 02:22.14 | *** join/#asterisk gopinsurg (cashmoney@dialup-4.224.222.132.Dial1.Cincinnati1.Level3.net) |
| 02:23.15 | cripito | :) yes... subx |
| 02:23.38 | *** join/#asterisk AgiNamu (~root@200.12.43.74) |
| 02:23.46 | labo | 2/dns AgiNamu |
| 02:24.12 | AgiNamu | huh? |
| 02:25.14 | AgiNamu | hey |
| 02:25.18 | AgiNamu | I'm buying a new Xeon |
| 02:25.23 | AgiNamu | should I get RHEL with EMT64? |
| 02:25.38 | AgiNamu | or is that just when you need mre memory? i..e, it's nothing faster.. |
| 02:27.11 | Umaro | Marlow: still haven't found it? |
| 02:27.35 | Marlow | Umaro : no .. the only thing i could find again bases on the lcr for i4l .. |
| 02:27.38 | Umaro | ~jbot root |
| 02:27.43 | jbot | i guess root is not a Good Thing to use when using IRC. Please use a different account. You will probably not be able to speak until change your user account. |
| 02:27.47 | Marlow | Umaro : and that is quite unlikely to help you |
| 02:27.56 | AgiNamu | yea, i know, but its a new machine |
| 02:27.58 | AgiNamu | no logins |
| 02:28.04 | Marlow | Umaro : but what you are looking for is basically LCR in asterisk |
| 02:29.45 | *** join/#asterisk doughecka_ (~Doug@adsl-18-107-211.sdf.bellsouth.net) |
| 02:29.49 | Umaro | Marlow: I am? |
| 02:29.55 | Marlow | Umaro : you are :) |
| 02:30.10 | Umaro | I thought LCR was just picking your cheapest rate |
| 02:30.19 | Marlow | Umaro : no |
| 02:30.39 | *** join/#asterisk Guills (~guills@S01060048548225f4.vc.shawcable.net) |
| 02:30.42 | Marlow | Umaro : lcr involves also handling different ways of connecting to those rates |
| 02:30.47 | Umaro | oic |
| 02:30.55 | Marlow | Umaro : call by call prefix, callback etc. |
| 02:31.03 | Marlow | Umaro : goes all under the term "LCR" |
| 02:31.46 | Marlow | didn't get too much sleep today |
| 02:32.08 | Guills | sleep is bad for you |
| 02:32.18 | Guills | play mmorpg games instead ;) |
| 02:36.48 | *** join/#asterisk fishboy1669 (fishboy166@cpc1-warr1-3-0-cust38.bagu.cable.ntl.com) |
| 02:36.57 | fishboy1669 | hi guys |
| 02:37.24 | Guills | hi |
| 02:38.17 | cripito | is u sleep much u lost weight :P so sleep is bad for hearth :D |
| 02:38.35 | wankel | -ENOPARSE |
| 02:41.06 | freat | anyone here running * on 64 bit? |
| 02:41.29 | tzanger | where is matt's libpri cvs repo? |
| 02:41.30 | tzanger | libpri-matt isn't on the normal cvs server |
| 02:42.18 | Qwell | Can anyone tell me the differences between a 7960 and 7960g? I'm sure this has been covered before, but its a fairly difficult google query |
| 02:42.48 | Qwell | I even checked the product lists at cisco.com, and it had no mention of 7960 |
| 02:42.54 | Qwell | (without the g, that is) |
| 02:43.06 | labo | If you look at them, the g is for global, which means that has images and icons instead of english text on the buttons. |
| 02:43.16 | labo | but its the same thing actually. |
| 02:43.18 | Qwell | thats literally it? |
| 02:43.29 | Qwell | Why is the g like 30-60 more? |
| 02:43.44 | labo | that i dont know. |
| 02:44.01 | hermie | royalties on the artwork :) |
| 02:44.16 | Qwell | hermie: I hope you're kidding, heh |
| 02:44.57 | Qwell | So, spending the extra money is definitely not worth it, if my entire team speaks (pseudo)English? |
| 02:45.09 | *** join/#asterisk ccfiel (~chatzilla@210.213.139.36) |
| 02:46.12 | hermie | Qwell: Engrish or Indlish? |
| 02:46.37 | fishboy1669 | any of u guys got any pointers at this issue |
| 02:46.39 | Qwell | hermie: both? indlirish? |
| 02:46.47 | Veto | fuck sip and nat...fuckit fuckit fuckit. |
| 02:46.52 | Veto | oh, evening everyone. |
| 02:46.55 | hermie | Qwell: Ingrish maybe? |
| 02:47.01 | fishboy1669 | my ztcfg come back ok but asterisk bitches when i try running |
| 02:47.09 | fishboy1669 | just put a x100p card in |
| 02:47.09 | Qwell | hermie: If you can add spanglish in there, you win |
| 02:47.32 | hermie | Qwell: Spingrish! :) |
| 02:47.32 | fishboy1669 | error is unable to get parameters |
| 02:47.40 | Qwell | That works. |
| 02:47.43 | fishboy1669 | unable to register channel 1 |
| 02:47.56 | fishboy1669 | load_module failed |
| 02:48.00 | fishboy1669 | any ideas |
| 02:48.08 | fishboy1669 | been messing for ages and fed up |
| 02:48.09 | Qwell | hermie: and, of course, 2 or 3 of us that speak plain old English |
| 02:48.11 | hermie | Qwell: there are advantages to heartland america ya know ;) |
| 02:48.50 | Qwell | hermie: I'm in Southern California, this is the mix we get. :p |
| 02:49.06 | fishboy1669 | yo guys could do with hand here |
| 02:49.10 | *** join/#asterisk NTJOCK (~brian@txshirts.com) |
| 02:49.13 | fishboy1669 | anyone got any ideas |
| 02:49.26 | hermie | where "culture" is defined as fights... I don't live _that_ far from where the Pistons play |
| 02:49.40 | hermie | the ironically named Palace of Auburn Hills |
| 02:49.46 | fishboy1669 | its 2:49am and i want to go to bed :( |
| 02:49.48 | *** join/#asterisk r0d3nt|m (RatMan@209-58-249-211.cust.telepacific.net) |
| 02:51.13 | hermie | 0h a 1337 d00d! |
| 02:51.27 | NTJOCK | hey guys |
| 02:51.44 | NTJOCK | Where is some good documentation on how to get a polycom phone working with Asterisk? |
| 02:51.53 | hermie | NTJOCK: the wiki |
| 02:51.56 | NTJOCK | I finally got my system up an drunning |
| 02:51.59 | NTJOCK | ok |
| 02:52.01 | NTJOCK | I'll check there |
| 02:52.02 | NTJOCK | htnaks |
| 02:52.03 | NTJOCK | :) |
| 02:52.52 | hermie | 1337 d00d, u 5p311 600d |
| 02:53.28 | gabriel1 | what would cause 2 iax2 clients on a local network to not see each other? |
| 02:53.55 | gabriel1 | or rather an iax2 client not to see asterisk. |
| 02:54.48 | Marlow | fishboy1669 : what hardware ? |
| 02:55.05 | Marlow | fishboy1669 : zaptel.conf and zapata.conf correctly configured ? ztcfg run ? |
| 02:57.07 | fishboy1669 | x100p |
| 02:57.11 | fishboy1669 | brand new dell box |
| 02:58.09 | fishboy1669 | suse 9.2 |
| 02:58.25 | fishboy1669 | and i have done the udev stuff adn the linux62 |
| 02:58.30 | fishboy1669 | linux26 |
| 02:58.44 | r0d3nt|m | PCIX slots ? |
| 02:58.57 | *** join/#asterisk imagmo (~imagmo@c-67-160-156-219.client.comcast.net) |
| 02:59.10 | fishboy1669 | whats a pcix? |
| 02:59.14 | fishboy1669 | pci |
| 02:59.19 | Marlow | fishboy1669 : what kind of dell box ? |
| 02:59.56 | Marlow | fishboy1669 : and what does ztcfg -v say ? does it find your card ? |
| 03:00.25 | fishboy1669 | dimension 2400c |
| 03:00.40 | fishboy1669 | yes no errors from ztcfg |
| 03:00.43 | *** part/#asterisk gabriel1 (~gabriels@12-216-224-186.client.mchsi.com) |
| 03:01.03 | Marlow | fishboy1669 : that should be ok .. pci-x is 100mhz/133mhz pci slots, but that's only in the servers, not dimension boxes |
| 03:01.14 | Marlow | fishboy1669 : but does it show the card ? |
| 03:02.31 | fishboy1669 | yes |
| 03:02.46 | fishboy1669 | shows the card no errors |
| 03:02.47 | Marlow | fishboy1669 : then you probably didn't configure zapata.conf correctly |
| 03:02.56 | fishboy1669 | maybe |
| 03:03.09 | Marlow | fishboy1669 : did you put fxs_ks there ? |
| 03:03.12 | fishboy1669 | i copied the one off my old bx |
| 03:03.23 | fishboy1669 | and i also tried editing new copy |
| 03:03.31 | fishboy1669 | as the cvs is diff time checkout |
| 03:03.55 | Marlow | fishboy1669 : doesn't matter where you got it from, it still needs to fit your setup :) |
| 03:04.21 | Marlow | fishboy1669 : that one channel, that you have in zaptel.conf and is configured there .. |
| 03:04.43 | Marlow | fishboy1669 : that one channel is the only one, that you should configure in zapata.conf .. everything else is rubish |
| 03:05.20 | redder86 | anyone here have a color fax machine? |
| 03:05.21 | fishboy1669 | yes fxs_ks is in there |
| 03:05.46 | Marlow | fishboy1669 : try to pastebin the output from the console during the call |
| 03:05.58 | Marlow | fishboy1669 : during the fail, i mean |
| 03:06.49 | Marlow | fishboy1669 : and include your zaptel and zapata.conf |
| 03:10.24 | Umaro | Marlow: debian? |
| 03:10.49 | Marlow | sure |
| 03:10.57 | Umaro | Marlow: I haven't used debian in like, 5 years.. probably the same stable packages as I used back then, LOL |
| 03:11.03 | Marlow | Umaro : 1.0.4 + bristuff + some others .. |
| 03:11.17 | Marlow | Umaro : unlikely .. i roll them myself :) |
| 03:11.17 | Umaro | 1.0.4?!?!?! |
| 03:11.31 | Umaro | When did this happen, and why didn't I hear about it? :( |
| 03:11.33 | Marlow | Umaro : asterisk-stable and asterisk-head |
| 03:11.55 | Chuji | Umaro : Can't you keep up with the mailing list? |
| 03:11.56 | Chuji | lol |
| 03:11.57 | Marlow | Umaro : i have a repository with custom debian packages for asterisk |
| 03:12.08 | Marlow | Umaro : http://www.marlow.dk/asterisk |
| 03:12.20 | fishboy1669 | marlow how do i pastbin |
| 03:12.31 | Umaro | Chuji: lol, no. I can only bear to read asterisk-users once a week, and I only get about 50 mails in before I have to stop |
| 03:12.36 | Marlow | fishboy1669 : http://pastbin.ca |
| 03:12.58 | Chuji | Umaro : Yeah, I feel you. I respond to about 5 a day |
| 03:13.00 | Chuji | that is my limit |
| 03:13.16 | Nethab | it's in the channel topic silly 1.0.4 released |
| 03:13.19 | Chuji | I let Critch get the rest |
| 03:13.19 | fishboy1669 | that link dont work can u check it |
| 03:13.33 | Umaro | Nethab: damn! 3 days ago! |
| 03:13.37 | fishboy1669 | ok got it |
| 03:13.45 | Umaro | sheesh man :/ |
| 03:13.59 | Umaro | 1.0.4 does realtime, then? |
| 03:14.15 | Nethab | i've found when i call people silly instead of dumbass it gets through a lot of barriers |
| 03:14.22 | Marlow | Umaro : dunno .. |
| 03:14.56 | Umaro | Nethab: you can call me dumbass next time, I can take it. |
| 03:15.33 | fishboy1669 | marlow http://pastebin.ca/4558 |
| 03:15.43 | Umaro | People who take personal offense to someone else being smarter than them have no place in the open source community, I say. ;) |
| 03:15.56 | Marlow | Umaro : i also just realised it today, when somebody mailed me, asking, when i'm going to update the packages :) |
| 03:16.17 | Marlow | Umaro : should have seen it before on the mirror-logs |
| 03:16.17 | Umaro | they need a release mailing list |
| 03:16.18 | Nethab | glad to see i'm in good company then |
| 03:16.40 | Nethab | i've got a nice autoload = no modules.conf going |
| 03:17.09 | Marlow | fishboy1669 : ok .. now add your zaptel.conf and zapata.conf to that .. |
| 03:17.10 | fishboy1669 | http://pastebin.ca/4559 |
| 03:17.34 | Chuji | umm, where is zaptel? It's not on ftp.digium.com |
| 03:17.53 | Chuji | 1.0.4 that is |
| 03:18.02 | *** part/#asterisk Nethab (~Nethab@mtvcafw.sgi.com) |
| 03:18.02 | Marlow | Chuji : ftp://debian.marlow.dk/mirrors/ftp.digium.com |
| 03:18.06 | fishboy1669 | http://pastebin.ca/4560 |
| 03:18.29 | cripito | hi guys.. did anyone have issues with deadlock in cvs head |
| 03:18.39 | Marlow | Chuji : the faster choice :) |
| 03:19.12 | Chuji | Marlow, is there a username/pass for that? |
| 03:19.32 | Chuji | I tried that earlier |
| 03:19.45 | Chuji | It wouldn't let me get past the root |
| 03:19.50 | Chuji | as anon |
| 03:20.00 | fishboy1669 | any ideas? |
| 03:20.11 | Marlow | Chuji : that should just let you in, anon .. |
| 03:20.19 | Marlow | Chuji : and you can also http to that site |
| 03:20.38 | Chuji | I tried, if I go a level deeper than root, it asks for pass |
| 03:20.47 | Marlow | fishboy1669 : zaptel and zapata are rubbish .. |
| 03:20.56 | fishboy1669 | in what way? |
| 03:21.28 | Marlow | Chuji : but if you use IE, you can forget ftp :) |
| 03:21.35 | Marlow | Chuji : IE ftp is broken |
| 03:22.47 | fishboy1669 | ? |
| 03:23.12 | Marlow | fishboy1669 : you didn't specify signalling in zapata.conf |
| 03:23.33 | Marlow | fishboy1669 : signalling=fxs_ks |
| 03:23.43 | Marlow | fishboy1669 : you did do it in zaptel.conf though |
| 03:24.38 | fishboy1669 | http://pastebin.ca/4561 |
| 03:24.46 | fishboy1669 | is my old conf that worked |
| 03:25.23 | fishboy1669 | on old manchie but not on this one |
| 03:25.38 | fishboy1669 | the copy didnt copy all the guff on th zaptel.conf |
| 03:25.40 | bjohnson | cripito: must be something specific to your hardware .. too many big installs use asterisk for this to be unknown |
| 03:25.49 | fishboy1669 | sory zapata.conf |
| 03:26.02 | fishboy1669 | i did specify the signalling further up |
| 03:26.52 | Marlow | fishboy1669 : i searched for it, couldn't find it .. |
| 03:27.05 | fishboy1669 | check the 4561 settings |
| 03:27.30 | fishboy1669 | ye dont bother with the other configs from elyer i cat the fiel but still didnt get all of it to copy paste |
| 03:27.43 | fishboy1669 | its putty from a win box |
| 03:27.48 | Marlow | fishboy1669 : check http://pastebin.ca/4562 |
| 03:27.58 | Marlow | fishboy1669 : that's a simple one .. |
| 03:29.30 | Marlow | Chuji : bet my ass, that you used something crappy like IE :) |
| 03:32.05 | Marlow | Chuji : IE is majorly broken in the way it does anonymous ftp |
| 03:32.13 | fishboy1669 | still same errors |
| 03:32.36 | Chuji | marlowe: Yeah, I just flipped over to ncftp |
| 03:32.40 | Chuji | I got it now |
| 03:33.29 | Marlow | Chuji : some muppet removed my mirror for not being avail .. |
| 03:33.34 | Marlow | Chuji : from the wiki .. |
| 03:33.46 | Marlow | Chuji : and I bet he checked availability the same way |
| 03:33.57 | Chuji | Heh |
| 03:34.15 | Marlow | Chuji : funny enough was it me, that created that page initially |
| 03:34.43 | Marlow | Chuji : because ftp.digium.com allways was overloaded, not avail etc. |
| 03:35.05 | fishboy1669 | any quick new ideas befor i call it a night? |
| 03:35.23 | Marlow | fishboy1669 : yeah .. pastebin the output from ztcfg -vvvvv |
| 03:35.27 | Marlow | fishboy1669 :) |
| 03:37.49 | fishboy1669 | http://pastebin.ca/4563 |
| 03:38.11 | fishboy1669 | looks fin and dandy to me? :( |
| 03:38.46 | Marlow | fishboy1669 : indeed .. |
| 03:39.20 | Marlow | fishboy1669 : the only thing might be the order in /etc/zaptel.conf, loading tonezone first, but i don't think that matters |
| 03:39.35 | Marlow | fishboy1669 : so basically this should work .. dunno what the trouble is .. |
| 03:39.45 | fishboy1669 | arse poo |
| 03:39.49 | fishboy1669 | :( |
| 03:39.50 | Marlow | fishboy1669 : are the zaptel drivers new enough ? new enough libpri ? |
| 03:40.00 | fishboy1669 | brand new |
| 03:40.07 | fishboy1669 | cvs check out yest |
| 03:40.24 | fishboy1669 | and the config fiels are copies of a working machine |
| 03:40.27 | Marlow | fishboy1669 : and no old crap flying around ? |
| 03:40.42 | fishboy1669 | no fresh linux install yest |
| 03:41.01 | Marlow | fishboy1669 : that sounds odd, indeed |
| 03:41.01 | fishboy1669 | only thing i did wrong was make install the zaptel with out doing make linux26 |
| 03:41.07 | Marlow | fishboy1669 : ah .. wait . |
| 03:41.14 | fishboy1669 | but did a make clean on everyting and re did um all |
| 03:41.17 | Marlow | fishboy1669 : what about rights on the devices ? |
| 03:41.25 | fishboy1669 | as in the udev |
| 03:41.28 | fishboy1669 | done that |
| 03:41.37 | fishboy1669 | don all the udev changes |
| 03:41.41 | Marlow | nah .. then i've got no clue :) |
| 03:41.51 | fishboy1669 | thanks for trying mate |
| 03:41.55 | fishboy1669 | valient effort |
| 03:42.14 | fishboy1669 | at least im am trying right things |
| 03:42.26 | fishboy1669 | such an arse as its nearly there |
| 03:42.28 | Marlow | fishboy1669 : tomorrow is a new day .. |
| 03:42.43 | fishboy1669 | mmm and a day closer to dead line |
| 03:42.45 | fishboy1669 | :( |
| 03:42.52 | fishboy1669 | cheers |
| 03:42.56 | fishboy1669 | have a gud un |
| 03:42.58 | fishboy1669 | night |
| 03:44.32 | bjohnson | the wiki page for gotoif shows this example but I get a syntax error .. is this right? exten => s,1,NoOp(${CALLERID}) |
| 03:45.36 | Silik0n | *yawn* |
| 03:46.06 | *** join/#asterisk autobus (~autobus@80.172.17.73) |
| 03:47.05 | autobus | hi people |
| 03:47.10 | autobus | i speak from portugal |
| 03:47.14 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-3365.nb.aliant.net) |
| 03:47.18 | autobus | its possible help me? |
| 03:47.45 | Qwell | file: y0 y0 y0 |
| 03:47.50 | file[laptop] | hi |
| 03:49.36 | file[laptop] | autobus: private messaging me is not a good thing |
| 03:49.51 | bjohnson | hmm .. that wasn't the line that caused errors .. this gives me errors: exten => s,3,GotoIf($[${ARG7} = ""]?macro-superdial,s,5) |
| 03:50.35 | bjohnson | can someone tell me correct syntax to check if ARG7 fed to a macro has a value? |
| 03:51.07 | bjohnson | ahhh ... missed the quotes |
| 03:53.45 | *** join/#asterisk brc_ (~brian@ip24-251-178-25.ph.ph.cox.net) |
| 03:55.54 | autobus | outside of my networks, the extencions not register |
| 03:55.58 | autobus | what is a problem |
| 03:56.10 | autobus | ? |
| 03:56.31 | autobus | i open 5060 port |
| 03:56.39 | autobus | but not solve |
| 03:56.48 | bjohnson | also 10000-20000 |
| 03:57.45 | bjohnson | I want to goto another context based on certain callerids. Do I have to use a gotoif for each callerid or is there a cleaner way of doing it? |
| 03:58.09 | autobus | not |
| 03:58.13 | autobus | the context is correct |
| 03:58.32 | autobus | i creat one context for outside extencions |
| 03:58.39 | autobus | of my network |
| 03:58.39 | bjohnson | autobus: also ports 10000-20000 |
| 03:58.48 | autobus | hum |
| 03:58.53 | autobus | i trie now! |
| 03:59.10 | brc_ | HAHAHHAHAHA http://apple.slashdot.org/comments.pl?sid=136893&cid=11437792&pid=11437792&threshold=-1&mode=nested&commentsort=0&op=Change |
| 03:59.21 | brc_ | solder resistor between lines 2 & 3 - Shuffle grows full color OLED touch-screen! |
| 03:59.24 | bjohnson | I have never been able to get sip to go through a NAT router .. it may be hardware dependant |
| 04:00.50 | bjohnson | brc_: this is better: stick bent paperclip in headphone jack - Steve Jobs comes to your home and cleans your car! |
| 04:00.58 | brc_ | wow! |
| 04:01.05 | brc_ | I'm going to buy ten! |
| 04:01.11 | brc_ | does it work more then once? |
| 04:01.18 | brc_ | or does the magic smoke get out after the first time |
| 04:01.46 | bjohnson | maybe you just need to do this first: open Shuffle and cover circuit board with cream cheese, insert in USB slot - $500 USD springs from CD drive! |
| 04:02.01 | *** join/#asterisk iMediax (~user@00045a809589.click-network.com) |
| 04:02.27 | cripito | :D |
| 04:04.12 | bjohnson | damn .. * is getting the ID of my Sipura fxo port instead of the telco line call |
| 04:04.28 | hermie | are there more Laws & Order now or CSIs... so many spin-offs |
| 04:04.46 | bjohnson | can someone help with callerid issues? I know this line has telco provided callerid |
| 04:05.53 | tzanger | hermie: yeah |
| 04:06.00 | tzanger | they were cool at first but now |
| 04:06.08 | Marlow | autobus : the thing with the callerid's is quite simple .. |
| 04:06.15 | tzanger | <-- watching gone in 60 seconds |
| 04:06.28 | *** join/#asterisk robbie (~rob@210.18.225.110) |
| 04:06.29 | PTG123 | <-- watching stargate sg1 :) |
| 04:06.38 | PTG123 | Tonight is the best tv night ever :) |
| 04:06.43 | robbie | hey Faithful |
| 04:06.51 | robbie | i finaly got the time |
| 04:06.56 | PTG123 | Star Trek Enterprise, Stargate Sg1, Stargate atlantis, BattleStar Gallactica and Monk |
| 04:06.58 | PTG123 | :) |
| 04:07.00 | tzanger | meh |
| 04:07.55 | Marlow | <-- watching the gumball 3000 movie |
| 04:09.12 | hermie | <--- has a mouse in the attic |
| 04:09.26 | bjohnson | <-- actually trying to configure * |
| 04:09.53 | hermie | bjohnson, what do you think this is, #asterisk? Oh wait.... |
| 04:10.10 | Veto | My extension in Dial looks like SIP/jdoe, what does my cell look like? IAX/1xxxxxxxxx? |
| 04:10.13 | bjohnson | <-- has a callerid problem |
| 04:10.39 | bjohnson | <-- has no idea what Veto is trying to ask |
| 04:10.56 | cripito | :D me neither |
| 04:11.06 | Veto | What does a external call look like in Dial? |
| 04:11.09 | cripito | bjohnson what is the prob with caller id? |
| 04:11.19 | bjohnson | Veto: how does it get to the pstn? |
| 04:11.27 | Veto | IAX2 |
| 04:12.22 | bjohnson | cripito: * is getting my SPA 3000 id instead of the incoming call pstn CID .. maybe I should doublecheck my sip.conf |
| 04:13.16 | hermie | bjohnson, I believe the wiki holds your answer |
| 04:13.19 | bjohnson | Veto: dial(IAX2/username:account@hostname/5551234) |
| 04:13.42 | hermie | bjohnson, you must persue the 8-fold path of Google and Wiki |
| 04:14.01 | bjohnson | hermie: spa specific? |
| 04:14.07 | cripito | :)) |
| 04:14.23 | hermie | bjohnson, you have just missed my point |
| 04:14.35 | hermie | I won't hold your hand, but your answer is there |
| 04:14.48 | cripito | well frustrated with a deadlock issue going 2 bed |
| 04:14.49 | bjohnson | here is my answer .. I set callerid in sip.conf |
| 04:14.52 | cripito | tomorrow is a hard day |
| 04:15.00 | hermie | in fact, you can probably use the "i'm feeling lucky" button |
| 04:15.00 | cripito | :D |
| 04:15.36 | bjohnson | cripito: good night |
| 04:16.42 | bjohnson | damn .. nope |
| 04:16.48 | bjohnson | must be a spa setting |
| 04:17.00 | hermie | bjohnson, I believe the wiki holds your answer |
| 04:17.21 | hermie | bjohnson, I searched and found it on the first page |
| 04:18.00 | Veto | bjohnson, thx |
| 04:18.02 | bjohnson | I love playing this game |
| 04:18.24 | Juggie | callerid=asreceived |
| 04:18.32 | Juggie | ^=hint |
| 04:20.01 | hermie | Juggie: it's a sip channel |
| 04:20.23 | hermie | ~google sipura asterisk callerid |
| 04:21.05 | Veto | bjohnson, I get a "everyone is busy", but if I dial direct the call goes through to 12223334444. verbose=9 they look the same. |
| 04:21.37 | iMediax | huh? |
| 04:22.52 | *** join/#asterisk psycodad (~obiwan@2001:4060:4419:b1:0:0:0:2) |
| 04:22.57 | bjohnson | hermie: I have been following the last post at http://www.voip-info.org/wiki-Sipura+3000 but I am NOT getting the callerid .. I'm getting the display name and User ID from the SPA fxo |
| 04:23.55 | bjohnson | Veto: must be something that is different. I haven't seen a "everyone is busy" error before |
| 04:24.12 | Manipura | bjohnson, JR168S, allatchina.com $150 on top of the price for the phones when buying 7 of them. |
| 04:24.34 | *** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com) |
| 04:25.00 | Manipura | Having troubles getting gsm to work |
| 04:25.06 | *** join/#asterisk Suspect` (~jterrero@66.28.34.177) |
| 04:25.24 | Manipura | Haven't tried getting IAX to work yet. |
| 04:26.41 | *** join/#asterisk jero (~boo@modemcable040.12-81-70.mc.videotron.ca) |
| 04:27.15 | iMediax | can't get callerid from where? |
| 04:27.37 | bjohnson | Manipura: 150 total or 150 each |
| 04:28.05 | *** join/#asterisk drkool (~drkool@210.211.144.70) |
| 04:28.06 | Manipura | 150 total. |
| 04:28.20 | tzanger | Call out Gouranga be happy!!! |
| 04:28.20 | tzanger | Gouranga Gouranga Gouranga .... |
| 04:28.20 | tzanger | That which brings the highest happiness!! |
| 04:28.22 | bjohnson | iMediax: callerid from Bell line isn't getting to my *. * seems to be getting the SIP device name. Now playing with wait times |
| 04:28.26 | tzanger | what the fuck kind of spam is that?! |
| 04:28.37 | tzanger | nothing odd in headers, no attachments... |
| 04:29.03 | iMediax | ok... was just going to suggest throwing in a Wait |
| 04:29.45 | bjohnson | iMediax: I have a 2s answer delay on the SPA and thought that would do it .. |
| 04:29.50 | bjohnson | now adding a wait to * |
| 04:30.06 | hermie | bjohnson: do a sip debug and pastebin what you get, along with pastebinging relevant sip.conf... wait() in Ast won't help |
| 04:30.08 | drkool | hi all when using "moniter" to record calls and mixing with SOX all i can hear is a very scratchy sound. listining carefully i can make out the actual phone conversation in the back ground. but the disturbance is very loud. Any ideas on how to eliminate |
| 04:30.14 | hermie | bjohnson: callerid is in first packet |
| 04:31.16 | drkool | i am at my wit's end .Hoping some here can help |
| 04:32.37 | bjohnson | hermie: should I put SPA answer delay back to 0? |
| 04:33.44 | bjohnson | should exten => s,2,NoOp(${CALLERID}) show the pstn supplied callerid? |
| 04:33.59 | *** join/#asterisk naouri (bonoi@d142-59-238-42.abhsia.telus.net) |
| 04:34.01 | hermie | bjohnson: no... at least if you're using Bellcore/Telcordia/(Whatever the hell they're callingthemselves now) CallerID, which comes between rings 1-2 |
| 04:34.12 | hermie | bjohnson: pastebin that SIP debug |
| 04:34.15 | *** join/#asterisk DigiTaL (~BaYt@61.68.60.72) |
| 04:34.23 | hermie | bjohnson: and we'll see what the SPA is sending |
| 04:35.41 | Juggie | you may as well see if asterisk sees the callerid, before u try sending it to your sip phone |
| 04:37.08 | hermie | huh? he's going PTSN -> SPA -> Asterisk |
| 04:37.18 | hermie | so what he needs is a sip debug |
| 04:38.02 | jontow | hmmm, ok.. now im 100% confused again |
| 04:38.21 | jontow | say i have Originate working from the manager API for SIP channels.. and Local channels.. when they're SIP |
| 04:38.33 | jontow | so, say i Originate with Channel: being Local/668 |
| 04:38.39 | jontow | if 668 is this: |
| 04:38.40 | jero | hello |
| 04:38.53 | jontow | exten => 668,1,Dial(${TRUNK}/15555551212|60) |
| 04:38.59 | *** part/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 04:39.10 | jontow | ... why exactly would the PRI catch a hangup about 1 second after executing that? |
| 04:39.50 | jontow | big point being, if i call the extension from a SIP phone it works great, but if i let the Manager API do it, it fails with a Hangup |
| 04:40.10 | jontow | i had it working earlier and now it doesn't and i'm just plain confused.. |
| 04:40.28 | *** join/#asterisk coppice (~chatzilla@200.192.17.210.dyn.pacific.net.hk) |
| 04:41.07 | jontow | -- Executing Dial("Local/668@default-3a9b,2", "Zap/G1/15555551212|60") in new stack .... -- Called G1/15555551212 ... -- Channel 0/23, span 1 got hangup |
| 04:41.14 | bjohnson | geez .. how do you clean up all the crap that debug spits out? |
| 04:41.41 | hermie | bjohnson: just thow it all up at nick.pastebin.ca |
| 04:41.56 | hermie | bjohnson: just everything you get between incoming all and answer |
| 04:42.58 | jontow | heh.. oh |
| 04:43.01 | jontow | thats a dumb one |
| 04:43.08 | bjohnson | I'm trying to find those .. /var/log/asterisk/messages is 227MB and I'm remote to the * server via ssh over dsl |
| 04:43.10 | jontow | callerid is invalid, so it doesn't give an error, it just silently dies.. |
| 04:43.21 | jontow | my cellphone doesn't accept it if it isn't numerical :) |
| 04:43.34 | jontow | my cellphone is the 15555551212 in question |
| 04:43.36 | jontow | (lame) :) |
| 04:43.44 | JerJer | Zap/g1/${EXTEN} |
| 04:43.47 | hermie | bjohnson: ok... get into the cli, run 'sip debug on' and paste the output of that |
| 04:46.32 | jontow | neat exercise in futility that was.. but at least my host monitoring works :) .. tested on 5 hosts now |
| 04:47.54 | bjohnson | http://nick.pastebin.ca/4564 |
| 04:48.07 | jero | can the CALLERIDNUM be non digit-only ? |
| 04:48.48 | jero | ie. can I do SetCIDNum(1-800-111-2222) |
| 04:48.48 | jontow | jero; .. going out a PRI, it seems that it can be digits only, here.. |
| 04:48.48 | jontow | i *just* had that issue, describe above. |
| 04:48.59 | jontow | i had callerid set to "svcmon machinename" |
| 04:49.08 | hermie | bjohnson: SPA isn't passing callerid |
| 04:49.11 | jero | okay thanks :) |
| 04:49.21 | jontow | it worked great for internal VoIP calls (to SIP phones, at least..), but when i tried to send that to my cell, the PRI silently hungup after 1 second of trying |
| 04:49.30 | jontow | .. it dialed the number, sent the data (silently) and failed. |
| 04:53.46 | *** join/#asterisk blitzrage (~blitzrage@d141-234-145.home.cgocable.net) |
| 04:56.32 | *** join/#asterisk C11- (~jonas@p508360C1.dip0.t-ipconnect.de) |
| 04:57.05 | *** join/#asterisk PTG123 (PTG123@ip68-106-19-249.ph.ph.cox.net) |
| 04:57.07 | *** join/#asterisk PTG123_ (PTG123@ip68-106-19-249.ph.ph.cox.net) |
| 04:59.04 | *** join/#asterisk PTG123 (Preston@ip68-106-19-249.ph.ph.cox.net) |
| 04:59.26 | *** part/#asterisk PTG123_ (PTG123@ip68-106-19-249.ph.ph.cox.net) |
| 05:01.22 | Smuggs | I installed asterisk and configured it to work with IAX at FWD. I need to configure a laptop on the same network through a linksys router to use a frontend to connect with other IAX networks |
| 05:02.54 | bjohnson | the SPA 3000 is not sending the pstn cid to * .. it is substituting it's own userid. I give up for tonight. |
| 05:04.14 | Juggie | bjohnson, what are you trying to accomplish |
| 05:04.33 | Juggie | are you trying to get the incomming caller id from a PSTN call on your sip phone? |
| 05:04.44 | Juggie | or are you trying to send caller id to an outgoing call |
| 05:05.43 | Veto | <-- confused. Direct dial to a # works, but a Dial in an extension comes back as "Everyone is Busy/Congested" |
| 05:05.55 | Veto | http://pastebin.ca/4565 |
| 05:06.00 | Smuggs | yo Juggie im poor. i have asterisk installed on this fc3 machine. i was under the assumption i would be able to use a software phone (headset) to make calls |
| 05:06.10 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 05:06.21 | Juggie | Smuggs, you are able to yes. |
| 05:06.34 | Juggie | xten as a sip client should do the trick |
| 05:06.34 | Smuggs | ok cool... just wondering if its another program seperate from askerisk |
| 05:06.43 | Smuggs | im sure ill find it if i keep reading |
| 05:07.00 | Nugget | asterisk can use a local soundcard as a device, but that's more of a novelty than anything. you'll really want to use a seperate softphone app. |
| 05:07.14 | bjohnson | Juggie: trying to get callerid from sip SPA 3000 fxo port to * to send certain caller to a different context |
| 05:07.28 | Juggie | you will need to get a sip client, xten is decent (its windows i assume u have a windows box) you'll need to configure sip.conf for your sip client, as well as xten and then make your call. |
| 05:07.36 | bjohnson | the SPA 3000 is sending the device info assigned from within it's web config |
| 05:08.04 | Juggie | is SPA3000 a sip phone? |
| 05:08.15 | Juggie | nm |
| 05:08.16 | bjohnson | a dual fxo/fxs unti |
| 05:08.18 | bjohnson | unit |
| 05:08.55 | bjohnson | it's something within the device config .. but I can't find it. It is supposed to work according to the voip wiki |
| 05:08.59 | Smuggs | i have two machines. one fc3 w/ asterisk and one laptop winxp pro. i wanted to make calls through the network to the asterisk box then out to somwhere else |
| 05:09.02 | bjohnson | I'm going to bed |
| 05:09.19 | Chuji | bjohnson : when you wake up tomorrow |
| 05:09.23 | Chuji | bjohnson : try this |
| 05:09.33 | Chuji | bjohnson : Works fine on my spa3k |
| 05:09.50 | Chuji | bjohnson : the bottom example is what you are after |
| 05:10.16 | Juggie | Smuggs, where is your outgoing conection comming from do you have a modem or whatnot in your asterisk box? |
| 05:10.16 | Chuji | http://www.voip-info.org/tiki-print.php?page=Sipura+3000 |
| 05:10.28 | Smuggs | Nugget, I was reading about the pulvor.communicator. wondering if im on the right path trying to use this app on the winxp laptop |
| 05:10.57 | Smuggs | modem > router > astericks box + laptop |
| 05:11.00 | brc_ | ~eyebeam |
| 05:11.04 | brc_ | ~xpro |
| 05:11.12 | brc_ | ~xlite |
| 05:11.13 | jbot | it has been said that xlite is at download xlite at: http://snipurl.com/5tgi | and see sample configs at http://snipurl.com/5tgj, or xlite is a free SoftPhone (software phone, requires no hardware) from xten inc, |
| 05:11.18 | Nugget | I have no idea what's available for a linux desktop. in win32 I hear that x-lite and sjphone are decent. |
| 05:11.22 | Nugget | linux is poo. :) |
| 05:11.28 | brc_ | yup |
| 05:11.33 | Funbags | anyone know why I am getting a really annoying clicking sound on my sipura 2000, even at dialtone |
| 05:11.53 | brc_ | faulty hardware |
| 05:12.06 | brc_ | try another phone, then replace the sipura |
| 05:13.06 | Smuggs | Juggie, i can install a modem. i have a few lying around. I figured I'd be able to configure it to work with the asterisk box |
| 05:13.19 | Smuggs | read somewhere its hard though because i need to know AT commands |
| 05:13.24 | Juggie | Smuggs, you need a special modem |
| 05:13.30 | Juggie | not just any modem will work with asterisk |
| 05:13.32 | Smuggs | jah i read about special modem |
| 05:13.45 | Juggie | you will have to search the wiki on that one |
| 05:13.55 | Smuggs | some dood said its possible to setup but spending $100 on special modem > 3 days of trying to config modem |
| 05:13.57 | Juggie | you need to get asterisk answering calls first |
| 05:14.06 | *** join/#asterisk firestrm (~vince@S010600047577bccd.gv.shawcable.net) |
| 05:14.09 | Juggie | it requires a special chipset |
| 05:14.19 | Juggie | or a certain chipset i should say |
| 05:15.25 | brc_ | which has been discontinued |
| 05:15.46 | Juggie | you cant even get them from digium? |
| 05:16.22 | firestrm | Juggie, what are we discussing? |
| 05:16.23 | Veto | why does CLI*> dial 1111 work and from the SIP phone at 1111, 1111 fails (thinking to myself) |
| 05:17.17 | Veto | who won the Ind vs. Mia game? |
| 05:17.27 | Juggie | firestrm i have NO idea i dont remember talking to you |
| 05:18.02 | firestrm | Juggie, you were discussing chipsets.. i was curious.. sorry.. |
| 05:18.09 | *** join/#asterisk asjoyner (~asjoyner@dargo.trilug.org) |
| 05:18.16 | *** join/#asterisk labo (~ariel@hankster.caliente.com.mx) |
| 05:18.37 | Juggie | oh, i just said asterisk requires a special modem which has a certain chipset |
| 05:18.53 | Juggie | you can get it from digium, you might find one that will work if you dig around but i dont know anything about that |
| 05:18.58 | Juggie | look on the wiki |
| 05:19.13 | firestrm | Juggie, oh, ya, you can get them for 29.00 if you know which one.. they are available.. i just bought one |
| 05:19.19 | asjoyner | In the latest stable build, just compiled and installed, asterisk is crashing when it tries to play music on hold (with custom: ... rawplayer) -- Is anyone else seeing this behavior? |
| 05:19.20 | Nugget | digium isn't selling those cards any more. |
| 05:19.33 | *** join/#asterisk odieflocon (~Odie@S01060011953994ee.cg.shawcable.net) |
| 05:19.38 | odieflocon | hello all. |
| 05:20.32 | Juggie | Nugget what do they sell now just the 4 port boards? |
| 05:20.36 | firestrm | Nugget, no but lots still do.. they re just recognised as clone cards. If you have an orig digium card you can clone the serial eeprom which contains the pci information over to the clone cards, and they will come up as orig hardware |
| 05:20.58 | Veto | ~paulc |
| 05:21.05 | Veto | jbot: last paulc |
| 05:21.17 | Veto | bah |
| 05:21.19 | Nugget | Juggie: correct, only the TDM400P now. |
| 05:21.40 | Nugget | firestrm: there aren't lots of digiums, just one digium. and that's all I was saying -- digium doesn't sell those cards any more. |
| 05:22.12 | Juggie | ~seen paulc |
| 05:22.13 | jbot | paulc <paulc@S010600062586a0b4.vc.shawcable.net> was last seen on IRC in channel #asterisk, 3h 57m 25s ago, saying: 'sudoer: give me a few minutes cos I'm just setting up a new IP600.. but I can probably send you a template file'. |
| 05:22.19 | Juggie | see easy |
| 05:22.29 | Veto | thx juggie |
| 05:22.37 | firestrm | Nugget, mine was made by encore.. $29.00 down the street at the local computer shop. as far as i can tell. digium just had special data put into the pci ident eeproms |
| 05:22.51 | Nugget | I'm aware that the clone cards are still available. |
| 05:24.16 | firestrm | im know somone who is in the process of cloning the txm400 fxs cards.. apparently there is nothing to them, although i have never seen them so this is just second hand knowlege.. |
| 05:25.02 | Juggie | thats fine but please support digium |
| 05:25.09 | tzanger | firestrm: do you feel that they're too expensive? |
| 05:25.21 | Nugget | I've never heard of a txm400. is it just a single port fxs? |
| 05:25.22 | tzanger | or do you just want to bite the hand that feeds you? |
| 05:26.03 | Juggie | i have a 4span t1/e1 card and a 1port t1/e1 so i'm in the clear ;) |
| 05:26.06 | JerJer | prolly pulling another atacomm |
| 05:26.15 | firestrm | tzanger, the reason i dont own one is that they are too epensive for my budget. maybe for medium size business, but im doing this on my own coin to learn.. |
| 05:27.22 | firestrm | tzanger, perhaps when i learn enough, i will show digium my appreciation by purchasing their hardware for any of my customers. |
| 05:27.43 | tzanger | firestrm: a TDM411P is what, $150? |
| 05:27.53 | firestrm | but for now i need to be able to do this on the cheap. |
| 05:27.55 | tzanger | I bet your video card cost more |
| 05:28.01 | Smuggs | jah im poor. teach me the way firestrm .. im just some dood in my basement trying to set this up |
| 05:28.04 | firestrm | nope.. $40.00 |
| 05:28.07 | tzanger | VOIP and cheap hardware don't mix |
| 05:28.15 | tzanger | I am just some dude doing this too |
| 05:28.26 | Juggie | work pays for my shit :) |
| 05:28.28 | tzanger | if you want to start playing, get a Sipura |
| 05:28.29 | Smuggs | your just some dood w/ more money than me |
| 05:28.46 | coppice | our food budget pays for my shit :-) |
| 05:28.55 | firestrm | <PROTECTED> |
| 05:29.26 | tzanger | firestrm: exactly. Find a PAP2NA or something then |
| 05:29.38 | Juggie | tzanger. if someone is ACTUALLY under a budget which doesnt allow for a 150 expenditure i can understand that, so long as they support the cummunity thats fine by me |
| 05:29.40 | Veto | hmm, dial <extension> works as root but not as user asterisk. |
| 05:30.06 | Juggie | more people learning asterisk isnt a bad thing |
| 05:30.07 | tzanger | I understand that too |
| 05:30.16 | tzanger | but asking to clone the card isn't cool |
| 05:30.25 | tzanger | it's not a rebranded card, it's one of Digium's own design |
| 05:30.33 | firestrm | if someone is going to offer me the same card as a tdm400 for $50.00 sorry, I'll take it.. that just market competition.. monopolies never last.. especially with engiunus ppl in places like china and india, who are willing to do anything for a buck |
| 05:30.46 | Juggie | tzanger, the original card was just a modem with special drivers |
| 05:30.47 | tzanger | firestrm: I agree |
| 05:30.55 | tzanger | Juggie: that's the X100P and it's FXO |
| 05:31.11 | Juggie | tzanger, thats all you need to learn |
| 05:31.23 | firestrm | but for a business.. i probbly would stay on the safe side a go digium.. i dont want to fix some clone hardware at 3:00am |
| 05:31.30 | tzanger | generally the same guy who will buy it on the cheap "to test it" is also the guy who'll clone the card to maximize profits |
| 05:31.41 | tzanger | firestrm: but you *just* asked for a clone TDM400P FXS module |
| 05:31.43 | tzanger | so which is it |
| 05:31.58 | tzanger | go clone because it's cheap or go digium to get the support |
| 05:32.26 | Juggie | tzanger, i dont support cloneing stuff that is digiums design |
| 05:32.37 | Juggie | however if someone wants to fin the X100P thats not digiums to learn |
| 05:32.38 | Juggie | then go nuts |
| 05:32.42 | tzanger | I understand your position, firestrm, I do. But you either need to save your nickles if this is something you really want to get in to, or start with something cheaper like the PAP2 |
| 05:32.44 | Juggie | *find |
| 05:32.46 | firestrm | for myself, i would take a clone, but if someone was paying me.. i would either have to have alot of sucessful field tesing of that clone board, or i would go with somehing else that has, like digium |
| 05:33.07 | Smuggs | ok mr. tzanger u sold me on this pap2 thing. what is it |
| 05:33.10 | tzanger | firestrm: yeah but what if they come to you and say "gee this is neat tech but it's just too pricey, are you sure there's nothing cheaper?" |
| 05:33.16 | *** join/#asterisk tris (~tristan@camel.ethereal.net) |
| 05:33.29 | tzanger | then you are faced with losing the sale or doing an end-run around digium |
| 05:33.45 | Smuggs | wow your right |
| 05:33.49 | Smuggs | im evil |
| 05:34.05 | tzanger | and I'm telling you from experience that the customer who's nickle and diming you at the start is ALSO and WITHOUT FAIL the same customer who will want your support and not want to pay for it |
| 05:34.19 | tzanger | jbot google PAP2 asterisk |
| 05:34.52 | tzanger | Smuggs: I never said you were evil |
| 05:34.54 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 05:35.01 | tzanger | but this is a numbers game |
| 05:35.02 | Smuggs | i was just letting u know |
| 05:35.13 | firestrm | tzanger, let me put it this way.. im a pilot.. thats my day job. now if i have to fly instrument, and im by myself, i'll take a single engine aircraft. but if im flying passengers.. its not my risk..multi engine no exceptions.. same as business.. yes there is cheaper, no i wont sell it to you because im the guy who is responsible for its reliablility (at least for a yean anyways) |
| 05:35.39 | firestrm | s/yean/year |
| 05:35.55 | tzanger | firestrm: understood, and if this is true it is commendable |
| 05:36.01 | Juggie | anyways support the hands that feeds you |
| 05:36.05 | *** join/#asterisk Darwin35 (~Darwin@c-24-3-241-22.client.comcast.net) |
| 05:36.26 | Juggie | thats all, i can understand testing, but production you go legit |
| 05:36.34 | tzanger | Not being a pilot myself I was under the impression that all multi-engine aircraft were flown more by instrument than visual |
| 05:37.18 | tessier__ | tzanger: Multi? no. |
| 05:37.31 | firestrm | Juggie, i'll try.. i just wish digium would cut us learners some slack.. its to their benifit that more learn. i think that they should allow someone to buy 1 (and only 1) card of what ever they sell as a student package, at cost.. |
| 05:37.37 | tessier__ | tzanger: Even airliners are usually IFR but not necessary. I fly twins VFR all the time. |
| 05:37.48 | Veto | any reason "*CLI> dial 1111" would work as "root" and not as "asterisk" if /var/spool/asterisk, /var/lib/asterisk, /var/run/autodial.ctl and /dev/zap were asterisk:asterisk? |
| 05:38.10 | tessier__ | firestrm: Where do you fly? |
| 05:38.16 | Nugget | all flight above 18000 feet is technically "instrument" flight, but no, there's nothing inherent in a multi-engine plane that makes it more difficult to fly under visual flight rules. |
| 05:38.21 | tzanger | firestrm: $150 for an FXS+FXO+support is not a bad deal |
| 05:38.28 | coppice | I used to design bits of aircraft |
| 05:38.30 | coppice | never trust one :-) |
| 05:38.35 | tzanger | interesting |
| 05:38.44 | tzanger | I never knew there wer eso many * pilots |
| 05:38.46 | firestrm | tzanger, no we even fly heavys visual <180000 feet if weather permits.. its much easier, and it takes strain off of an allready busy system.. |
| 05:38.54 | tzanger | coppice: hahaha lots of softdsps in them eh? :-p |
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| 05:39.22 | *** join/#asterisk labo (~ariel@hankster.caliente.com.mx) |
| 05:39.27 | Veto | firestrm, that's mighty fair of you @ 180k ft |
| 05:39.38 | Juggie | hah |
| 05:39.42 | Veto | :) |
| 05:39.43 | Juggie | 18k perhaps |
| 05:39.47 | Juggie | 180k is a little high |
| 05:39.55 | Brixius | firestrm: What cost, someone put hard effort and time in designing that card, people forget that, they think the "cost" on a card is what it takes to have a circuit board made and componants mounted on it, that's not the case. |
| 05:40.00 | coppice | it use to worry me watching the fitters put cabling into the ducts with their boots. |
| 05:40.00 | firestrm | tzanger, if $150.00 is true then they have dropped their prices.. last time i checked for what i need, i was looking at $280.00 by the time i get it into canada.. |
| 05:40.14 | firestrm | veto.. sorry typo, 18k feet |
| 05:40.17 | tzanger | firestrm: the dev kit light? |
| 05:40.42 | *** join/#asterisk NormAst (~NormAst@207.245.0.131) |
| 05:40.50 | Veto | firestrm: lol, no worries...I've been chasing typos lately! |
| 05:41.09 | firestrm | tzanger, dev kit lite, last i checked, landed in canada would cost me $280.00 |
| 05:41.11 | tzanger | yup |
| 05:41.13 | tzanger | $195US sorry |
| 05:41.40 | tzanger | if you don't need the FXO interface it's cheaper |
| 05:42.04 | Juggie | firestrm. why not just use a x100p you can get one for 20$ |
| 05:42.15 | tzanger | FXO ports are always expensive and I'm not sure why, I don't think Part68 is that hard to achieve nor test for (I work for an industrial motion control manufacturer, we do UL testing all the time) |
| 05:42.20 | tzanger | Juggie: he wants the FXS |
| 05:42.27 | Juggie | oh woops |
| 05:43.09 | odieflocon | does anybody know if Digium is going to make analog boards with more ports then 4? |
| 05:43.10 | firestrm | tzanger, im still concidering it.. im just i just moved, so things are tight right now.. but i want to play with my new toy.. |
| 05:43.25 | Smuggs | let me play w/ it too |
| 05:43.32 | Veto | firestrm: what do you fly? |
| 05:43.43 | coppice | odieflocom: they did some work on one a long time ago, bit I think they dropped it |
| 05:43.52 | odieflocon | hmmm. |
| 05:43.56 | Smuggs | veto a better question would be "what gets you high" |
| 05:44.02 | odieflocon | it would be nice if I could get a board with 8 or 12 ports on it. |
| 05:44.04 | tzanger | firestrm: I hear ya |
| 05:44.09 | firestrm | veto, anything i can log time on :) but usually you see me in the right seat of an AC-A320 |
| 05:44.19 | InfraRed | 05:43 < Smuggs> veto a better question would be "what gets you high" |
| 05:44.27 | InfraRed | i can think dope |
| 05:44.27 | Veto | lol smuggs |
| 05:44.33 | Juggie | odieflocon. for more ports you add another board or get a t1/e1 board with a channel bank |
| 05:44.33 | coppice | odieflocom: quite. 4 is a bit limiting |
| 05:44.44 | firestrm | veto, i started off with CMA flying a beech 1900d |
| 05:44.51 | firestrm | them moved to AC |
| 05:45.04 | Veto | firestrm: my brother flies a319 as cpt |
| 05:45.10 | *** join/#asterisk DigiTaL (Jalepeno@61.68.2.83) |
| 05:45.14 | odieflocon | juggie, but I can only add 4 boards to a system. |
| 05:45.23 | firestrm | veto, what airline? |
| 05:45.32 | Veto | firestrm: AmWst |
| 05:45.40 | Brixius | firestrm: how about a sipura 2000 it has 2 fxs ports and is only $75, or the 3000, 1 fxs and 1 fxo it's $99 |
| 05:45.44 | odieflocon | or I have to get a $1,000. Channel Bank. |
| 05:45.51 | Veto | firestrm, sorry...saw the a320. |
| 05:46.19 | odieflocon | the thing that bugs me is that these cards are limiting. |
| 05:47.01 | odieflocon | it would make my life alot easier if I could put 8 ports on one card. |
| 05:47.25 | *** join/#asterisk chrisf0rd (~chris@cvg-165-100-203.cinci.rr.com) |
| 05:47.28 | coppice | odieflocom: they were working on a full length PCI taking 12 or 16 (forget which) of the little FXS/FXO modules. I think that would have suited a lot of people |
| 05:47.48 | chrisf0rd | Hello |
| 05:47.50 | odieflocon | then I could build a nice soho box. a whole lot cheaper. |
| 05:48.00 | chrisf0rd | I have a very interesting question |
| 05:48.09 | odieflocon | I could sell a lot of those. |
| 05:48.14 | chrisf0rd | I have Voice pluse working on the inbound side |
| 05:48.18 | chrisf0rd | I can call in |
| 05:48.30 | chrisf0rd | but I have had nothing but trouble configuring my out bound |
| 05:48.44 | chrisf0rd | I can not seem to get passed All Cirtus are busy |
| 05:48.56 | blankman | odieflocon, you can look at some of the other boardvendors for higher density ... |
| 05:49.04 | chrisf0rd | can anybody lay some info on me |
| 05:49.14 | odieflocon | yeah, but I would really like to support digium. |
| 05:49.33 | coppice | blankman: like what other vendors? |
| 05:49.52 | chrisf0rd | Any advise would be well recieved |
| 05:50.13 | odieflocon | there are others out there if you look |
| 05:50.17 | chrisf0rd | 3 hours at this and still cannot figuare it out did everything Voice pluse said to do on thier web site |
| 05:50.20 | odieflocon | on the * website |
| 05:50.46 | odieflocon | l8r all |
| 05:50.54 | odieflocon | time for World of Warcraft. |
| 05:50.57 | coppice | odieflocom: voicetronix have a larger card, but it has some limitations. I don't know any others |
| 05:51.01 | blankman | coppice, I forget the name, but there was two or three other vendors that came through the lab here when we were testing... |
| 05:51.51 | blankman | coppice, there was a 12 and 16 port version. The issue is that not all the features of * are supported then ... and you have to monkey with stuff alot more ... Least we had to when we did testing ... We ended up buying 1 and 4 port t1 boards from digium. |
| 05:52.39 | blankman | ~seen drkool |
| 05:52.40 | jbot | drkool <~drkool@210.211.144.70> was last seen on IRC in channel #asterisk, 1h 21m 24s ago, saying: 'i am at my wit's end .Hoping some here can help'. |
| 05:53.00 | coppice | the 12 port dialogic is sort of supported, but I would be crazy to use one |
| 05:53.13 | chrisf0rd | Am I voiced can you see what I am typing |
| 05:53.50 | Smuggs | jah |
| 05:54.02 | blankman | coppice, yeah, crushing an ant with a cannonball :-) |
| 05:54.31 | blankman | Has anyone gotten the new IAX encryption to work? |
| 05:54.35 | coppice | no, its latency just sucks |
| 05:54.47 | blankman | I was messing with it today and couldn't get it working * to *. |
| 05:55.17 | blankman | Why is Mark encrypting the "whole" packet instead of just the signaling? |
| 05:55.35 | coppice | blankman: i don't think it is intended to work yet. as far as I know it is just some plumbing right now |
| 05:55.46 | chrisf0rd | I set up Voice Pulse today and am getting in bound calls but I cannot make outbound calls is there a switch I am not using right |
| 05:56.17 | chrisf0rd | I keep getting All Circuts are Busyy |
| 05:56.20 | Brixius | blankman: when you say whole packet, you mean data portion of packet, correct, not ip headers too? |
| 05:56.54 | blankman | Brixius, yeah, the payload for the frame or mini frame as the case would be ... |
| 05:57.55 | blankman | Has anyone heard of a provider for VoIP that is provider agnostic yet? |
| 05:58.21 | blankman | Meaning they let you choose which PSTN provider to use on-the-fly? |
| 05:59.25 | chrisf0rd | When somebody gets a chance I need help please..I am dying here. (-; I am completely stumped |
| 06:00.26 | Veto | so I can direct dial a number on my sip phone XYYYZZZNNNN, but if I put a Dial XYYYZZZNNNN in my extensions.conf, i get an error from my IAX provider. |
| 06:01.05 | Veto | I picked a bad week to stop sniffing glue... |
| 06:01.29 | Brixius | Veto, what do your 2 dial command's look like? |
| 06:01.48 | *** join/#asterisk meshugga (philip@loeblich.linuxteam.at) |
| 06:01.57 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 06:02.08 | *** join/#asterisk freat[laptop] (~freat@node-40242662.mdw.onnet.us.uu.net) |
| 06:02.20 | Veto | If anyone has a chance, it's extension 1113...I've cut my extensions.conf down to the minimum: http://pastebin.ca/4567 (brixius) |
| 06:02.36 | Veto | line 39 |
| 06:02.45 | jetscreamer | greetings my minions |
| 06:02.51 | Veto | direct dial work for that same number brixius |
| 06:03.35 | Brixius | veto, why the extry 2 comma's at the end of the dial command? |
| 06:03.45 | Brixius | s/extry/extra |
| 06:04.12 | Veto | brixius, trying to make it work...same effect with none or even with a Dial,XYYYZZZNNN |
| 06:05.01 | Veto | brixius, I was trying to fully qualify the call (with empty args) |
| 06:05.56 | Brixius | veto, what happens if you change it to goto(default,12815387577,1) |
| 06:06.01 | freat[laptop] | exten => _1NXXNXXXXXX,2,Dial(IAX2/XXXX@voipjet/${EXTEN}) |
| 06:06.22 | freat[laptop] | my prio 1 just sets the CID |
| 06:06.46 | firestrm | Veto, sorry we were talking about your amwest flying brother when i got a phone call.. I dont know much about flying for amwest, they are mostly southern USA are they not? |
| 06:07.57 | *** join/#asterisk lattice (~lattice@dsl017-054-176.sfo1.dsl.speakeasy.net) |
| 06:08.15 | chrisf0rd | brb |
| 06:08.17 | *** part/#asterisk chrisf0rd (~chris@cvg-165-100-203.cinci.rr.com) |
| 06:08.17 | Veto | brixius, it F**King work, thx! why does that work and not the dial, if you have a sec??? |
| 06:09.02 | Brixius | not sure, I don't know why your dial string isn't working, let me test something simlear on my box. I just figured that would be a work around. |
| 06:09.06 | Veto | firestrm, yea...no biggie...I just saw the a320 in your bird string and it set off a buzzer. |
| 06:09.24 | coppice | Veto: one of the *really* big nasties in * is when you do the wrong thing it never ever gives you a clue what you did wrong |
| 06:09.58 | firestrm | coppice, tell me about it.. i spent 4 hours on a typo today.. |
| 06:10.00 | Veto | coppice, indeed. I will not even APPROACH my nat issues...we're staying to basics. |
| 06:10.24 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 06:10.24 | *** join/#asterisk beto75 (~hav@201.128.177.84) |
| 06:10.41 | beto75 | hello guys |
| 06:11.16 | Brixius | I've spent many hours tring to fix something, going on many tangents just to find a type in my origional config hours later. |
| 06:11.32 | Brixius | sometime's day's later |
| 06:11.39 | Brixius | :( |
| 06:11.56 | beto75 | excuse me , guys I have heard horror stories about motherboard compatibility and the TDM400 stuff,, I need to have some of them , but ,, what motherboard suits? can i put several of those (3) in the same board? |
| 06:12.39 | firestrm | beto75, from what ive "read" the more important issue with mobo's is irq sharing.. |
| 06:12.39 | Veto | brixius, I was on the right track with the orig, right? |
| 06:12.42 | pimpwell | what's TDM? |
| 06:12.58 | Nukemizer | Can you have Astrisk be un NAT'ed but use SIP softphones offsite behind routers being NAT'D ? |
| 06:13.02 | Veto | Time Division Multiplexing? |
| 06:13.08 | pimpwell | k |
| 06:13.24 | Veto | pimpwell, that's from memory...may be wrong. |
| 06:13.47 | coppice | Nah, TDM is just short for tedium |
| 06:13.47 | firestrm | beto75, some of the cheaper boards, will do stupid things like irq miltiplexing.. those are the ones to stay away from |
| 06:13.49 | Veto | Nukemizer, maybe...i've had hell with it. |
| 06:14.22 | Nukemizer | so really SIP is only good on the LAN ? yikes |
| 06:14.26 | *** join/#asterisk Sitxu (~sitxu@200.82.228.164) |
| 06:15.07 | Brixius | as far as I can tell, I'm setting up the same type of thing with voip jet on my system to see what happens. |
| 06:15.21 | firestrm | beto75, when everything starts up and post display's irq useage, you want all the digium cards on different irq's , and ESPECIALLY dont share irq's with nic cards.. |
| 06:15.23 | *** join/#asterisk LeAfAr (~unix@16stb16.codetel.net.do) |
| 06:15.32 | firestrm | thats all know.. |
| 06:16.17 | Veto | nukemizer, it's just weird. I have nat'ed sip phone@1111. I can dial my DID (IAX) and get my phone, can dial <num> from phone and get <num>. Can't dial myself. |
| 06:16.53 | Veto | nukemizer, and 1112 can't dial 1111, but both can dial the world. |
| 06:16.58 | firestrm | bbiab.. |
| 06:17.11 | Brixius | veto: it works for me, I have exten => 6048,1,dial(IAX2/nnn@voipjet/18005551212) |
| 06:17.23 | Nukemizer | Veto: that is odd |
| 06:17.48 | Veto | brixius, I get: (sec to get msg) |
| 06:17.57 | beto75 | firestrm: I only have 1 NIC on a dual processor ,, I thinnk is not those cheap board,, I hope it may work :) |
| 06:18.37 | Brixius | veto: I've seen some weird things in the past, try deleating the entire line, then recreating it. |
| 06:18.45 | Veto | excuse the 4 lines: |
| 06:18.50 | Veto | -- Call accepted by 216.118.117.46 (format ulaw) |
| 06:18.50 | Veto | <PROTECTED> |
| 06:18.50 | Veto | <PROTECTED> |
| 06:18.51 | Veto | <PROTECTED> |
| 06:18.51 | Veto | <PROTECTED> |
| 06:19.57 | wolfson | did you mean to call 1? |
| 06:20.30 | coppice | of course they are all busy/congested. its winter. they all have flu :-) |
| 06:20.56 | Veto | wolfson, that's apparently a voipjet channel or such...get same/similar (1-4) on a direct dial that completes. |
| 06:21.03 | Brixius | I have a feeling there is some weird character in the line, perhap's something that's unprintable in vi or whatever editor you are using. |
| 06:21.21 | *** join/#asterisk thetalon (~thetalon@pcp05736786pcs.norstn01.pa.comcast.net) |
| 06:21.28 | *** join/#asterisk Ac|d-ra|n (~feo@224tdev09.codetel.net.do) |
| 06:21.35 | wolfson | veto, can you paste the cli log of the dial command? |
| 06:22.07 | Nukemizer | If one were to attempt port forwarding, what port ranges are required. |
| 06:22.15 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 06:22.46 | Veto | brixius, :set list in vi shows nothing weird. |
| 06:23.11 | Veto | wolfson, completed direct dial from sip or failed via Dial(xxx)? |
| 06:23.47 | Brixius | veto what does the called line on the console show, (the line just before the call accepted line. |
| 06:27.13 | Brixius | wolfson: the iax2/voipjet/1 means it's the first channel to voipjet, not the # dialed. mine show's iax2/voipjet/5, but the called # shows the full # passed to them. |
| 06:27.27 | Veto | birxius, wolfson: http://pastebin.ca/4569 |
| 06:27.53 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 06:28.43 | Veto | first is sip -> extension 1113 via Dial(XXXXXXXXXXX), second is via direct dial from sip phone to XXXXXXXXX |
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| 06:29.27 | Veto | birxius suggestion of goto(default,xxxxx,1) works for extension 1113. |
| 06:30.57 | *** join/#asterisk jterrero (~jterrero@66.28.34.177) |
| 06:31.03 | Brixius | it works, but it's really a work around, not really a fix per say. |
| 06:32.36 | Veto | brixius, I retyped the exten => 1113,1,Dial(IAX2/XXX@voipjet/12815387577) by hand and same issue. |
| 06:34.01 | Veto | so the workaround says, goto default and do what you need to do for this (xxxyyyzzznnnn) number @ 1st step, which happens to grab my voipjet stuff? |
| 06:34.16 | Brixius | yep |
| 06:34.32 | Brixius | if you didn't want to set callerid you could goto step 2 instead |
| 06:35.14 | Veto | gotcha...but something is still fuxored in my extensions.conf if it works for you and not me. (1.0.4 on rhel3) |
| 06:36.33 | Brixius | I'm running CVS-HEAD-01/12/05-12:28:13 on rh9 |
| 06:37.21 | *** join/#asterisk Inv_arp (junya@adsl-8-230-72.mia.bellsouth.net) |
| 06:37.26 | Brixius | does your show dialplan show it correctly? |
| 06:37.29 | Veto | I need to redo my extensions.conf anyways...I've been mangling the sample one for a week. only way I made the postbin readable was with a grep -v "^;" |
| 06:37.31 | *** part/#asterisk thetalon (~thetalon@pcp05736786pcs.norstn01.pa.comcast.net) |
| 06:38.08 | Veto | brixius, emm...I'm to noob to know how to find/show/report my dialplan. |
| 06:38.27 | Brixius | show dialplan from the console |
| 06:39.07 | Brixius | ex: mine shows '6048' => 1. dial(IAX2/834@voipjet/18005551212) [pbx_config] |
| 06:39.25 | Brixius | doh, |
| 06:39.30 | Veto | . |
| 06:39.33 | Veto | erase! |
| 06:39.55 | Brixius | you can't use it with out the hash anyway's |
| 06:40.01 | Veto | mine looks like: '1113' => 1. Dial(IAX2/123@voipjet/12815387577) [pbx_config] |
| 06:40.30 | Veto | right, it's 256bit anyways(?) |
| 06:41.24 | Veto | just habit on your/my part...and that's a partner of mine who's out on the town (the number) so we can call him and if he answers...make fun of him =P |
| 06:41.28 | Brixius | ya, and it's md5 |
| 06:41.48 | Veto | btw, that's with your goto fix. |
| 06:41.59 | Veto | let me see without. |
| 06:42.46 | Brixius | I don't see why it won't work, have you tried any other phone #'s in the dial command to test it with? |
| 06:42.47 | Veto | I lied, that was with Dial(), this is with your fix: |
| 06:42.52 | Veto | <PROTECTED> |
| 06:43.19 | Veto | Dial() fails, goto works...who knows. |
| 06:44.44 | Brixius | I don't have any clue as to why that would be, and I can't do alot of testing on my system cause it's in production and if I break it, people start to get mad really fast.... |
| 06:44.55 | Veto | understood. |
| 06:45.21 | Veto | hey man, thanks a ton for the help. I was stumped with what I thought (and apparently is) proper semantics. |
| 06:45.56 | Veto | if you need any sendmail/postfix/bind help...let me know...I owe you one. |
| 06:46.57 | Brixius | I reinstalled my test/lab system on fedora yesterday and don't have it all set back up yet. I'm thinking about upgrading my prod box to fedora, but have to make sure there are no gotya's in the process. |
| 06:47.51 | Veto | or nis/nas/firewall/etc...but not * :) |
| 06:48.21 | Veto | just had some issues with deploying a fc box as a stopgap, ended up rolling back. |
| 06:50.17 | *** join/#asterisk meshugga (~philip@62.99.211.227) |
| 06:50.33 | Veto | if you use autofs (which we do quite a bit), be careful...it likes to export with --no-report-v2 and --no-report-v3 (or the like) |
| 06:50.54 | Brixius | Veto, what firewall's do you work on iptables/ipchains for the most part? |
| 06:51.15 | Veto | iptables, but have experience with pix |
| 06:51.52 | Veto | well, i've had to deal with most of them...but I don't know them. |
| 06:52.10 | Brixius | ok, I havn't really used iptables, and I have some exp with pix(I don't like them), I usually use Checkpoint |
| 06:52.32 | Veto | which ckpt? |
| 06:52.44 | Brixius | ngai r55 |
| 06:53.10 | Brixius | or edge for the soho's |
| 06:53.24 | Veto | ugh... |
| 06:55.15 | Veto | i'm a bigot, I admit. I really like OSS FW's. I like FBSD ipfilter and linux iptables. |
| 06:55.34 | suma | hi |
| 06:55.43 | suma | i'm having problem with TDM400P |
| 06:56.08 | suma | whenever i receive call, once the call is finished, it is not automatically hanging up |
| 06:56.09 | Veto | i deal with customers pix as required: since most of them think PIX = my world is safe. |
| 06:56.53 | Brixius | they've been mislead by cisco and drank too much coolaid... |
| 06:57.21 | Veto | but the cisco coolaid runs strong in the pix line Luke. |
| 06:58.38 | nullogic | hi |
| 06:58.45 | Veto | suma, I'd love to help...but I don't know jack about your issue :/ |
| 06:59.11 | coppice | PIX was bought in by cisco, so it was probably cisco drinking the pix coolaid that made them buy it in the first place :-) |
| 06:59.14 | nullogic | the one thing positive i can say about the pix (well fwsm) is that they have the port density issue solved... |
| 06:59.42 | Veto | coppice, indeed. I still think it's bsd on the floppy/boot device! |
| 06:59.55 | Brixius | this is true, I hear that cisco has some new stuff comming out with there self defending networks, I'm kindof anxious to see what it's all about. A friend of mine who worked for CP and hated cisco pix was drawn away becsuse of some of the stuf he saw in the new cisco geer. |
| 07:00.27 | nullogic | i work with both and can say that cisco is finally getting into the security game.. |
| 07:00.37 | coppice | self defending networks? The new seven samurai line? |
| 07:00.40 | nullogic | some real neat stuff coming soon.. |
| 07:00.43 | nullogic | haha |
| 07:01.38 | subx | ANyone want to help with a broadvoice system? |
| 07:01.58 | nullogic | help how? |
| 07:02.10 | subx | well I can not get calls in or out |
| 07:02.26 | subx | and when I try to update the chan_sip.c it blows errors |
| 07:02.44 | nullogic | i a real newbie but asterisk -vvvvvv -g -dddddd -cr has really help me |
| 07:02.56 | nullogic | s/i/i'm |
| 07:03.29 | subx | I got the ATA186 to connect with the system but no broadvoice connection, yet |
| 07:04.12 | Brixius | Someone e-mailed me a bunch of saying's Here's one I like "Some people are like Slinkies.....not really good for anything, but you still can't help but smile when you see one tumble down the stairs. " |
| 07:04.46 | subx | hey that is how I feel about myself at this moment! ') |
| 07:05.15 | Brixius | this one is good too. "In the 60's, people took acid to make the world weird. Now the world is weird and people take Prozac to make it normal." |
| 07:05.36 | nullogic | subx, what does the the logs show? |
| 07:05.37 | subx | keep'em comin' |
| 07:05.56 | Brixius | "Men have two emotions: Hungry and Horny. If you see him without an erection, make him a sandwich! " |
| 07:06.00 | *** join/#asterisk ptblank (~MURDER1@68-169-173-102.lmdaca.adelphia.net) |
| 07:06.42 | nullogic | ok, I have a weird problem.. When I dial my cell, I get the following: -- Executing Dial("SIP/9993-ef0d", "Zap/g1/6725687319") in new stack |
| 07:06.42 | nullogic | -- Called g1/6725687319 |
| 07:06.42 | nullogic | -- Zap/1-1 is ringing |
| 07:06.42 | nullogic | -- Channel 0/1, span 1 got hangup |
| 07:06.42 | nullogic | -- Hungup 'Zap/1-1' |
| 07:06.44 | nullogic | == No one is available to answer at this time |
| 07:07.56 | Brixius | And on that note, good night all |
| 07:08.06 | nullogic | gnite |
| 07:08.36 | nullogic | cmon you asterisk wizards.. |
| 07:09.04 | Veto | night brixius, thx again! |
| 07:09.52 | Veto | nullogic: I think I have your fix. |
| 07:10.35 | suma | Veto: I'm from the UK it is connected to BT line, do you need any info |
| 07:10.36 | nullogic | ok |
| 07:11.02 | Veto | nullogic, try this: goto(default, XXXYYYZZZZ,1) instead of the Dial(). |
| 07:12.10 | Veto | suma, I really don't know anything about tele hardware :/ |
| 07:12.34 | nullogic | huh? shouldn't this exten => _NXXXXXXXXX,1,Dial,Zap/g1/${EXTEN} ; handle all outbound calls? |
| 07:13.51 | Veto | Ugh, you would think so...but I just went through this for an extension. I had exten => 1113,1,Dial(<Number>), which failed...while 1113,1,Goto(default,<number>,1) worked. |
| 07:14.35 | Veto | In your case, it's quite different...that IS your outbound, not an extension. |
| 07:14.43 | nullogic | right |
| 07:15.24 | nullogic | its weird, I get inbound calls and can call local sip to sip but get the above when I dial out |
| 07:16.12 | Veto | I had your same error while using the Dial(<num>) until I changed to the workaround of Goto(default,<num>,1). It drove me crazy as my semantics were correct. |
| 07:16.30 | Veto | but you don't have a redirection to apply it to. |
| 07:17.23 | *** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it) |
| 07:17.58 | Veto | ~seen paulc |
| 07:18.00 | jbot | paulc <paulc@S010600062586a0b4.vc.shawcable.net> was last seen on IRC in channel #asterisk, 5h 53m 12s ago, saying: 'sudoer: give me a few minutes cos I'm just setting up a new IP600.. but I can probably send you a template file'. |
| 07:19.18 | nullogic | ~seen Shido6 |
| 07:19.20 | jbot | shido6 is currently on #asterisk (15h 9m 2s). Has said a total of 6 messages. Is idling for 13h 55m 59s |
| 07:22.13 | *** part/#asterisk sudoer (~sudoer@65.75.148.190) |
| 07:23.28 | *** join/#asterisk pulu (~chatzilla@65.77.78.3) |
| 07:23.42 | *** join/#asterisk Lethol (~lethol@201.128.129.125) |
| 07:27.27 | nullogic | any suggestions? |
| 07:28.46 | Lethol | can someone hook me up with a 7905 sip firmware? pls msg me if possible |
| 07:31.20 | twisted | jesus. |
| 07:31.23 | twisted | that was a long movie. |
| 07:31.29 | twisted | * twisted is back (gone 156:07:55) |
| 07:31.34 | twisted | oh well. |
| 07:33.59 | *** join/#asterisk Sitxu (~sitxu@200.82.228.164) |
| 07:35.39 | pulu | when you guys make extensions for large numbers of diff. international countries, do you usually stick each one in their own context? |
| 07:40.31 | *** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it) |
| 07:41.46 | *** join/#asterisk KleinJonp___ (~chatzilla@dsl-213-023-212-200.arcor-ip.net) |
| 07:43.15 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:47.18 | subx | nullogic, I can now make calls... |
| 07:47.48 | subx | did you add the informaiton in your extensions.conf |
| 07:47.59 | subx | that did the trick for me in regards to placing calls |
| 07:48.13 | nullogic | cool |
| 07:48.54 | nullogic | no i did not.. that was not the right answer |
| 07:55.33 | subx | I am not receiving any information using asterisk -vvvvvv -g -dddddd -cr when I try to call into the system, what could that be? |
| 07:56.12 | subx | I should have prefaced that, I am using a broadvoice SIP connection. |
| 07:58.26 | subx | anyone? |
| 07:58.38 | subx | please??? |
| 07:58.42 | Veto | what did you watch twisted? |
| 08:02.01 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-78.lv.lv.cox.net) |
| 08:02.30 | *** join/#asterisk BurnedOutGeek (nobody@res-66-191-172-161.spa.sc.charter.com) |
| 08:05.40 | firestrm | anyone know why i would be missing the first 1 sec of audio from a playback command? |
| 08:06.08 | Veto | I get that sometimes, do you have a Ring or a Wait in there? |
| 08:06.15 | firestrm | its very consistant, and doesnt matter if i do a wait first |
| 08:07.05 | firestrm | although is i play 5 sec of blank audio, it wil not lose the next playback command |
| 08:07.21 | Veto | I've heard it, no idea why, I'd guess it's something to do with answer,wait,ring? |
| 08:07.44 | Veto | Is it your initial .gsm? |
| 08:07.52 | firestrm | its very strange, allmost like i need 1 second of blank audio in all my sound files. |
| 08:08.01 | Veto | or all your files? |
| 08:08.22 | firestrm | yes the inital one does it.. as long as i do a playback after playback.. no loss in the second, third etc |
| 08:09.08 | firestrm | and it seems to be consistant over all interfaces, zap/iax/sip |
| 08:09.09 | Veto | I get it in my intial .gsm, fixed it with (sec) Wait,1 then Answer |
| 08:09.42 | firestrm | so your doing wait(1) wait(1) playback ? |
| 08:10.53 | *** join/#asterisk delphiuk (~delphi@host81-155-71-170.range81-155.btcentralplus.com) |
| 08:12.16 | Veto | actually, I'm doing: |
| 08:12.18 | Veto | exten => s,1,Ringing |
| 08:12.18 | Veto | exten => s,2,Wait,1 |
| 08:12.19 | Veto | exten => s,3,Background(gcg-menu) |
| 08:13.08 | firestrm | hmm, i dont know the ringing command.. im doing the same thing except answer rather than ringing |
| 08:13.26 | firestrm | Veto, a friend sent me a very cool datasheet on a SIP<->GSM gateway. you can use any gsm cellphone with asterisk.. |
| 08:13.48 | firestrm | now how cool is that?! |
| 08:14.00 | Mavvie | url? |
| 08:14.02 | Veto | I put the ringing in so they hear something familiar when they call, a ring, before they hit the menu. |
| 08:14.10 | Veto | firestrm, whoa! |
| 08:14.40 | firestrm | Mavvie, no url, just hard copy datashher, he is going to get set up as a dealer.. |
| 08:14.54 | Veto | ANY gsm phone? Cingular, T-Mobile? |
| 08:15.34 | firestrm | any phone, apparently it sends the imei as the callerid, so you can use that for access control in you extensions.conf |
| 08:16.08 | Veto | so it requires work on the phone, which isn't a bad thing. |
| 08:16.11 | Veto | ? |
| 08:16.19 | firestrm | 5W approx 5km radius |
| 08:16.36 | Mavvie | oh, that way. |
| 08:16.53 | Mavvie | now you have to make your own SIM cards :-) |
| 08:17.01 | Veto | oh, so it's an RF interface. |
| 08:18.15 | firestrm | yes it handles all of the gsm gprs signaling and registration and handoff.. its basicly the same equp as mobile provisers use, the only catch is that it can only handle 5 users at a time.. |
| 08:18.31 | implicit | gsm does not do 'gprs' signalling lol |
| 08:18.36 | *** join/#asterisk Pressy (~pressy@p548104C0.dip.t-dialin.net) |
| 08:18.37 | firestrm | yes i know i cant spell.. its my keyboard.. i swear :) |
| 08:18.46 | implicit | ss7 map (mobile access part) |
| 08:19.04 | coppice | GSM-A actually. |
| 08:20.32 | firestrm | ok, well i dont claim to understand all about gsm.. hey im a pilot.. you want expertise lets talk aircraft:) but i do know that you can use a regular gsm cellphone to talk to an * box with it, and it says it has handoff and GPRS capibilities |
| 08:21.08 | implicit | cool |
| 08:21.11 | coppice | firestrm: there are a lot of things that say they do that. most are just GSM modems |
| 08:21.28 | coppice | to be legal they would have to be |
| 08:21.40 | Veto | bah, I've been good this week, wife is 'with child', it's late, she's asleep, I've been drinking...off to find some pr0n. |
| 08:21.45 | implicit | firestrm: to be nice they would have to not be |
| 08:23.07 | firestrm | my friend claims to have one being shipped to him.. so i will test and report back once hardware is in hand.. |
| 08:23.38 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 08:24.55 | firestrm | hey i did find a url on the datasheet, www.2n.cz |
| 08:26.28 | coppice | seen those before. they are GSM modems. |
| 08:27.04 | firestrm | http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html is the exact model i have ds for. |
| 08:27.30 | firestrm | coppice, so it wont work for gsm calls? |
| 08:28.04 | coppice | depends what you mean. It lets a VoIP channel call into a public GSM network. |
| 08:28.37 | firestrm | coppice, but not the other way around? |
| 08:28.41 | Veto | coppice, you dealt with many sip+nat issues? |
| 08:28.50 | firestrm | gsm phone -> * |
| 08:29.20 | implicit | ipv6 is the way to go |
| 08:29.27 | Veto | firestrm, so it's using a gsm /modem/ to be a hardware device to *, like GSM/1 ? |
| 08:29.29 | coppice | well, it lets you go both ways. its just a GSM modem with a VoIP interface tacked on. |
| 08:29.33 | firestrm | i was ready to slash this afternoon over sip+nat |
| 08:30.16 | Veto | from *CLI> dial 1111 works, from 1111 dial <world> works, from 1111 dial 1112 fails...I'm going nuts. |
| 08:30.46 | firestrm | Veto, i think so.. but not mobile savy enough to really know.. the guy who is importing them, is expert at wireless and gives me the impression they are VERY cool |
| 08:31.40 | firestrm | coppice, so i wonder how they get the handoff to other modem capibility, or if that needs to be on the server.. |
| 08:31.43 | Veto | firestrm, that would be handy for med/lrg business to deploy as long as it didn't jack their normal phone operations. |
| 08:31.45 | coppice | someone selling them tries to make them sound cool. that sounds kinda familiar :-) |
| 08:32.00 | Veto | does it take a seperate SIM? |
| 08:32.19 | coppice | of course it takes a SIM. its a GSM modem |
| 08:32.35 | Veto | right, but a * "adjusted" sim? |
| 08:32.48 | firestrm | Veto, no, at least thats what im told.. but who knows.. |
| 08:32.51 | *** part/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 08:33.16 | coppice | you gets the SIM from the carrier you subscribe to. nothing adjusted about it |
| 08:33.29 | firestrm | Veto, thats why we need to buy hardware to play with.. you can never believe the salesman/datasheet |
| 08:34.50 | coppice | some people are using those units in .eu to link * and ISDN PBXs to the GSM networks. I was told they work OK. |
| 08:34.50 | firestrm | is there not some sort of lock on gsm phones to lock you to the carrier though,, that might be the hitch in the plan. |
| 08:35.16 | coppice | only phones sold the in the US, and a couple of other places, are locked. A GSM SIM works in any phone that is not locked |
| 08:35.22 | Veto | dammit, back to pr0n and beer...I've had enough *+nat+sip this week. later dudes ! |
| 08:35.33 | pkwong | unlocking phones is not difficult. |
| 08:35.58 | pkwong | contrary to popular belief, if you have one unlock code, you can unlock all the phones of the same model.. |
| 08:36.16 | pkwong | at least it works with the .eu motorolas.. |
| 08:36.19 | coppice | depends on the model. the info for many models is easy to get |
| 08:36.31 | firestrm | pkwong, i thought they did it by using the imei as the seed for the key |
| 08:36.32 | veto_ | as signoff, I hear gsm treo 650 is avail jan-26 |
| 08:36.42 | pkwong | fire: that's what i thought too.. |
| 08:36.48 | coppice | the locking is brain dead |
| 08:36.52 | *** join/#asterisk burton27_ (mimx@w201.ljudmila.org) |
| 08:36.57 | pkwong | i bought an unlock code for my v525 and used the same code to unlock 3 phones. |
| 08:37.05 | firestrm | lol |
| 08:37.10 | pkwong | so who knows.. |
| 08:37.46 | redder86 | coppice: hi. looks like you've been busy. I notice recent releases. |
| 08:37.52 | firestrm | thats like the seed/key on vehicle ECM's.. (which i like to hack) they use the same seed/key for every vehicle of the same year/model |
| 08:38.19 | pkwong | yeah.. there are lots of little tricks with all that stuff. |
| 08:38.22 | pkwong | for example.. |
| 08:38.33 | pkwong | ford vehicles with the stupid little keypad on the door.. |
| 08:38.44 | pkwong | ya know how to get the unlock code? |
| 08:39.02 | delphiuk | does the username entry in the sip.conf file get used if you are using a budgetone 100 phone? |
| 08:39.20 | pkwong | stick your head under the dash on the drivers side.. there's a big sticker with the code attached to the firewall in 24 point. |
| 08:39.34 | redder86 | coppice: pre10 doesn't have any of the pseudomodem stuff, does it? |
| 08:40.16 | coppice | do you mean the stuff for letting HylaFAX attach? |
| 08:40.16 | firestrm | pkwong, lol, i know.. isnt it stupid.. i guess they assume all crooks are retards |
| 08:40.25 | redder86 | coppice: yeah :-) |
| 08:40.43 | pkwong | yeah. |
| 08:40.52 | pkwong | i bought my truck used. |
| 08:40.55 | coppice | firestrm: no. they assume most customers don't care enough for it to matter |
| 08:41.51 | firestrm | pkwong, you can set mileage on a ford speedo using a serial port.. its just rs232 |
| 08:41.53 | coppice | redder86: the core modem emulation is in there. The channel driver code to make it talk with Hylafax is a unicall module, which is also on my FTP site |
| 08:42.08 | pkwong | hah.. nice. |
| 08:42.36 | redder86 | coppice: oh, that's great |
| 08:42.57 | firestrm | pkwong, i know someone who built a speedo zapper so he could turn back the milage before going into the dealer for service.. save on leasing charges.. |
| 08:42.58 | redder86 | coppice: so I should be able to use it, then? |
| 08:42.59 | *** join/#asterisk elric (fsck@ppp114-10.static.internode.on.net) |
| 08:43.02 | coppice | its rather preliminary, and there are no instructions for how to use it yet |
| 08:43.31 | redder86 | what are the instructions? |
| 08:43.33 | pkwong | heh.. now that's nice. |
| 08:43.35 | redder86 | :-) |
| 08:43.59 | implicit | ~seen ozjames79 |
| 08:44.01 | jbot | ozjames79 <~james@CPE20320889-1842-1.gex.ncable.net.au> was last seen on IRC in channel #asterisk, 7d 22h 52m 57s ago, saying: 'think i will go with php'. |
| 08:44.20 | elric | how would I implement this set up? IAX Softphone ----> Asterisk -----> POTS, i want to make calls from my desktop to the outside world through my asterisk box. |
| 08:44.20 | firestrm | pkwong, he gave me the info on how to do just about every manufacturer/model/year right up to 2003 in a big database.. |
| 08:44.31 | *** join/#asterisk infinii (~wchan@66.146.150.178) |
| 08:44.33 | pkwong | heh.. nice. |
| 08:44.55 | firestrm | pkwong, doesnt do me much good though, my truck has analog odometer.. |
| 08:45.10 | elric | i have a digiumtdm400p card with 2 fxo and fxs ports. |
| 08:45.41 | redder86 | elric: with the dialplan? |
| 08:45.50 | firestrm | pkwong, woops, now how did that propritary manufacturers database get out.. |
| 08:46.01 | pkwong | haha.. |
| 08:46.26 | elric | redder86, alright |
| 08:48.04 | firestrm | pkwong, you know on the z-somthing bmw's with the launch control.. in this database it shows how to turn off recording number of times its used.. apparently if you do it more than 3 times the warrenty is void, but this disables counting |
| 08:48.18 | pkwong | you're kiddin. |
| 08:49.13 | firestrm | nope. |
| 08:49.13 | infinii | doncha hate obd-2? |
| 08:49.13 | pkwong | yeah. |
| 08:49.13 | pkwong | obd-2 is good and bad. |
| 08:49.13 | firestrm | pkwong, its some sequence that if done right will launch the beemer at max exceleration. |
| 08:49.14 | pkwong | i was thinking of getting a beemer.. |
| 08:49.17 | firestrm | pkwong, just wait for obd-3.. big brother WILL be watching.. |
| 08:49.26 | pkwong | ahh. yes.. as if they don't already. |
| 08:50.37 | elric | what would cause this --> Jan 22 19:41:31 NOTICE[6637]: chan_iax2.c:5405 socket_read: Rejected connect attempt from 192.168.1.15 |
| 08:50.40 | firestrm | one of my regular fliers, is vp of somthing a bosch automotive. he is telling me that the insurance companies are lobbying and will probbly get an interface to "bill" you for every time you speed |
| 08:51.10 | redder86 | elric: a firewall |
| 08:51.41 | firestrm | they will plug in when you renew, and bill you for everytime you exceed the national max speed limit (110kph in canada) |
| 08:51.46 | pkwong | that's gonna be a tough one to pass. |
| 08:52.05 | pkwong | so i have an * question.. |
| 08:52.22 | pkwong | what's the benefit of using a digiumesque card over a cisco as 5300? |
| 08:52.32 | pkwong | i can't think of any. |
| 08:52.32 | elric | redder86, i dont have one... i dont have ipf or ipfw filtering on the FreeBSD box * is running on. |
| 08:52.38 | firestrm | pkwong, they dont have to pass anything. if the capibility is there, they just make it a condition of insurance discounts.. dont want to be read, dont get the discount.. |
| 08:52.51 | pkwong | heh. ok. |
| 08:53.05 | firestrm | scary.. |
| 08:53.13 | pkwong | i'm still grappling with the t1 card vs. media gateway hardware solution. |
| 08:53.26 | pkwong | it seems that the t1 cards aren't really cheaper per port. |
| 08:53.27 | redder86 | elric: a firewall on 192.168.1.15 |
| 08:53.34 | delphiuk | elric: the client iax connect specifying an invalid context in it connect string |
| 08:54.08 | *** join/#asterisk burton27_ (mimx@w201.ljudmila.org) |
| 08:54.11 | pkwong | i got clocked doing 177 in a 65 this summer. |
| 08:54.34 | pkwong | and i didn't go to jail! talk about lucky. |
| 08:54.34 | elric | alright, i will check both |
| 08:54.56 | firestrm | pkwong, my record 205 mph in montana..500.00 fine (yes they will fine you in montana for that fast) |
| 08:55.03 | delphiuk | of it's a firewall issue, would it not reach * at all? |
| 08:55.12 | delphiuk | *if |
| 08:55.18 | pkwong | nice. |
| 08:56.07 | firestrm | i still have the ticket.. that was when i was much younger, returning my boss's car from being shown at sema.. it was a very short trip |
| 08:56.45 | implicit | was it worth $500? :) |
| 08:56.59 | redder86 | how did they catch you at 200 mph? |
| 08:57.02 | firestrm | you bet... i was racing my boss at the time too :) |
| 08:57.04 | pkwong | so.. anyone here have any opinions of the digium t1 cards vs. the cisco as 5300? |
| 08:57.09 | implicit | i'm surprised they caught you |
| 08:57.21 | firestrm | redder86, radio'd to the next county.. |
| 08:57.25 | implicit | as5300 sounds quite a bit better |
| 08:57.28 | elric | it registers properly and iax2 show peers lists it. |
| 08:57.29 | implicit | and it does everything in hardware |
| 08:57.40 | pkwong | yeah. i agree. |
| 08:57.55 | pkwong | i just don't know if there's a reason why anyone would go with the card. |
| 08:58.31 | *** join/#asterisk burton27 (mimx@w201.ljudmila.org) |
| 09:00.02 | coppice | what's the launch control in a BMW? |
| 09:02.19 | firestrm | coppice, you rev up the engine and do some sequence of events, and it changes the shift pattern for a nice smokeshow with max exceleration.. but you only get to do it 3 times and you warrenty is void |
| 09:03.28 | firestrm | coppice, http://www.q.co.za/2001/2001/11/16-bmw.html |
| 09:03.59 | coppice | so you only get three attempts to get laid? :-) |
| 09:04.55 | firestrm | lol |
| 09:05.14 | firestrm | or you know someone who can reset the counter |
| 09:08.02 | coppice | so, there should be even more dead BMWs lining the motorway shoulders |
| 09:08.23 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 09:08.31 | *** join/#asterisk zoa (zoa@82.103.76.147) |
| 09:11.38 | firestrm | time for sleep, long day.. |
| 09:12.56 | suma | hi any help here for my tdm400p ? |
| 09:14.29 | suma | when i receive call or place call using tdm400p, it is not hanging up the line when the other party hangs up the call |
| 09:14.42 | suma | it was working perfect when i use my x100p card |
| 09:21.59 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 09:21.59 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:25.33 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 09:46.54 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 09:46.54 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:55.14 | *** join/#asterisk lele (~lele@rivendell.windmill.it) |
| 09:58.59 | *** part/#asterisk jero (~boo@modemcable040.12-81-70.mc.videotron.ca) |
| 10:02.26 | *** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62) |
| 10:09.06 | *** join/#asterisk abbas_ (~nid@203.128.19.138) |
| 10:12.03 | suma | hi |
| 10:12.31 | abbas_ | hi |
| 10:12.47 | suma | familier with tdm400p ? |
| 10:13.16 | abbas_ | have a little knowledge ask me if i know i will tell u |
| 10:13.28 | suma | when i receive call or place call using tdm400p, it is not hanging up the line when the other party hangs up the call |
| 10:14.56 | abbas_ | show me dial plan |
| 10:15.23 | suma | exten => s,3,Dial(SIP/12345,60,t) |
| 10:15.38 | suma | exten => s,1,Dial(SIP/${CALLERIDNUM}) |
| 10:15.38 | suma | exten => s,2,Wait |
| 10:15.38 | suma | exten => s,3,Dial(SIP/12345,60,t) |
| 10:16.18 | abbas_ | mmm u have not added hangup |
| 10:16.37 | abbas_ | just a minute |
| 10:16.44 | suma | is that compulsory ? |
| 10:17.08 | *** join/#asterisk Vahram (~Vahram@xalt.xter.net) |
| 10:17.52 | Vahram | Hi people, anyone have success story building h323 channel on Fedora Core 3? |
| 10:20.45 | abbas_ | exten => 08449869305,6,HangUp |
| 10:21.32 | abbas_ | suma exten => s,4,HangUp |
| 10:21.32 | suma | yes |
| 10:21.42 | suma | is it not optional one ? |
| 10:21.50 | abbas_ | suma exten => s,4,HangUp put this in dial plzn |
| 10:22.49 | suma | thanks abbas |
| 10:23.04 | abbas_ | u r welcome |
| 10:25.21 | *** join/#asterisk Tili (~Tili@202-133-65-226-dialup.sat.net.pk) |
| 10:25.38 | pauldy | I'm here hoppping for suggestions on sip providers that offer did and unlimited nationwide long distance in the us for a reasonable price were I can use asterisk |
| 10:25.59 | abbas_ | nufone |
| 10:26.06 | pauldy | hit me up if anyone know of a good one besides broadvoice I did find that one on my own |
| 10:26.13 | pauldy | nufone supports asterisk? |
| 10:26.23 | pauldy | kewl I will have to check that out |
| 10:26.27 | abbas_ | yes |
| 10:27.27 | abbas_ | pauldy can u pls help me to test a calling card application |
| 10:27.44 | abbas_ | i tell u a accessnumber and u need to make a test call |
| 10:28.03 | pauldy | I don't have phone line access |
| 10:28.14 | abbas_ | ok |
| 10:28.14 | abbas_ | Suma u |
| 10:28.17 | pauldy | thus the need for phone service |
| 10:28.31 | pauldy | all I have is broadband internet no voice |
| 10:29.04 | abbas_ | Suma u there ? |
| 10:29.26 | pauldy | yea and nufone doens't do unlimited |
| 10:29.51 | postel | pauldy: well, try get some PSTN access, at least for emergency access |
| 10:29.59 | abbas_ | i dont know but they used to do |
| 10:30.28 | abbas_ | postel can u pls make a test calll? |
| 10:30.29 | postel | pauldy: if your house start burning most likely elec would go down after the wiring melts |
| 10:30.29 | pauldy | postel thought about a cell phone for that |
| 10:30.40 | postel | abbas_: depends, on what? |
| 10:30.55 | pauldy | if my apartment burned down I would just hop through a wall they are paper thin |
| 10:31.17 | abbas_ | i tell u a US number i m using it as access number of my calling card platform |
| 10:31.47 | pauldy | plus I would rather have a cell phone without service for emergencies |
| 10:32.03 | pauldy | that way I don't have to use it inside the apt while it is burning down |
| 10:32.32 | pauldy | which doesn't make sense to have a dedicated pots line for that scenario either but if it helps ya sleep at night I guess |
| 10:39.37 | *** join/#asterisk abbas (~nid@203.128.19.94) |
| 10:40.00 | abbas | hi |
| 10:41.18 | abbas | postel u still there? |
| 10:41.40 | *** join/#asterisk KleinJonp____ (~chatzilla@dsl-213-023-225-206.arcor-ip.net) |
| 10:49.21 | *** join/#asterisk Guest99306 (~YeN@210.186.104.16) |
| 10:50.16 | suma | hi abbas |
| 10:50.18 | suma | u there ? |
| 10:50.25 | abbas | yes |
| 10:50.33 | suma | i tried with hangup |
| 10:50.38 | suma | it doesn't help |
| 10:51.11 | *** join/#asterisk r0d3nt-m (~RatMan@wsip-24-234-241-78.lv.lv.cox.net) |
| 10:51.46 | abbas | sorry i dnt know else |
| 10:51.58 | abbas | can u help me in testing my caling card platform |
| 10:52.07 | abbas | u need to dial a US access numbee |
| 10:52.11 | abbas | number |
| 10:52.55 | suma | when i try calling my ip phone connected to tdm400p through my mobile it answers, when i hangup my mobile, then i get busy tone on my ip phone instead of hangup |
| 10:53.00 | suma | yes tell me abbas |
| 10:53.08 | abbas | ok |
| 10:53.17 | suma | can i call anywhere in the world for FREE ? |
| 10:53.32 | abbas | no |
| 10:53.42 | suma | give me some offers |
| 10:53.43 | abbas | i will ask u to dial another US number only |
| 10:53.45 | abbas | :) |
| 10:53.50 | suma | that will encourage me to test right ;) |
| 10:53.52 | abbas | thats just a test |
| 10:53.58 | abbas | hahaha |
| 10:53.59 | suma | yes |
| 10:55.21 | suma | sip calls not accepted ? |
| 10:55.49 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 10:56.46 | *** join/#asterisk robin_sz (~robin@adsl.redpoint.org.uk) |
| 10:57.14 | suma | yes |
| 10:57.17 | suma | i dunno |
| 11:03.27 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 11:03.48 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-183-50.dsl.scarlet.be) |
| 11:03.58 | *** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 11:04.58 | pulu | abbas: stupid question, but why can't you dial it yourself? is it your only phone line? |
| 11:06.50 | abbas | pulu have not other line |
| 11:07.04 | pulu | i hate when that happens |
| 11:07.58 | abbas | so anyone gonna help me in testing? |
| 11:08.36 | pulu | well even with the magic of asterisk it's still gonna cost all of us to do it, but i need to test something else anyway, so give me the area code and 2 min so I can set it up with diax |
| 11:08.57 | abbas | ok |
| 11:10.48 | markit | hi :) I opened but 3401, mark closed it as fixed, but I've found that it does not work :( any bug-marshall here to assist me? |
| 11:10.55 | markit | but=bug |
| 11:15.13 | *** join/#asterisk HearT (~Man_in_BL@81.212.13.230) |
| 11:15.41 | *** join/#asterisk syper (~anon@203-206-52-57.dyn.iinet.net.au) |
| 11:18.14 | syper | Need help setting up an SPA-2000. I can get a dial-tone, but cannot dial-out. |
| 11:20.42 | pulu | syper: if you do sip debug from the console what shows? |
| 11:26.14 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 11:30.44 | *** join/#asterisk psycodad (~obiwan@2001:4060:4419:b1:0:0:0:2) |
| 11:40.19 | *** join/#asterisk switch_ (~switch@61.206.115.5.user.ad.il24.net) |
| 11:40.58 | syper | pulu: how do i do sip debug exactly? |
| 11:41.22 | pulu | syper: from the console type "sip debug" without the quotes |
| 11:41.34 | postel | "how do i get the console" |
| 11:41.48 | syper | what he said ^ |
| 11:41.49 | postel | "what is asterisk" |
| 11:41.56 | syper | i am not using asterisk |
| 11:42.02 | pulu | uhh |
| 11:42.02 | postel | "who am i" |
| 11:42.17 | postel | "why does it always rain on me" |
| 11:42.24 | syper | you aren't helping postel, please go back to idleing |
| 11:42.30 | pulu | i don't think an spa-2000 has alot to do with anything around here without asterisk |
| 11:42.50 | *** join/#asterisk ncjp_ (~switch@61.206.115.5.user.ad.il24.net) |
| 11:43.17 | pulu | i have two spa-3000's sitting on my desk waiting for me to take them out of their boxes and set them up, but that will be with asterisk |
| 11:43.25 | pulu | which of course is the title of this channel |
| 11:43.30 | pulu | i'm not being very helpful either |
| 11:43.32 | pulu | i better idle |
| 11:43.45 | *** join/#asterisk DoT|cobain (cobain@adsl077.r24-Dbg2.Vie.AT.KPNQwest.net) |
| 11:43.53 | DoT|cobain | Hello everybody |
| 11:43.56 | syper | well maybe you can point me to a more specific IRC channel? I assumed if you can set up asterisk you would know a fair bit |
| 11:45.16 | DoT|cobain | is there anyone who knows about chan_sccp ? |
| 11:46.59 | *** join/#asterisk kentster (~kentster@vpn1.ccstg.com) |
| 11:47.18 | *** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com) |
| 11:48.56 | DoT|cobain | did anyone here set up a cisco 7905G phone with chann_sccp oder chan_skinny ? |
| 11:53.45 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 11:54.32 | *** join/#asterisk kentster (~kentster@vpn1.ccstg.com) |
| 11:56.20 | *** join/#asterisk franck (~franck@202.62.1.34) |
| 11:56.46 | franck | Hi |
| 11:57.37 | franck | when I start asterisk, it takes over my sound card and I cannot play anything. How do I stop asterisk to use the sound card |
| 11:58.02 | RaYmAn-Bx | franck: make it not load oss/alsa modules |
| 11:58.26 | franck | RaYmAn-Bx: do I remove the config files or I rem all the lines? |
| 11:59.24 | RaYmAn-Bx | just put change load to noload for the chan_oss and chan_alsa modules in modules.conf |
| 12:00.17 | franck | ok |
| 12:00.49 | *** join/#asterisk ckruetze (~ckruetze@i3ED658DA.versanet.de) |
| 12:02.41 | DoT|cobain | ok... i have the following problem: i set up asterisk with the chan_sccp module and the 7905G is able to register with asterisk an shows text and all this, i can call the phone from my softphone (SIP) on the computer and it also shows my callerid in the display, but i cant dial on the 7905G, i hear a dialtone an the dtmf signals of the keys... but it wont dial and simply play the busy tone |
| 12:02.48 | franck | cool, working! thx |
| 12:03.34 | DoT|cobain | after the dial procedure and the busy tone the phone shows "Network congestion or error" |
| 12:05.45 | *** join/#asterisk cjk (~cjk@80.92.75.186) |
| 12:07.58 | *** join/#asterisk coppice (~chatzilla@200.192.17.210.dyn.pacific.net.hk) |
| 12:15.35 | markit | anyone here using chan_mISDN? |
| 12:16.23 | *** part/#asterisk franck (~franck@202.62.1.34) |
| 12:16.59 | visik7 | markit what's the difference between chan_misdn and chan_modem_i4l ? |
| 12:17.54 | *** join/#asterisk h4mm3r` (~h4mm3r@81-208-60-202.fastres.net) |
| 12:18.35 | *** join/#asterisk HjemmeRoyK (~roy@110.80-203-29.nextgentel.com) |
| 12:21.08 | HjemmeRoyK | morning, morons |
| 12:22.05 | coppice | why are you talking to yourself? |
| 12:22.25 | coppice | I have another problem with your ATA box |
| 12:24.31 | *** join/#asterisk gopinsurg (cashmoney@dialup-4.224.135.236.Dial1.Cincinnati1.Level3.net) |
| 12:24.59 | gopinsurg | Good Morning |
| 12:26.07 | HjemmeRoyK | coppice: ok? |
| 12:26.08 | HjemmeRoyK | what |
| 12:29.55 | *** join/#asterisk chachoo (~root@209.234.83.19) |
| 12:30.09 | *** part/#asterisk chachoo (~root@209.234.83.19) |
| 12:30.37 | coppice | RoyK: I make calls. I chat. Everything is fine. I press the start button on the FAX machine. The ATA crashes. No pinging. No browsing. Nothing until I power cycle the ATA :-) |
| 12:31.56 | *** join/#asterisk Tommmo (~tps@203.62.181.52) |
| 12:31.56 | coppice | to be more specific, I press the start button. about 2 seconds of calling tone passes. then it crashes |
| 12:31.56 | Tommmo | hi, i'm trying to use asterisk with openH323. |
| 12:31.56 | *** join/#asterisk kaffemand (~martin@cpe.atm2-0-1101141.0x50a4a2de.bynxx12.customer.tele.dk) |
| 12:31.56 | Tommmo | when i place a call, i get the following: |
| 12:31.56 | Tommmo | Jan 22 22:08:33 WARNING[30146]: channel.c:1901 ast_request: No channel type registered for 'OH323' |
| 12:32.23 | *** join/#asterisk Che (~A--she@62.139.11.5) |
| 12:32.33 | Tommmo | in extensions.conf I'm using : exten => 132000,1,Dial(OH323/132000@203.62.181.59:1720) |
| 12:32.40 | Tommmo | do i need to somehow define OH323 somewhere else? |
| 12:33.52 | *** join/#asterisk Leland (~leland@ws2.discpro.org) |
| 12:33.54 | Leland | hi all |
| 12:34.42 | Leland | anyone available for a quick question? |
| 12:36.53 | HjemmeRoyK | coppice: is that a normal analog fax you're having? |
| 12:37.25 | *** join/#asterisk ard (ard@gw-uunet-office.telegraaf.net) |
| 12:38.01 | coppice | yeah. I plugged it into the ATA, and made voice calls with its handset OK. when I press the start button to transfer a fax, after about 2 seconds of calling tone BOOM! |
| 12:38.17 | HjemmeRoyK | wtf? |
| 12:38.24 | HjemmeRoyK | that never happened to me.... |
| 12:38.33 | coppice | have you sent faxes? |
| 12:38.42 | HjemmeRoyK | let me check |
| 12:38.49 | HjemmeRoyK | I haven't sent any faxes. no t.38 support. |
| 12:38.49 | suma | when i try calling my ip phone connected to tdm400p through my mobile it answers, when i hangup my mobile, then i get busy tone on my ip phone instead of hangup |
| 12:38.52 | HjemmeRoyK | but no hangups |
| 12:38.56 | coppice | I tried it several times. same thing every time |
| 12:39.38 | *** join/#asterisk fishboy1669 (fishboy166@cpc1-warr1-3-0-cust38.bagu.cable.ntl.com) |
| 12:39.43 | fishboy1669 | morning guys |
| 12:39.48 | coppice | well, it doesn't hang until you try to send a fax. I used it for quit a while with just voice |
| 12:40.30 | suma | fax ? |
| 12:40.41 | suma | My TDM is connected to my landline |
| 12:40.49 | suma | when i call my landline, it rings and i can talk |
| 12:41.01 | suma | if I hangup my call from the mobile |
| 12:41.11 | Leland | having a bit of an issue with asterisk between the voip provider and Cisco ITS/CME ... the inbound call from the provider hits asterisk, which passes the call to the Cisco .. the phone rings, but there is no audio .. same for outbound calls. Both peers are configured with canreinvite=no ... any ideas? |
| 12:41.18 | suma | the TDM400P should also hang up right |
| 12:41.30 | suma | instead i get busy tone on the phone |
| 12:41.40 | suma | which is annoying ! |
| 12:43.24 | HjemmeRoyK | coppice: try this the b release http://karlsbakk.net/fw/ |
| 12:43.32 | HjemmeRoyK | coppice: iirc the first release was a brown paper bag |
| 12:49.12 | gopinsurg | morning |
| 12:49.27 | coppice | RoyK: I upgraded. same thing |
| 12:49.50 | HjemmeRoyK | fuck |
| 12:49.58 | *** join/#asterisk zotz (~zotz@24.244.133.136) |
| 12:50.18 | HjemmeRoyK | can you send me an email about it please, along with an ethereal dump if you get that far? I need to email Yoda |
| 12:50.43 | coppice | try ethereal, I will |
| 12:53.31 | suma | where will i get support for tdm400p ? |
| 12:53.37 | suma | i was very happy with x100p |
| 12:53.59 | suma | but i have been recomended to buy tdm400p |
| 12:54.06 | suma | not ended in stupid issues |
| 12:54.23 | HjemmeRoyK | suma: four analog lines? |
| 12:54.35 | HjemmeRoyK | suma: tdm400p is supported in * |
| 12:54.40 | suma | yes |
| 12:54.57 | suma | but it is not hanging up when the caller drops the line |
| 12:56.00 | suma | i hear "hang up tone" on the phone instead of phone hang up |
| 12:56.06 | suma | i use cisco 7960 |
| 12:56.29 | suma | i have connected only one analog line |
| 12:57.36 | HjemmeRoyK | in your dialplan, do you Hangup() or just Congestion? |
| 12:59.43 | suma | exten => s,1,NoOp(${CALLERIDNUM}) |
| 12:59.43 | suma | exten => s,2,Wait |
| 12:59.43 | suma | exten => s,3,Dial(SIP/12345,60,t) |
| 12:59.43 | suma | exten => s,4,Hangup |
| 13:00.05 | suma | The application is still executing Dial application |
| 13:00.17 | suma | it is not returning to the next priority |
| 13:02.15 | HjemmeRoyK | exten => s,105,Hangup |
| 13:02.17 | HjemmeRoyK | perhaps |
| 13:02.40 | HjemmeRoyK | asdf |
| 13:02.44 | HjemmeRoyK | test |
| 13:04.16 | *** join/#asterisk HjemmeRoyK (~roy@110.80-203-29.nextgentel.com) |
| 13:04.29 | *** join/#asterisk Luhiwu (~marsosa@200.63.87.246) |
| 13:05.06 | Luhiwu | hello, i'm having problems trying to park calls, anyone can help? |
| 13:06.46 | *** join/#asterisk loick (~loick@ATuileries-151-1-26-74.w82-123.abo.wanadoo.fr) |
| 13:07.11 | kaffemand | hey .. I'm trying to get asterisk to autodial out, but when I place a file in spool/outgoing, I get "Unable to open /var/spool/asterisk/outgoing/outgoing.tmp: Permission denied, deleting" in the log |
| 13:07.47 | kaffemand | I've tried to chmod it to 666, and to chown it to the asterisk user |
| 13:08.19 | kaffemand | how should I set up the permissions? |
| 13:09.03 | fishboy1669 | hi guys anyone any idea on this http://pastebin.ca/4577 |
| 13:09.19 | fishboy1669 | its not the config fiels udev or make linux26 |
| 13:10.15 | fishboy1669 | how do i check what irq's stuff is using |
| 13:12.00 | markit | suma: AFAIK, if the call is answered, the flow is not to the next priority, it exits the context. But if you provide hangup priority (as HjemmeRoyK suggested), THEN the flow goest there |
| 13:12.29 | markit | suma: is something that confused (and confuses) me a lot also :) |
| 13:20.01 | *** join/#asterisk Nuttah (~andrew@amber.interdart.co.uk) |
| 13:20.07 | Leland | has anyone ever tried the Bicom administration front-end to asterisk? (PBXware) -- or indeed has anyone ever gotten a response from Bicom about their product pricing ?? |
| 13:20.11 | Nuttah | heya guys |
| 13:20.35 | Nuttah | anyone give me a hand with asterisk_addons? |
| 13:21.19 | HjemmeRoyK | rotfl |
| 13:21.22 | HjemmeRoyK | Das Leben des Jesus :) |
| 13:21.33 | HjemmeRoyK | I hope there is an English translation of that |
| 13:22.33 | Nuttah | do I actually have to compile asterisk_Addons? |
| 13:22.57 | pimpwell | what is a wiki |
| 13:23.04 | pimpwell | is it an acronym? |
| 13:23.15 | pimpwell | or just a stupid word for help file |
| 13:23.25 | HjemmeRoyK | ~wiki |
| 13:23.27 | suma | markit: i want to hangup the call, nothing else, a simple x100p does brilliant job. I spent $130 on this and a real *** |
| 13:23.32 | HjemmeRoyK | ~wiki? |
| 13:23.34 | jbot | i guess wiki is http://www.voip-info.org |
| 13:24.03 | HjemmeRoyK | pimpwell: a wiki is a documentation site where everyone can change/add stuff |
| 13:24.06 | markit | suma: hang up by code? hanging up the phone is not ok? |
| 13:24.46 | suma | mmmmm |
| 13:25.00 | suma | omg, how to explain this |
| 13:25.13 | suma | what asterisk will send to a SIP phone to end the call ? |
| 13:25.23 | suma | BYE , SIP msg right ? |
| 13:25.49 | suma | i'm saying the rtp stream in still with asterisk rather than the BYE msg |
| 13:26.06 | HjemmeRoyK | FUCKOFF SIP/2.0 |
| 13:26.08 | HjemmeRoyK | :) |
| 13:26.35 | suma | the TDM400P card begs whatever the sound it gets from ZAP line |
| 13:26.52 | suma | rather than the end call msg |
| 13:27.04 | coppice | sounds like the perfect SIP response to telemarketing calls |
| 13:28.19 | suma | HjemmeRoyK: You had good ? |
| 13:28.23 | suma | HjemmeRoyK: You had food ? |
| 13:28.32 | pimpwell | if I want the worst quality call, for volume calling, what protocol to use? |
| 13:28.45 | pimpwell | less bandwidth |
| 13:28.58 | pimpwell | the most economic |
| 13:29.05 | suma | don't use TDM400P cards |
| 13:29.11 | suma | that is a stupid card |
| 13:29.29 | pimpwell | I don't think I need a card for what I am doing |
| 13:29.31 | *** join/#asterisk datareactor (datareacto@203.81.215.93) |
| 13:29.36 | coppice | suma: it has nothing to do with the card. the cards are all dumb |
| 13:29.56 | suma | asterisk is stupid in this case ? |
| 13:30.12 | coppice | maybe its just you. |
| 13:30.13 | suma | i mean asterisk with TDM400P is stupid ? |
| 13:30.39 | suma | come on |
| 13:30.52 | suma | i was nicely playing with X100p with just $5 card |
| 13:30.59 | suma | it was working brilliant |
| 13:31.04 | Chuji | pimpwell : lpc10 |
| 13:31.04 | pimpwell | Web Interface >>> Asterisk >>> T1 >> VoicePlus is my setup |
| 13:31.16 | coppice | You want it to hang up on busy detect. have you actually bothered to configure it thus? |
| 13:31.22 | gopinsurg | web interface??? wtf? |
| 13:31.32 | pimpwell | ya, PHP |
| 13:31.39 | Chuji | pimpwell : But don't use it. You sound like darth vader |
| 13:31.48 | gopinsurg | Did you write it yourself? |
| 13:32.00 | suma | coppice: busydetect=yes in zapata.conf ? |
| 13:32.03 | pimpwell | I havent even setup anything, I'm all theory now |
| 13:32.14 | pimpwell | but, I figure I just drop a message in spool when it's time |
| 13:32.19 | Chuji | heh, hope you aren't busy for a couple months |
| 13:32.24 | pimpwell | I'm doing all automated calls |
| 13:32.30 | pimpwell | .mp3 or whatever |
| 13:32.33 | coppice | suma: do you have the right tone set configured for yuor location? |
| 13:33.01 | suma | loadzone=uk <----- i'm in the UK |
| 13:33.25 | datareactor | can i forward sip calls to other voip server |
| 13:33.26 | suma | defaultzone= |
| 13:33.37 | suma | fxsks=1-4 |
| 13:33.38 | suma | loadzone=uk |
| 13:33.38 | suma | defaultzone=uk |
| 13:33.43 | *** join/#asterisk rtcg (~rtcg@bdsl.66.15.181.96.gte.net) |
| 13:33.45 | pimpwell | chuki: what will take a month? |
| 13:33.51 | pimpwell | chuji* |
| 13:34.55 | Chuji | months |
| 13:34.58 | Chuji | (s) |
| 13:35.13 | Chuji | creating a viable asterisk solution |
| 13:35.17 | gopinsurg | How is everyone using * ? |
| 13:35.33 | suma | coppice: do i need to verify anything else ? |
| 13:36.01 | rtcg | 'Mornin everyone! Has anyone ever had firefly work in all aspects save the passing of the actual voice traffic? |
| 13:37.57 | datareactor | rtcg cannot get your question ? |
| 13:38.30 | pimpwell | connectivity via SIP and IAX... SIP and IAX are defined as "Terminators?" |
| 13:38.39 | pimpwell | and lpc10, etc are codecs? |
| 13:39.18 | Chuji | no |
| 13:39.28 | Chuji | iax2 and sip are protocols |
| 13:39.33 | Chuji | as is h323 |
| 13:39.48 | Chuji | but yes, lpc10, g729, g711u etc, are codecs |
| 13:40.32 | datareactor | can i forward sip calls to other asterisk server |
| 13:41.07 | Chuji | datareactor : Yes |
| 13:43.08 | datareactor | can you give example exten => will look like ? |
| 13:44.37 | *** join/#asterisk santiago (~santiago@63.245.86.101) |
| 13:45.19 | Chuji | datareactor : Well, it's not that straight forward, you need to use dial the second Asterisk server and have it issue a reinvite |
| 13:46.09 | Chuji | So your dial string would be nothing more than a Sip call. Then the second asterisk server would take the reinvite and contact the originating device directly |
| 13:46.54 | Chuji | Now, if you want true routing/forwarding you might look at SER too |
| 13:47.05 | Chuji | but that's a different animal all together |
| 13:47.54 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 13:48.37 | datareactor | Chuji thanks for the explaination |
| 13:49.51 | datareactor | but should i users should be made on both servers ? |
| 13:50.01 | pimpwell | chuji, you don't think setting up my asterisk server will be easier then most? |
| 13:50.12 | markit | suma: are you using asterisk CVS head or stable? |
| 13:50.36 | markit | suma: I remember something about UK problems listed in mantis... you could check |
| 13:50.37 | pimpwell | all I am doing is sending a mp3 to stream t voiceplus |
| 13:50.43 | pimpwell | to* |
| 13:50.52 | Chuji | y? |
| 13:51.21 | pimpwell | I have customers who need the call |
| 13:51.26 | pimpwell | at certain times |
| 13:51.43 | Chuji | won't be too bad |
| 13:52.07 | pimpwell | just trying to decide the lowest bandwith way |
| 13:52.10 | Chuji | The thing about * is that you start with one idea, then you end up finding all of the other things that it can do.... |
| 13:52.18 | Chuji | and poof, there goes many months of your life |
| 13:52.45 | rtcg | Chuji: HAHAHAHA that is so true. |
| 13:52.57 | pimpwell | I'll keep that in mind |
| 13:53.03 | Chuji | pimpwell : Well, voicepulse is only going to support so many codecs |
| 13:53.18 | Chuji | pimpwell : gsm may be your best route with them |
| 13:53.26 | Chuji | pimpwell : That is cell quality |
| 13:53.31 | pimpwell | heard of clearpath? |
| 13:53.46 | pimpwell | they contacted me cause I left a message on the shoutout board on voip-info |
| 13:53.55 | pimpwell | regarding multiple calls at once |
| 13:54.05 | Chuji | No, I haven't actually |
| 13:54.15 | pimpwell | www.clearpath1.com |
| 13:54.23 | pimpwell | scary |
| 13:54.43 | Chuji | Are you using voicepulse because they are local numbers? |
| 13:54.51 | Chuji | Cuz voipjet has the best rates right now |
| 13:55.01 | pimpwell | no, because I need to send mulitple calls at once |
| 13:55.13 | Chuji | Hah, yeah, I would have forgone putting up a site at all rather than what they did |
| 13:55.14 | pimpwell | and they seem to be the only one who will work with me |
| 13:55.51 | Chuji | Well, any per minute co will allow multiple calls. i.e. voipjet, nufone, iax.cc |
| 13:55.51 | pimpwell | I can't daisy chain the calls, I need possibly to send 50 at the same moment |
| 13:56.08 | Chuji | look at those other three too |
| 13:56.16 | pimpwell | thx |
| 13:56.39 | Chuji | /msg jerjer with your idea |
| 13:56.47 | Chuji | he'll quote you some rates |
| 13:56.57 | Chuji | ~jerjer |
| 13:57.07 | mortehu | Why does iax_get_event stop returning new events after while, even though I can see in the strace output new packets are coming in? |
| 13:57.33 | Chuji | ~nufone |
| 13:57.34 | jbot | methinks nufone is Visit http://www.nufone.net for an excellent, native IAX termination service. |
| 13:58.12 | *** join/#asterisk derfer (Babar@54-119.a2f.dsl.net4all.net) |
| 13:58.15 | rtcg | where is port 4569 specified for use in the * code? I can't find that port listed ANYWHERE in my config files. Is it hard coded? |
| 13:58.17 | derfer | hello |
| 13:58.21 | pimpwell | chuji: my idea was to not have to pay a phone bill and just a service like voiceplus per call. I got the idea because I saw the little softphones and thought to myself, I will never need a copper line again and pay a monthy stipend. Am I thinking correctly? |
| 13:58.32 | fishboy1669 | hi guys anyone any ideas on this i been at it three days now and getting down http://pastebin.ca/4579 |
| 13:58.40 | rtcg | what is port 4569 used for? (if anything?) |
| 13:58.53 | fishboy1669 | x100p card done the udev and the make linux26 |
| 13:59.06 | fishboy1669 | and configs are right as they are off a working system |
| 13:59.28 | fishboy1669 | think is someting to do with compile but have deleted everyting and copletely re insatlled |
| 14:01.20 | rtcg | hmm.. 4569 appears to be |
| 14:01.35 | rtcg | used in the iax.conf |
| 14:01.46 | Chuji | rtcg : yeah, you can set it there |
| 14:01.46 | Nuttah | fish, what does ztcfg -vv give you? |
| 14:01.55 | rtcg | I must've changed my default to something else. |
| 14:01.57 | fishboy1669 | nuttah comes back fine |
| 14:02.10 | fishboy1669 | just noticed when i do a make clean in zaptel i get error |
| 14:02.26 | Nuttah | whats the error? |
| 14:02.28 | fishboy1669 | i did mess up first time round and did make with out linux26 |
| 14:02.28 | Chuji | pimpwell : Yes, you still have to pay "per call" though. All of the per minute plans are like that. There is no monthly base rate charge |
| 14:02.43 | fishboy1669 | got a feeling there is the first make make isntall stuff lurking somwhere |
| 14:02.43 | pimpwell | ya, so I'ts perfect for m |
| 14:02.44 | pimpwell | e |
| 14:02.49 | fishboy1669 | god know where though |
| 14:02.58 | Nuttah | pastbin the make error |
| 14:03.02 | pimpwell | I just need to be able to send many out at once, and load share once it's too much for my system |
| 14:04.10 | fishboy1669 | nuttah http://pastebin.ca/4580 |
| 14:04.31 | fishboy1669 | i did read this which looks same issue http://www.marko.net/asterisk/archives/0207/0053.html |
| 14:04.33 | Nuttah | useful |
| 14:05.00 | fishboy1669 | ? home u are refering to the error code as usefull! |
| 14:05.06 | Nuttah | aye |
| 14:05.26 | Chuji | pimpwell : Yeah, shouldn't be a problem. |
| 14:05.30 | Nuttah | and that archive is 2 years old |
| 14:05.30 | fishboy1669 | guess i need to go through the makefile and see if i can figure where it puts everything can and clean it all out |
| 14:05.36 | *** join/#asterisk HD (~Henk@82-136-197-93-mx.xdsl.tiscali.nl) |
| 14:05.53 | Chuji | pimpwell : There is a company at my colo that streams all of the nascar races over nextel phones |
| 14:06.05 | Chuji | pimpwell : He's doing the same thing with a streaming mp3 server |
| 14:07.37 | Nuttah | fish: what versions are you trying to compile at the moment? |
| 14:08.25 | fishboy1669 | straight from cvs |
| 14:08.37 | Nuttah | so latest everything? |
| 14:08.39 | fishboy1669 | new checkout last night and new agin this morn |
| 14:08.40 | fishboy1669 | yes |
| 14:08.46 | fishboy1669 | asterisk zap and pri |
| 14:09.00 | Nuttah | didnt think you had any e1 or t1 cards |
| 14:09.10 | fishboy1669 | i dont |
| 14:09.14 | fishboy1669 | x100p |
| 14:09.23 | Nuttah | i'm no using 1.0.4 yet, just recompiled 1.0.3 zap and asterisk |
| 14:09.30 | Nuttah | then dont install libpri? |
| 14:09.36 | fishboy1669 | ok |
| 14:09.57 | fishboy1669 | i have a bri does that need pri i guess not else it would be libbri lol |
| 14:10.12 | Nuttah | isdn card? |
| 14:10.17 | fishboy1669 | just install it from habit |
| 14:11.02 | Nuttah | i'd suggest staying on 1.0.3 personally. |
| 14:11.04 | fishboy1669 | the zaptel makefiel has some stuff i can play with at the end |
| 14:11.20 | fishboy1669 | how do i cvs check out that ver? |
| 14:12.18 | Nuttah | not sure, but you can download the tarball from ftp://ftp.asterisk.org/pub/asterisk |
| 14:12.28 | Nuttah | its has older versions |
| 14:12.44 | fishboy1669 | ok |
| 14:13.08 | *** join/#asterisk marlowe (~marlowe@bmw.princetonhost.com) |
| 14:14.25 | fishboy1669 | whats the command for checking if i have confilicting irq |
| 14:16.33 | postel | fishboy1669: cat /proc/interrupts |
| 14:18.08 | fishboy1669 | aaaaaaaaaaggggggggggghhhhhhhhhhhhhh irq 11 used by uhci_hdc and wcf_xo |
| 14:18.15 | fishboy1669 | will cause issue? |
| 14:18.20 | *** join/#asterisk PakiPenguin (~info@202.147.173.206) |
| 14:18.29 | PakiPenguin | hello everyone |
| 14:18.44 | fishboy1669 | sorry uhci_hcd |
| 14:19.31 | *** part/#asterisk santiago (~santiago@63.245.86.101) |
| 14:20.07 | markit | anyone here using chan_mISDN? |
| 14:21.24 | derfer | hello i am student and i need help for make a demo with asterisk someone can help me pls ? |
| 14:21.52 | PakiPenguin | ~docs |
| 14:21.53 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:22.00 | derfer | thx |
| 14:22.02 | PakiPenguin | here you go derfer |
| 14:22.54 | rtcg | Well, I guess I don't need BOTH sides of the conversation! Firefly can pass voice traffic TO the TDM40 but voice FROM the TDM40 doesn't make it to the firefly client. This is all INTERNAL - no public IPs in use here. It's all on the same network so there shouldn't be any..firewalling issues. |
| 14:23.17 | rtcg | any pointers as to where I should look? |
| 14:27.00 | *** join/#asterisk BBRodriguez (~BBRodrigu@pD9EA7F9F.dip.t-dialin.net) |
| 14:29.31 | *** join/#asterisk DoT|cobain (cobain@adsl077.r24-Dbg2.Vie.AT.KPNQwest.net) |
| 14:31.13 | *** join/#asterisk BBRodriguez (~BBRodrigu@pD9EA7F9F.dip.t-dialin.net) |
| 14:31.45 | mortehu | Can it be right that my sound card plays back at 7817 Hz even though I asked ALSA for 8000 Hz? |
| 14:32.16 | mortehu | When I stream from Asterisk, I need to drop a lot of packets to not fall too far behind. I time Asterisk's data to 8000.73 Hz. |
| 14:34.50 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-3365.nb.aliant.net) |
| 14:35.23 | *** join/#asterisk uunot (~teliax@c-67-166-37-218.client.comcast.net) |
| 14:38.01 | rtcg | Does 'reload' reload the iax.conf too? |
| 14:39.01 | Chuji | yeah |
| 14:42.26 | derfer | someone know sip software for win32 ? |
| 14:42.39 | Chuji | softphone? |
| 14:42.46 | derfer | yep |
| 14:42.55 | Chuji | xlite? |
| 14:43.05 | Chuji | sjphone |
| 14:43.05 | derfer | xlite work with asterisk ? |
| 14:43.16 | Chuji | sure, any sip phone will |
| 14:43.24 | Chuji | well, that's not totally true |
| 14:43.27 | Chuji | but most yes |
| 14:43.33 | derfer | thx |
| 14:44.03 | Chuji | Here is a nice list |
| 14:44.04 | Chuji | http://www.voip-info.org/tiki-print.php?page=VOIP+Phones |
| 14:44.34 | rtcg | ok, I STILL don't get why sometimes port 5036 is referenced and sometimes 4569 is referenced! |
| 14:45.18 | file[laptop] | there were two different versions of IAX, IAX1 and IAX2 |
| 14:45.24 | file[laptop] | IAX1 used 5036 and IAX2 uses 4569 |
| 14:45.29 | BBRodriguez | Hi people |
| 14:45.30 | rtcg | what gives with this one sided conversation...... |
| 14:45.33 | *** join/#asterisk SIPMAN (~eugeniod@225stb15.codetel.net.do) |
| 14:45.34 | rtcg | hi bbr |
| 14:45.53 | rtcg | ok...so my iax.conf should use port 4569? |
| 14:45.56 | fishboy1669 | hi rtcg |
| 14:45.58 | SIPMAN | hello |
| 14:46.02 | file[laptop] | it will use it, you can't override it |
| 14:46.06 | rtcg | cuz it's currently set to 5036. |
| 14:46.17 | rtcg | Greetings everyone! |
| 14:46.30 | BBRodriguez | I have lots of outgoing SIP channels, in dialplan i need to check the next available channel and dial it, does anybody know how to do that ? |
| 14:46.34 | rtcg | Greetings and salutations! some terrific pig. |
| 14:47.05 | SIPMAN | i have problem with Asterisk and Snom Proxy Server |
| 14:47.27 | SIPMAN | NAT retransmition problem |
| 14:47.31 | SIPMAN | any idea? |
| 14:47.57 | *** join/#asterisk JmanA9 (~josh@h207.182.40.69.ip.alltel.net) |
| 14:48.04 | JmanA9 | hello |
| 14:48.27 | JmanA9 | in order to place a call, do you have to have a sip phone, or can you use a microphone? |
| 14:48.43 | rtcg | Greetings Jman |
| 14:49.19 | rtcg | Just a plain microphone in thin air works wonders. |
| 14:49.36 | Chuji | rtcg : You can use the mic on yoru linux box |
| 14:49.44 | *** join/#asterisk zoa (zoa@82.103.76.147) |
| 14:49.46 | zoa | yo |
| 14:49.59 | JmanA9 | ok, i'll give it a try :) |
| 14:50.01 | Chuji | oops |
| 14:50.04 | rtcg | hee hee heee. :) |
| 14:50.04 | Chuji | meant JmanA9 |
| 14:50.19 | Chuji | JmanA9 Check out http://www.voip-info.org/wiki-Asterisk+tips+console |
| 14:50.42 | rtcg | JmanA9: check out http://www.voip-info.org/tiki-print.php?page=VOIP+Phones too. |
| 14:50.52 | JmanA9 | thanks :) |
| 14:50.59 | Chuji | Yeah, a MUCH better solution ;) |
| 14:51.27 | rtcg | Yeah, someone special posted that link a while back.. |
| 14:51.29 | nullogic | JmanA9: I use firefly ( http://www.virbiage.com/firefly/download/ ) for simple tests.. |
| 14:51.29 | *** join/#asterisk pimpwell (~pimpwell@ool-44c6ab45.dyn.optonline.net) |
| 14:51.52 | rtcg | nullogic: do you get two way conversations on your firefly? cuz I don't. :) |
| 14:51.56 | JmanA9 | i just don't want to have to spend any money if i don't have to, which is why i probably woun't buy a voip phone just yet |
| 14:52.10 | rtcg | JmanA9: go after a softphone till you get it working. |
| 14:52.25 | fishboy1669 | jman xten xlite is ok |
| 14:52.30 | rtcg | You can use your mic, but you'll STILL NEED A PHONE (hard or soft) I perfer hard myself..but that's another topic. |
| 14:52.53 | fishboy1669 | lol carefull wat u say rtg |
| 14:52.54 | JmanA9 | so there's no way i can do this without a phone? |
| 14:53.09 | rtcg | phone can mean manythings |
| 14:53.17 | rtcg | computer software phone = phone. |
| 14:53.19 | JmanA9 | ok |
| 14:53.30 | JmanA9 | well, i just downloaded firefly, i'll give that a go |
| 14:53.40 | fishboy1669 | jman with out some sort of phone how u gonna speak? |
| 14:53.53 | JmanA9 | microphone |
| 14:53.54 | rtcg | telepathy...it's the wave of the future. |
| 14:53.57 | JmanA9 | lol |
| 14:54.01 | fishboy1669 | lmfao |
| 14:54.31 | fishboy1669 | does * have telepahy module yet? |
| 14:54.38 | HjemmeRoyK | chan_telepathy |
| 14:54.38 | JmanA9 | lol |
| 14:55.05 | fishboy1669 | wikid |
| 14:55.25 | BBRodriguez | I have lots of outgoing SIP channels, in dialplan i need to check the next available channel and dial it, does anybody know how to do that ? |
| 14:55.56 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 14:56.39 | cripito | BBR i am interested on that too. |
| 14:57.47 | BBRodriguez | cripito: we need a dialplan |
| 14:58.24 | HjemmeRoyK | dialplans are for chickens |
| 14:58.39 | rtcg | bock bock |
| 14:59.49 | cripito | yes or an appl that do the work for us |
| 15:00.01 | bjohnson | well .. I've reviewed http://voip-forum.tmcnet.com/voip-forum/forum/forum_posts.asp?TID=1707&PN=2&TPN=1 , http://voxilla.com/forum-viewtopic-t-1335.html , and http://www.voip-info.org/tiki-print.php?page=Sipura+3000 .. but I just can't get my SPA 3000 to forward the CID from the PSTN to *. I guess next step is to plug in a CID phone and confirm I'm getting the CID service I'm paying for (although I checked this a while ago) |
| 15:00.01 | BBRodriguez | HjemmeRoyK: how do you recommend to handle it ? |
| 15:00.03 | cripito | i am thinking more in a appl with the channels in db |
| 15:00.36 | Chuji | bjohnson : Yeah, that is fore sure |
| 15:00.39 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 15:00.45 | Chuji | bjohnson : I have mine passing cid without problems |
| 15:01.03 | Chuji | bjohnson : Did you follow those directions I told youlast night? |
| 15:01.19 | HjemmeRoyK | BBRodriguez: exactly what are you trting to do? |
| 15:01.23 | bjohnson | I also have a second SPA 3000 that I might test to see if it is device specific |
| 15:01.48 | bjohnson | Chuji: yes .. that is one of the links I provided .. the voxilla ones are the source for that wiki post |
| 15:02.07 | BBRodriguez | HjemmeRoyK: have over thousand outgoing sip channels, need to check the next available channel, before dialing |
| 15:02.31 | bjohnson | I can't get into the office till later so I'll have to put it off for a while |
| 15:03.23 | BBRodriguez | cripito: there's ${AVAILCHAN} |
| 15:04.24 | HjemmeRoyK | BBRodriguez: is this to an ITSP or something? |
| 15:04.36 | BBRodriguez | HjemmeRoyK: yes |
| 15:05.03 | BBRodriguez | cripito: http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail |
| 15:07.01 | *** join/#asterisk ToyMan (~stuq@user-12lcqq2.cable.mindspring.com) |
| 15:09.47 | bjohnson | BBRodriguez and cripito: I made a superdial macro that I will post to pastebin that I think will do what you guys are looking for |
| 15:10.04 | cripito | thanks bj |
| 15:10.49 | cripito | usually what happen is that u have $ in certains sip channels and the system must be able 2 pick the ones that are available... |
| 15:11.09 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfkgn.dialup.mindspring.com) |
| 15:11.15 | bjohnson | http://pastebin.ca/4581 |
| 15:12.28 | *** join/#asterisk zwi (~zwi@216.88.131.43) |
| 15:13.05 | bjohnson | with that macro .. in extensions.conf for an outgoing call you just list multiple lines like: exten => s,1,Macro(superdial,IAX2/voipjet/${tfnumber},,,,voip,${MAXVOIPCALLS},,,voipjet) and it will take the first one that is available |
| 15:13.27 | bjohnson | it also works for incoming like so .. |
| 15:13.43 | bjohnson | exten => s,3,Macro(superdial,${PHONE3},15,Ttm,,pstn,${MAXPSTNCALLS},,,,u${GENERALVM}) |
| 15:13.48 | bjohnson | exten => s,4,Macro(superdial,${PHONE1},15,Ttm,,pstn,${MAXPSTNCALLS},,,,u${GENERALVM}) |
| 15:13.53 | bjohnson | exten => s,5,Voicemail(b${GENERALVM}) |
| 15:14.27 | rtcg | OK..well it AIN'T firefly, cuz now may other IAX phone that WAS working is now unworkful.. only one side of the conversation there too. |
| 15:14.46 | bjohnson | and then goes to u vm if one times out .. otherwise tries next extension (eg if busy) .. if all are busy or unavailable .. it gets to the last voicemail busy |
| 15:14.56 | bjohnson | now I have to go till tonight |
| 15:18.52 | cripito | thanks bj |
| 15:18.59 | cripito | i will see the macro |
| 15:26.35 | eKo1 | These Clipcomm FXO gateways suck donkey balls. |
| 15:26.36 | rtcg | What search terms should I google for when only one party can talk on an IAX connectiont? |
| 15:27.26 | freat | rtcg: what's happening? |
| 15:28.05 | freat | rtcg: is it basically one-way audio? is it the called person who can't here the callee? |
| 15:28.52 | rtcg | The IAX softphone client can send audio to the hard phone connected to the FXO ... but the softphone never receives any audio. |
| 15:28.53 | freat | rtcg: if Asterisk is bridging then releasing the calls, try keeping * in the middle by putting notransfer=yes into their settings in iax.conf |
| 15:30.19 | freat | that softphone may have trouble talking directly to the hardphone. putting notransfer=yes in there will keep asterisk in the middle |
| 15:30.22 | rtcg | ook... adding notransfer=yes to the context in iax.conf |
| 15:30.35 | freat | yeah |
| 15:30.44 | freat | I had that same problem |
| 15:31.27 | rtcg | well, the thing is, I *HAD* it working with Steven sokhol's(sp) IAXphone. then this AM I tried firefly.... |
| 15:31.37 | freat | when * bridges calls, it actually tries to get the two clients to talk directly to each other. kinda cool... but not always what you want |
| 15:31.43 | rtcg | firefly had problems. So I messed with ports in firefly and IAX.CONF |
| 15:31.57 | freat | 'messed with ports'??? |
| 15:32.09 | rtcg | everything SHOULD be back to the way it was. (only played with one line) only NOW not even iax phone works. |
| 15:32.23 | freat | did you restart * ? |
| 15:32.25 | rtcg | teh sokol one doesn't even work any more. |
| 15:32.37 | rtcg | heck yeah...all the way down to the wcfxo drivers |
| 15:33.06 | freat | are you sure that your softphone is configured right? |
| 15:33.14 | rtcg | modprobe -r wcfxo |
| 15:33.14 | rtcg | modprobe -r wcfxs |
| 15:33.14 | rtcg | modprobe -r wcfxo |
| 15:33.14 | rtcg | modprobe -r wcfxs |
| 15:33.14 | rtcg | modprobe -r zaptel |
| 15:33.14 | rtcg | modprobe zaptel |
| 15:33.16 | rtcg | modprobe wcfxs |
| 15:33.18 | rtcg | modprobe wcfxo |
| 15:33.20 | rtcg | #screen -c /etc/asterisk/screenrc |
| 15:33.22 | rtcg | sleep 3 |
| 15:33.24 | HjemmeRoyK | ~pastebin? |
| 15:33.25 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 15:33.25 | rtcg | asterisk -cvvv |
| 15:33.27 | freat | hhe |
| 15:33.28 | *** join/#asterisk oDin (BuLuT@81.212.12.120) |
| 15:33.56 | rtcg | Um... it was configured correctly before (sorry about the paste) it was small tho.) |
| 15:34.05 | freat | rtcg: sounds like you just need to get yourself back to the original setup. |
| 15:34.26 | rtcg | ok....attempting to verify that I am on the original setup. bbiab. |
| 15:34.36 | freat | rtcg: you don't ever need to modify that port= in the [general] of iax.conf |
| 15:35.06 | freat | rtcg: that doesn't really get used anyways, from what i hear the IAX2 port is hard-coded anyways |
| 15:35.26 | HjemmeRoyK | ~lart rtcg for not using the pastebin |
| 15:36.02 | freat | hehe |
| 15:38.17 | rtcg | I don't even know what lart is.....(IRC NOOBIE after 3 years of off and on use) |
| 15:38.29 | *** join/#asterisk Mehmet (seF@mstr195175-20657.dial-in.ttnet.net.tr) |
| 15:39.00 | rtcg | I did figure out enough to get the "I'm too lame to read the BitchX docs" off my config....AND I deleted all the vulgar signoffs...PHEW. |
| 15:40.03 | wankel | quite a list of accomplishments for a mere three years of your life |
| 15:40.11 | JerJer | when in doubt, whip it out |
| 15:40.28 | JerJer | :) |
| 15:40.41 | file[laptop] | potato? |
| 15:40.48 | fishboy1669 | carrots |
| 15:40.53 | JerJer | corn |
| 15:41.20 | rtcg | Oh yeah. I figure a year per project....that should be enough. |
| 15:41.22 | tzanger | tired of the same old crap, eat more corn! |
| 15:41.27 | freat | rutabaga |
| 15:41.35 | rtcg | ruhbarb. |
| 15:41.38 | rtcg | rubarb |
| 15:41.40 | fishboy1669 | sprouts |
| 15:41.42 | rtcg | rubharb |
| 15:41.49 | fishboy1669 | brocokly |
| 15:41.55 | tzanger | hahahahha |
| 15:41.59 | tzanger | brocokly? |
| 15:42.02 | tzanger | HAHAHAHHAHAHA |
| 15:42.24 | fishboy1669 | peas |
| 15:42.28 | wankel | heh. that's great. |
| 15:42.28 | subx | This ihas to be the most social channel I have ever seen. |
| 15:42.38 | rtcg | It are not. |
| 15:42.47 | tzanger | how about broccoli? |
| 15:42.48 | wankel | really? it's pretty on-topic for irc :) |
| 15:42.50 | fishboy1669 | knot |
| 15:42.51 | subx | lol thanks |
| 15:43.00 | tzanger | steak and eggs... not a bad breakfast |
| 15:43.05 | tzanger | I prefer the southern breakfasts |
| 15:43.16 | fishboy1669 | bacon sandwich |
| 15:43.29 | wankel | i think my usual channel is on-topic maybe 1% of the time. then again, we don't let anyone come in to ask stupid questions anymore, so that helps :) |
| 15:43.32 | tzanger | bacon, eggs, saussage, grits, orange, coffee, orange juice, pancakes, biscuits and gravy... fuck man I need a southern Belle |
| 15:43.53 | fishboy1669 | you let me get away with asking stupid questions! |
| 15:43.54 | wankel | mmmmm, gris. |
| 15:43.58 | wankel | grits even |
| 15:43.59 | rtcg | we all need a southern belle. |
| 15:44.04 | wankel | fishboy: this isn't my usual channel :) |
| 15:44.21 | fishboy1669 | aha |
| 15:44.30 | fishboy1669 | penguin |
| 15:44.41 | rtcg | Well I'm back to my original config and I am still southern belleless. I mean untalkful. |
| 15:45.01 | fishboy1669 | i just want anything in a skirt |
| 15:45.11 | fishboy1669 | within reason of course |
| 15:45.19 | rtcg | You know..... this isn't all bad.. this way I can call home and tell the wife...THINGS...but I won't be able to hear her reply. |
| 15:45.27 | fishboy1669 | jacky stallone is out of the question |
| 15:45.34 | wankel | rtcg: sounds ideal |
| 15:45.36 | eKo1 | Somehow, 'southern belle' reminded me of Bellsouth. |
| 15:45.38 | rtcg | LOL hairy scottsman! eeeewwwww |
| 15:45.59 | rtcg | HA! in those three years... I also learnt how to clear the screen. |
| 15:46.11 | fishboy1669 | thats a kilt man kilt not a skirt |
| 15:46.11 | rtcg | for wife related purpleses. |
| 15:46.27 | file[laptop] | no food for file, darn |
| 15:46.28 | wankel | have you figured out how to turn on the crypto signing for your messages yet? |
| 15:46.43 | rtcg | you who? |
| 15:46.49 | wankel | rtcg: you |
| 15:47.36 | rtcg | I am unclueful as to what crypto signing is. And which messages would I want to be puttful on? |
| 15:48.05 | wankel | just to make sure everyone knows they're really coming from you. just switch to signing mode to by typing "/sign 2" |
| 15:48.20 | wankel | signing mode 2, that is |
| 15:49.00 | rtcg | well, /signing mode 2 isn't workful. |
| 15:49.12 | wankel | no, just "/sign 2" |
| 15:49.28 | wankel | i don't think the bitchx idiots could figure out how to spell signing out the whole way |
| 15:49.41 | rtcg | <PROTECTED> |
| 15:49.56 | wankel | aw, hell. stupid bitchx. |
| 15:50.01 | rtcg | hell ya |
| 15:50.04 | rtcg | ! I agree! |
| 15:50.50 | markit | what does disconnect => *0 in features.conf is mean to do? I thought was for hangin up, but seems not to work.. is it a bug, or does it does something different? cvs head |
| 15:50.54 | rtcg | cool! My eastern belle just brought me sustinance.!!! |
| 15:51.05 | *** join/#asterisk FryGuy (fryguy@c-67-174-57-164.client.comcast.net) |
| 15:51.12 | rtcg | I don't know what it is...I just eat without asking...it's better that way. |
| 15:51.25 | fishboy1669 | btw just so u all know judge jules rocks |
| 15:52.06 | fishboy1669 | tried and tested euphoria |
| 15:52.07 | fishboy1669 | mmmmm |
| 15:52.20 | *** join/#asterisk karbon (~karbon@wbar9.lax1-4-11-199-179.dsl-verizon.net) |
| 15:52.51 | fishboy1669 | wow fit bird with well good legs just walked past my window shorteste denim skirt ever and its january here |
| 15:52.54 | rtcg | ok....working softphone configure and tested outside this network USED to work. Gonna POTS them and get them to test inbound callage. |
| 15:53.51 | karbon | whats up moonwick! |
| 15:54.53 | fishboy1669 | sky |
| 15:54.59 | tzanger | callage? who do you think you are, Pauly Shore? |
| 15:55.10 | rtcg | prolly not. |
| 15:55.39 | *** join/#asterisk LUTOR_ASI (~newbie@icable-docsis2-108.ibw.com.ni) |
| 15:55.52 | LUTOR_ASI | hi |
| 15:56.32 | *** join/#asterisk doughecka_ (~Doug@adsl-18-85-228.sdf.bellsouth.net) |
| 15:56.54 | LUTOR_ASI | i have a noise between pstn and xpro, and then get no comunication, |
| 15:56.56 | *** join/#asterisk jcims (~chatzilla@cpe-69-135-121-57.columbus.rr.com) |
| 15:57.05 | LUTOR_ASI | i'm using alaw codec, |
| 15:57.36 | LUTOR_ASI | when i dial from pstn to xpro i can listen perfectly , but in the other way, i get no sound ... |
| 15:57.40 | LUTOR_ASI | can somebody help me.. |
| 15:58.51 | fishboy1669 | check your phone.conf and your sip.conf for codec settings |
| 15:59.07 | Chuji | LUTOR_ASI : Anything sit between xpro and Asterisk? |
| 15:59.16 | Chuji | i.e. NAT, Firewall, etc |
| 15:59.17 | fishboy1669 | do u have any nat stuff in there |
| 16:00.43 | LUTOR_ASI | no, xpro is my lan.. |
| 16:01.09 | Chuji | Then as fishboy1669 said, make sure there is no NAT settings in your sip.conf |
| 16:01.53 | Chuji | do a sip show channel <chan> and zap show channel <chan> while they are connected. Make sure they are using the codecs you think |
| 16:02.04 | LUTOR_ASI | i have my phone.conf mode=inmediate, format=slinear, echocancel=medium |
| 16:02.09 | beto75 | Leland I did receive information from Bicom , in fact I was on a demo of their softswitch product |
| 16:05.12 | LUTOR_ASI | i get from the zap show channel, default law=ulaw |
| 16:06.13 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 16:06.13 | LUTOR_ASI | sometimes i get comunication, and then lose it, but i usually get no comunication |
| 16:07.40 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 16:12.23 | Chuji | LUTOR_ASI : Try another softphone |
| 16:12.34 | Chuji | LUTOR_ASI : Could be a PC error, not Asterisk |
| 16:14.54 | LUTOR_ASI | but if would be a PC error, why do i get a perfect comunication dialing from PSTN to xpro |
| 16:15.24 | LUTOR_ASI | but from XPRO to PSTN i get no sound.. |
| 16:18.54 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-3365.nb.aliant.net) |
| 16:19.16 | drumkilla | file[laptop]: !!! |
| 16:19.25 | Qwell | drumkilla: !!! |
| 16:19.28 | file[laptop] | drumkilla: !!! |
| 16:19.30 | drumkilla | Qwell: !!! |
| 16:19.36 | Qwell | file[laptop]: !!! |
| 16:19.49 | file[laptop] | Qwell: !!! |
| 16:19.52 | drumkilla | group hug! |
| 16:20.07 | drumkilla | nice! |
| 16:20.09 | file[laptop] | muffins! |
| 16:20.18 | markit | hi drumkilla, I've re-opened bug 3401... maybe the patch is ok, but doesn't solve the problem |
| 16:20.20 | drumkilla | file[laptop]: you see my message for you in the 1.0.4 email? |
| 16:20.30 | drumkilla | markit: yeah, I saw that |
| 16:20.56 | Qwell | drumkilla, file[laptop]: Where do you guys go when you need to get new phones and stuff? |
| 16:21.12 | drumkilla | heh ... depends, I guess |
| 16:21.18 | file[laptop] | drumkilla: ah yes that, saw it |
| 16:21.35 | Qwell | The only two places I know, suck |
| 16:21.38 | file[laptop] | Qwell: depends, I usually call in favors and get stuff cheaper |
| 16:22.15 | drumkilla | I use mostly Zap stuff at home ... |
| 16:22.18 | wankel | anyone know if there's a reason mwi and qualify don't work with realtime or if it's just work that hasn't been done yet? |
| 16:22.43 | Qwell | my work is gonna be getting a couple 7960's, no clue where to go to get them though |
| 16:22.44 | file[laptop] | wankel: it's because the stuff from realtime doesn't stick around |
| 16:22.57 | file[laptop] | Qwell: oh I go to Jon Putnam @ http://www.gtsinc.biz/ for Cisco phones |
| 16:23.05 | wankel | right. is that intentional, desired behavior, or is it just not well integrated yet? |
| 16:23.07 | file[laptop] | he's treated me well... |
| 16:23.20 | Nugget | ewwww. |
| 16:23.25 | drumkilla | I wish i had money to buy a Cisco 7960 :( |
| 16:23.29 | Nugget | I don't want to buy hardware from a website that has music. |
| 16:23.39 | file[laptop] | wankel: the MWI and qualify stuff has to be modified... or realtime has to be expanded... it's just the way it is |
| 16:23.48 | [Sim] | *yawn* |
| 16:23.53 | wankel | okay, so no one's against the idea of it working, then. |
| 16:23.56 | file[laptop] | wankel: MWI and qualify count on the info of the peer sticking around in memory, with realtime - it doesn't |
| 16:23.57 | eKo1 | drumkilla: You want one for yourself? |
| 16:23.59 | Qwell | file[laptop]: Do I actually have to call to get pricing and such? |
| 16:24.11 | drumkilla | eKo1: yeah, i don't have any SIP devices |
| 16:24.12 | file[laptop] | Qwell: just e-mail him... the prices change as he gets stock and stuff |
| 16:24.16 | Qwell | ahh |
| 16:24.29 | wankel | is anyone working on it, or should i? i'm kinda screwed without it :) |
| 16:24.43 | markit | I'm using VoiceMailMain().. the vm-login seems not good to me... "Asterisk Mail. Mailbox?" should be "Asterisk Mail. What mailbox number do you want?" or something like that.. or am I wrong? |
| 16:25.03 | Qwell | file[laptop]: Can you PM me an address? |
| 16:25.12 | file[laptop] | markit: it says, "Comedian Mail. Mailbox?" |
| 16:25.21 | file[laptop] | Qwell: yes |
| 16:25.41 | markit | file[laptop]: it says what I wrote, if you read sounds.txt from CVS head |
| 16:25.55 | file[laptop] | markit: ooh did we get new sounds? |
| 16:26.10 | gambolputty | as if Chris Rock or Gallagher are going to be using * voicemail anytime soon. |
| 16:26.17 | markit | file[laptop]: are you using "stable"? there are new sounds in head, of course :) |
| 16:26.47 | file[laptop] | at any given day I go through stable and head a few times |
| 16:26.52 | markit | in any case, the message should be better... you don't know what is required... seems to wait for a "yes" or "no" ;) |
| 16:27.05 | file[laptop] | markit: then replace it with one you like |
| 16:27.34 | wankel | a terse "Mailbox?" is typical for VMB systems |
| 16:27.42 | markit | file[laptop]: of course I know I can do whatever I want, just discussing the item here... I want to know if my error, or I should file a bug to mantis |
| 16:27.43 | wankel | that's what meridian mail says |
| 16:28.00 | file[laptop] | you people are crazy |
| 16:28.07 | file[laptop] | utterly, utterly, crazy |
| 16:28.45 | markit | wankel: I was trying that command after some time, and I did not remembered what I had to enter... |
| 16:29.01 | drumkilla | file[laptop] is the crazy one ... |
| 16:29.11 | markit | wankel: I was wondering "do I have to enter 1 for yes, 0 for no?" |
| 16:30.00 | wankel | seems obvious enough to me :) |
| 16:30.05 | mikegrb | yes |
| 16:30.12 | file[laptop] | in the asterisk world, nothing is obvious |
| 16:30.21 | file[laptop] | when an error spits out the reason, people still don't catch on |
| 16:30.35 | *** join/#asterisk loick_ (~loick@ATuileries-151-1-29-61.w82-123.abo.wanadoo.fr) |
| 16:30.36 | mikegrb | obviously markit has never used any voicemail system or has an amazing stellar IQ |
| 16:30.38 | file[laptop] | drumkilla: I'm very bitter... I'm turning into Brian |
| 16:30.57 | drumkilla | file[laptop]: that's what happens when you hang around here too much |
| 16:31.03 | file[laptop] | drumkilla: indeed |
| 16:31.09 | markit | mikegrb: thanks, from your reply, I can guess we have the same IQ, but I'm more polite than you :) |
| 16:31.23 | mikegrb | markit: I doubt it |
| 16:32.49 | markit | adding a simple "number" (like "mailbox number") would greatly make thing easier to remember, a lot more user-friendly |
| 16:33.09 | file[laptop] | I've never had someone who couldn't understand to enter their mailbox number |
| 16:33.17 | *** join/#asterisk CoNaN (~GirL_15_@pD958DA95.dip.t-dialin.net) |
| 16:33.34 | wankel | file: yes, but when was the last time an asterisk error message actually included the reason? :) |
| 16:33.55 | markit | file[laptop]: mmm maybe, since I'm providing the italian translation of the sounds, in italian sounds less obvious |
| 16:34.17 | file[laptop] | wankel: if you read the message and think, you figure it out |
| 16:34.35 | wankel | so far i've figured it out after reading the source, most of the time :) |
| 16:35.00 | file[laptop] | I've never had to read the source to figure out an error, unless it's a huge critical error that should never happen |
| 16:37.27 | Qwell | How would an IP phone work over a VPN? |
| 16:37.38 | HjemmeRoyK | I'm try to set it up right now..... |
| 16:37.45 | HjemmeRoyK | with an ATA that supports simple ipsec |
| 16:38.11 | *** join/#asterisk sarumont (~sarumont@pcp04563795pcs.dallas01.ga.comcast.net) |
| 16:38.26 | wankel | @$^T@#$ cable modem |
| 16:38.31 | [Sim] | qwell: if you have enough bandwidth and little latency/jitter it will just work over a vpn :) |
| 16:38.36 | wankel | file: i didn't find "auto-congesting" or "no path to translate" immediately obvious the first time. |
| 16:38.55 | [Sim] | type of vpn (tcp/udp, crypto etc) may have effect on quality ofcourse |
| 16:39.10 | Qwell | [Sim]: I'm almost certain it'll be insanely tricky for me. The tunnel runs in Windows(which is in VMWare) |
| 16:39.13 | file[laptop] | wankel: no path to translate? you really didn't clue in on that? |
| 16:39.20 | wankel | i used openvpn with no crypto and the additional latency wasn't noticable |
| 16:39.27 | [Sim] | qwell: yikes! :-) |
| 16:39.39 | Qwell | gonna have to hack up some ugly routes |
| 16:40.06 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 16:40.22 | wankel | file: no path to translate what? the sip uri? the rtp stream? no, i didn't know what it meant. was plenty clear in the source, though. |
| 16:41.03 | Qwell | hmm, I could have a * install locally on the Linux box, have the phone connect to that, and do IAX to the primary? |
| 16:41.41 | wankel | qwell: if the other end supports iax, sure. |
| 16:41.49 | Qwell | wankel: I'll own the other end |
| 16:42.08 | [Sim] | that will make things easier if NAT is your problem |
| 16:42.25 | *** join/#asterisk lele (~lele@rivendell.windmill.it) |
| 16:45.14 | bjohnson | Qwell: definitely .. I do it all the time. My SIP fxs ports connect to * and * uses iax to connect to my home * or my voip providers |
| 16:45.49 | Qwell | bjohnson: the routing is still gonna be fairly tricky for me, since the tunnel is in vmware |
| 16:46.48 | *** join/#asterisk Zeeek (~Zeeek@80.125.80.38) |
| 16:48.43 | BBRodriguez | Hi people, does anybody know why ChanIsAvail kicks the call out of context ? |
| 16:49.33 | bjohnson | BBRodriguez: did you look at my superdial macro? |
| 16:49.48 | BBRodriguez | bjohnson: where is it ? |
| 16:50.11 | *** join/#asterisk Hakan (Jaz@AC9FA92B.ipt.aol.com) |
| 16:50.15 | elric | when I try and make a call with my softphone iax2 debug gives me an error saying, No Authority Found.. what might be causing this? |
| 16:50.16 | *** join/#asterisk HenryTheBIG (~Asterisk@ool-182c098b.dyn.optonline.net) |
| 16:50.27 | bjohnson | check your irc logs back 1 hour 30 minutes for pastebin link and usage instructions |
| 16:50.35 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 16:50.39 | HenryTheBIG | Hi |
| 16:50.47 | BBRodriguez | bjohnson: I desperately want to look at your superdial macro !!! |
| 16:50.57 | bjohnson | elric: you don't have a matching secret/config in sip.conf or iax.conf |
| 16:51.12 | bjohnson | BBRodriguez: check your irc logs back 1 hour 30 minutes for pastebin link and usage instructions |
| 16:51.30 | elric | bjohnson ok i will check on that |
| 16:52.23 | BBRodriguez | bjohnson: Thank you, looking at pastebin |
| 16:52.25 | bjohnson | elric: to start .. just make a guest account .. should be one (but commented out) in the sample configs |
| 16:52.51 | *** join/#asterisk fearnor (~alex@66.250.55.66) |
| 16:53.26 | bjohnson | BBRodriguez: if you like it, do everyone a favour and post it to the wiki (with usage instructions) so I don't have to take the time |
| 16:53.53 | *** join/#asterisk MuRat (~NumLocK@mstr195175-30021.dial-in.ttnet.net.tr) |
| 16:54.22 | Zeeek | what does it do? |
| 16:54.25 | Nukemizer | is there a way to make asterisk start on boot, and still get CLI access ? or must asterisk alwys be started from command line ? |
| 16:54.39 | Zeeek | Nuke see safe_asterisk |
| 16:54.46 | Nukemizer | thanks :) |
| 16:55.00 | Zeeek | bj what does this incredible macro do ? |
| 16:55.22 | BBRodriguez | bjohnson: I like it and i'll post it, but you don't check for next available channel in your superdial |
| 16:55.35 | n00b101 | When I use a 7940 hold feature it takes 30 seconds for the musichold to be heard |
| 16:55.55 | Zeeek | the band has to warm up and smoke a joint first! |
| 16:56.08 | n00b101 | if i do it on a xten softphone it works instantly |
| 16:56.22 | BBRodriguez | Hi people, does anybody know why ChanIsAvail kicks the call out of context ? |
| 16:56.23 | n00b101 | What is causing the Cisco phone to delay |
| 16:56.26 | Corydon76-home | Yeah, well, the Cisco phone is a piece of shit, to put it mildly |
| 16:56.32 | Zeeek | heh |
| 16:56.45 | Corydon76-home | I'd call it a boat anchor, but that would be insulting to boat anchors |
| 16:57.17 | n00b101 | I like it - using SIP image, though |
| 16:57.33 | wankel | good. don't use the skinny images. |
| 16:57.59 | Nukemizer | might disolve the phone quickly, making the phone unacceptable for even boat anchor |
| 16:58.06 | n00b101 | SIP image works fine, except for thje delay using musiconhold |
| 16:58.25 | BBRodriguez | Please, anyone ? ChanIsAvail drops the call out of current context, does anybody know why ? |
| 16:58.44 | Corydon76-home | Yeah, Grandstream works fine, too, except it doesn't have that delay |
| 16:58.44 | BBRodriguez | does anybody have a working example of ChanIsAvail usage ? |
| 16:59.05 | Corydon76-home | and the GS doesn't cost more than an industrial blender |
| 16:59.22 | wankel | heheh. my customers would laugh at me if i tried to sell them a grandstream |
| 16:59.47 | wankel | things looks like a $20 radio shack PSTN phone |
| 16:59.50 | Corydon76-home | Our customers cut us out of the loop and just buy Grandstreams directly |
| 16:59.52 | wankel | err, POTS phone |
| 17:00.00 | Nukemizer | is there a way to have the status of active phones show on a console like an auto-upateding "sip show peers" ? |
| 17:00.24 | JonR800 | your customers buy GS phones??? you need new customers. |
| 17:00.30 | Corydon76-home | Nukemizer: show channels |
| 17:00.40 | Nukemizer | tnx |
| 17:01.09 | Corydon76-home | New customers? What for? |
| 17:01.12 | *** join/#asterisk renato (~renato@200165045110.user.veloxzone.com.br) |
| 17:01.29 | JonR800 | Because they're obviously cheap |
| 17:01.32 | Corydon76-home | Just because they aren't willing to drop $500+ on a Cisco doesn't make them bad customers |
| 17:01.41 | wankel | $500? |
| 17:01.57 | Corydon76-home | It just puts more money in their budgets to buy programming from us |
| 17:02.03 | JonR800 | black market prices wankel. |
| 17:02.10 | bjohnson | BBRodriguez: that's right .. you just list the next channel to check in your exten priorities |
| 17:02.19 | JonR800 | they're buying them from columbian drug lords. |
| 17:02.21 | Corydon76-home | wankel: yeah, most businesses won't buy from eBay... retail only... |
| 17:02.36 | wankel | jon: ah, that explains it. they probably have coke packed into the handsets. |
| 17:02.42 | *** join/#asterisk calvinhp (~calvinhp@dhcp065-029-088-222.indy.rr.com) |
| 17:02.50 | Zeeek | anyone hooked up the Bellster yet ? |
| 17:03.02 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 17:03.12 | bjohnson | Corydon-w: IP500s are $180 retail |
| 17:03.29 | Corydon76-home | bjohnson: Yeah, that's Polycom, not Cisco |
| 17:03.34 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 17:03.38 | bjohnson | ciscos are about $220 |
| 17:03.39 | *** join/#asterisk Smuggs (~Smuggs@S0106000c41a28668.cg.shawcable.net) |
| 17:03.39 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 17:03.41 | Corydon76-home | bjohnson: and we have customers with Polycom's too |
| 17:04.04 | ctooley | bjohnson, no IP500's are $279 retail, the going rate for them is $180-$190 |
| 17:04.07 | Corydon76-home | bjohnson: on eBay maybe |
| 17:04.08 | *** join/#asterisk nathano (~nigeriand@adsl-68-95-251-105.dsl.rcsntx.swbell.net) |
| 17:04.24 | wankel | you don't pay $500 for a cisco direct from cisco. |
| 17:04.38 | nathano | hello |
| 17:04.49 | wankel | well, not unless you're an idiot. |
| 17:05.03 | Zeeek | nathano lo |
| 17:05.07 | Corydon76-home | wankel: yeah, that's why I don't buy Cisco phones |
| 17:05.27 | Corydon76-home | Besides, they're pieces of crap |
| 17:05.46 | wankel | why, exactly? |
| 17:06.14 | *** join/#asterisk PakiPenguin (~info@202.147.173.206) |
| 17:06.18 | bjohnson | ctooley: http://www.tritechcoa.com .. ip500 for $180 |
| 17:06.22 | PakiPenguin | hello everyone |
| 17:06.28 | Corydon76-home | They're impossible to configure correctly by using the phone console |
| 17:06.28 | Zeeek | hey pki |
| 17:06.35 | Zeeek | up late ? |
| 17:06.48 | wankel | bjohnson: right. that's what he meant by "going rate." the MSRP (which no one ever pays) is like $280 |
| 17:06.50 | PakiPenguin | nah Zeeek: been busy with family all day |
| 17:06.50 | nathano | I am new to Asterisk and was wandering how one could make calls using an Option Globtrotter PCMCIA GSM card and Asterix |
| 17:06.55 | JonR800 | uhhhh |
| 17:07.01 | JonR800 | phone console? |
| 17:07.22 | nathano | I think so the Phone connects via a PCMCIA port |
| 17:07.36 | wankel | corydon: i've set several up from the console alone, but no one sane would try to scale that. that's what the tftp server is for. |
| 17:07.46 | Smuggs | sup i got asterisk working |
| 17:07.57 | Smuggs | any magic numbers i can call to test it? |
| 17:08.01 | wankel | when you have 5000 phones, you don't want to configure them all by pressing buttons. |
| 17:08.03 | Corydon76-home | wankel: everytime we've tried to configure it, most options are grayed out |
| 17:08.05 | JonR800 | bingo wankel .. i wouldn't set more than one up from console.. lol |
| 17:08.16 | wankel | corydon: mehehh. did you read the manual? |
| 17:08.25 | Corydon76-home | wankel: yep |
| 17:08.32 | wankel | evidently not all of it |
| 17:08.41 | Corydon76-home | wankel: yep, all of it |
| 17:08.51 | wankel | okay, did you _understand_ what you read? |
| 17:08.57 | Corydon76-home | wankel: yep |
| 17:09.02 | wankel | then you can't follow directions, i guess. |
| 17:09.07 | pulu | hahhaa |
| 17:09.13 | Corydon76-home | We unlocked the phone, tried it, still greyed out |
| 17:09.16 | pulu | people are so friendly in this channel |
| 17:09.47 | wankel | pulu: he didn't ask "how do i make the greyed out options on a cisco work?" he came in saying "ciscos are total pieces of shit." |
| 17:09.57 | wankel | apparently because he can't figure out how to work them |
| 17:10.23 | nathano | So is it possible to make voice calls using a Option Globtrotter PC GSM card with Asterix? |
| 17:10.24 | ctooley | bjohnson, I realize that's the going rate, that's what I sell them for too. That does not make it retail |
| 17:10.29 | Corydon76-home | And if a genius can't figure it out, maybe it's a piece of crap |
| 17:11.13 | wankel | the simple often eludes the genius |
| 17:11.45 | ctooley | Genius usually need not declare itself either. |
| 17:11.59 | Corydon76-home | ctooley: yeah, well |
| 17:12.07 | drumkilla | Corydon76-home has submitted *a lot* of code to the asterisk code base ... |
| 17:12.24 | nathano | any takers on the globtrotter GSM card and Asterix? |
| 17:12.29 | renato | Hi! Newbie question: The only option I have to connect normal phones on a Asterisk PBX is throuhg the TDM400P whith FXS modules? |
| 17:12.45 | Corydon76-home | ctooley: I don't necessarily agree with the practice of attaching the word "genius" to an IQ range, rather than attaching it to accomplishments, but that's how it is |
| 17:12.50 | *** join/#asterisk fheese (~fheese@dsl-084-057-005-128.arcor-ip.net) |
| 17:12.58 | ctooley | drumkilla, submitting software to the project does not make a genius, nor someone cable of administration of equipment. I know all to many people capable of one and not the other. |
| 17:13.00 | Qwell | renato: You could also use something like a Sipura SPA-2000, but the TDM is pretty good |
| 17:13.36 | renato | Qwell: I'm concerned about prices. Which is the best option? |
| 17:13.43 | Qwell | renato: I'd go with the TDM, really |
| 17:13.44 | ctooley | I'm not trying to infer that Corydon76-home is an idiot or anything of the light, but arrogance gained through one accomplishment does not ensure success in another. |
| 17:14.04 | nathano | rento: So its not possible or would a new channel driver be required? |
| 17:14.10 | bjohnson | BBRodriguez: http://pastebin.ca/4589 (with usage instruction included |
| 17:14.49 | Corydon76-home | ctooley: that's certainly obvious in this channel |
| 17:15.02 | Nukemizer | since sip phones do not like NAT ( from what i have been watching here) do IAX phones fair any better when looking for a phone that supports NAT ? |
| 17:15.14 | Nugget | NAT blows goats. |
| 17:15.20 | renato | Qwell: In this case, I'll have a cost of near $75 dolar per phone. This is considered a reasonable price for a PBX solution? Whe have hundreds of extension in my company |
| 17:15.29 | Corydon76-home | IAX was specifically designed with NAT in mind |
| 17:15.31 | Nugget | but yeah, IAX is generally less of a pain in the ass with NAT than SIP is. |
| 17:15.38 | Qwell | renato: hundreds? Why not just get IP phones then? |
| 17:15.52 | Qwell | (and $75 is pretty damn cheap, AFAIK) |
| 17:15.59 | renato | IP phones are expensives also |
| 17:16.14 | Corydon76-home | Qwell: because IP phones are all more expensive than $75? |
| 17:16.17 | Nukemizer | lol |
| 17:16.27 | Qwell | How many PCI cards do you think will fit in one system, exactly? |
| 17:16.36 | Qwell | 6 tops? 6*4=24 phones |
| 17:16.38 | Nukemizer | SWEET ! |
| 17:16.38 | wankel | deploying ip phones may also require infrastructure upgrades... new switches with QOS support, new cabling, etc. |
| 17:16.40 | fearnor | renato: if you have lots of extensions, t400p+channel banks+ADSI phones = excellent solution. |
| 17:16.51 | fearnor | qwell: welcome to "channel banks". google it. |
| 17:16.54 | fearnor | thanks bye. |
| 17:16.55 | renato | Qwell. I was considering migrate from our proprietary PBX solution. |
| 17:16.57 | Corydon76-home | Qwell: that's why you use T1 ports and channel banks |
| 17:17.13 | Nukemizer | Nugget, Coryddon - Thanks |
| 17:17.18 | Qwell | Hey, I was answering the question he asked, no need to flame me |
| 17:17.20 | Nugget | glad I could help! :) |
| 17:17.31 | renato | fearnor: many tanx I'll google it |
| 17:17.42 | Nukemizer | you guys are awsome - |
| 17:17.48 | Smuggs | yo could u please post a few test numbers I can use with asterisk. im excited i got it working w/ "dial 500" but no one answered. i'm wondering if someone can actually hear me using my mic |
| 17:17.55 | Nugget | without me it's just "awseo"! |
| 17:18.21 | renato | Qwell: many tanks too |
| 17:18.45 | Zeeek | awe |
| 17:19.17 | Qwell | Nugget: aweso> |
| 17:19.31 | Corydon76-home | Smuggs: nobody answered, because it's Saturday, and Digium is closed |
| 17:19.45 | Smuggs | jah i know that |
| 17:19.50 | *** join/#asterisk AgiNamu (~AgiNamu@200.12.43.74) |
| 17:19.57 | nathano | what FXS modules can be used with a GSM PCMCIA card to make calls |
| 17:19.57 | Smuggs | Corydon-w, gimme your number i just wanna see if this worx |
| 17:20.02 | Zeeek | Smuggs sign up for FWD or IAXTEL |
| 17:20.02 | Qwell | Smuggs: You could get FWD account or something, and call an 800 number |
| 17:20.06 | Smuggs | oh ok |
| 17:20.09 | *** join/#asterisk calvinhp_ (~calvinhp@dhcp065-029-088-222.indy.rr.com) |
| 17:20.18 | Zeeek | great minds and all that... |
| 17:20.18 | Qwell | erm, iaxtel is the one that does 800, isn't it? whichever |
| 17:20.27 | Corydon76-home | 700 |
| 17:20.34 | Nukemizer | I have one pc with a test phone on it and it keeps doing this - Peer '203' is now TOO LAGGED! then seconds after - Peer '203' is now REACHABLE! |
| 17:20.36 | Zeeek | between those two, ât least one is usually up |
| 17:20.52 | Nukemizer | the lan is good the PC is an HP - POS ( new i mide add ) |
| 17:20.59 | Qwell | I haven't had luck with fwd yet |
| 17:21.02 | Nukemizer | (might add) |
| 17:21.18 | Corydon76-home | Nukemizer: point of sale? |
| 17:21.22 | Zeeek | Qwell in what way? It works, mostly |
| 17:21.27 | AgiNamu | Did someone here want to talk about my IAX2 ATAs / Phones? |
| 17:21.38 | PakiPenguin | i want some :p |
| 17:21.43 | Nukemizer | piece of Sh.. |
| 17:21.43 | Zeeek | has anyone used Bellster yet ? |
| 17:21.43 | PakiPenguin | AgiNamu:) |
| 17:21.44 | Qwell | Zeeek: dunno, I only tried it for a day or two, and couldn't get out anywhere. I forget exactly |
| 17:21.59 | renato | fearnor: Is this possible to use t400p+channel banks+common telephones? I whant to proceed the migration with minimun adicional cost. |
| 17:22.08 | Zeeek | Qwell I just use it to test. Echo test, time test, 800 numbers, calling myself etc |
| 17:22.25 | Qwell | Zeeek: Thats what I was doing. Didn't have a mic for the echo test, and I couldn't hit 800 |
| 17:22.32 | Zeeek | Paki talk to Farfon |
| 17:22.44 | fearnor | renato: yes |
| 17:22.49 | Nukemizer | with XP no less , can not keep then in Linux . always bitching "can not run our games" arrrgg |
| 17:22.49 | Zeeek | 18005551212 -tell me |
| 17:23.15 | renato | fearnor: tank yoy. I'm going to study this |
| 17:23.17 | PakiPenguin | Zeeek: yes i will , theycost too much :p |
| 17:23.31 | Zeeek | *no shipping tho' :) |
| 17:23.46 | PakiPenguin | Zeeek: convergence is shifting to Lahore now |
| 17:23.49 | fearnor | anyone here has a contact at chiwanese cointract manufacturing company? ;) |
| 17:23.53 | *** join/#asterisk chachoo (~root@209.234.83.19) |
| 17:24.01 | Zeeek | Paki where is that? |
| 17:24.09 | Nukemizer | if you use the FXO modules versus the T1 won't you be getting a whole bunch of echo when making phone calls ? |
| 17:24.14 | PakiPenguin | its around 400KM from their current place :p |
| 17:24.30 | fearnor | nuke: adit 600 channel bank, done/done. |
| 17:24.35 | Zeeek | since all I k,now is my phones are stuck in OUTGOING customs! |
| 17:24.37 | fearnor | no echo problems ever since i switched to adit |
| 17:24.54 | PakiPenguin | Zeeek: customs needs a lot of bribes here , how may did you order? |
| 17:25.04 | Zeeek | two for the moment |
| 17:25.12 | Zeeek | the bribes are on them though, not me |
| 17:25.21 | PakiPenguin | your location Zeeek? |
| 17:25.27 | Zeeek | Paris |
| 17:25.46 | PakiPenguin | ah, you should've told me before ( a cousin of mine just left, he could've carried those for you) |
| 17:25.51 | PakiPenguin | no customs :p no shipping |
| 17:25.54 | Nukemizer | if you have POTS lines to begin with where would you be puttin the channel bank ? |
| 17:26.07 | Zeeek | Pkai BLOODY HELL!!! |
| 17:26.18 | Zeeek | you have many cousins, yes? |
| 17:26.42 | PakiPenguin | lol , not that many , but one lives in paris and works for nestle :p |
| 17:26.52 | Zeeek | ah ha |
| 17:27.06 | wankel | the problem with VOIP hardware is that you can't just put it in a condom and get someone to swallow it before going through customs |
| 17:27.13 | Nukemizer | i guees i should have asked in this way, if you are strting with POTS and use FXO your going to get echo ? right ? unless you use the T1 interface ? |
| 17:27.16 | PakiPenguin | peire or something is the brand he manages i dont know :p too non-geekish for me :p |
| 17:27.33 | Zeeek | wankel if its small it wont cause problems though |
| 17:27.42 | Zeeek | like an IAXy, that's pretty easy |
| 17:28.26 | wankel | yeah. 5300s were harder to get through customs sometimes. luckily, switching the voip blades with mica modem blades usually did the trick |
| 17:28.48 | fearnor | like customs has any freaking idea what mica and voip blades are |
| 17:29.05 | PakiPenguin | :) |
| 17:29.09 | Zeeek | they loved my woodwinds synthesizer |
| 17:29.12 | wankel | they seemed to in the caribbean |
| 17:29.17 | wankel | much to my surprise |
| 17:29.17 | *** join/#asterisk ReaL (~flyywoman@81.212.12.103) |
| 17:29.19 | fearnor | wankel: neat |
| 17:29.46 | wankel | i guess cable & wireless gave them a hit sheet of things to look for or something |
| 17:29.49 | *** join/#asterisk calvinhp (~calvinhp@dhcp065-029-088-222.indy.rr.com) |
| 17:30.00 | PakiPenguin | lol, they do here too :( |
| 17:30.25 | wankel | the dumbass US customs people just wanted to pull things apart with crowbars to look for bombs |
| 17:30.52 | ardor | if i want to use gsm only its |
| 17:30.55 | Zeeek | wankel coming in? |
| 17:31.02 | wankel | returing |
| 17:31.06 | wankel | returning, even |
| 17:31.06 | ardor | disallow=all ... then allow=gsm ? |
| 17:31.12 | PakiPenguin | yes |
| 17:31.28 | *** join/#asterisk nathano (~nigeriand@adsl-68-95-251-105.dsl.rcsntx.swbell.net) |
| 17:31.44 | Zeeek | years ago I had to register an advanced radio I took to Europe and then brought back |
| 17:31.56 | Zeeek | same with all out instruments |
| 17:32.08 | Nukemizer | fearnor: looking at adit 600 - that is a pretty serious echo solver. you use that for small applications ? |
| 17:32.35 | Zeeek | who has heard about the new Sipura cheap SIP phones? Are they working well with * ? |
| 17:32.50 | *** part/#asterisk nathano (~nigeriand@adsl-68-95-251-105.dsl.rcsntx.swbell.net) |
| 17:32.57 | *** join/#asterisk ManxPower (~eric@adsl-35-239-85.msy.bellsouth.net) |
| 17:33.54 | Zeeek | someone had one - wondered how well it worked |
| 17:35.03 | PakiPenguin | Zeeek: link me to it please |
| 17:35.21 | *** join/#asterisk Getty (torsten@metaluna4.de) |
| 17:35.53 | Zeeek | dunno just go to sipura - I think voxilla has them on offer |
| 17:36.11 | *** join/#asterisk moonwick (~moonwick@core.dump.net) |
| 17:36.50 | wankel | fearnor: you just use the adit as a channel bank? what's up with the voip features? |
| 17:38.07 | wankel | ah, voip gateway is new in 9.0 and not covered in that older pdf |
| 17:48.07 | bjohnson | zeeEK: ManxPower has one |
| 17:49.25 | wankel | manx said nice things about them. i'm still waiting for mine. |
| 17:49.26 | Smuggs | i have modified default extensions.conf and iax.conf to register with IAXTEL networks. iax.conf (register => Smuggs:****@iaxtel.com) Though when i launch asterisk (asterisk -vvvvg) I get a registration errors (chan_iax2.c:6466 socket_read: Reg of Smuggs rejected: Reg refused). Wondering what ports if any I need to open on my routers firewall. Possibly if you have an account I'd be able to substitute so I can properly test this out |
| 17:49.30 | bjohnson | Nukemizer: I use SPA 3000 and X100P for fxo and do not get echo |
| 17:50.22 | bjohnson | wankel: I think Aginamu's would be similar .. but his are IAX |
| 17:51.06 | bjohnson | Smuggs: Iaxtel has been unreliable for me for weeks .. and I coulnd never get it to do 800 numbers anyway |
| 17:51.29 | Smuggs | bjohnson, ok so I guess FWD is my only other choice for testing |
| 17:51.43 | wankel | not sure i want to push IAX all the way out to the desk |
| 17:52.13 | bjohnson | wankel: why not? It would sure be easier for external users too. |
| 17:52.38 | wankel | not real happy about the density it seems i can get with asterisk |
| 17:52.48 | wankel | i'd rather just use it for services where possible |
| 17:52.56 | bjohnson | how come nobody is oohing and awwing at my superdial macro? |
| 17:53.17 | *** join/#asterisk Tili (~Tili@202-133-65-98-dialup.sat.net.pk) |
| 17:54.43 | bjohnson | Smuggs: there is always other choices .. and * box at home for instance |
| 17:54.44 | *** join/#asterisk rtcg (~rtcg@bdsl.66.15.181.96.gte.net) |
| 17:54.47 | ManxPower | Gave one what? |
| 17:54.52 | ManxPower | Have one what? |
| 17:54.57 | wankel | new spa |
| 17:55.00 | bjohnson | 841 |
| 17:55.05 | ManxPower | Ah. Yes. Two of them actually. |
| 17:55.23 | bjohnson | zeeek was looking for an evaluation .. but he left |
| 17:55.32 | rtcg | Well I figured it out!!! in an unsolutionful way.. |
| 17:55.42 | ManxPower | But at the moment I'm trying to figure out why iptables is NATing everything except the SPA-3k |
| 17:56.00 | rtcg | When I have the IAX softphone REGISTER with the * server, the TDM zap ports can't send audio to the softphone. |
| 17:56.13 | bjohnson | ManxPower: port 5061? |
| 17:56.23 | ManxPower | 5060 |
| 17:56.34 | rtcg | If I don't have the softphone register, everything seems to work fine. |
| 17:56.49 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 17:57.22 | *** join/#asterisk freat[laptop] (~freat[lap@node-40242662.mdw.onnet.us.uu.net) |
| 17:57.53 | ManxPower | I only have one iptables rule. MASQ everything. |
| 18:01.56 | rtcg | So, does anyone have a clue as to why * would stop sending VOICE traffic to a softphone if the softphone registers with the * server? |
| 18:02.32 | ManxPower | rtcg, Is the phone behind NAT? |
| 18:02.59 | rtcg | there are no public IP addresses in use. |
| 18:03.40 | rtcg | The asterisk server in question is behind a firewall and the softphone is behind the same firewall. |
| 18:03.59 | rtcg | The asterisk server in question is NOT the firewall nor is it the default gateway on the network. |
| 18:05.33 | Schism | anyone have an update about astricon europe / astricon training there? |
| 18:05.50 | freat[laptop] | rtcg: didn't you say yourself that you had it working earlier, then you messed with the config files and now it doesn't work? |
| 18:06.17 | ManxPower | rtcg, no idea then |
| 18:06.45 | rtcg | well..I have it working again... what BROKE it was having the softphone REGISTER with the * server. |
| 18:07.08 | rtcg | Now that I'm no longer REGISTERing the softphone with the * server...I have two way communications again. |
| 18:07.24 | freat[laptop] | rtcg: it sounds like you don't have a lot of customizations in your files. archive your configs to another folder then make the samples again |
| 18:07.55 | wankel | niiice. 24-30" here tonight. should be a fun one. |
| 18:08.11 | rtcg | well...I actually DO have * in use heavily..I'm just trying to add this softphone feature. |
| 18:08.16 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 18:08.17 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 18:08.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:08.28 | czero | wankel:) |
| 18:08.54 | rtcg | I have an extensive extensions.conf and zapata.conf to do distinctive ring detection => context$x |
| 18:09.06 | rtcg | but I can TRY the default configs just for grins.... |
| 18:09.19 | rtcg | should I still try the default configs? |
| 18:09.25 | *** part/#asterisk loick_ (~loick@ATuileries-151-1-29-61.w82-123.abo.wanadoo.fr) |
| 18:09.50 | freat | I think you should... get the soft phone working that way, then start rolling back the changes you made until it breaks |
| 18:10.08 | rtcg | Softphone NEVER worked when registered. |
| 18:10.16 | rtcg | Softphone worked with NOT registered. |
| 18:10.28 | freat | what softphone are you using? |
| 18:10.29 | rtcg | Earlier I was unclueful as to WHY it used to work and not now.. |
| 18:10.52 | *** join/#asterisk dg1nsw_ (~dg1nsw@saturn2.franken.de) |
| 18:10.57 | rtcg | Sokol IAXPhone workes unregistered. Firefly does NOT work either way (registered or not) |
| 18:11.26 | rtcg | what piece of the * configuration deals with the registering of softphones? |
| 18:11.32 | freat | did you follow the directions on voip-info for those phones? there are some good documentations there |
| 18:12.16 | freat | http://www.voip-info.org/wiki-Asterisk+phones |
| 18:12.22 | rtcg | hm.. I'm on the pages aBOUT each of those phones. but... |
| 18:12.25 | rtcg | let me go there. |
| 18:12.43 | freat | soft phones are devices, like everything else... |
| 18:13.00 | freat | if it's IAX, then you set it up in iax.conf |
| 18:13.04 | freat | if it's sip.... |
| 18:14.30 | rtcg | it's IAX... and I've been in and out of the iax conf MANY times today. |
| 18:14.32 | Lethol | anyonw know whos selling $8-$10 cisco service contracts? |
| 18:14.40 | BBRodriguez | Please, someone tell me what am i doing wrong ? http://pastebin.ca/4590 |
| 18:14.56 | freat[laptop] | pastebin your iax.conf file |
| 18:15.30 | czero | 8-10$ cisco contracts? |
| 18:15.53 | freat[laptop] | http://pastebin.ca/ |
| 18:16.04 | freat[laptop] | then post the link here |
| 18:16.24 | rtcg | oh...here's a stupid issue that I'll eventuallyhave to deal with. During all my restarting of asterisk (unload kernel modules and reload...start *) my FXO zap channels got swapped...so line two was line one and line one was line two. |
| 18:16.27 | Lethol | czero: the wiki talks bout them, but trying to get one from CDW didnt work |
| 18:17.28 | Lethol | im stuck trying to get a cisco 7905 sip firmware.. |
| 18:18.17 | rtcg | GREAT!!! Now firefly works as long as it does NOT register.. So I have *2* working softphones....but only so long as they don't register with * |
| 18:18.56 | rtcg | once they register...it's game over until I restart * |
| 18:19.08 | BBRodriguez | ChanIsAvail() always returns the same value, please someone check my dialplan out http://pastebin.ca/4590 |
| 18:19.24 | freat[laptop] | what does 'iax2 show peers' tell you? |
| 18:19.42 | *** join/#asterisk LUTOR_ASI (~newbie@icable-docsis2-108.ibw.com.ni) |
| 18:19.47 | *** join/#asterisk eschvoca (~eschvoca@dailyglen.istop.com) |
| 18:20.02 | LUTOR_ASI | hi |
| 18:20.06 | rtcg | hi |
| 18:20.10 | eschvoca | hi |
| 18:20.26 | freat[laptop] | rtcg: do "iax2 show peers" at the server console. you should see your soft phones there |
| 18:20.39 | *** join/#asterisk mitcheloc (~mitcheloc@ca-fullerton-69-166-193-228.vnnyca.adelphia.net) |
| 18:20.41 | LUTOR_ASI | i 'd like to configure asterisk to call from xpro to pstn..but i have a problem |
| 18:20.53 | LUTOR_ASI | i only get one way communication.. |
| 18:21.40 | wankel | yeah, good luck getting ahold of one of the cheap cisco contracts :) |
| 18:22.17 | mitcheloc | use ams.net and contact someone there, i'm pretty sure they can help with the contract |
| 18:22.18 | rtcg | I only get one way communication too. :) iax2 show peers shows me that my softphone has registered it's ip address on port 4560 and it's unmonitored. |
| 18:23.16 | Smuggs | hehe it works. fwd 4 life nigga |
| 18:24.02 | wankel | the contract does exist: |
| 18:24.26 | wankel | CON-SNT-CP7960 8x5xNBD Svc, IP Phone 7960, Mgr Set (w/User Lic) N/A $8 |
| 18:24.50 | *** join/#asterisk nullogic (~nullogic@216.24.172.242) |
| 18:25.01 | *** join/#asterisk nighty- (~nighty@sushi.rural-networks.com) |
| 18:25.13 | eschvoca | Anyone know how to ring with a different tone for long distance? |
| 18:25.14 | Qwell | 7960 license? |
| 18:25.18 | *** part/#asterisk nighty- (~nighty@sushi.rural-networks.com) |
| 18:25.52 | wankel | yeah, lethol was looking for one |
| 18:26.00 | wankel | cdw couldn't find it, apparently |
| 18:27.36 | rtcg | eschvoca: are you talking about distinctive ring? oh...if the incoming call is long distnace THEN do a distinctive ring! oh... |
| 18:27.58 | mitcheloc | wankel: ams.net |
| 18:28.00 | eschvoca | rtcg, yes |
| 18:28.09 | mitcheloc | they've got the best prices on everything |
| 18:28.11 | mitcheloc | =) |
| 18:28.11 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 18:28.47 | eschvoca | rtcg, if incoming is long distance (I know the expression for that), then Dial an internal extension with a different kind of ring! |
| 18:29.08 | eschvoca | There is Playtone but how does it interact with Dial? |
| 18:29.27 | mitcheloc | anyone buy a 7940 lately? i'm wondering what they go for now? |
| 18:29.47 | mitcheloc | i bout a bunch a long time ago for less then $210 |
| 18:29.56 | mitcheloc | so i was wondering if they are cheaper now |
| 18:30.07 | netsurfer | anyone know of a good call center module, like ICD only with decent documentation ? |
| 18:30.11 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-20-118.d4.club-internet.fr) |
| 18:30.36 | BBRodriguez | ChanIsAvail() always returns the same value, please someone check my dialplan out http://pastebin.ca/4590 |
| 18:31.24 | wankel | mitcheloc: they want 3xMSRP for the support contract :P |
| 18:32.15 | netsurfer | BBRodriguez - im quite a n00b to this, but shouldnt u have , instead of & |
| 18:32.37 | netsurfer | actually, ignore that.. im wrong |
| 18:33.39 | BBRodriguez | netsurfer: http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail |
| 18:34.31 | *** join/#asterisk oFf (~Hale22__-@217.131.174.217) |
| 18:34.47 | BBRodriguez | netsurfer: & is the needed character, i think, after first run, $AVAILCHAN doesn't get updated |
| 18:35.27 | BBRodriguez | ChanIsAvail() always returns the same value, i think my dialplan is wrong somehow http://pastebin.ca/4590 |
| 18:36.18 | mitcheloc | has anyone received this error while modprobeing wcfx0??? nmi received for unknown reason 31 on cpu 0 |
| 18:36.22 | visik7 | what's the difference between chan_misdn and chan_modem_i4l ? |
| 18:36.57 | mitcheloc | my problem is very similar to this http://lists.digium.com/pipermail/asterisk-users/2004-December/079256.html |
| 18:40.48 | *** join/#asterisk Druken (Druken@CPE00023f0862f9-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 18:41.27 | wankel | mitcheloc: yep, looks very similar. |
| 18:42.12 | mitcheloc | wankel: sorry? do you know how to fix it? |
| 18:42.20 | *** part/#asterisk bkw_ (~brian@65.38.28.146) |
| 18:42.21 | *** part/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 18:43.04 | wankel | nope. i suspect it's an issue with the kernel support for your chipset, or possibly a zaptel driver issues with it. |
| 18:43.36 | Leland | evening... anyone know of a common reason that an inbound call would ring fine, but there's no audio? If I set the phone itself to talk to the provider directly it works fine, but not via asterisk. |
| 18:44.31 | *** join/#asterisk abbas_ (~nid@203.128.19.200) |
| 18:44.54 | freat[laptop] | Leland: is it SIP? |
| 18:44.54 | Leland | both the phone and the sip provider peer lines in sip.conf are set to "canreinvite=no" (which I saw on the wiki somewhere.. but that didn't appear to fix the problem) |
| 18:45.17 | Leland | freat[laptop]: the phone itself will do SIP or H323... same behaviour on both... provider is SIP |
| 18:45.35 | freat[laptop] | Leland: sounds like a NAT issue. |
| 18:45.41 | Leland | no nat involved. |
| 18:45.56 | freat[laptop] | your phone has a public IP? |
| 18:45.58 | Leland | yes |
| 18:46.13 | wankel | any firewalls involved? |
| 18:46.15 | Leland | no |
| 18:46.29 | Leland | at least not affecting the IPs concerned |
| 18:47.19 | rtcg | wow....Leland's problem sounds....somehow familiar. |
| 18:47.20 | *** join/#asterisk JmanA9 (~josh@h207.182.40.69.ip.alltel.net) |
| 18:47.22 | Leland | it's as if once the call is setup, the RTP is trying to go direct from the phone to the provider rather than relaying through asterisk |
| 18:47.45 | Leland | but I don't have a facility to trace to see if that's actually what's happening, unfortunately |
| 18:47.50 | wankel | run ethereal on the asterisk box and take a look at what's going on |
| 18:47.55 | freat[laptop] | does * release the call? |
| 18:47.57 | *** join/#asterisk devi0us (devi0us@west.philly.ghetto.org) |
| 18:48.08 | rtcg | what about the notransfer=yes is THIS a good place for it? |
| 18:48.20 | rtcg | (I'm not wise...just likewise) |
| 18:48.30 | rtcg | (and even THAT is debatable) |
| 18:48.34 | Leland | * does not release the call |
| 18:48.42 | freat[laptop] | ok |
| 18:48.58 | freat[laptop] | yeah sounds like your RTP is getting messed up |
| 18:49.10 | Leland | I take it that means that after it's setup, the call is trying to go direct peer to peer from the phone to the provider then ? |
| 18:49.30 | freat[laptop] | yeah but I think that only applies to IAX protocol... |
| 18:49.58 | pimpwell | is this what I need: ftp://ftp.asterisk.org/pub/asterisk/asterisk-1.0.4.tar.gz |
| 18:50.05 | pimpwell | thats the main installation? |
| 18:50.29 | netsurfer | yes pimpwell |
| 18:50.35 | pimpwell | x |
| 18:50.37 | pimpwell | thx |
| 18:50.51 | Leland | freat[laptop]: so any work-arounds that you're aware of ? |
| 18:51.01 | netsurfer | pimpwell - depending on ur hardware u may need more |
| 18:51.21 | freat[laptop] | I'm not sure what the problem is... |
| 18:51.23 | pimpwell | just a debian box |
| 18:51.27 | Leland | hmm |
| 18:51.36 | *** join/#asterisk Umaro (~umaro@c-24-22-76-14.client.comcast.net) |
| 18:51.39 | Umaro | Hey guys |
| 18:51.44 | Leland | I get exactly the same thing if I use H323 between the phone and asterisk as well.. |
| 18:51.47 | Leland | weird |
| 18:51.58 | freat[laptop] | like wankel suggested, sniffing traffic would be a good idea |
| 18:52.13 | netsurfer | pimpwell - try: apt-get install asterisk |
| 18:52.17 | netsurfer | works great |
| 18:52.27 | netsurfer | u wont get 1.0.4 though |
| 18:52.58 | netsurfer | pimpwell - debian testing release has asterisk 1.0.2 in apt |
| 18:53.39 | Umaro | trying to compile current CVS.. wish me luck! ;) |
| 18:53.49 | Umaro | too bad 1.0.4 doesn't seem to have realtime in it |
| 18:54.13 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 18:54.23 | Leland | blah.. the UK debian mirror keeps changing their flipping directory structure... totally screws up the sources.list |
| 18:54.24 | drumkilla | Umaro: realtime will be in the next major release - 1.2 |
| 18:55.03 | Umaro | drumkilla: :( |
| 18:55.16 | drumkilla | it won't be too long |
| 18:55.26 | f00b3r | mmm |
| 18:55.27 | f00b3r | weekend |
| 18:55.34 | mitcheloc | does anyone here know anything about a kernel error saying that perhaps i have a strange power mode enabled? similar to this other asterisk users problem: http://lists.digium.com/pipermail/asterisk-users/2004-December/079256.html |
| 18:55.41 | Umaro | drumkilla: not to make you promise, but do you think the current CVS is stable enough that it won't reduce my * to a puddle of telephony goo? |
| 18:55.59 | f00b3r | howdy Umaro |
| 18:56.08 | Umaro | f00b3r: hey :) |
| 18:56.09 | freat[laptop] | Umaro: running HEAD is not recommended for production |
| 18:56.17 | drumkilla | Umaro: I wouldn't recommend running head in production unless you're willing to have problems |
| 18:56.33 | Umaro | Aww.. |
| 18:57.06 | netsurfer | maybe thats why my moh isnt working :o\ |
| 18:57.27 | *** join/#asterisk afrosheen (~chatzilla@txprotoa17.august.net) |
| 18:57.32 | afrosheen | hey gang |
| 18:57.43 | Umaro | wankel: yah.. I guess i'll have to wait though.. my client might not understand the benefits of realtime if the * crashes :/ |
| 18:57.48 | freat[laptop] | is realtime where you put iax users in a db? |
| 18:57.54 | drumkilla | Umaro: agreed |
| 18:58.00 | wankel | put various things in a db |
| 18:58.19 | f00b3r | damn, that will be so nice |
| 18:58.25 | wankel | my only other option is to generate static config files from the db and kick asterisk every time something changes |
| 18:58.34 | Umaro | I've been looking at figuring out SER |
| 18:58.37 | drumkilla | you can do it in 1.0, but it's limited |
| 18:58.41 | drumkilla | and you still have to issue a reload |
| 18:58.48 | Umaro | for a frontend to my * servers.. pretty cool stuff |
| 18:58.53 | blitzrage | yo yo yo |
| 18:58.55 | freat[laptop] | yeah that's the next step I want to go to... use odbc to connect to a custom db. provides for excellent application integration possibilities... |
| 18:59.04 | afrosheen | Umaro: AMP is pretty nice now |
| 18:59.13 | blitzrage | drumkilla: !!! |
| 18:59.17 | blitzrage | file[laptop]: !!! |
| 18:59.35 | wankel | umaro: i've had some problems with nat using ser. asterisk likes to talk directly back to phones, and the firewall only has NAT open for the ser proxy port |
| 18:59.43 | mitcheloc | anyone know about that issue? it's with an fxo card...it's acting stupid =/ |
| 18:59.45 | file[laptop] | blitzrage: ??? |
| 18:59.45 | f00b3r | I was planning on trying out AMP this weekend |
| 18:59.53 | *** join/#asterisk firestrm (~vince@S010600047577bccd.gv.shawcable.net) |
| 18:59.54 | f00b3r | of course my office got broken into thursday night |
| 18:59.57 | blitzrage | file[laptop]: you ??? my !!!'s?! |
| 18:59.59 | afrosheen | f00b3r: it's come a long way in 3 months |
| 19:00.01 | f00b3r | that seriously put me back a few days |
| 19:00.03 | file[laptop] | blitzrage: yesssssssssss |
| 19:00.10 | afrosheen | it supports zap, sip and iax trunking now |
| 19:00.11 | blitzrage | file[laptop]: I challenge you to a dual! |
| 19:00.12 | drumkilla | how dare you?! |
| 19:00.15 | file[laptop] | hold me hold me hug me hug me! |
| 19:00.16 | blitzrage | lol |
| 19:00.20 | afrosheen | blitzrage: a dual processor? |
| 19:00.23 | drumkilla | blitzrage: I think you mean duel ;) |
| 19:00.26 | blitzrage | afrosheen: yes.... :) |
| 19:00.31 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 19:00.34 | blitzrage | drumkilla: yes... yes I due :D |
| 19:00.44 | wankel | so i'm trying to just have the phones tlak directly to asterisk now, but asterisk's nat keepalives don't work with realtime... so now i'm digesting chan_sip.c |
| 19:00.55 | blitzrage | yes... I used the wrong word on purpose :D |
| 19:01.08 | file[laptop] | blitzrage: welcome to the club |
| 19:01.19 | blitzrage | file[laptop]: do I get fringe benefits? |
| 19:01.20 | firestrm | mmmmmmm... ipod... |
| 19:01.21 | f00b3r | iPod is like crack |
| 19:01.26 | f00b3r | I can't leave home without it |
| 19:01.27 | file[laptop] | blitzrage: you get a hug! |
| 19:01.29 | afrosheen | no, crack is cheap |
| 19:01.31 | blitzrage | file[laptop]: lol |
| 19:01.34 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.rr.com) |
| 19:01.35 | f00b3r | hahah |
| 19:01.36 | wankel | heh. i won another ipod at work thursday for sitting through a four-hour all-hands meeting. |
| 19:01.47 | blitzrage | file[laptop]: shhhh... I'm a dCAP now |
| 19:01.53 | mitcheloc | wankel: send it here if you have an extra |
| 19:01.55 | afrosheen | wankel: ebay it |
| 19:02.00 | mitcheloc | afrosheen: shhh |
| 19:02.02 | file[laptop] | blitzrage: eeek |
| 19:02.06 | wankel | which is good, i suppose, since my current ipod is permanently wired into the truck's 12V power since the battery lasts about 12 minutes. |
| 19:02.28 | *** join/#asterisk ManxPwr (~eric@adsl-35-245-42.msy.bellsouth.net) |
| 19:02.28 | mitcheloc | wankel: nahh you want to give it to a friend |
| 19:02.35 | drumkilla | wankel: like me!! |
| 19:02.46 | mitcheloc | * mitchel |
| 19:03.01 | wankel | drumkilla: would nat keepalives magically start working with realtime? :) |
| 19:03.22 | drumkilla | wankel: We might be able to work something out, haha |
| 19:03.31 | afrosheen | anyone know about linux permissions and mounting |
| 19:04.30 | freat | woah |
| 19:04.30 | drumkilla | haha |
| 19:04.30 | drumkilla | ride the waaave! |
| 19:04.30 | wankel | holy netsplit, batman |
| 19:04.31 | mitcheloc | heh teh zelazny server went down |
| 19:04.31 | afrosheen | kerrrunch |
| 19:04.31 | *** join/#asterisk Mike9 (~sturdee@sun.mikesweb.com) |
| 19:04.31 | *** part/#asterisk beto75 (~hav@201.128.177.84) |
| 19:04.31 | wankel | i haven't seen one of those since i stopped using efnet :) |
| 19:04.33 | afrosheen | this happens every saturday |
| 19:04.46 | mitcheloc | heh why? |
| 19:05.18 | afrosheen | it's beer and pizza day at the freenode complex |
| 19:05.19 | robin_sz | afrosheen: trains. |
| 19:05.19 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) [NETSPLIT VICTIM] |
| 19:05.19 | *** join/#asterisk devi0us (devi0us@west.philly.ghetto.org) [NETSPLIT VICTIM] |
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| 19:05.19 | *** join/#asterisk file (~symlink@mctn1-3365.nb.aliant.net) |
| 19:05.19 | *** join/#asterisk LOT (~Methos@S0106000f6694b86f.ed.shawcable.net) [NETSPLIT VICTIM] |
| 19:05.19 | *** join/#asterisk DrukenWrk (Druken@67.69.139.226) [NETSPLIT VICTIM] |
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| 19:05.20 | mitcheloc | heh |
| 19:05.24 | freat | we |
| 19:05.24 | freat | weeeeeeeeee |
| 19:05.43 | robin_sz | afrosheen: they have to split the net in Ohio, wher eit crosses a train track, due to a train every saturday |
| 19:05.58 | netsurfer | lol |
| 19:06.04 | afrosheen | robin_sz: lol |
| 19:06.16 | mitcheloc | are you serious? |
| 19:06.16 | afrosheen | wireless cantenna link |
| 19:06.21 | afrosheen | INCOMING WHOO WHOOOOO |
| 19:06.44 | f00b3r | lol |
| 19:06.46 | mitcheloc | thats weird...why don't they just raise the antenna a little higher |
| 19:06.58 | robin_sz | heh |
| 19:07.01 | afrosheen | fcc regulations |
| 19:07.14 | firestrm | afrosheen, faa regs |
| 19:07.16 | afrosheen | michael powell himself complained about it |
| 19:07.30 | afrosheen | 'this pole is too high, we fine you 2 million dollars' |
| 19:07.32 | *** join/#asterisk fishboy1669 (fishboy166@cpc1-warr1-3-0-cust38.bagu.cable.ntl.com) |
| 19:07.39 | mitcheloc | hide it in a tree |
| 19:07.52 | afrosheen | where birds will nest in it? naw |
| 19:07.52 | robin_sz | they tried that |
| 19:07.52 | f00b3r | the little blinkey light on top is expensive |
| 19:07.54 | mitcheloc | trains aren't that tall |
| 19:07.55 | firestrm | afrosheen, cant have aircraft flying into the network.. imagine the netspit that would cause ::) |
| 19:08.01 | robin_sz | it upset the local enviromentalist groups |
| 19:08.12 | afrosheen | nestenna |
| 19:08.16 | robin_sz | worried about RF radiation and nesting birds |
| 19:08.27 | afrosheen | it's cool, you get free fried eggs |
| 19:08.34 | firestrm | lol |
| 19:08.51 | firestrm | hard boiled i think |
| 19:08.55 | afrosheen | See here, it's perfectly harmless...pass the salt |
| 19:09.15 | f00b3r | PETA dosen't like fried eggs |
| 19:09.16 | robin_sz | weirdest thing I ever saw was in finland ... bridges for cars on the ice that they have to pull out of the way when a ferry comes past |
| 19:09.19 | f00b3r | and they look at you funny when you offer it to them |
| 19:09.28 | mitcheloc | any tall enough buildings in the area? damn, that really sucks...they should have two then, one a mile down the tracks so they can switch between them when the train comes |
| 19:09.58 | robin_sz | mitcheloc: tall buildings? in ohio? |
| 19:10.01 | afrosheen | robin_sz: an ice bridge? |
| 19:10.07 | robin_sz | mitcheloc: you are lucky they have lectric! |
| 19:10.11 | firestrm | robin_sz, there are many wierd things in finland.. |
| 19:10.36 | robin_sz | afrosheen: yeah .. on the frozen sea in the Aaland islands .. |
| 19:11.05 | robin_sz | the mark roads out and put aluminium or wooden bridges over the broken ice where the ferries come through |
| 19:11.17 | robin_sz | they have to move them when a ferry wants to come past |
| 19:11.54 | firestrm | robin_sz, it may be cold, but finnish women are hot :) |
| 19:11.59 | wankel | NYC still has a few rotary bridges |
| 19:12.05 | robin_sz | now, that bit is true |
| 19:12.13 | wankel | the whole bridge turns parallel to the river around a pivot when something is passing |
| 19:12.38 | robin_sz | firestrm: finnish women are scary. yes they are hot, boy .. do NOT argue with them |
| 19:12.49 | blitzrage | robin_sz: lol |
| 19:12.56 | file[laptop] | blitzrage: who got them, WHO |
| 19:12.59 | firestrm | robin_sz, lol i know.. |
| 19:13.04 | blitzrage | file[laptop]: Steve and Olle |
| 19:13.11 | file[laptop] | blitzrage: ah |
| 19:13.11 | blitzrage | file[laptop]: they are very cool |
| 19:13.16 | file[laptop] | blitzrage: where are you? |
| 19:13.22 | blitzrage | file[laptop]: Kansas City, MO |
| 19:13.31 | firestrm | robin_sz, im married to a danish girl.. same things.. dont mess with vikings... |
| 19:13.39 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-39-201.no.no.cox.net) |
| 19:13.41 | blitzrage | file[laptop]: Steve just installed the development tools and is attempting to get Asterisk installed on it |
| 19:13.42 | robin_sz | firestrm: eek :) quite. |
| 19:13.48 | blitzrage | firestrm: thats true :D |
| 19:14.00 | file[laptop] | blitzrage: ah |
| 19:14.18 | sung | hi |
| 19:14.25 | robin_sz | firestrm: do the danish do the 'molten tin' ceremony at new year? |
| 19:14.27 | file[laptop] | it only takes a minute girl, to fall in love! to fall in love! |
| 19:14.49 | blitzrage | file[laptop]: a minute girl? hrmmmm... |
| 19:15.00 | blitzrage | file[laptop]: never met one of those, lol |
| 19:15.09 | firestrm | robin_sz, is been reffered to but ive never seen it.. but i have been forced to eat lootfisk |
| 19:15.18 | robin_sz | firestrm: and survived? |
| 19:15.33 | robin_sz | just |
| 19:16.12 | *** join/#asterisk RDF (~leonardo@S010600055d210201.vs.shawcable.net) |
| 19:16.13 | firestrm | robin_sz, for the most part... the best part is the schnaps afterward.. and then lighing the alchol left over in the bottle to let the ghost out |
| 19:16.25 | robin_sz | heh |
| 19:16.44 | robin_sz | firestrm: how unusual, a nordic custom that involves alcohol ;)) |
| 19:16.51 | blitzrage | robin_sz: lol |
| 19:17.18 | robin_sz | finland is famous for winter sports |
| 19:17.37 | blitzrage | hrmmmm... is it too early to start drinking? |
| 19:17.41 | RDF | Anyone seen a case where a .msg or .wav can be recorded played and included in a Background dial plan but does not play and get a return error of Unable to open greeting.wav (format unknown): No such file or directory |
| 19:17.42 | robin_sz | the top 3 are drinking, wifeswapping and suicide ... usually in that order |
| 19:17.51 | RDF | ? |
| 19:18.14 | file[laptop] | RDF: don't put the extension, just use greeting |
| 19:18.17 | RDF | It is there with the rest of the gsm files in /var/lib/asterisk/sounds |
| 19:19.28 | blitzrage | RDF: yah, what file said |
| 19:19.52 | RDF | I dont knowwhat you mean it is a s start line in s,4 |
| 19:20.06 | file[laptop] | you're using Background(greeting.wav) |
| 19:20.10 | file[laptop] | just use Background(greeting) |
| 19:20.10 | blitzrage | RDF: s/Background(greeting.wav)/Background(greeting) |
| 19:20.14 | RDF | exten => s,4,Background(greeting) |
| 19:20.15 | wankel | he means leave ".wav" off of it |
| 19:20.18 | blitzrage | hrmmm |
| 19:20.19 | RDF | Thats what i have |
| 19:20.31 | blitzrage | is it in /var/lib/asterisk/sounds? |
| 19:20.37 | freat[laptop] | if I got asterisk stable from CVS, I'm getting the latest thing right? 1.0.4 ? and cvs update keeps it the latest and greatest stable right? |
| 19:20.37 | RDF | yes |
| 19:20.40 | blitzrage | oh yah, you said that already |
| 19:20.40 | RDF | i can play it |
| 19:21.09 | blitzrage | freat[laptop]: you're getting the latest changes to the stable branch - post 1.0.4 |
| 19:21.14 | RDF | I wonder if some other service is interfearing with it playing when asterisk is running |
| 19:21.30 | freat[laptop] | blitzrage: thx |
| 19:21.34 | RDF | I can though punch in the extensions and thay work |
| 19:21.39 | *** part/#asterisk mitcheloc (~mitcheloc@ca-fullerton-69-166-193-228.vnnyca.adelphia.net) |
| 19:22.44 | blitzrage | wiki isdown |
| 19:22.53 | blitzrage | ~wiki |
| 19:22.59 | blitzrage | ~asterisk wiki |
| 19:23.00 | jbot | asterisk wiki is, like, http://www.voip-info.org/wiki-Asterisk |
| 19:23.11 | blitzrage | hrmmm... doesn't jbot report back a status somehow? |
| 19:23.15 | RDF | in fact, let me try one of the existing built in gsm files and see what happens. |
| 19:23.24 | *** join/#asterisk _RaYmAn_ (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 19:23.32 | firestrm | robin_sz, my best dont mess with danes story was from before i married my wife, i had to move a 30 cubic foot freezer.. i was waiting for a bodybuilder friend to help me move it onto the truck. My wife to bee said, lets just move it ouselves.. i snickered and said sure.. what the heck. imageine my suprise when she succeeded in picking her end up. |
| 19:23.47 | blitzrage | ahhh |
| 19:23.48 | blitzrage | ~wiki irc |
| 19:23.49 | freat[laptop] | silly question... but I've seen the wiki go down before. not a big deal to me, but, is it backed up? |
| 19:23.59 | robin_sz | right .. so having found a Makefile that can build and install zaptel on debian (unlike the currrent one) ... hermie maybe? |
| 19:24.06 | robin_sz | firestrm: hehe |
| 19:24.07 | *** join/#asterisk Frantic (~ab@68-175-94-179.nyc.rr.com) |
| 19:24.12 | firestrm | robin_sz, i only realized later that she had picked up the apprentice end.. Danish women are surprisingly strong for their size.. |
| 19:24.15 | blitzrage | wiki returns! :) |
| 19:24.43 | firestrm | robin_sz, i know imediatly that i was in serious trouble if i EVER messed with her.. |
| 19:24.59 | freat[laptop] | blitzrage: you run the wiki? |
| 19:25.05 | blitzrage | lol, no :) |
| 19:25.15 | blitzrage | asteriskdocs.org |
| 19:25.16 | file[laptop] | blitzrage runs around in circles |
| 19:25.18 | robin_sz | firestrm: yeah. my mate is married to a finn ... I know exactly what you mean |
| 19:25.23 | freat[laptop] | haha |
| 19:25.41 | blitzrage | its true(tm) |
| 19:25.46 | freat[laptop] | anyone know if the wiki is backed up or at least on RAID-5? it would be a shame to lose all that info... |
| 19:25.46 | Frantic | anyone knows about this: ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
| 19:26.03 | firestrm | robin_sz, when they are mad enough for violence, they dont just slap you.. they mess you up.. |
| 19:26.09 | file[laptop] | eek |
| 19:26.12 | blitzrage | Frantic: you don't have zaptel.conf or zapata.conf configured right |
| 19:26.15 | file[laptop] | wifi going, insane |
| 19:26.34 | robin_sz | firestrm: exactly. the phrase "too hot to handle" comes to mind |
| 19:26.38 | firestrm | file, ask wifi if i can go too |
| 19:26.50 | blitzrage | file[laptop]: I hate pineapples - so I suppose that is a valid weapon |
| 19:26.59 | Frantic | blitzrage: I have- digium was able to fix it once- if I take the card out and put it back it works ok- they say something about VendorID change they did |
| 19:27.13 | RDF | play filename.gsm right from / it should play? the path should be set and is required? |
| 19:27.33 | blitzrage | Frantic: oh I know how to fix that. Open zconfig.h and near the bottom you can tell the card to use all the VendorIDs |
| 19:27.40 | blitzrage | Frantic: I think that'll fix it anyways |
| 19:27.40 | Frantic | blitzrage: I just re-did zaptel with 1.0.4 |
| 19:27.53 | firestrm | robin_sz, Im the only one who has the keys to unlock the garage deadbolt... for a good reason, its my bunker for when my wife is mad |
| 19:28.02 | file[laptop] | blitzrage: VON? |
| 19:28.05 | blitzrage | Frantic: open zconfig.h and find that section, you'll have to comment it |
| 19:28.06 | Frantic | blitzrage: wow- let me check |
| 19:28.12 | file[laptop] | I can't remember who is and isn't going now |
| 19:28.13 | blitzrage | uncommen it * |
| 19:28.29 | freat[laptop] | break it |
| 19:28.50 | firestrm | robin_sz, not that that happens much, but just like Nukes, it only takes one event to mess you up :) |
| 19:28.59 | blitzrage | file[laptop]: ummmm odd... where did I mention VON? |
| 19:29.00 | freat[laptop] | breaking stuff is cool |
| 19:29.16 | file[laptop] | blitzrage: nowhere, it was a question as towards whether you will be there or not |
| 19:29.28 | blitzrage | file[laptop]: you know what is REALLY freaky? I was just about to say that I think I might be going. |
| 19:29.34 | blitzrage | file[laptop]: that scares me |
| 19:29.38 | file[laptop] | blitzrage: I'm psychic |
| 19:29.49 | robin_sz | firestrm: they are fun creatures in the summer, but in the winter .. something changes .. they go .. dark. best avoid upsetting them |
| 19:29.59 | file[laptop] | blitzrage: I've gone to hell two times at the exact same time as Brian, it was freaky |
| 19:30.05 | blitzrage | file[laptop]: actually, I have more of a feeling that I can communicate my thoughts to other people and not that you're psychic. Its my skill, not yours |
| 19:30.14 | file[laptop] | blitzrage: nope - all mine |
| 19:30.25 | blitzrage | file[laptop]: perhaps a combo skill then |
| 19:30.42 | freat[laptop] | abababupdownselectstart |
| 19:30.44 | file[laptop] | blitzrage: perhaps |
| 19:30.44 | firestrm | robin_sz, lol, you have true understanding of the creature in question.. kinda like the shadow creatures in bab5 :) |
| 19:30.45 | blitzrage | file[laptop]: I was talking to Olle last night and he might try to figure out a way to get me down to VON |
| 19:30.59 | file[laptop] | blitzrage: excellent |
| 19:31.09 | blitzrage | Frantic: let me know if that thing works for you |
| 19:31.18 | robin_sz | firestrm: yeah, but at least that had the decency to look mean. these one just look blonde and babe-like :) |
| 19:31.23 | blitzrage | file[laptop]: sounds like it's going to be a really good time :) |
| 19:31.28 | file[laptop] | blitzrage: it is. |
| 19:31.28 | blitzrage | file[laptop]: all sorts of people there |
| 19:31.37 | file[laptop] | yup |
| 19:31.46 | freat[laptop] | where are you guys going? |
| 19:32.05 | blitzrage | hopefully San Jose |
| 19:32.07 | Frantic | <blitzrage> I make clean and building- now restarting |
| 19:32.18 | Frantic | <blitzrage> I'll let you know in a minute |
| 19:32.30 | file[laptop] | blitzrage: paulc is going too :) |
| 19:32.33 | firestrm | blitzrage, which von? toronto? if so i might be trying to fill an aircraft for the trip across canada.. the VON bus !! |
| 19:32.40 | freat[laptop] | anyone here in Chicago? |
| 19:32.53 | file[laptop] | firestrm: Spring VON in San Jose, CA |
| 19:33.09 | file[laptop] | in the month known as March |
| 19:33.15 | *** join/#asterisk dontmsgme (~none@m810f36d0.tmodns.net) |
| 19:33.18 | file[laptop] | between the days known as the 7th and the 10th |
| 19:33.38 | firestrm | file(laptop) even better.. i'll fly the SJ Von bus :) |
| 19:33.56 | file[laptop] | my flight is just peachy |
| 19:33.59 | blitzrage | firestrm: personally, for VON Canada, I wouldn't bother. I went this year and it sucked |
| 19:34.18 | blitzrage | file[laptop]: sweet! I want to meet paulc |
| 19:34.29 | firestrm | blitzrage, good to know.. it will be my first VON, i dint want to get a bad impression |
| 19:34.30 | dontmsgme | Is there a website for VON in san jose |
| 19:36.03 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 19:36.03 | *** topic/#asterisk is Asterisk: The Open Source PBX || DUNDi: http://www.dundi.com #dundi || 1.0.4 Released |
| 19:36.04 | firestrm | cool |
| 19:36.15 | blitzrage | Frantic: can you email me the info of the errors you were getting before? |
| 19:36.16 | ardor | ~jbot whats up |
| 19:36.17 | jbot | nothing much mate, you ? |
| 19:36.26 | Frantic | <blitzrage> sure |
| 19:36.35 | Frantic | <blitzrage> what's your email |
| 19:36.48 | blitzrage | Frantic: leif (at) leifmadsen (dot) com |
| 19:36.50 | firestrm | i would stop in seattle for customs, so i could pick up seattle ppl. |
| 19:37.08 | file[laptop] | I get to fly... Moncton -> Toronto -> Denver -> San Jose |
| 19:37.13 | RDF | blits, something is wrong with my files being played in Background(filename) I can replace my working sound file in place of the built in ones included in /asterisk/sounds and it will play. I create my own sound file names and its dead silence. Ever seen a case of this? |
| 19:37.15 | blitzrage | anyone know about faking a zaptel timer on OSX? |
| 19:37.30 | RDF | blitzrage so this is a little strange. |
| 19:37.34 | file[laptop] | blitzrage: good luck? |
| 19:37.36 | *** join/#asterisk robf_ (robf@user-24-236-86-244.knology.net) |
| 19:37.40 | blitzrage | RDF: is it mono, 8bit? |
| 19:37.48 | RDF | how do I know? |
| 19:38.06 | RDF | by default rec is recording in what mode? |
| 19:38.11 | blitzrage | RDF: good question - I'm going to assume its not then. Use sox to convert the file to the appropriate format. Check the wiki |
| 19:38.32 | blitzrage | RDF: if you're using Astersk Record() you shouldn't hvae to do anything though |
| 19:38.36 | RDF | It should be mono,8bit ? |
| 19:38.38 | freat[laptop] | blitzrage: also, don't mp3s have to not have id3 tags I think...? |
| 19:38.45 | blitzrage | freat[laptop]: yep, thats also true |
| 19:38.48 | RDF | its used sox rec |
| 19:39.03 | blitzrage | RDF: sorry, not positive. Check the man pages |
| 19:39.10 | RDF | k |
| 19:39.25 | freat[laptop] | RDF: are you just recording stuff for menus? if so, there's some directions on the wiki about creating an extension to dial to do that |
| 19:39.32 | blitzrage | sheesh, with the number of questions I"ve been answering lately I should have op status, lol |
| 19:39.54 | RDF | still, you are saying it should recorded as mono, 8bit? |
| 19:40.01 | blitzrage | RDF: pretty sure |
| 19:40.03 | file[laptop] | blitzrage: you don't know the magic? all you have to do is ask. |
| 19:40.04 | freat[laptop] | wiki wiki |
| 19:40.05 | RDF | k |
| 19:40.08 | RDF | good enough |
| 19:40.09 | RDF | :) |
| 19:40.18 | firestrm | i have been issued a challenge by my wife.. she has many deaf friends(she does sign language interpretation) and she want me to make my asterisk TTY aware.. |
| 19:40.25 | firestrm | any ideas? |
| 19:40.26 | blitzrage | file[laptop]: just kidding around. Having op status just means people /msg you more :) |
| 19:40.37 | file[laptop] | people msg me regardless |
| 19:40.37 | blitzrage | file[laptop]: although I'd like to be able to change the topic... thats about the only perk I see :D |
| 19:40.40 | file[laptop] | crazy buggers |
| 19:40.42 | blitzrage | file[laptop]: yep, me too |
| 19:40.55 | freat[laptop] | firestrm: that's a cool idea |
| 19:40.55 | blitzrage | file[laptop]: I need to implement one of those message queueing scripts or something |
| 19:41.11 | *** join/#asterisk uunot (~teliax@c-67-166-37-218.client.comcast.net) |
| 19:41.14 | firestrm | freat(laptop) i thought so too.. |
| 19:41.35 | file[laptop] | ahhhhhh I have Monday off |
| 19:41.40 | file[laptop] | and Wednesday, and Thursday |
| 19:41.42 | file[laptop] | and then Monday and Tuesday |
| 19:41.57 | freat[laptop] | firestrm: if you use a lossless codec, will TTY get passed from one phone to the other via * ? |
| 19:42.46 | firestrm | freat(laptop), it should, its more of a problem detecting that its TTY, vs fax voice or modem |
| 19:43.19 | fishboy1669 | carrot |
| 19:43.32 | freat[laptop] | firestrm: would be nice if it could recognize the TTY so that you could code extensions for TTY |
| 19:43.32 | blitzrage | file[laptop]: I'm more like a carrot - good for the eyes, lol |
| 19:43.32 | firestrm | arugula |
| 19:43.48 | *** join/#asterisk hacim (micah@micha.hampshire.edu) |
| 19:43.58 | file[laptop] | blitzrage: blasphemy |
| 19:44.13 | hacim | at the moment, what is the cheapest card I can get to hook my PTSN line up to my computer so I can participate in bellster? ;) |
| 19:44.20 | firestrm | freat(laptop) i agree.. it has usefulness for business too, no more seperate line for TTY |
| 19:45.00 | blitzrage | hey, if anyone needs Asterisk consulting, check my website :D |
| 19:45.10 | freat[laptop] | firestrm: integrate * TTY recognition with instant messaging... |
| 19:45.29 | f00b3r | blitzrage: seriously, what is your website? |
| 19:45.30 | firestrm | freat(laptop), Carol (my wife) thinks that she might even be able to get funding do develop such a thing |
| 19:45.33 | file[laptop] | I've gotta stay... can't run away... |
| 19:45.36 | hacim | anyone know where I can get cheap Analog Line/FXO cards? |
| 19:45.40 | f00b3r | keeping some consultans queued up if I fail misserably |
| 19:46.00 | firestrm | freat(laptop), awesome idea.. |
| 19:46.08 | RDF | BTW, can any other modem other then a x100p be used? |
| 19:46.09 | blitzrage | f00b3r: www.leifmadsen.com - still working on the site as I'm in transition so its a bit of a mess, but my contact info is there |
| 19:46.17 | freat[laptop] | firestrm: yeah for a business, you could have * ring agents the same way it normally does, except that if it's TTY, they communicate with an IM window that pops up |
| 19:46.30 | f00b3r | cool thanks |
| 19:46.42 | blitzrage | f00b3r: thank, you! |
| 19:47.00 | LUTOR_ASI | what codec a PSTN line uses? |
| 19:47.00 | firestrm | freat(laptop), could even do mailbox for TTY that way |
| 19:47.07 | pimpwell | whats the best way to check if asterisk installed correctly? |
| 19:47.14 | blitzrage | file[laptop]: you know whats funny, I guess I'm dCAP #1 (#3 if you count Olle and Steven) |
| 19:47.25 | f00b3r | connect with a sip phone and dial 1000 |
| 19:47.27 | file[laptop] | blitzrage: and what does it get you? |
| 19:47.28 | f00b3r | if the demo is enabled |
| 19:47.33 | *** join/#asterisk mesh (meshuga@c-24-21-94-74.client.comcast.net) |
| 19:47.38 | freat[laptop] | firestrm: I'm starting to work on a project to integrate * and instant messaging for the university... would be good to keep in touch as we're gonna be looking for ideas for funding some projects as well |
| 19:47.38 | blitzrage | file[laptop]: ummmm.... cheap entry to Astricon? :D |
| 19:47.38 | file[laptop] | a potato! |
| 19:47.51 | pimpwell | any software way of checking? |
| 19:47.53 | file[laptop] | no no no, a potato! |
| 19:48.02 | blitzrage | file[laptop]: I could go for a baked potato with diced tomato's, chives and sour cream right now |
| 19:48.17 | file[laptop] | blitzrage: then go get one you lazy bugger |
| 19:48.20 | *** join/#asterisk waddy (waddy@66.90.92.190) |
| 19:48.20 | f00b3r | pimpwell: sure, use a softphone |
| 19:48.24 | freat[laptop] | firestrm: shoot me an email so i've got yours. rsenykoff (at) harrislogic (dot) com |
| 19:48.57 | firestrm | freat[laptop], will do.. |
| 19:49.11 | blitzrage | file[laptop]: I'm not in my own house |
| 19:49.30 | czero | Mmmmmmmmmmmm food |
| 19:49.37 | czero | thikn I need to venture outside |
| 19:49.44 | file[laptop] | blitzrage: that's why you run out the door screaming, "POTATO!", and head to the nearest venue of consuming food |
| 19:49.45 | czero | blitzrage is the storm bad over your way |
| 19:50.06 | freat[laptop] | We've had quite a bit of snow here in Chicago so far |
| 19:50.14 | freat[laptop] | can't get the car out of the alley |
| 19:50.15 | blitzrage | czero: not here in KC, MO, but I'm afraid of my flight getting cancelled tonight |
| 19:50.31 | czero | Mmm yeah I'd saw no fly home tonight :) |
| 19:50.38 | blitzrage | czero: crap |
| 19:50.42 | czero | we've got aboput 3inches down and -30 |
| 19:50.57 | blitzrage | czero: eeeeesh! |
| 19:50.59 | Schism | hey, if I am setting up a trunk to broadvoice, do I really need their patch? |
| 19:51.00 | blitzrage | -30C right? |
| 19:51.05 | czero | my wife went to NC yesterday and they where saying on the AC webside ALOT of flight would be canceled today |
| 19:51.07 | file[laptop] | so come on baby do it to me do it to me good now do it to me slowly |
| 19:51.11 | czero | blitzrage: yeah |
| 19:51.20 | freat[laptop] | -30 C or F? |
| 19:51.26 | blitzrage | either way thats damn cold |
| 19:51.26 | czero | C |
| 19:51.29 | *** join/#asterisk h4mm3r` (~h4mm3r@81-208-60-202.fastres.net) |
| 19:51.30 | blitzrage | file[laptop]: LOL |
| 19:51.32 | freat[laptop] | ahh ok |
| 19:51.37 | firestrm | freat(laptop), message sent |
| 19:51.40 | czero | yeah after -20(c/f) nothing matters :) |
| 19:51.41 | freat[laptop] | cool |
| 19:51.43 | wankel | no snow here yet |
| 19:51.48 | wankel | ~metar kpvd |
| 19:51.49 | czero | wankel where r u |
| 19:52.13 | *** join/#asterisk Umaro (~umaro@c-24-22-76-14.client.comcast.net) |
| 19:52.16 | wankel | -9C here |
| 19:52.27 | file[laptop] | -20C here |
| 19:52.35 | czero | never as cold near water |
| 19:52.40 | Schism | this is great about the weather, but isn't this the * channel? |
| 19:52.41 | firestrm | ~metar cyxx |
| 19:52.42 | blitzrage | ~metar yyz |
| 19:52.49 | *** join/#asterisk dant (~dan@81-86-69-213.dsl.pipex.com) |
| 19:52.52 | wankel | yeah, the water keeps us a bit warm |
| 19:52.55 | czero | wankel they have a dealer ship in RI or you have to go up to PEabody? |
| 19:53.14 | wankel | the dealership here is evil. i got to citysidegarage.com |
| 19:53.26 | firestrm | wankel how you get jbot to give metar? |
| 19:53.31 | *** join/#asterisk DrPete (~Pete@82-40-26-136.cable.ubr04.uddi.blueyonder.co.uk) |
| 19:53.35 | file[laptop] | all along the shore, we sit together in the calm of a summer breeze |
| 19:53.38 | wankel | ~metar cyyz |
| 19:53.40 | czero | I _had_ a range rover got it in Peabody when I lived in NH |
| 19:53.43 | file[laptop] | I move a little closer, and I slip my arms from you arms |
| 19:53.44 | firestrm | nevermind jbot was just slow.. |
| 19:53.45 | wankel | you have to give the full four-letter ICAO code |
| 19:54.00 | czero | but have a 99 disco now |
| 19:54.02 | file[laptop] | and hold me tight in your arms... tell me that you love me too knowing that you care |
| 19:54.03 | wankel | yeah, i dunno why it's so slow |
| 19:54.05 | file[laptop] | makes me feel alive |
| 19:54.20 | wankel | jbot's mostly a functioning infobot, but whoever hacked it broke karma |
| 19:54.24 | firestrm | ~metar CYYJ |
| 19:54.49 | czero | ~metar CYHZ |
| 19:54.59 | wankel | yow. i'm surrounded by canadians. |
| 19:55.05 | file[laptop] | yes, yes you are |
| 19:55.13 | czero | blitzrage: http://aircanada.com/en/news/travel_advisory8.html |
| 19:55.21 | blitzrage | ~metar CYYZ |
| 19:55.26 | firestrm | its an invasion.. all your asterisk belong to canuck |
| 19:55.27 | blitzrage | czero: thanks, checking |
| 19:55.31 | wankel | i just did toronto! pay attention! :) |
| 19:55.39 | netsurfer | ~metar EGAA |
| 19:55.40 | blitzrage | wankel: oops! :) |
| 19:55.47 | blitzrage | too many windows, lol |
| 19:55.58 | *** join/#asterisk mrempire (~user1@h71032.upc-h.chello.nl) |
| 19:55.59 | Schism | great |
| 19:56.02 | wankel | metar's not that hard once you get the hang of the unspecified bits at the end |
| 19:56.03 | Schism | as long as someone can |
| 19:56.08 | Schism | hey, if I am setting up a trunk to broadvoice, do I really need their patch? |
| 19:56.11 | *** join/#asterisk securAX (~lmonstat@adsl-67-119-205-233.dsl.pltn13.pacbell.net) |
| 19:56.14 | file[laptop] | MY GENERATION! |
| 19:56.28 | firestrm | wankel, what is karma? |
| 19:56.44 | czero | karma is what kicks my ass on a daily basis :) |
| 19:56.56 | czero | ~metar EBUD |
| 19:57.07 | wankel | with a normal infobot, you say things like foo++ and foo-- and it accumulates karma points for foo |
| 19:57.20 | wankel | but this one's not working except in private messages for some reason. it seems to be the ~ hack |
| 19:57.24 | file[laptop] | blitzrage: never ending story! POTATO |
| 19:57.43 | file[laptop] | turn around... look at what you see... in your face the mirror of your dreams... |
| 19:57.44 | firestrm | wankel, i see |
| 19:57.46 | wankel | it'll look up codes by itself, actually |
| 19:57.53 | wankel | ~airport code for south bend |
| 19:57.55 | blitzrage | wankel: yes... karma would be VERY nice in here |
| 19:58.16 | file[laptop] | ooh it's almost 4PM, I wonder what's on |
| 19:58.18 | wankel | wow. people must be pounding the noaa web sites |
| 19:58.42 | blitzrage | wankel: I started on compiling for infobot - but stuck at the point when I need my Google API code - their servers were down, so now its just sitting at the screen waiting for me :) |
| 19:58.44 | wankel | hmm... no, other infobots are still fast. |
| 19:59.07 | wankel | blitzrage: you can just leave the google plugin out |
| 19:59.13 | *** join/#asterisk Spooch (~rath@p549A1740.dip0.t-ipconnect.de) |
| 19:59.19 | blitzrage | wankel: yah, but I want it :) |
| 19:59.24 | wankel | 'tis handy |
| 19:59.25 | RDF | wankel, why |
| 19:59.25 | blitzrage | wankel: so will just wait for google |
| 19:59.34 | blitzrage | I'm in no rush for it |
| 19:59.36 | wankel | rdf: why what? |
| 19:59.37 | mrempire | can anyone tell me what hardware i need to make voip calls at both ends? |
| 19:59.44 | blitzrage | mrempire: none |
| 19:59.53 | mrempire | Huh? |
| 19:59.54 | blitzrage | other than a computer |
| 19:59.54 | RDF | wankel, why do you think noaa site is being pounded? |
| 20:00.13 | blitzrage | mrempire: use Asterisk and softphones, no HW required (ok... speakers and mic too) |
| 20:00.14 | czero | blitzrage: on the GTAA site most flights depayed not canceled so u have a chance |
| 20:00.16 | mrempire | where do i connet my phone then |
| 20:00.29 | *** join/#asterisk drgalaxy (~brian@adsl-69-149-120-216.dsl.lbcktx.swbell.net) |
| 20:00.33 | wankel | rdf: well, jbot's weather module is slow as balls, and it is snowing a lot. i deduced (incorrectly) that noaa's web sites were running slow. it must just be jbot, though, because other bots are pulling the noaa data way faster. |
| 20:00.33 | blitzrage | czero: coolness. I have a site here I'm going to load up here with status of my flight |
| 20:00.35 | f00b3r | mrempire: use a softphone |
| 20:00.52 | RDF | mr, how new are you to this? |
| 20:01.10 | mrempire | My mother cannot use the softphone |
| 20:01.15 | mrempire | ;( |
| 20:01.20 | RDF | wankel, there is no nation wide events other then the snow storm to my knowledge that would be bogging down noaa web site. |
| 20:01.40 | wankel | rdf: me either. i assumed it was the snow storm. like i said, though, it seems to just be jbot. |
| 20:01.50 | mrempire | Can i have a normal phoe setup at one side |
| 20:02.04 | f00b3r | lol |
| 20:02.06 | f00b3r | sweetg |
| 20:02.07 | *** join/#asterisk thefallen (PolarBear@thefallen.user) |
| 20:02.09 | czero | bluetooth worked pretty good |
| 20:02.13 | wankel | mrempire: if you pay for PSTN access from an ITSP |
| 20:02.14 | f00b3r | s/sweetg/sweet |
| 20:02.14 | drgalaxy | I am looking for a way to wire my cell phone's headset jack to an FXS. Can anyone provide any pointers? |
| 20:02.37 | wankel | galaxy: uh... google for "hybrid interface" |
| 20:02.38 | RDF | wankel, personally you ask people on the street what noaa is and thay say...dont know. I really dont think its a function of the snow storm. most people just turn on the tv to see what the local weather report is. |
| 20:02.42 | PatrickDK | drgalaxy, you don't |
| 20:02.47 | wankel | galaxy: but really, just go buy a $10 phone at radio shack |
| 20:02.48 | PatrickDK | but you can get an adaptor |
| 20:03.11 | Umaro | drgalaxy: do NOT buy a $10 phone at radio shack |
| 20:03.13 | wankel | rdf: yeah, no one cares much about science in government anymore. |
| 20:03.14 | Umaro | buy the $20 phone |
| 20:03.16 | RDF | dr you need to know what FXS stands for before asking a question like that :) |
| 20:03.16 | drgalaxy | you are missing the point... want to use one cell phone to call the other and access my * box |
| 20:03.22 | Nivex | heck for $20 you can get a cordless from Wal-Mart with a 2.5" headset port on it |
| 20:03.25 | mrempire | can I connect the phone to asterisk wankel |
| 20:03.33 | drgalaxy | have free mobile to mobile calling.. can get free voip calling if I can call a 2nd phone |
| 20:03.37 | PoWeRKiLL | evening :) |
| 20:04.01 | wankel | empire: if you want to call your mother, you can use a softphone on your end and call out through asterisk, through an ITSP like nufone, to your mother's normal phone. |
| 20:04.08 | *** join/#asterisk Moc_ (~mochouina@modemcable212.49-80-70.mc.videotron.ca) |
| 20:04.10 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 20:04.21 | Moc_ | Im back... damn power faillure hehe |
| 20:04.30 | wankel | empire: or, you can get a hardware IP phone like a sipura and connect it to asterisk with your ethernet switch, and then do everything else the same |
| 20:04.52 | drgalaxy | free mobile to mobile + asterisk + voip account == FREE ANYTIME CELL |
| 20:04.52 | mrempire | the rates at nufone are more expesive than my telco |
| 20:04.54 | wankel | OR you could get a sipura for both yourself AND your mother and have them both talk to your asterisk box, though you may have NAT problems |
| 20:04.59 | RDF | drgalaxy you cannot interface cell to cell with asterisk. thats what cell towers are for :) |
| 20:05.34 | drgalaxy | RDF: I can call one cell with the other and hand off a data call. |
| 20:05.46 | wankel | rdf: nono. he wants to set his cell up to auto-answer and plug the headset into asterisk. |
| 20:05.56 | RDF | dr, there is no such thing as free in the cell bissiness. All cell frequencies are regulated by the FCC and are OWNED by varios commerical providers though a bid process. |
| 20:05.58 | file[laptop] | argh? |
| 20:05.59 | drgalaxy | RDF: you can buy GSM to PBX interfaces.. but they are like $500 |
| 20:06.17 | Leland | well.. finally got the audio to work right.. now just need to work out why the heck dtmf isn't working |
| 20:06.20 | drgalaxy | RDF: I have a cingular plan for my family.. we all have free mobile to mobile calling as long as the other side is cingular |
| 20:06.23 | RDF | But still cell frequencies are regulated |
| 20:06.24 | markit | blitzrage: are you aware of the existance of mini-itx boards? www.mini-itx.com |
| 20:06.24 | file[laptop] | A-Tuin[Play]: will you shut that off? |
| 20:06.41 | blitzrage | markit: yeppers |
| 20:06.42 | file[laptop] | A-Tuin[Play]: page |
| 20:06.47 | Nivex | well, you can interface a cell phone to an FXS port on an ATA with a cellsocket |
| 20:06.54 | czero | ok off to McD's biab good luck with your flight blitzrage |
| 20:06.56 | RDF | dr, sure...but you cannot create your own...and again its not free. you are still paying a monthly fee. |
| 20:06.57 | Nivex | but those are well over $100 |
| 20:07.07 | Leland | file[laptop]: for information, the way I got the audio stream to workw as putting "nat=yes" in the sip.conf for both peers, even though they're not really nat'ing. |
| 20:07.07 | blitzrage | markit: but the Mac Mini was just released today, so it has that coolness factor |
| 20:07.11 | blitzrage | czero: thanks, lates |
| 20:07.11 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 20:07.15 | drgalaxy | RDF: $10/month more per extra phone on the plan |
| 20:07.21 | PatrickDK | drgalaxy, http://www.cellsocket.com/ |
| 20:07.26 | drgalaxy | Nivex: thank you for a positive answer! so much negativity |
| 20:07.31 | file[laptop] | Leland: hrm? ...okay |
| 20:07.36 | BoRiS | file!!!!!!!! |
| 20:07.36 | *** join/#asterisk gopinsurg (cashmoney@dialup-4.224.186.20.Dial1.Cincinnati1.Level3.net) |
| 20:07.37 | Leland | weird.. but still |
| 20:07.38 | RDF | drgalaxy, that plan is cheaper because of marketing. |
| 20:07.43 | wankel | leland: if you had to put nat=yes in sip.conf, then they're either being nat'd or they're misconfigured to use the wrong IP address in their headers. |
| 20:07.52 | mrempire | Wankel , i will ahve a look for sipura |
| 20:07.57 | file[laptop] | BoRiS!!!!!!!!!!!!!!!!!! |
| 20:08.01 | veto_ | hmm, why am I getting a 407 proxy authenticaion required from * to one of my phones on 5060(sip)? |
| 20:08.02 | BoRiS | Wasssssssssup? |
| 20:08.03 | mrempire | thanks |
| 20:08.06 | drgalaxy | RDF: yes, but you can add "free calls to other cingular customers" for cheap ($10/mon) |
| 20:08.07 | file[laptop] | BoRiS: nada, u? |
| 20:08.27 | Leland | wankel: all devices are on external network with public IPs and there are no natting devices in between them |
| 20:08.31 | file[laptop] | veto_: asterisk is saying, "hey! give me authentication details!" |
| 20:08.36 | BoRiS | Just waking up, waiting for my lunch to be delivered :) |
| 20:08.42 | wankel | leland: something's misconfigured, then. nat=yes means "use the ip address you got this packet from, not the address in the packet" |
| 20:08.49 | file[laptop] | BoRiS: mmm food |
| 20:08.56 | DrPete | anyone use debain with asterisk? |
| 20:09.00 | wankel | if you run ethereal on the * box you could track down who's putting the wrong address where |
| 20:09.06 | file[laptop] | DrPete: yes it works fine |
| 20:09.19 | RDF | wishes his stepdad had actually invested in McCaw Cellular in its infancy when it was the only cell company in the world. Craig Mccaw was the godfauther of the cell biz. Step dad told me he was with Craig when thay turned on the first cell test site in Chicogo in 1978 :) |
| 20:09.35 | Leland | wankel: yea.. that makes sense.. what appeared to be happening was sip provider was using its own IP and IP phone was using it's own, but I wanted asterisk to link between both in the middle.. like a proxy |
| 20:09.36 | veto_ | file[laptop], okay, but why? How do I turn that off? |
| 20:09.42 | RDF | he has some very very interesting stories to tell about it. |
| 20:09.45 | file[laptop] | veto_: you don't, it's the way SIP works |
| 20:09.53 | wankel | leland: normally, asterisk does. it bridges the two calls. |
| 20:10.06 | DrPete | file[laptop]: Yeah, I have it running now, I just wondered if you use the debian package or you compliled it? |
| 20:10.08 | Leland | so the call setup would go fine, but it would then attempt to get the phone and the sip provider to talk direct to each other. |
| 20:10.10 | file[laptop] | veto_: initially your device sends no username/password or anything, asterisk challenges for the data, your device then resends with an Authorization header which contains the username/password |
| 20:10.15 | Leland | wankel: that's the problem.. it wasn't bridging |
| 20:10.15 | Schism | hey, if I am setting up a trunk to broadvoice, do I really need their patch? |
| 20:10.15 | file[laptop] | DrPete: always compile it from CVS |
| 20:10.17 | RDF | Imagine being at the first cell site in the world that would be cool. |
| 20:10.23 | wankel | even if it wasn't... well, you said there were no firewalls, so the ITSP and the phone should be able to talk directly to one another. |
| 20:10.34 | file[laptop] | I pray, seeing you in heaven... one day! |
| 20:10.51 | Leland | wankel: they can talk directly to each other if I configure the authentication on the phone for the provider... |
| 20:10.57 | veto_ | file[laptop], gotcha...so it's registering to make the call on top of just plain being registered as a sip peer? |
| 20:11.02 | Leland | authentication to asterisk is different usernames etc. |
| 20:11.08 | Leland | hence the desire to bridge. |
| 20:11.09 | RDF | btw, anyone here has worked in the technical aspects of cell? |
| 20:11.11 | file[laptop] | veto_: it's not registering when you make a call |
| 20:11.15 | *** join/#asterisk jarrod (jarrod@dipole.informationwave.net) |
| 20:11.17 | DrPete | file[laptop]: ahh, oki. I was thinking of doing that, but I have a backport running now. How do you surgest I do it? |
| 20:11.21 | wankel | leland: the RTP isn't authenticated, afaik. |
| 20:11.25 | jarrod | yo.. should I run ext2 or ext3 on my asterisk filesystem? |
| 20:11.31 | Leland | no but the sip session is |
| 20:11.33 | file[laptop] | veto_: just the way SIP works, it has to send the username/password so asterisk verifies it... but SIP doesn't work like that |
| 20:11.42 | RDF | jarrod, it should not matter. Im running ext3 |
| 20:11.52 | file[laptop] | veto_: SIP initially sends the packet to start a call with no username/password, so asterisk has to ask the device "uh... send your username/password too" |
| 20:11.54 | PatrickDK | I much perfer reiserfs over ext3 |
| 20:11.56 | BBRodriguez | Hi people, i want to limit number of outgoing calls thrue sip channel using setgroup, but the conecpt escapes me, can anybody point me to an example ? |
| 20:12.06 | RDF | PatrickDK why is that |
| 20:12.06 | jarrod | are you pushing a lot of traffic through your *, rdf? |
| 20:12.08 | PatrickDK | don't run ext2, unless your doing some kind of 50megs or smaller partition |
| 20:12.09 | wankel | leland: the sip sessions were all to asterisk. if it was trying to get them to talk to one another, all it would change is the rtp endpoints. |
| 20:12.10 | file[laptop] | DrPete: just compile it from CVS... |
| 20:12.20 | PatrickDK | rdf, reiserfs handles inodes alot better |
| 20:12.22 | RDF | jarrod, no traffic :) |
| 20:12.47 | Leland | wankel: then I really don't know what the problem was ... maybe just a fluke (or a bug) |
| 20:12.49 | wankel | reiserfs is getting fairly stable but i'd still trust ext3 more for production. |
| 20:12.57 | PatrickDK | basically, if you have a few large files, ext3 is better |
| 20:12.58 | wankel | leland: hard to tell. gotta look at the packets. |
| 20:13.02 | DrPete | file[laptop]: I guess what I am asking, is, will it update my current install, or do I have to remove the package and compile the csv |
| 20:13.05 | PatrickDK | if you have tons of small files, reiserfs is better |
| 20:13.12 | file[laptop] | DrPete: it'll overwrite the installed files |
| 20:13.20 | Leland | need to work out this dtmf problem now.. |
| 20:13.22 | file[laptop] | DrPete: you can keep your configuration though |
| 20:13.24 | *** join/#asterisk PyroSteve (~steve@wsip-68-14-203-254.no.no.cox.net) |
| 20:13.28 | Leland | tried all dtmf modes and none of them work |
| 20:13.29 | veto_ | file[laptop], okay I see. first INVITE goes out, 407 comes back, ACK's with info then resends the INVITE, which completes. |
| 20:13.33 | PyroSteve | hey guys |
| 20:13.43 | RDF | has anyone tested the range of wifi phones yet? |
| 20:13.51 | PyroSteve | are there any Voice Pulse COnnect techs here |
| 20:14.01 | file[laptop] | veto_: your problem is elsewhere if that's going fine |
| 20:14.01 | Leland | RDF: I've used one of them |
| 20:14.03 | PatrickDK | rdf, I get about 1/4mile out of my wisip |
| 20:14.05 | wankel | rdf: i managed to throw mine about 100 feet, but i don't have a very good arm. |
| 20:14.20 | PyroSteve | rofl |
| 20:14.21 | RDF | Leland which one was it line of sight? |
| 20:14.49 | Leland | can't remember the model.. it's the one sold by Communitech for Net2PHone (but I don't use it for net2phone) |
| 20:15.03 | *** join/#asterisk Penfold (~mike@gerethia.altrion.org) |
| 20:15.07 | RDF | Patrick..really thats impressive. Line of sight? nothing in between the handset and your base? |
| 20:15.08 | veto_ | file[laptop], still trying to debug why I can't dial extensions through NAT so I broke you ethereal. (spa841's) |
| 20:15.22 | Leland | it works okay when it works.. but after about 15 mins or so, if it's idle, it stops sending/receiving IP entirely so it just sits there. |
| 20:15.28 | PatrickDK | rdf, just shingles |
| 20:15.34 | file[laptop] | veto_: there's info out there about NAT and stuff... just voip-info... |
| 20:15.36 | PatrickDK | antenna mounted in attic |
| 20:15.41 | Leland | if it's not idle, then it works fine |
| 20:15.47 | RDF | Pat, how high up? |
| 20:16.00 | RDF | what was the Gain of the antenna? |
| 20:16.02 | PatrickDK | but then I was using 12db gain antenna, and a 500mw amp/28db receive gain, around 35 feet high |
| 20:16.08 | file[laptop] | BoRiSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSSS |
| 20:16.30 | RDF | Pat, thats fairly impressive but I can see why. |
| 20:16.44 | veto_ | file[laptop] oh yea, I've been all over it...believe me. nat=yes, canreinvite, qualify, forwarding SIP ports, setting RTP range, forwarding RTP range, etc. |
| 20:16.45 | PatrickDK | I know, the antenna's on the phones suck |
| 20:16.55 | RDF | Pat, who makes the reciver and antenna? |
| 20:16.56 | Leland | I get the impression that cisco's version of rfc2833 isn't really rfc2833.. since it doesn't seem to work with anything else using rfc2833 |
| 20:16.57 | PatrickDK | you reallyshould use a amp/preamp on them |
| 20:17.07 | PatrickDK | hmm, highgain I think, let me check |
| 20:17.12 | RDF | okay |
| 20:17.50 | RDF | 12 db is really high. |
| 20:18.06 | blitzrage | my friend has 15dB antennaes |
| 20:18.10 | file[laptop] | didn't Pulver bring down the price on 'da Wisip? |
| 20:18.12 | blitzrage | for wifi |
| 20:18.13 | drgalaxy | each connector is .5db loss and each 20-25 feet is 1 db |
| 20:18.20 | file[laptop] | yeah down to $199... haha |
| 20:18.24 | RDF | what coax is used? |
| 20:18.27 | drgalaxy | RG8 |
| 20:18.29 | PatrickDK | drgalazy, depends on what cable ya use |
| 20:18.36 | RDF | ahh thik rg-8 |
| 20:18.37 | drgalaxy | true, thats just a general rule |
| 20:18.39 | RDF | thick |
| 20:18.49 | PatrickDK | I am using Times LM900 |
| 20:18.49 | RDF | I hav 100 feet of rg-8 |
| 20:19.25 | RDF | But the best for those frequencies is Hardline. |
| 20:19.48 | drgalaxy | RDF: how is that better than RG8? |
| 20:20.28 | JerJer | LMR is designed with very high frequencies in mind |
| 20:20.33 | blitzrage | someone kick A-Tuin|work |
| 20:20.37 | blitzrage | errr... A-Tuin[Play] |
| 20:20.44 | file[laptop] | I haven't the power Captain! |
| 20:20.55 | blitzrage | file[laptop]: another advantage for an @ |
| 20:21.08 | *** join/#asterisk WizardOne (~wizard@h40n3c1o848.bredband.skanova.com) |
| 20:21.11 | twisted | *yawn* |
| 20:21.13 | file | drumkilla!!!!!!!!!!!!!!!!!!!!!!!! |
| 20:21.13 | twisted | what's up? |
| 20:21.16 | file | twisted, TWISTED! |
| 20:21.19 | file | twisted: kick A-Tuin[Play] |
| 20:21.20 | WizardOne | lo |
| 20:21.23 | twisted | file, why? |
| 20:21.25 | drgalaxy | is there a noticable difference between LMR900 and RG8 for wifi? |
| 20:21.29 | blitzrage | twisted: y0 y0 |
| 20:21.34 | twisted | hey blitzrage |
| 20:21.36 | file | twisted: because his away system is flooding the channel every now and then |
| 20:21.37 | Moc_ | hi twisted |
| 20:21.44 | PatrickDK | ah, there it is, the RF LINX 500mw 2.4 APC amp |
| 20:21.51 | file | quiet him! |
| 20:21.52 | PatrickDK | 23db receive gain |
| 20:21.52 | Moc_ | patch is working... I find it very cool... |
| 20:22.02 | twisted | file, ah. |
| 20:22.07 | wankel | thbbt. 500mw. |
| 20:22.08 | PatrickDK | got it from fab-corp.com |
| 20:22.09 | *** mode/#asterisk [+q A-Turin[play]!*@*] by twisted |
| 20:22.11 | veto_ | just to verify, sip:jdoe@1.2.3.4:5060 is valid, correct? I saw somewhere that said the jdoe part needed to be a number. |
| 20:22.18 | twisted | problem solved |
| 20:22.23 | blitzrage | scweet |
| 20:22.26 | PatrickDK | wankel, I only needed it to cover 5acres, not the whole city |
| 20:22.30 | RDF | drgalaxy, hardline is used for VHF and above. It has the lowest db loss of any coax for those frequencies. It is used alot in cell towers. It is also very expensive and hard to bend. Very stiff outer aluminum coat and is over one inch in thickness. |
| 20:22.30 | wankel | :) |
| 20:23.05 | JerJer | that's the worse thing you can do |
| 20:23.08 | twisted | i hate trying to make this decision |
| 20:23.10 | RDF | wank, i have my licence to. Also have my FCC licence. |
| 20:23.11 | PatrickDK | wankel, anything over 1w has to be apc though |
| 20:23.22 | twisted | it's cold as all get out outside |
| 20:23.23 | twisted | yet |
| 20:23.28 | twisted | i need to take out the garbage and go to the store |
| 20:23.28 | JerJer | you ampilfy the good with the bad |
| 20:23.42 | f00b3r | hardline is a motherfucker |
| 20:23.44 | PatrickDK | yep, the point of installing filters |
| 20:23.51 | f00b3r | I did cell tower installations for 3 years |
| 20:23.53 | JerJer | twisted: its 9 degress and 8 inches of blowing snow out here |
| 20:23.56 | f00b3r | back in my electrician days |
| 20:23.58 | Schism | unlicensed is unlicensed |
| 20:24.00 | RDF | f00, I have not worked with it but can imagine how frigen stiff it is. |
| 20:24.00 | twisted | JerJer, yah, but are you going outside? |
| 20:24.01 | wankel | i was shopping for ones with channel-selectable filters |
| 20:24.08 | JerJer | twisted i've done two wireless installs today |
| 20:24.15 | PatrickDK | channel selectable? that is just silly |
| 20:24.15 | twisted | JerJer, heh |
| 20:24.30 | f00b3r | RDF: yeah, it sucked, especially for dense systems |
| 20:24.32 | wankel | filters out only one channel, auto gain control, etc. should work fairly well. |
| 20:24.36 | f00b3r | like 9 cambles per sector |
| 20:24.39 | f00b3r | er cables |
| 20:24.41 | nullogic | how do I set the ani on a outbound call? |
| 20:24.43 | RDF | foo, you mean with little room to work around. |
| 20:24.51 | wankel | oops. starting to snow. better go run get food and water so i don't have to go out tonight or tomorrow. |
| 20:24.55 | f00b3r | right |
| 20:25.13 | f00b3r | especially in some of the places we built these things |
| 20:25.16 | f00b3r | on buildings |
| 20:25.21 | f00b3r | inside grain silos |
| 20:25.22 | drgalaxy | I could wear a short sleeved shirt outside here |
| 20:25.26 | f00b3r | oin church steeples |
| 20:25.30 | f00b3r | on |
| 20:25.43 | nullogic | anyone? |
| 20:25.52 | JerJer | nullogic: |a |
| 20:26.34 | drgalaxy | nullogic if I remember.. the CO sets the ANI for calls coming into them |
| 20:26.49 | JerJer | show application SetCallerID |
| 20:26.52 | JerJer | next |
| 20:26.59 | blitzrage | kram: !!! |
| 20:27.04 | kram | hi blitz |
| 20:27.14 | blitzrage | JerJer: you need more enthusiasm on your _next_ |
| 20:27.27 | JerJer | that takes too much work |
| 20:27.34 | RDF | f00, but the nice thing with that type of work is you feel a sence of acomplishment :) |
| 20:27.39 | blitzrage | kram: how are you? I'm still at Steven's and hoping my flight back to Toronto doesn't get cancelled |
| 20:27.41 | JerJer | that caps lock key is sooooo far away |
| 20:27.47 | blitzrage | JerJer: lol |
| 20:27.57 | f00b3r | lol |
| 20:27.59 | f00b3r | true |
| 20:28.07 | f00b3r | but I sure as hell dont miss waking up at 4am |
| 20:28.10 | f00b3r | in pain |
| 20:28.12 | f00b3r | during the winter |
| 20:28.14 | wankel | jerjer: you could make it a script, but that's waaaaay too much work. |
| 20:28.16 | markit | kram: any hope of having more information about how to setup/use the new "native" assisted transfer? I need them to keep testing it |
| 20:28.22 | RDF | blitzrage we are sending the rain from BC to Toronto :) |
| 20:28.28 | f00b3r | doing VoIP in a warm office is the way to go ;) |
| 20:28.28 | RDF | in the form of snow |
| 20:28.33 | JerJer | quick everybody pile on kram :) |
| 20:28.51 | blitzrage | RDF: unfortunately its now frozen and a blizzard |
| 20:29.10 | JerJer | whoever is tracking this channel needs to see who says 'kram' the most *evil grin* |
| 20:29.18 | RDF | it has rained and rained and rained from the pinapple express. The jet stream was carring it into alberta. |
| 20:29.36 | blitzrage | JerJer: hrmmmm... I actually wonder :) |
| 20:29.42 | blitzrage | file: damn you! |
| 20:30.25 | file[laptop] | FOOOOOOOOOOD |
| 20:30.27 | blitzrage | someone needs to install one of those channel stats programs for #asterisk |
| 20:30.33 | JerJer | 327 people in room....how many bots? |
| 20:30.43 | DrPete | file[laptop]: Oki, thanks, so I dont need to bother removing the package? Should i backup my config files? Will it overwrite them? |
| 20:30.44 | twisted | file[laptop], excellent idea |
| 20:30.46 | freat | ~jbot weather |
| 20:30.46 | JerJer | blitzrage: there is one somewhere |
| 20:30.48 | chipig | don't look at me, Im not a bot. |
| 20:31.02 | scrubb | order some for me |
| 20:31.20 | JerJer | yeah let the poor chinamen freeze his rice off |
| 20:31.20 | twisted | er |
| 20:31.20 | twisted | brow |
| 20:31.39 | f00b3r | dude, chinamen is not the prefered nomenclature |
| 20:31.41 | twisted | JerJer, i can offer him some sake |
| 20:31.57 | JerJer | ok how about slant? |
| 20:32.04 | f00b3r | heh |
| 20:32.04 | Nugget | s nugget@dazed:~/irc/#asterisk>zgrep kram * | grep -v "<kram" | cut -d '<' -f 2 | cut -d '>' -f 1 | cut -d ' ' -f 1 | sort | uniq -c | sort -nr | head -4 14:31:56 |
| 20:32.08 | Nugget | <PROTECTED> |
| 20:32.08 | f00b3r | it was a movie quote |
| 20:32.10 | Nugget | <PROTECTED> |
| 20:32.13 | Nugget | <PROTECTED> |
| 20:32.15 | Nugget | <PROTECTED> |
| 20:32.16 | f00b3r | The Big Lebowski |
| 20:32.24 | file[laptop] | Nugget: try muffin |
| 20:32.38 | *** part/#asterisk securAX (~lmonstat@adsl-67-119-205-233.dsl.pltn13.pacbell.net) |
| 20:32.40 | blitzrage | JerJer: you know where that site is? |
| 20:32.50 | JerJer | nope |
| 20:32.59 | Nugget | s nugget@dazed:~/irc/#asterisk>zgrep muffin * | cut -d '<' -f 2 | cut -d '>' -f 1 | cut -d ' ' -f 1 | sort | uniq -c | sort -rn | head -4 14:32:38 |
| 20:33.00 | JerJer | i just remember i made the list for some reason |
| 20:33.02 | Nugget | <PROTECTED> |
| 20:33.05 | Nugget | <PROTECTED> |
| 20:33.07 | Nugget | <PROTECTED> |
| 20:33.10 | Nugget | <PROTECTED> |
| 20:33.14 | file[laptop] | muffintastic! |
| 20:33.27 | twisted | Nugget, do tiwsted :P |
| 20:33.30 | twisted | er twisted |
| 20:33.39 | twisted | i'm curiouis |
| 20:33.47 | JerJer | yeah n00bs like to pile on twisted, i've notced |
| 20:34.06 | veto_ | motherfucking son of a bitch, my 2 days of NAT issues were resolved by commeting out callerid="blah"...all that work for something so stupid. |
| 20:34.06 | blitzrage | check blitzrage t00 |
| 20:34.12 | Nugget | s nugget@dazed:~/irc/#asterisk>zgrep twisted * | grep -v "<twisted" | cut -d '<' -f 2 | cut -d '>' -f 1 | cut -d ' ' -f 1 | sort | uniq -c | sort -rn | head -4 |
| 20:34.16 | Nugget | <PROTECTED> |
| 20:34.18 | Nugget | <PROTECTED> |
| 20:34.21 | Nugget | <PROTECTED> |
| 20:34.23 | Nugget | <PROTECTED> |
| 20:34.31 | JerJer | veto_: callerid has nothing to do with NAT |
| 20:34.33 | twisted | geez |
| 20:34.55 | JerJer | bkw_ is winning all kinds of awards and he's not even here to accept them |
| 20:34.57 | *** join/#asterisk freat (~freat@node-40242662.mdw.onnet.us.uu.net) |
| 20:35.09 | twisted | JerJer, you beat me to the punch :) |
| 20:35.14 | twisted | about cid/NAT |
| 20:35.18 | veto_ | jerjer, I know...I have NO idea why commenting that out works...I had callerID="John Doe <1111>" and commenting it out makes the calls complete. |
| 20:35.28 | file[laptop] | veto_: wrong format |
| 20:35.34 | file[laptop] | "John Doe" <1111> |
| 20:35.34 | veto_ | jerjer, I probably haven't had a nat issue since the first night....it was my damned sip.conf |
| 20:35.38 | JerJer | perhaps whomever you are sending them to requires a valid CID format |
| 20:35.45 | *** join/#asterisk mrempire (~user1@h71032.upc-h.chello.nl) |
| 20:35.53 | veto_ | i feel dumber for doing all that when I didn't have a nat issue. |
| 20:35.53 | twisted | or none, in the case of privacy |
| 20:36.16 | *** join/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net) |
| 20:36.18 | Nugget | You don't need the quotes when you set callerid. |
| 20:36.27 | twisted | sure don't |
| 20:36.38 | ardor | exten => _X.,1,blah |
| 20:36.38 | ardor | exten => XXXX,2,blah_4_Digits |
| 20:36.38 | ardor | exten => XXXXXXX,2,blah_7_Digits |
| 20:36.42 | ardor | sould that work? |
| 20:36.43 | twisted | i have many a line like callerid=Some Dude <555-555-1212> |
| 20:36.51 | Nugget | ardor: yes. |
| 20:37.00 | JerJer | but blah is not an application |
| 20:37.09 | ardor | Nugget: Freaking awesome |
| 20:37.15 | Nugget | I use that trick for console logging |
| 20:37.21 | Umaro | ardor: _ |
| 20:37.22 | Nugget | ; Incoming calls from voicepulse account |
| 20:37.23 | Nugget | [vpcontext] |
| 20:37.23 | Nugget | exten => _.,1,NoOp(Incoming Call from voicepulse ${CALLERID} for ${EXTEN}@${SIPDOMAIN}) |
| 20:37.26 | veto_ | thanks for all the help ove rthe last few days whileI pulled my hair out...I now have a working phone system! |
| 20:37.26 | Nugget | exten => _NXXNXXXXXX,2,LookupCIDName |
| 20:37.28 | Nugget | exten => _NXXNXXXXXX,3,Goto(default,inbound,1) |
| 20:37.31 | Nugget | ^ like that |
| 20:37.33 | twisted | yeh |
| 20:37.53 | mrempire | wankel, is it possible to connect 2 asterisk at both ends trough the internet and each asterisk is then connected to local telco |
| 20:38.10 | *** join/#asterisk freat[laptop] (~freat[lap@node-40242662.mdw.onnet.us.uu.net) |
| 20:38.45 | twisted | well crap - i still need to go to the store |
| 20:38.46 | Silik0n | thats like one of the main ideas behind VoIP |
| 20:38.52 | twisted | bbl |
| 20:39.13 | jarrod | do most with multiple pris have multiple servers terminating sip sessions and then those boxen communicating with the asterisk server with pri cards via iax? |
| 20:40.01 | Silik0n | jarrod theres a variety of solutions that work in the situation and it depends on # of PRIs vs Number of SIP peers |
| 20:40.22 | jarrod | terminating hundreds of sip phones with ~15:1 ratio on pri channels |
| 20:40.35 | *** join/#asterisk kaffemand (~martin@cpe.atm2-0-1101141.0x50a4a2de.bynxx12.customer.tele.dk) |
| 20:40.47 | *** join/#asterisk sudoer (~sudoer@65.75.148.190) |
| 20:40.53 | sudoer | does anyone use firefly? |
| 20:41.26 | sudoer | do the actual IM messages go through the main * server before goign to reciever or is it direct |
| 20:41.32 | kaffemand | hey .. |
| 20:41.44 | Silik0n | jarrod: i'd prolly have SER do the SIP part... |
| 20:42.05 | freat | sudoer: there's directions for firefly on the wiki |
| 20:42.21 | sudoer | ok |
| 20:42.22 | bjohnson | god damn!! |
| 20:42.26 | jarrod | ill check that |
| 20:42.36 | Silik0n | then have the PRIs spread out on a couple of boxes... for redundancy sake |
| 20:42.46 | jarrod | yea |
| 20:42.51 | kaffemand | I've got some trouble .. When I dial out from my budgetone, through my asterisk to my provider, the connection is made, but disconnected after less than a second .. |
| 20:42.54 | bjohnson | my callerid problem is from my Ultraswitch 100 fax/phone/data switch that is supposed to forward through caller id info |
| 20:42.57 | jarrod | its only 3ghz xeons with a single quad card in each |
| 20:43.00 | kaffemand | http://pastebin.ca/4593 |
| 20:45.51 | bjohnson | anyone familiar with Ultracom Ultra 100 fax switches? I need a manual. |
| 20:48.26 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 20:49.41 | *** part/#asterisk eKo1 (~bernd@63.245.57.70) |
| 20:52.21 | RDF | found out somethign very interesting. asterisk is interfearing with sox play. |
| 20:56.26 | ManxPwr | RDF, Well if you use chan_oss or chan_alsa then other apps won't be able to open the sound device. |
| 20:56.41 | markit | anyone with asterisk cvs Head, that know something about "disconnect => *0" in features.conf? |
| 20:57.36 | mrempire | which software can i use on windows to create call via aterisk |
| 20:57.48 | mrempire | Asterisk |
| 20:57.51 | Silik0n | <PROTECTED> |
| 20:58.12 | mrempire | telnet? How can I talk audio then? |
| 20:58.15 | Silik0n | or are you looking for a softphone client for windows |
| 20:58.25 | mrempire | Yes softphone |
| 20:58.40 | mrempire | Do you know a good one |
| 20:58.41 | Silik0n | creating calls doesnt require a softphone, but in that case ytry firefly |
| 20:58.42 | RDF | Manx, yea thats what I figure. |
| 20:58.44 | Silik0n | or x-lite |
| 20:58.53 | PTG123 | xpro |
| 20:58.57 | markit | mrempire: you can put a well formed text file in the spool that asterisk uses for outgoing calls, or interface witht he telnet protocol to asterisk "manager" and issue the apropriate command sequence (I to this way with Delphi) |
| 20:58.59 | Silik0n | firefly is good for IAX and SIP calls, X-Lite is sip only, both are free |
| 20:59.02 | PTG123 | is the best one i have found, if you don't mind spending $30 |
| 20:59.15 | PTG123 | ~firefly |
| 20:59.16 | jbot | methinks firefly is http://virbiage.com/firefly/download/firefly-thirdparty.exe |
| 20:59.42 | PTG123 | firefly does sip? |
| 21:00.01 | mrempire | Which one uses less Traffic? |
| 21:00.02 | RDF | Manx, can you explain why when recording a gsm file and use it as a background it will not play? I can play it with the play command. also, when replacing that in the Background app with one of the built in gsm files it does work when i call the zap channel. I suspect it has to do with sample rate? I have some warning messages about it. |
| 21:00.05 | *** join/#asterisk Smuggs (~Smuggs@S0106000c41a28668.cg.shawcable.net) |
| 21:00.19 | RDF | googling Unable to open greeting.wav (format unknown): No such file or directory |
| 21:00.25 | RDF | err thats not it. |
| 21:00.26 | RDF | :) |
| 21:00.28 | mrempire | On one side I have a 28k8 modem |
| 21:00.39 | PTG123 | mrempire: which softphone? |
| 21:00.41 | RDF | sox: Unable to set audio speed to 8000 (set to 8018) |
| 21:00.46 | RDF | yea thats it :) |
| 21:01.08 | PTG123 | mrempire, xpro uses g729, not gonna get less bw usage then that |
| 21:01.10 | mrempire | Yes whicj softphone with what protocoll for less traffic |
| 21:01.19 | PTG123 | mrempire, xpro is the one you want |
| 21:01.24 | *** join/#asterisk jero (~boo@modemcable040.12-81-70.mc.videotron.ca) |
| 21:02.33 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 21:02.34 | mrempire | Thanks PTG123, I will go and look for it now |
| 21:02.43 | PTG123 | mrempire, xten.com |
| 21:02.48 | drgalaxy | RDF did you use sox to resample your files? |
| 21:02.59 | mrempire | is it free? |
| 21:03.02 | drgalaxy | RDF I had the best luck using standard mono wav files and resampling |
| 21:03.04 | RDF | dr, no im creating a new greeting file. |
| 21:03.23 | PTG123 | no, but if you need to use it from a modem its probably the one you will n eed |
| 21:03.24 | RDF | it is in my start application when a call comes in. |
| 21:03.24 | drgalaxy | sox wavefile.wav -r 8000 outfile.gsm resample -ql |
| 21:03.26 | PTG123 | you can try xlite with the GSM codec |
| 21:03.32 | PTG123 | on the same website |
| 21:03.36 | PTG123 | but g729 is gonna work best |
| 21:03.41 | Silik0n | mrempire X-Pro is like $30... g.729 is not a feww codec....x-lite works just fine with GSM |
| 21:03.48 | Silik0n | s/feww/free/ |
| 21:03.53 | RDF | dr, are you saing I should first record as a wav then convert it and it will work? |
| 21:04.15 | drgalaxy | RDF : that is the only way I could get audio files from my recording studio (nuendo) into the PBX |
| 21:04.15 | mrempire | Is g729 codes the best compression? |
| 21:04.24 | Nukemizer | my Digium cards did not come this wekend and I wanted to test a card, dos anyone know if a Dailog |
| 21:04.31 | Nukemizer | card would work |
| 21:04.43 | drgalaxy | RDF : you cannot confuse GSM encoded wav files with a raw GSM stream |
| 21:04.56 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
| 21:04.57 | PTG123 | mrempire, by far, and the best quality |
| 21:05.11 | PTG123 | mrempire, GSM uses like 50% more bandwidth, and its not as good of quality |
| 21:05.30 | RDF | drgalaxy okay I will give that a try. this is all fairly interesting :) |
| 21:05.31 | PTG123 | mrempire, next best one is speex, quality ok, bw usage double to triple that of g729 |
| 21:05.53 | *** join/#asterisk letherglov (~letherglo@8036aa5e.resnet.ucsd.edu) |
| 21:06.05 | mrempire | A have a limited bandwith(28k8) is that enough? |
| 21:06.19 | PTG123 | i have done testing with g729 on modems, worked suprisingly well |
| 21:06.23 | drgalaxy | RDF what are you using to record your audio? |
| 21:06.27 | PTG123 | so if you want to pull it off, spend $30 on xpro |
| 21:06.34 | letherglov | the DVSD modems use G.729 |
| 21:06.37 | RDF | drgalaxy rec on the command line |
| 21:06.50 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-39-201.no.no.cox.net) |
| 21:07.06 | drgalaxy | RDF teeheehee you can spice it up with audacity.sf.net (for free) |
| 21:07.14 | RDF | :) |
| 21:07.18 | drgalaxy | mix in some creative commons music.. make you sound like a movie star |
| 21:07.21 | mrempire | I wiil go and check the site xten.com, Thanks |
| 21:07.30 | RDF | heheh |
| 21:08.00 | RDF | dr, sure but for now would want to get this going :) my wife who is a audiophile would love it when I eventually show her. |
| 21:08.29 | drgalaxy | RDF cool. be sure to check out "convert wav audio files for use in Asterisk" on the wiki |
| 21:08.37 | RDF | okay |
| 21:08.37 | RDF | :) |
| 21:09.39 | veto_ | anyone know if the YAP usb headset is a true usb audio device? (which could be used with another softphone) |
| 21:10.16 | drgalaxy | wikis have brought about a new term.. RTFW |
| 21:10.36 | Druken | STFW |
| 21:10.46 | drgalaxy | LMWO |
| 21:11.03 | Druken | lmwo ? |
| 21:11.33 | Druken | oh.... |
| 21:12.42 | mrempire | xpro costs $60 now?? thta's not $30!! |
| 21:15.52 | hermie | kram round? |
| 21:16.51 | PTG123 | what does LNP stand for? |
| 21:17.52 | robin_sz | Luser Numbering Protocol? |
| 21:18.32 | *** join/#asterisk ` (~PaPaX@pD9E49A1E.dip.t-dialin.net) |
| 21:18.44 | drgalaxy | lugubrious neutered packrats |
| 21:18.45 | PTG123 | Local Number Portability perhaps |
| 21:20.34 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 21:21.14 | robin_sz | so ... having failed to get a conference set up on one omachine at location A, I managed it at location B, presumably I can just set up an IAX link between the two boxes, and convince * to route call to the conference over to box B right? |
| 21:21.15 | bjohnson | anyone use fax/data/phone switches to route faxes to fax machine before getting to a fxo port (ie keep faxes out of the voip system)? |
| 21:23.20 | *** join/#asterisk lancey (Shady@support.net1.cc) |
| 21:23.22 | lancey | hi guys! |
| 21:23.25 | Schism | bjohnson: like an auto detect box? |
| 21:23.46 | lancey | anyone knows what that Cisco ATA debug message means : Failed to extract UID from RxMsg |
| 21:23.56 | lancey | i don't get any sound using SIP to connect it to Asterisk |
| 21:24.12 | lancey | connecting it to another asterisk works, though, what could be the reason? |
| 21:25.25 | Schism | codecs configured the same on the * machines? |
| 21:25.36 | lancey | yes |
| 21:25.42 | lancey | sip.conf is absolutely the same |
| 21:25.46 | lancey | firewall was turned off.... |
| 21:25.51 | *** join/#asterisk vexorg (~vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 21:25.56 | Schism | same ver of *? |
| 21:26.02 | lancey | nopez |
| 21:26.06 | lancey | slightly newer cvs |
| 21:26.16 | lancey | both are cvs-head |
| 21:26.31 | Schism | do you see anything on the * console? |
| 21:26.37 | lancey | everything is normal |
| 21:26.42 | lancey | like a normal call |
| 21:26.44 | Schism | hmm |
| 21:26.51 | bjohnson | Schism: yes |
| 21:27.10 | lancey | :))) |
| 21:27.30 | Schism | bjohnson: never used one w/ *, but using it as intended, they never seemed to work right |
| 21:27.31 | lancey | i do see "anything", too :) |
| 21:27.47 | Schism | hehe |
| 21:27.52 | *** join/#asterisk tih (~tih@athene.hamartun.priv.no) |
| 21:27.55 | Schism | that's about as far as I can troublshoot * :-P |
| 21:28.03 | lancey | :)) |
| 21:28.06 | lancey | me too :)))) |
| 21:28.11 | Schism | what other WARNING / NOTICE messages u get? |
| 21:28.32 | lancey | <PROTECTED> |
| 21:28.32 | lancey | <PROTECTED> |
| 21:28.32 | lancey | <PROTECTED> |
| 21:28.32 | lancey | <PROTECTED> |
| 21:28.32 | lancey | <PROTECTED> |
| 21:28.40 | lancey | everything is fine :) |
| 21:28.48 | Schism | hmm |
| 21:29.13 | Schism | have you tried calling your ata186 from the console? |
| 21:29.24 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 21:29.31 | lancey | nopez |
| 21:29.41 | Schism | that might eliminate some variables |
| 21:29.42 | ctooley | how od I turn off CDR's all together? |
| 21:29.46 | RDF | Manx, are you here? Audio quality is okay its to bad the conversion sounded a little better :) |
| 21:29.52 | robin_sz | xorcom++ # highly amusing |
| 21:29.58 | lancey | Schism how would i hear anything |
| 21:30.04 | lancey | the * box is 500 km away from me |
| 21:30.07 | lancey | and with no sound at all :) |
| 21:30.22 | Schism | I thought you couldn't get a call to complete |
| 21:30.25 | RDF | lancey, whats the problem? |
| 21:30.39 | robin_sz | and bugger me, it does what youd expect |
| 21:30.47 | Schism | what? |
| 21:30.59 | lancey | RDF: when connecting Cisco ATA to *, i don't get any sound |
| 21:31.02 | lancey | using SIP |
| 21:31.10 | lancey | connecting the ATA to another * is fine |
| 21:31.11 | RDF | i see |
| 21:31.36 | lancey | both with same sip.conf |
| 21:33.19 | Schism | anyone have any luck w/ using a broadvoice trunk? |
| 21:33.24 | RDF | lancey well you are further then me :) |
| 21:33.35 | RDF | getting my fricken greeting to work is a pain. |
| 21:33.40 | Umaro | ok guys, who here is using SER with NAT? |
| 21:33.49 | lancey | :) |
| 21:33.55 | lancey | auch, ser |
| 21:34.00 | lancey | auch, nat :) |
| 21:34.19 | lancey | i hate both of these 3-letter shits ;) |
| 21:35.21 | RDF | http://pastebin.ca/4594 |
| 21:35.31 | RDF | my gsm file problem :) |
| 21:36.17 | lancey | [23:33] <RDF> lancey well you are further then me :) |
| 21:36.18 | lancey | ops |
| 21:36.30 | lancey | 001 Jan 21 13:28:28 WARNING[904]: format_wav.c:159 check_header: Unexpected freqency 8018 |
| 21:36.40 | lancey | trying to use windows-created .wav file? |
| 21:36.57 | lancey | 007 Jan 21 13:28:38 WARNING[904]: file.c:790 ast_streamfile: Unable to open greeting (format unknown): No such file or directory |
| 21:37.05 | lancey | is the file in /var/lib/asterisk/sounds/ ? |
| 21:37.13 | lancey | can the asterisk process access it? |
| 21:37.52 | RDF | lancey, dam I have the same problem as you! |
| 21:38.14 | RDF | look at my pastebin same thing. |
| 21:38.23 | lancey | RDF: i'm looking at it |
| 21:38.23 | lancey | :) |
| 21:38.29 | lancey | what do you think i'm talking about? |
| 21:38.30 | lancey | :) |
| 21:38.36 | lancey | i don't have such a problem |
| 21:38.37 | lancey | :) |
| 21:38.40 | RDF | I was away looking at asterisk site. |
| 21:38.46 | lancey | just suggesting some things fo you :) |
| 21:38.58 | RDF | yes the file is in /asterisk/sounds |
| 21:39.24 | RDF | I can play the built in sound files though Background application when dialing in via zap but not my own. |
| 21:39.41 | lancey | yes |
| 21:39.46 | lancey | how did you create the file? |
| 21:40.01 | RDF | I have done what Manx sugested by converting a wav file to gsm with his aproach but nothing helped. |
| 21:40.16 | lancey | well |
| 21:40.20 | lancey | the file is .wav / |
| 21:40.20 | lancey | ?? |
| 21:40.38 | lancey | as far as i see |
| 21:40.43 | RDF | rec greeting.gsm -r 8000 vol 1.5 |
| 21:41.17 | RDF | then what manx wanted me to do was... |
| 21:41.39 | RDF | sox greeting.wav -r 8000 greeting.gsm resample -ql |
| 21:41.44 | RDF | That did not work |
| 21:42.00 | lancey | so what did you do next? |
| 21:42.23 | lancey | as far as i see from the pastebin, you DO have a .wav file |
| 21:42.24 | RDF | My very first time I recorded was doing a rec greeting.gsm -r 8000 vol 1.5 and that did not work. |
| 21:42.32 | RDF | yea |
| 21:42.35 | RDF | I know |
| 21:42.39 | lancey | how did you create it? |
| 21:42.58 | RDF | rec greeting.wav -r 8000 vol 1.5 |
| 21:43.03 | *** join/#asterisk xcoyote (~coyote@201.128.119.188) |
| 21:43.20 | lancey | xm |
| 21:43.26 | lancey | could you send me that file |
| 21:43.55 | RDF | ohh i can also play those files and hear them though my headphone. |
| 21:44.01 | xcoyote | i have a problem while trying to use my own function in extensions.conf it returns 'pbx_extension_helper: No application 'IVRMusicRequest' for extension .... any suggestion about this kind of problem |
| 21:44.02 | xcoyote | ? |
| 21:44.08 | lancey | probably it's not in 8 KHz |
| 21:44.33 | RDF | I was thinking the same. what switch is needed to set to 8khz? |
| 21:44.44 | lancey | xcoyote obviously * doesn't now anything about such application |
| 21:44.49 | lancey | RDF: dunno, never used rec :) |
| 21:44.54 | lancey | rec --help ? |
| 21:44.57 | RDF | -r 8000 should be it |
| 21:45.01 | lancey | maybe |
| 21:45.08 | RDF | lancey what do you use? |
| 21:45.20 | drgalaxy | RDF why don't you install sox and convert your file? |
| 21:45.23 | lancey | maybe -r 8192 |
| 21:45.35 | lancey | RDF: i record them on windows :) |
| 21:45.38 | RDF | drgalaxy because it already is and has been done. |
| 21:45.53 | pimpwell | 100 minute long calls a day, using SIP adds up to how much transfer a day? |
| 21:46.11 | xcoyote | i know that |
| 21:46.29 | xcoyote | any better suggestion about this kind of message? |
| 21:46.46 | lancey | xcoyote nopez :/ |
| 21:46.49 | RDF | okay what about esd? |
| 21:47.11 | lancey | RDF: could you send me that greeting.wav? |
| 21:47.20 | lancey | DCC or mail? |
| 21:47.52 | RDF | let me make another one for your vieing it has some personal info on it. |
| 21:47.58 | lancey | ok |
| 21:48.14 | lancey | mail it to root[at]net1.cc |
| 21:48.26 | lancey | or dcc-send it to me |
| 21:50.43 | *** part/#asterisk xcoyote (~coyote@201.128.119.188) |
| 21:51.57 | BBRodriguez | does anyone know how to use SetGroup/CheckGroup for outgoing channels ? Please ? |
| 21:52.12 | BBRodriguez | point me to an example, please |
| 21:52.41 | RDF | lancey, check your mail |
| 21:53.12 | lancey | k |
| 21:53.30 | lancey | still not here :) |
| 21:53.49 | RDF | mmm |
| 21:54.02 | RDF | let me put on tcpdump and see if its going out. |
| 21:55.10 | BBRodriguez | Can anyone please tell me, how to use SetGroup instead of outgoinglimit=X in sip.conf ??? please |
| 21:58.03 | BBRodriguez | how do i restrict the number of outgoing calls on single SIP channel ?? anyone ? please |
| 21:58.04 | RDF | lancey, are you running the channel? because when I send you mail I see this in the channel 3:55:41.259923 IP pbx.vanjet.com.32963 > server1.net1.cc.smtp: . ack 424 win 5840 <nop,nop,timestamp 1152422 510905381> |
| 21:58.09 | DrPete | well I am having a go at compliing asterisk, heh. Is it safe run stable? I mean keep upgrading? |
| 21:58.15 | markit | RDF: my wav are created with Kwave, a 44.100Hz sampling, 1 track, no compression, resolution di 16 bit, ?linear two's complement? |
| 21:58.20 | RDF | no i mean i see the uiu You have new email |
| 21:58.37 | markit | RDF: then sox file.wav -r 8000 -c 1 outfile.gsm resample -ql |
| 21:59.11 | RDF | markit manks did not show me the -c 1 part. I will try again. |
| 21:59.32 | bjohnson | anyone have an example of answering a call and allowing use of outgoing lines for matching callerids or some other kind of authentication? |
| 21:59.35 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 21:59.56 | markit | RDF: but do you have the correct wav? not all wav are the same... (i.e. the "linear two complement" is important, AFAIR) |
| 22:01.02 | Moc_ | hi kram |
| 22:01.38 | twisted | kram :) |
| 22:01.56 | lancey | RDF i was out for a while |
| 22:01.58 | lancey | lemme check |
| 22:02.05 | lancey | nothing |
| 22:02.31 | Schism | anyone get any of the digium drivers working on OSX yet? |
| 22:02.58 | RDF | lancey, wierd. okay let me try what markid said and hope this works. |
| 22:03.03 | *** join/#asterisk zotz (~zotz@24.244.133.136) |
| 22:03.21 | lancey | k |
| 22:05.03 | RDF | does not work.... |
| 22:05.11 | RDF | ckick....no sound file playing. |
| 22:05.33 | markit | RDF: I produce the italian translation of the asterisk sounds... there must be something wrong in your setup |
| 22:05.41 | markit | RDF: do you have kwave? |
| 22:07.13 | robin_sz | hmmm .. as I join an empty meetme, I get "you are currently the only.. " etc then abut 20ms of on-hold music ... where the rest of it gone? |
| 22:08.05 | RDF | okay hold on....does Background application play the wav by default? it is complaining is it not present but its there. Why in the world do I feel there is some kind of sharing violation. |
| 22:08.27 | RDF | no, I do not have kwave |
| 22:09.09 | lancey | RDF it first says the header is not ok |
| 22:09.10 | derfer | exten => 206,1,GotoIf($[${CALLERIDNUM} = 303]?3:2) |
| 22:09.30 | derfer | i can remplace variable by another function ? |
| 22:09.34 | RDF | lancey yea so what would that mean? |
| 22:10.00 | derfer | exten => 206,1,GotoIf($[Answer = 303]?3:2) |
| 22:10.14 | derfer | i will make my if |
| 22:10.28 | derfer | if user press 5 on the phone |
| 22:10.33 | derfer | for make a menu |
| 22:10.34 | lancey | RDF i think the .wav file is weird |
| 22:10.49 | RDF | Um, I would agree :) |
| 22:10.58 | lancey | :) |
| 22:11.07 | RDF | Lancey, again it would play built in native gsm files but not my own. |
| 22:11.27 | lancey | yup |
| 22:11.28 | lancey | so |
| 22:11.31 | RDF | I have a es1370 sound card. Running slack 10.1 |
| 22:11.44 | lancey | this has nothing to do with it |
| 22:11.47 | lancey | the file is not OK |
| 22:11.54 | RDF | I know thought I would mention it. |
| 22:11.54 | lancey | it's probably not 8 KHz |
| 22:11.54 | RDF | :) |
| 22:11.56 | lancey | or is windows-style |
| 22:12.27 | RDF | I think there was some mention of that on a google search. |
| 22:12.32 | markit | RDF: gimme your e-mail, I will send you a wav file that works for sure (in italian) so you can do some tests |
| 22:13.36 | RDF | I wonder if there is a sound related channel on this network :) |
| 22:16.53 | *** join/#asterisk zeek (~unknow@gw.dhivehinet.net.mv) |
| 22:18.20 | derfer | pls how i can receive value preset of user on the phone ? |
| 22:18.40 | Schism | what? |
| 22:19.03 | derfer | sorry for my bad english |
| 22:19.18 | derfer | i will make gotoif on a variable |
| 22:19.28 | RDF | i have sb128 es1370 |
| 22:19.32 | derfer | this variable contain a value |
| 22:19.48 | derfer | value preset on the phone by on the user |
| 22:20.07 | RDF | markit, need to take dog outside brb okay ? |
| 22:20.22 | markit | RDF: ok |
| 22:20.29 | RDF | :) |
| 22:20.34 | RDF | brb in 10-15 |
| 22:20.45 | gopinsurg | Do you guys think I could use a P3 350 to install and test a the IP features with * ? |
| 22:21.01 | markit | RDF: ok, but I'm not a sound/linux expert.. I had help from others to make my sb work |
| 22:21.02 | file | sure why not |
| 22:21.07 | file | asterisk can run on a P75 fyi |
| 22:21.34 | gopinsurg | will I get bad audio quality on it? |
| 22:21.42 | file | you'll be fine |
| 22:24.56 | lancey | anyone knows anything about Cisco ATA saying Failed to extract UID from RxMsg |
| 22:25.02 | lancey | and not willing to send the voice |
| 22:25.03 | lancey | :) |
| 22:33.46 | PyroSteve | I need help with VP COnnect account !! |
| 22:34.04 | PyroSteve | My incoming calls wont work anymore |
| 22:34.13 | PyroSteve | my account balence was $000 |
| 22:34.19 | PyroSteve | so I added money to it |
| 22:34.26 | PyroSteve | and it still wont work |
| 22:34.36 | PyroSteve | it gives me a fast busy |
| 22:34.42 | PyroSteve | outgoing works fine |
| 22:35.07 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 22:35.07 | PyroSteve | help ! |
| 22:35.31 | lancey | :) |
| 22:35.47 | lancey | be more descriptive, please |
| 22:36.12 | lancey | what happens on the console |
| 22:36.14 | lancey | what does it say? |
| 22:36.17 | yashax | twisted: are you there? i had a quick question for you. Trying to remember if you were the one that told me that you live in georgai? |
| 22:36.17 | PyroSteve | nothing! |
| 22:36.23 | lancey | do you get incoming call from VP Connect? |
| 22:36.27 | PyroSteve | no |
| 22:36.34 | PyroSteve | it cant be on my end |
| 22:36.36 | *** join/#asterisk patdk (patrickdk@dyn-19-218.myactv.net) |
| 22:36.38 | PyroSteve | its been working |
| 22:36.42 | PyroSteve | nothing has changed |
| 22:36.49 | lancey | well the problem is probably with VP Connect |
| 22:36.54 | lancey | why not contact them? |
| 22:36.54 | PyroSteve | plus my account balence was 0.00 !! |
| 22:37.07 | PyroSteve | cause they dont work on weekends |
| 22:37.31 | PyroSteve | i was hoping to find an employee here |
| 22:37.58 | twisted | yashax, no. i do not live in georgia. |
| 22:38.48 | *** join/#asterisk rikstahh (~rick@81-178-241-190.dsl.pipex.com) |
| 22:39.09 | PyroSteve | is there any voice pulse connect admins hanging around ? |
| 22:39.39 | yashax | twisted: thanks, sorry, There was someone who was a speaker during the last conference and we spoke then and I forgot his name, trying to get a hold of him for a small job in Atlanta |
| 22:39.47 | lancey | like VoicePulse? |
| 22:39.49 | lancey | :) |
| 22:42.36 | czero | yashax lots of qualified people here though if they don;t have to be local to Altanta |
| 22:43.01 | *** join/#asterisk heLp (FaTHeR@144.138.194.170) |
| 22:45.00 | PyroSteve | <PROTECTED> |
| 22:45.05 | PyroSteve | oops |
| 22:45.50 | czero | :) |
| 22:46.00 | lancey | L:)) |
| 22:46.11 | lancey | be sure it's VoicePulse |
| 22:46.12 | lancey | :) |
| 22:48.39 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 22:50.42 | zeek | I have done this but still I don't get music on hold |
| 22:50.42 | zeek | <PROTECTED> |
| 22:50.42 | zeek | <PROTECTED> |
| 22:50.42 | zeek | <PROTECTED> |
| 22:50.42 | zeek | I have the mpg123 installed. What do it need to check? |
| 22:54.14 | *** join/#asterisk JmanA9 (~josh@h207.182.40.69.ip.alltel.net) |
| 22:54.20 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 22:54.22 | JmanA9 | hello |
| 22:55.25 | BBRodriguez | how do i restrict the number of outgoing calls on single SIP channel ?? anyone ? please |
| 22:55.35 | *** join/#asterisk cursor (~kevin@andromeda.office.cursor.biz) |
| 22:57.33 | *** join/#asterisk dontmsgme (~none@adsl-64-168-164-90.dsl.lsan03.pacbell.net) |
| 22:57.37 | dontmsgme | 'Lo. |
| 22:57.52 | cursor | hello |
| 22:59.20 | uunot | BBRodriguez: investigate setgroup() and checkgroup() |
| 22:59.49 | *** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au) |
| 23:00.59 | dontmsgme | If you make your own VOIP service how much does each DID cost you |
| 23:01.15 | JmanA9 | whats a good guide on setting up * for freeworlddialup? |
| 23:01.24 | JmanA9 | i've tried the guides on the support page, but it just doens't work for some reason |
| 23:01.54 | asjoyner | dontmsgme: depends on where you're getting your DIDs from -- in the case of a small to medium size ISP getting DIDs over PRIs, in the neighborhood of $0.25 a piece |
| 23:02.33 | uunot | asjoyner: course the PRI is a litle more than $0.25 ;) |
| 23:02.42 | czero | :) just a bit more |
| 23:02.46 | asjoyner | uunot: he just asked about the DIDs :) |
| 23:02.55 | BBRodriguez | uunot: I'm looking for working example of SetGroup |
| 23:03.00 | uunot | ah, innocent omission :) |
| 23:03.18 | uunot | BBRodriguez: tried the wiki? |
| 23:03.27 | BBRodriguez | uunot: but the one in wiki is for incoming calls only... the concept of using it for outgoing escapes me |
| 23:03.58 | *** join/#asterisk Sitxu (~sitxu@200.82.228.164) |
| 23:04.10 | uunot | ? works, pretty much, the same way. just put it in the context for the outgoing dial plan. |
| 23:05.50 | derfer | someone can help me ? |
| 23:06.18 | Sitxu | hi, how do i test * 1st time, without fxs/fxo? |
| 23:07.10 | asjoyner | quick question for someone familiar with any of the various VOIP providers (ala Broadvoice, Voicepulse, Vonage, etc) - how many concurrent outbound calls do they allow? just 1? 2? more? |
| 23:07.52 | RDF | good question |
| 23:08.11 | asjoyner | of course I mean with particular reference to their "unlimited" plans |
| 23:08.13 | RDF | But most importantly it may be more a factor of bandwith. |
| 23:08.35 | asjoyner | RDF: at least on my end (and I doubt on their end), bandwidth is not a factor in my equation. :) |
| 23:08.59 | robin_sz | ah ha! ... I have a small firewall problem .. I can see udp packets passing inwards over my NAT router, but they fail to escape |
| 23:09.15 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 23:09.29 | asjoyner | robin_sz: sounds like you don't have any particular rule that will allow them to come inwards |
| 23:10.02 | asjoyner | robin_sz: particularly if you're not explicitly allowing it, and it's.. ah I always confuse them, symetric NAT vs <insert other type of NAT here> |
| 23:10.25 | asjoyner | robin_sz: some NATs will handle passing RTP streams with out any real mangling, Linux and most NATs do not |
| 23:11.14 | asjoyner | robin_sz: but don't take my word for it alone, I am going mostly on my impression from reading / other conversations, not actual practical knowldge in this case |
| 23:11.21 | robin_sz | asjoyner: well, I have a rule to pass them from the external ADSL line into the box on the internal net |
| 23:11.34 | asjoyner | robin_sz: then your rule is incorrect? :) |
| 23:11.49 | robin_sz | asjoyner: I can see them arriving and leaving again with tcpdump on the box |
| 23:11.51 | RDF | markit did u send it? |
| 23:12.11 | robin_sz | asjoyner: I guess it is ... hmmm |
| 23:12.14 | RDF | markit, also did you make that gsm file? |
| 23:12.16 | asjoyner | robin_sz: okay, by that you mean packets come in one interface, and go out the other interface... so what pray-tell is the problem, then? |
| 23:13.06 | robin_sz | asjoyner: packets from the big wide world come in, hit the NAT box, get passed to the * box, they leave the * box, but never make it to the big wide world |
| 23:13.37 | asjoyner | robin_sz: are you sure the asterisk box knows where to send them? |
| 23:13.45 | robin_sz | asjoyner: which is odd, because softphones make it out |
| 23:14.11 | robin_sz | asjoyner: yeah, tcpdump shows them addressed to the righ thost |
| 23:14.21 | asjoyner | robin_sz: and they hit the internal interface and then get dropped? |
| 23:14.30 | robin_sz | seems so ... |
| 23:14.32 | robin_sz | weird |
| 23:14.43 | asjoyner | robin_sz: what specifically does your firewall do with outbound traffic that it doesn't know what to do with? default of drop? |
| 23:14.57 | robin_sz | drop I suspect |
| 23:15.04 | asjoyner | robin_sz: what type of firewall? |
| 23:15.32 | robin_sz | Alcatel 'speedtouch pro' adsl modem/router |
| 23:15.38 | robin_sz | linuxy thing underneath |
| 23:15.44 | uunot | asjoyner: if you interested, i use teliax, they've been great |
| 23:16.02 | asjoyner | uunot: know their policy on concurrent outbound calls? |
| 23:16.21 | asjoyner | robin_sz: how are you configuring the rules? iptables "underneath", or some web interface on top? |
| 23:16.42 | derfer | how i can make a macro who respond you have push 9985140 if user push on the xlite 9985140 |
| 23:16.44 | uunot | asjoyner: think it 2 on the res plans, 4 on the corp, and no limit on PAYG |
| 23:16.51 | derfer | someone can help me ? |
| 23:16.53 | robin_sz | asjoyner: some command line thing on the router |
| 23:17.36 | asjoyner | robin_sz: ah, not something I'm familiar with, but if I had to guess it's probably related to the fact that it's not creating a "reverse" allow mapping for that incoming rule that allows the RTP traffic through to * |
| 23:18.20 | RDF | since when does vi use 50% of cpu resources. |
| 23:18.21 | RDF | mmm |
| 23:18.44 | asjoyner | derfer: exten => 9985140,Macro(yourmacro) for more information check http://www.voip-info.org |
| 23:18.49 | robin_sz | asjoyner: thats what it looks like to me as well ... |
| 23:19.04 | markit | OT: beware paypal messages about your incomplete data, there is a link to a fake site that steals your password! |
| 23:19.19 | robin_sz | no? really |
| 23:19.24 | markit | (deyshengs.netfirms.com) |
| 23:19.24 | asjoyner | robin_sz: my condolences. You should throw away that router, and use iptables firewalling, with the * linux box as your gateway :) |
| 23:19.59 | robin_sz | asjoyner: would love to ... if I ever get transparent bridging to work on it I'll bypass the b*stard :) |
| 23:20.06 | markit | robin_sz: I know it's a well know problem, but I've got that e-mail right now, and I think I'm not the only one they sent it, so... |
| 23:20.35 | asjoyner | robin_sz: alcatel DSL modem / router, right? |
| 23:20.40 | robin_sz | markit: I get upwards of ten of those a day ... |
| 23:20.43 | robin_sz | asjoyner: yeah |
| 23:20.49 | markit | (and I know a lot of people uses / likes paypal here) |
| 23:21.08 | markit | robin_sz: really? it's the first one I've got |
| 23:21.09 | asjoyner | robin_sz: chuck it and buy a Zoom, do half bridging so it can do PPPoA (or PPPoE if godforbid it's required) and chuck the alcatel junk :) |
| 23:21.09 | robin_sz | markit: but this is a geek channel, geeks dont fall for that stuff |
| 23:22.05 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-20-118.d4.club-internet.fr) |
| 23:22.10 | robin_sz | markit: at least 10. and ebay ones ... and bank ones .. and ... over 3000 spams a day (thank $deity for spamassassin!) |
| 23:22.15 | asjoyner | robin_sz: I work as the sysadmin for a medium sized ISP, deal with these issues all day long. :) Most of them can be remedied by using a better DSL modem :) |
| 23:22.16 | markit | robin_sz: geek can easely ignore my warnings, but some newbie could he here... just he is not reading the channel, since it's already in that fake site entering his info ;) |
| 23:22.28 | robin_sz | as |
| 23:22.37 | robin_sz | asjoyner: yeah, I can imagine :) |
| 23:22.40 | Qwell | I get those ones from capital one, etc, at work, and I work for a competing bank |
| 23:22.47 | RDF | I need to creat some kind of voip-pstn dial in account. Calling my zap via cell for testing is getting a little expensive. |
| 23:22.51 | Qwell | never gotten a fake from "my bank" though |
| 23:22.56 | *** part/#asterisk cursor (~kevin@andromeda.office.cursor.biz) |
| 23:23.34 | *** join/#asterisk RDF (~leonardo@S010600055d210201.vs.shawcable.net) |
| 23:23.52 | robin_sz | markit: whatever. if everyone reported every phishing scam on every IRC channel .. the place woudl grind to a halt .. I expect anyone interested would be on #phishing |
| 23:25.00 | *** join/#asterisk Preston (Preston@ip68-106-19-249.ph.ph.cox.net) |
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| 23:25.09 | *** join/#asterisk empire667 (~user1@h71032.upc-h.chello.nl) |
| 23:25.46 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-3365.nb.aliant.net) |
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| 23:28.45 | RDF | . |
| 23:28.58 | *** join/#asterisk forrestc{hm} (~forrestc@iMach.com) [NETSPLIT VICTIM] |
| 23:29.07 | RDF | markit recived your file..in italian :) |
| 23:29.49 | *** join/#asterisk cyberlord (ghixm34@ip68-1-36-78.pn.at.cox.net) [NETSPLIT VICTIM] |
| 23:32.14 | markit | of course, I told you |
| 23:32.21 | markit | it's for testing |
| 23:32.21 | *** join/#asterisk dontmsgme (~none@adsl-64-168-164-90.dsl.lsan03.pacbell.net) |
| 23:35.42 | RDF | markit, it plays them all |
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| 23:36.03 | markit | RDF: ok, so what is the problem? |
| 23:36.05 | RDF | but not when asterisk is excepting a incoming zap call |
| 23:36.11 | DrPete | Do you guys restart apache every day? |
| 23:36.36 | RDF | Background will not play MY gsm files but plays default ones with the install :) |
| 23:36.36 | markit | RDF: are you setting some language=xx somewhere? |
| 23:36.44 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) [NETSPLIT VICTIM] |
| 23:36.56 | RDF | language=xx? |
| 23:36.56 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) [NETSPLIT VICTIM] |
| 23:37.13 | markit | RDF: xx= your country.. I've language=it, for instance |
| 23:37.27 | markit | RDF: but since you ask, I guess you don't have that set ;) |
| 23:37.44 | *** join/#asterisk ManxPower (~eric@adsl-222-11-77.msy.bellsouth.net) |
| 23:38.03 | RDF | how does that have anything to do with playing the start script Background file? |
| 23:38.03 | markit | RDF: you mean that if you substitute a standard asterisk file with one of yours (same location, same name), it's not played? |
| 23:38.06 | *** join/#asterisk kajtzu (~kajtzu@shell1.fi.basen.net) [NETSPLIT VICTIM] |
| 23:38.06 | *** join/#asterisk robin_sz (~robin@adsl.redpoint.org.uk) |
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| 23:38.17 | robin_sz | doh |
| 23:38.20 | RDF | the default included gsm files do play. my own does not. |
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| 23:38.31 | RDF | in Background () |
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| 23:38.52 | markit | RDF: I mean, if you take a default file that plays, and overwrite it with your own, does it work or not? just to be sure |
| 23:39.18 | *** join/#asterisk burton27_ (mimx@w201.ljudmila.org) |
| 23:39.22 | RDF | markit, no never done that before and why would I |
| 23:39.28 | markit | RDF: (are you running asterisk as root, or do you have the right read permission for your gsm files?) |
| 23:39.36 | *** join/#asterisk rikstahh (~rick@81-178-241-190.dsl.pipex.com) [NETSPLIT VICTIM] |
| 23:39.39 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) [NETSPLIT VICTIM] |
| 23:39.49 | RDF | running asterisk as root |
| 23:40.22 | RDF | su |
| 23:40.22 | RDF | :) |
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| 23:40.25 | *** join/#asterisk Spooch (~rath@p549A1740.dip0.t-ipconnect.de) [NETSPLIT VICTIM] |
| 23:40.25 | *** join/#asterisk r0d3nt|m (RatMan@209-58-249-211.cust.telepacific.net) |
| 23:40.26 | RDF | markit, i can hear slight hising of the file..so it is playing |
| 23:40.32 | markit | what is the exact Background() command you use? |
| 23:40.41 | *** join/#asterisk fgkjasldkjfalksd (~mike@mike.totton.ac.uk) [NETSPLIT VICTIM] |
| 23:40.44 | RDF | exten => s,2,Answer() |
| 23:40.44 | RDF | exten => s,3,NoOp(${CALLERID}) |
| 23:40.44 | RDF | exten => s,4,Background(greeting) |
| 23:40.44 | RDF | exten => t,1,Goto(s,4) |
| 23:41.25 | markit | if you use my gsm file, does it work? |
| 23:41.48 | RDF | have not tried. this is eating up my cell cost with these test calls ;) |
| 23:42.11 | RDF | but will give it a shot |
| 23:42.16 | RDF | this one time |
| 23:42.20 | RDF | brb |
| 23:42.21 | JmanA9 | i'm having a problem calling extensions using firefly |
| 23:42.26 | markit | RDF: ? if you use an extension to call a context with that, does it work? |
| 23:42.30 | JmanA9 | in the terminal, i can type dial 1 or dial 2, and the phone rings |
| 23:42.46 | JmanA9 | but in firefly, if i try to call 1 or call 2, it says no authority found |
| 23:42.53 | JmanA9 | anyone know whats wrong? |
| 23:43.30 | *** join/#asterisk ast_user (~root@interactive.mediasat.ro) |
| 23:44.35 | ast_user | hello, i am having some problems with asterisk, chan_h323, IAX and codecs... i am using GSM codec for transport between two asterisk boxes (IAX) and G726 codec for h323 endpoints... |
| 23:44.44 | ast_user | it seems that no codec translations are made |
| 23:44.53 | hermie | ast_user, running an IRC client as root is a _really_ bad idea |
| 23:45.03 | ast_user | sorry for that |
| 23:45.18 | ast_user | any ideeas concerning my problem ? |
| 23:45.19 | hermie | ast_user, don't be sorry to me, I'm not the one who'll get hacked |
| 23:46.17 | markit | anyone got the new native assisted transfer working? |
| 23:46.38 | RDF | markit, your gsm worked on Background |
| 23:47.33 | markit | RDF: so I think your recording level is too low, it happend to me also, sound boards are tricky, are you using alsa? |
| 23:50.43 | RDF | snd 30852 0 (autoclean) |
| 23:50.43 | RDF | wcfxo 8384 0 |
| 23:50.43 | RDF | zaptel 175904 0 [wcfxo] |
| 23:50.43 | RDF | 3c59x 25648 1 |
| 23:50.44 | RDF | es1370 24716 1 |
| 23:50.46 | RDF | gameport 1420 0 [es1370] |
| 23:50.47 | RDF | soundcore 3396 4 [snd es1370] |
| 23:50.50 | RDF | agpgart 43940 0 (unused) |
| 23:51.29 | RDF | channel die? |
| 23:51.45 | markit | RDF: you have some problems with your irc program also |
| 23:51.53 | *** join/#asterisk [Sim] (florian@clio.obsimref.com) [NETSPLIT VICTIM] |
| 23:51.57 | markit | you start pasting in a private window, and you end int his channel one |
| 23:51.59 | RDF | runing bitchx :) |
| 23:52.16 | markit | RDF: they will fire you for not using pastebin,so beware not to do it again |
| 23:52.29 | RDF | :) |
| 23:52.45 | RDF | brb |
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| 23:57.05 | drgalaxy | RDF hasn't got his wav file playing in ast yet I see |
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| 23:58.22 | firestrm | lots of netsplits today |
| 23:58.38 | JmanA9 | yeah |
| 23:58.47 | JmanA9 | well, i was fiddling around with my softclients, and they are perfectly registered |
| 23:59.06 | JmanA9 | but when i try to call a working extension, i've tried using 1 and *1 for extension 1, and this message appears in the console: |
| 23:59.06 | JmanA9 | Jan 22 18:58:22 NOTICE[9282]: pbx.c:1335 pbx_extension_helper: Cannot find extension context 'sip' |
| 23:59.12 | JmanA9 | anyone know what's causing that? |
| 23:59.42 | firestrm | JmanA9, your sip is registering in contect sip.. |