| 00:00.37 | drmessano | mvanbaak |
| 00:00.48 | drmessano | Paladine clearly stated they didnt want to spend the money |
| 00:00.55 | drmessano | [TK]D-Fender suggested an X100p |
| 00:01.01 | drmessano | I referred them to a link |
| 00:01.49 | mvanbaak | NOOOOOOOOOOO |
| 00:01.53 | mvanbaak | not a X100p |
| 00:02.07 | mvanbaak | way better to get an ATA for that then |
| 00:02.16 | drmessano | For $90 |
| 00:02.27 | mvanbaak | ah well |
| 00:02.42 | mvanbaak | at least I dont have to work with it |
| 00:05.43 | tzafrir_home | X100P is cheaper than most FXO ATAs. I do agree about the cheap part |
| 00:07.49 | mvanbaak | I tried a X100P onec |
| 00:07.59 | mvanbaak | s/onec/once/ |
| 00:08.09 | mvanbaak | I still regret it |
| 00:08.21 | mvanbaak | replaced it with a TDM400 wildcard |
| 00:08.41 | mvanbaak | man, my life started to have meaning again |
| 00:09.17 | alrs | x100p is tolerable if you use it with oslec |
| 00:09.28 | alrs | and is fine if you just want a timing source |
| 00:10.11 | mvanbaak | ztdummy ? |
| 00:10.27 | alrs | mvanbaak: I'd rather have an x100p than rely on ztdummy |
| 00:10.43 | alrs | mvanbaak: and an x100p you can use as a timing source for Asterisk in a Xen domu |
| 00:10.48 | Paladine | drmessano, I can't find that card in the UK, just the Wildcard version |
| 00:10.50 | mvanbaak | alrs: as long as you dont run xen and/or virtualbox that will work |
| 00:11.25 | mvanbaak | alrs: yeah, in 1 xen domu |
| 00:11.29 | mvanbaak | not in multiple |
| 00:11.51 | alrs | allegedly it will work in multiple if the ztxen patch ever surfaces |
| 00:12.00 | alrs | but yes, now it only works in 1 |
| 00:12.00 | mvanbaak | it wont |
| 00:12.11 | mvanbaak | because ztdummy works fine in xen |
| 00:12.28 | mvanbaak | and if you have multiple zaptel cards in the host you can assign one to a vm |
| 00:12.40 | alrs | yeah, I've done just that |
| 00:12.48 | mvanbaak | me too |
| 00:13.06 | mvanbaak | some E1 cards in the host |
| 00:13.30 | mvanbaak | and some domU's that claim a specific E1 card |
| 00:13.35 | mvanbaak | works great |
| 00:13.56 | *** join/#asterisk trukosh (n=trukosh@port-195-158-176-60.dynamic.qsc.de) |
| 00:14.06 | mvanbaak | combine that with a redundant E1 failover switch |
| 00:14.10 | mvanbaak | and there you go |
| 00:19.37 | tzafrir_home | mvanbaak, ztdummy works fine? what kernel? Any patches needed? |
| 00:20.13 | tzafrir_home | The ztxen patch? |
| 00:22.06 | tzafrir_home | alrs, how do you use the x100p card with xen? expose the card directlyto that specific guest? |
| 00:23.10 | trukosh | Hi, i have a minipci isdn card(HFC). I want to connect an NTBA. I think it must be A1<->B2, B1<->A2, A2<->B1, B2<->A1, as that was it before when cable-colours are right - but that seems so wrong to me ... Anybody, who really knows? |
| 00:24.55 | mvanbaak | tzafrir_home: 1.4 svn worked fine here |
| 00:25.32 | *** join/#asterisk RoyK (n=roy@91.149.19.93) |
| 00:25.34 | tzafrir_home | trukosh, what card, exactly? |
| 00:26.56 | tzafrir_home | Maybe I'm missing something, but can't you just use a flat "ethernet" cable? |
| 00:27.05 | *** join/#asterisk etfonhomey (n=chatzill@74-143-197-5.static.insightbb.com) |
| 00:29.15 | trukosh | Cameronet ISDN-MiniPCI Card based on Cologne-Chip HFC-S.. |
| 00:34.36 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:34.48 | mvanbaak | flat ethernet cable should work fine |
| 00:36.00 | *** join/#asterisk etfonhomey_ (n=chatzill@74-143-197-6.static.insightbb.com) |
| 00:36.41 | nhuisman_work | flat ? |
| 00:37.03 | nhuisman_work | what is that? |
| 00:37.31 | mvanbaak | normal cat5 cable |
| 00:37.31 | trukosh | At the moment i soldered a cabl at the card and NTBA is directly next to it. And: I have no tools to crimp. |
| 00:37.34 | mvanbaak | non-cross |
| 00:38.25 | mvanbaak | orangewhite/orange/greenwhite/blue/bluewhite/green/brownwhite/brown |
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| 00:38.47 | *** part/#asterisk hardwire (n=bip@rdbck-7085.palmer.mtaonline.net) |
| 00:38.50 | nhuisman_work | i've never heard it called a flat cable before |
| 00:38.52 | nhuisman_work | i've heard patch |
| 00:39.11 | trukosh | The Card and the are NTBA labled with A1-B2 - i just not sure about the right pairs .. |
| 00:39.18 | nhuisman_work | anyways... |
| 00:39.22 | nhuisman_work | i'm off |
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| 02:27.16 | MrTelephone | why would it be a bad thing to disable nonce checking? |
| 02:28.15 | Paladine | drmessano, managed to find one of the x100p SE cards, had to order it from hong kong though so it will take 3 or 4 days to arrive |
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| 02:31.29 | tzafrir_home | "SE"? |
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| 02:35.34 | MrTelephone | anyone here program chan_sip.c? |
| 02:36.07 | *** join/#asterisk kisu (n=kisu@211.117.76.198) |
| 02:38.10 | tzafrir_home | MrTelephone, better ask specific questions |
| 02:42.15 | MrTelephone | its a step question |
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| 03:07.45 | lmadsen | evening all |
| 03:07.57 | katsup | anyone know of a setting in asterisk that would possibly block a sip refer transfer |
| 03:08.14 | katsup | hi lmadsen |
| 03:08.27 | lmadsen | can't think of anything |
| 03:10.10 | katsup | ok, let me describe the situation. there are two * boxes. When calling through 1* to the 2*, then with the 2*, preforming a sip refer transfer, the transfer fails |
| 03:10.24 | katsup | but if I stay only inside 2*, the transfer works |
| 03:10.50 | lmadsen | refer being attended transfer, or blind? |
| 03:10.55 | katsup | blind |
| 03:11.08 | lmadsen | what version of asterisk? |
| 03:11.16 | katsup | 1.4 |
| 03:11.20 | lmadsen | 1.4..................? |
| 03:11.24 | lmadsen | latest? |
| 03:11.26 | lmadsen | svn? |
| 03:11.28 | lmadsen | 1.4.17? |
| 03:11.31 | lmadsen | 1.4.0? |
| 03:11.32 | katsup | latest is on 2* |
| 03:11.35 | katsup | not sure about 1* |
| 03:11.38 | lmadsen | that's not an answer :) |
| 03:11.43 | lmadsen | who is 1*? |
| 03:11.44 | katsup | haha |
| 03:11.57 | katsup | let me ask and I'll come back |
| 03:12.06 | lmadsen | seriously.... |
| 03:12.12 | katsup | can you think of anything offhand that would cause it though? |
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| 03:12.15 | lmadsen | I'll try and reproduce here |
| 03:12.17 | lmadsen | no |
| 03:12.29 | lmadsen | I'm pretty sure it works on my ABE boxes |
| 03:12.39 | lmadsen | I'm gonna try because I need to test for a client anyways |
| 03:12.51 | katsup | 1.4.11 |
| 03:12.54 | katsup | on 1* |
| 03:13.07 | lmadsen | how are the boxes connected? |
| 03:13.21 | katsup | standard cable connection |
| 03:13.26 | lmadsen | that's not an answer |
| 03:13.26 | katsup | *ISP cable |
| 03:13.32 | lmadsen | I mean via Asterisk |
| 03:13.42 | lmadsen | I'm assuming layers 1-4 are fine |
| 03:14.10 | katsup | can i pm? |
| 03:14.35 | lmadsen | no, just msg here |
| 03:14.44 | lmadsen | so everyone can benefit |
| 03:14.53 | katsup | they are not connected in the asterisk |
| 03:15.03 | lmadsen | then how do you expect the transfer to work? |
| 03:15.07 | katsup | 1* is just directing the call to 2* |
| 03:15.13 | lmadsen | then they are connected somehow |
| 03:15.21 | katsup | :) |
| 03:15.27 | lmadsen | how are they connected? |
| 03:15.33 | katsup | i have to look into 1* more |
| 03:15.38 | katsup | i'll get back to you |
| 03:15.40 | lmadsen | you need more information |
| 03:16.12 | lmadsen | you don't even have the basic topology understood |
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| 03:51.25 | UnixDog | ok |
| 03:51.32 | UnixDog | whats not going on here |
| 03:51.47 | UnixDog | I dont see dial plan running across the screen |
| 03:52.40 | lmadsen | it's friday night -- people wen tout |
| 03:53.18 | [hC] | UnixDog: set verbose 10 ? |
| 03:53.29 | [hC] | lmadsen: happy belated bday, not sure if you saw my msg the other day |
| 03:53.46 | [hC] | after reorganizing my office and carrying a heavy ass tv and two chairs upstairs, its time to depart to the bar myself |
| 03:53.47 | [hC] | :) |
| 03:54.00 | lmadsen | [hC]: heh, I just got back :) |
| 03:54.05 | lmadsen | sorry, I missed the message, but thanks :) |
| 03:54.16 | [hC] | you're also +3h so that makes sense |
| 03:54.42 | lmadsen | true |
| 03:54.57 | lmadsen | was considering moving to vancouver though |
| 03:56.30 | [hC] | ooo really |
| 03:56.42 | [hC] | something here for you? need a job? :) |
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| 04:17.51 | lmadsen | [hC]: I'm self employed, but I do consulting :) |
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| 04:21.45 | J4k3 | vancouver sounds nice |
| 04:21.50 | J4k3 | well, areas around it |
| 04:21.52 | J4k3 | cities = bleh |
| 04:22.28 | lmadsen | I like downtown |
| 04:22.34 | lmadsen | on the west side |
| 04:22.47 | lmadsen | my buddy lives at Granville and something (if I'm remembering that name right) |
| 04:23.18 | file | Union? |
| 04:23.48 | file | probably not, I just randomly chose that word |
| 04:23.59 | d3wayne | *_* |
| 04:24.10 | lmadsen | union is toronto :) |
| 04:24.17 | jameswf-home | such violence |
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| 04:41.23 | UnixDog | <PROTECTED> |
| 04:41.23 | UnixDog | <PROTECTED> |
| 04:41.23 | UnixDog | <PROTECTED> |
| 04:41.26 | UnixDog | having issues |
| 04:42.15 | UnixDog | [Jan 11 23:41:39] WARNING[1173]: chan_sip.c:3674 sip_write: Asked to transmit frame type 2, while native formats is 0x100 (g729)(256) read/write = 0x2 (gsm)(2)/0x2 (gsm)(2) |
| 04:42.23 | jameswf-home | ~pb |
| 04:42.24 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 04:42.31 | UnixDog | why is it not trans coding |
| 04:42.42 | UnixDog | I have g729 |
| 04:48.02 | *** part/#asterisk Blinkiz (n=niklas@h-89-233-204-231.wholesale.rp80.se) |
| 04:49.10 | UnixDog | ok I am on sip my cousin is on iax2 |
| 04:49.21 | UnixDog | and the error above is what we get |
| 04:54.43 | UnixDog | its working now |
| 04:54.49 | UnixDog | gawd |
| 04:55.34 | jameswf-home | lmao |
| 04:55.35 | denon | UnixDog: sounds like my current luck |
| 04:55.46 | denon | first time this asterisk box has been rebooted in almost exactly a year .. |
| 04:55.49 | denon | and its not coming back up |
| 04:55.56 | denon | I'm remote, of course |
| 04:56.10 | UnixDog | he had to set his cliennt to gsm only |
| 04:56.17 | UnixDog | and then it works fine |
| 04:56.33 | jameswf-home | bending glass this would be awesome stoned |
| 04:57.36 | jameswf-home | you need a robt |
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| 04:58.55 | denon | blargh |
| 04:58.58 | denon | check-forced |
| 04:59.06 | denon | leave my filesystem alone your silly linux config |
| 04:59.08 | denon | you |
| 05:02.20 | jameswf-home | I love this... I have installed a zillion systems and this is the first time using xyz card I normally use abc card and its not working wah wah |
| 05:04.03 | *** join/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
| 05:04.10 | denon | so quit buying sagnoma cards :) |
| 05:04.54 | UnixDog | sangoma cards rock |
| 05:04.57 | UnixDog | and they work |
| 05:05.06 | UnixDog | I set them up all the time |
| 05:05.21 | jameswf-home | no with sangoma cards I could see that cause they are all f**d up but any zaptel card will be the same |
| 05:06.50 | jameswf-home | stupid wanpipe crap wtf quit trying to be different |
| 05:07.19 | UnixDog | ? |
| 05:07.30 | UnixDog | what issue you having |
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| 05:08.10 | UnixDog | I find sangoma cards to be better then digium cards in performance |
| 05:08.21 | meshuga | hey anyone have the latest polycom 650 firmware? |
| 05:08.30 | UnixDog | 2.2.0 ? |
| 05:08.40 | UnixDog | its the latest polycom firmware |
| 05:08.46 | meshuga | i saaw bootrom 4.0 came out |
| 05:08.54 | UnixDog | its been out |
| 05:09.01 | UnixDog | 3 months now |
| 05:09.05 | meshuga | you got it? |
| 05:09.07 | UnixDog | your behind |
| 05:09.20 | meshuga | i havent done a polycom implementation lately :) |
| 05:09.25 | jameswf-home | speeking of obscure upgrades need to upgrade kde |
| 05:09.33 | UnixDog | some where I have to dig it up and shove it to a server |
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| 05:11.14 | meshuga | anyone else have it? i've tried 3 or 4 ftpd and the polycom doesnt seem to connect at all |
| 05:11.31 | meshuga | and tftp timing out like its natted even tho its local |
| 05:11.37 | jameswf-home | tried polycom? |
| 05:11.50 | meshuga | they only give current to resellers |
| 05:12.08 | meshuga | and the prev is oct 06 which is what i got |
| 05:12.19 | jameswf-home | maybe for a reason |
| 05:16.08 | jameswf-home | meshuga: http://yum.trixbox.org/centos/5/RPMS/firmware-polycom-2.2.0-1.noarch.rpm |
| 05:16.32 | jameswf-home | use rpm2cpio to extract |
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| 05:21.24 | UnixDog | ~ book |
| 05:21.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
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| 06:08.28 | jameswf-home | ping |
| 06:09.17 | russellb | pong? |
| 06:09.24 | jameswf-home | bong? |
| 06:09.31 | AndyGraybeal | when i talk about asterisks to my g/f ... she keeps making fun of me and saying i'm learning new ass trix.......... |
| 06:09.53 | AndyGraybeal | she says.. your learning new ass tricks with the phone now?! |
| 06:09.58 | jameswf-home | whats your old ass trix |
| 06:09.59 | AndyGraybeal | i'm like.. GArGHh shuttup |
| 06:10.29 | AndyGraybeal | :) |
| 06:11.14 | jameswf-home | the difference between a wife and girlfriend... miy wife knows not to ask... I start tech talking her eyes glaze over |
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| 06:12.43 | russellb | AndyGraybeal: yes that is the #1 most unfortunate mis-pronunciation of the word :) |
| 06:14.10 | AndyGraybeal | russellb: :) looks like there's binary version for opensuse 1.4.15 or 1.4.17.. i'm not sure which.. but either way, i think i'll be off to a good start... damn this dialup connection thouhg |
| 06:14.31 | russellb | cool |
| 06:14.39 | russellb | though you should just compile from source |
| 06:14.41 | russellb | it's not that hard :) |
| 06:15.08 | AndyGraybeal | true... some guy over in #suse says, someone already bothered to make the binaries, so why compile? |
| 06:15.24 | AndyGraybeal | i don't know what to do :) |
| 06:15.26 | russellb | depends how up to date you want to be, heh |
| 06:15.36 | russellb | if they have 1.4.17, use that |
| 06:15.43 | russellb | if they're on 15, i'd compile it ... |
| 06:15.45 | AndyGraybeal | i just want to play |
| 06:15.53 | russellb | then just install what they have |
| 06:15.55 | russellb | it doesn't matter |
| 06:15.58 | AndyGraybeal | and make pd patchies for it :) |
| 06:16.12 | russellb | well, when you get to that point, you _have_ to compile it |
| 06:16.24 | russellb | to get all that stuff, it's all development code |
| 06:16.57 | AndyGraybeal | this guy in #suse says they have 1.4.17.. but i don't think they do i think it's 1.4.15.. so i'm gonna download just the asterisk-zaptel stuff and see what happens |
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| 06:17.30 | AndyGraybeal | russellb: ah... wait a second.. i'd have to compile asterisk from scratch to make pd stuffs? |
| 06:17.47 | drmessano | Whats so bad from compiling from source? lol |
| 06:17.51 | russellb | AndyGraybeal: |
| 06:17.52 | AndyGraybeal | then i'm going to do that.. the goal really is to make pd stuffs for asterisk |
| 06:17.52 | drmessano | err |
| 06:17.54 | russellb | AndyGraybeal: correct |
| 06:17.58 | drmessano | Whats so bad about compiling from source? lol |
| 06:18.14 | AndyGraybeal | learning asterisk is secondary, but necessary, and if compiling is necessary then i will compile the latest devs... |
| 06:18.14 | russellb | AndyGraybeal: i just write the jack interfaces ... they aren't even in the main development tree, much less a release |
| 06:18.22 | AndyGraybeal | drmessano: i'm scared of the source luke! |
| 06:18.32 | russellb | AndyGraybeal: it's in one of my personal developer branches ... it's about as experimental as it gets, heh |
| 06:18.33 | drmessano | It's easy |
| 06:18.42 | drmessano | Dont use someone elses binaries |
| 06:18.52 | drmessano | Do it once, and you're set for life |
| 06:18.56 | russellb | this is the code you want to use ... $ svn co http://svn.digium.com/svn/asterisk/team/russell/jack asterisk-jack |
| 06:19.12 | russellb | $ cd asterisk-jack && ./configure && make && sudo make install && sudo make samples |
| 06:19.12 | russellb | done |
| 06:19.13 | russellb | :) |
| 06:22.00 | drmessano | All the cool kids compile from source.. you know you want to |
| 06:22.07 | drmessano | Come on.. everyone else is doing it |
| 06:22.14 | drmessano | Peer pressure rocks |
| 06:22.33 | russellb | AndyGraybeal: i haven't even finished everything with my jack code, heh |
| 06:22.44 | russellb | AndyGraybeal: one interface is working, the other still needs some debugging |
| 06:23.00 | *** join/#asterisk AndyGraybeal_ (n=andy@node216.36.251.72.1dial.com) |
| 06:24.00 | AndyGraybeal_ | okay... so i'm compiling from source |
| 06:24.02 | AndyGraybeal_ | looks liek to get the zaptel drivers working i'll need my linux sources... which i don't have |
| 06:24.04 | AndyGraybeal_ | or atleast ii don't think i have |
| 06:27.18 | [TK]D-Fender | AndyGraybeal_, This is usually where your choice for "binary" keeps circling around to bite you in the ass :) |
| 06:27.29 | drmessano | lol |
| 06:28.08 | AndyGraybeal_ | i get like anxiety attack whenver i compile anything |
| 06:28.25 | russellb | AndyGraybeal_: lol |
| 06:28.44 | [TK]D-Fender | KDE4 looks perrrrty |
| 06:28.53 | drmessano | At least you won't ever have to deal with "This is release 7 of the same version of the same app because we keep compiling it wrong, sorry" |
| 06:30.51 | UnixDog | is it official yet |
| 06:31.02 | UnixDog | it is looking nice |
| 06:31.18 | UnixDog | but I thought it was still a few months away |
| 06:32.24 | drmessano | Released today |
| 06:32.32 | drmessano | Yesterday now |
| 06:34.00 | *** join/#asterisk Abydos313 (n=abydos31@adsl-76-214-13-12.dsl.lsan03.sbcglobal.net) |
| 06:34.52 | MrTelephone | russellb how come your up so late |
| 06:34.57 | MrTelephone | nvrmd |
| 06:35.03 | AndyGraybeal_ | MrTelephone: :) |
| 06:35.27 | MrTelephone | whats happening |
| 06:36.05 | AndyGraybeal_ | waiting for all my repodata to come down so i can install the 'kernel-sources' for opensuse10.2 right now... go go dialup |
| 06:36.29 | AndyGraybeal_ | this only takes like 45 minutes |
| 06:36.45 | [TK]D-Fender | AndyGraybeal_, no broadband available where you are? |
| 06:36.46 | AndyGraybeal_ | this isn't even to download the kernel-sources! |
| 06:37.40 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:37.55 | AndyGraybeal_ | [TK]D-Fender: cellular broadband and sattilite are options, but i'm sketched out by sattilite and we don't have a south facing horizon... cellular reception is very poor here, and cable is about 3 miles down the road. |
| 06:38.38 | [TK]D-Fender | AndyGraybeal_, Can't pay to run a wire? |
| 06:39.03 | AndyGraybeal_ | run a wire? where to? |
| 06:39.31 | AndyGraybeal_ | i dont' think there is an option to run a wire somewhere |
| 06:39.57 | AndyGraybeal_ | as far as i understand it... we'd need to get a higher population density in my area for the cable people to even think about getting any closer |
| 06:40.12 | MrTelephone | why opensuse? |
| 06:40.31 | AndyGraybeal_ | MrTelephone: i'm not sure, it's the only one i could get to work with all my hardware |
| 06:40.59 | AndyGraybeal_ | MrTelephone: and that means.. that it pretty much worked at install..... i don't know how to mess around with much of the hardware |
| 06:41.03 | *** join/#asterisk mkl1525 (n=qwertz@89.246.179.53) |
| 06:41.06 | AndyGraybeal_ | i tried most distros too |
| 06:41.26 | AndyGraybeal_ | i mean... "mess around getting the software to work with the hardware" |
| 06:41.49 | MrTelephone | what kind of hardware |
| 06:42.31 | AndyGraybeal_ | i don't even remember at this point, but i think that somewhere between my firewire card and my soundcard |
| 06:42.52 | AndyGraybeal_ | opensuse was the only o/s to work with my firewire card right at the install, same with my soundcard |
| 06:43.13 | AndyGraybeal_ | i had better luck actually with the pc-card version of my soundcard than the pci version! |
| 06:43.13 | MrTelephone | i know what you mean though |
| 06:43.18 | MrTelephone | sucks fighting with drivers for shit |
| 06:43.54 | drmessano | Sat and VoIP do not mix |
| 06:43.55 | AndyGraybeal_ | i'm not very good at any of this stuff, i just want to program |
| 06:43.55 | drmessano | lol |
| 06:44.00 | drmessano | Thats another good reason |
| 06:44.04 | AndyGraybeal_ | it's pretty much the only thin that keeps me happy |
| 06:44.25 | AndyGraybeal_ | drmessano: yea, sat just seems like a really horrible fix |
| 06:44.47 | drmessano | Yep................ ................................lots of latency |
| 06:45.25 | AndyGraybeal_ | MrTelephone: i like your nick .... MrT :) |
| 06:45.48 | drmessano | Now I want some MrT cereal |
| 06:45.57 | AndyGraybeal_ | i;'m headed to sleep.... can't stand waiting for this repos to download |
| 06:46.01 | drmessano | I pity the fool who don't eat my cereal |
| 06:46.03 | AndyGraybeal_ | thanks for the help so far |
| 06:46.11 | drmessano | Chow AndyGraybeal_ |
| 06:47.38 | MrTelephone | thanks, it was a last minute nick thing |
| 06:47.55 | MrTelephone | show how handicapped i am.. |
| 06:47.57 | MrTelephone | shows |
| 06:48.19 | MrTelephone | then when I join a channel about computers I'm screwed |
| 06:48.24 | MrTelephone | i have to change it to MrComputer |
| 06:51.53 | MrTelephone | you know Im trying to fix a sip issue on a client but the only way to recreate the problem is to wait 4 days :( |
| 07:18.31 | mkl1525 | Hi, having problems with queues: our agents have snom phones and when they want to pause they press a key where 998 is called to do a PauseQueueMember. problem: when a caller is in the queue the caller is connected to the extension that does the Pause cause the phone sends a "Got SIP response 302 "Moved Temporarily" back from 10.2.4.53" - anybody know how to hold the caller in the queue? |
| 07:22.37 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583048.dsl.bell.ca) |
| 07:31.45 | [TK]D-Fender | mkl1525, Your attempt to pause the member clearly did not work, and the redirect is the phones fault for being forwarded |
| 07:32.21 | [TK]D-Fender | mkl1525, * can't make a phone respond with a redirect. |
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| 07:56.51 | mkl1525 | [TK]D-Fender thanks, found a way to block 302 on the snoms will try it |
| 07:57.59 | [TK]D-Fender | mkl1525, You should stop that person from forwarding in the first place, or better yet, find out where you failed to properly pause the member |
| 08:04.48 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582797.dsl.bell.ca) |
| 08:08.16 | mkl1525 | [TK]D-Fender put a the extension and a log to http://pastebin.com/m463189e1 maybe you could have a look - when the agent presses the key for "998" extension is ringing with the caller |
| 08:09.40 | [TK]D-Fender | mkl1525, Why is the phone FORWARDING to the logoff option? |
| 08:10.20 | [TK]D-Fender | mkl1525, and clearly ${AGENTBYCALLERID_${CALLERID(number)}} is not evaluating |
| 08:12.09 | [TK]D-Fender | mkl1525, And that is a key part of the problem... your queue CALLER is the one whose caller-id is being used to try to match against the agent. |
| 08:12.16 | [TK]D-Fender | mkl1525, You caller is not your agent |
| 08:12.42 | [TK]D-Fender | mkl1525, So forwarding to 998 = completely wrong idea. |
| 08:14.30 | mkl1525 | thanks, the problem is the phone has no forward setting set that should forward the call to the extension the agent wants to call, will have a talk with snom if they know the problem |
| 08:15.08 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:15.26 | [TK]D-Fender | mkl1525, No, the phone is clearly doing this forward. keep checking what selective & automatic means it might be using to do this. |
| 08:15.28 | mkl1525 | so atm the caller is forwarded to what the agent should get |
| 08:16.18 | [TK]D-Fender | mkl1525, ${AGENTBYCALLERID_${CALLERID(number)}} <- when the CID is your customer and not the agent this isn't going to get you anywhere |
| 08:18.24 | mkl1525 | I know the problem is I don't know why the caller comes in this extension (and not the agent), caller should stay in the queue and wait for the next agent being available |
| 08:19.02 | [TK]D-Fender | mkl1525, the problem is that phone is forwarded. It should not have been. Go fix it. |
| 08:23.20 | dacs | anyone got FWD account working with *? can you help me |
| 08:25.32 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 08:27.12 | [TK]D-Fender | dacs, Whats the problem? |
| 08:28.37 | TJNII | In an agi, if I'm using WAIT FOR DIGIT to get input from the user, will calline EXEC BACKGROUND (file) work as expected where it will not block WAIT FOR DIGIT so the user can interrupt playback by pressing a key, or will it block like EXEC PLAYBACK (file)? |
| 08:33.54 | *** join/#asterisk ToTo (n=ToTo@207.176.6.156) |
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| 08:51.20 | dacs | [TK]D-Fender: i need help setting up my account with * |
| 08:51.29 | [TK]D-Fender | dacs, go follow their guides |
| 08:51.58 | dacs | [TK]D-Fender: its talking about iax and i don't have one |
| 08:52.11 | dacs | [TK]D-Fender: maybe you have a diffrent link? |
| 08:52.22 | [TK]D-Fender | dacs, www.fwdnet.net |
| 08:52.29 | [TK]D-Fender | dacs, how about THEIR site..... |
| 09:07.17 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 09:07.33 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
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| 09:27.39 | *** join/#asterisk Al_WinKiller (i=Alex_Win@83.139.12.190) |
| 09:27.45 | badcfe | i am making an ivr in the asterisk dialplan and want the SayNumber to compose arabic numbers from my arabic numeric sound files. now, the problem is that, as i just learned, the arabic composed numbers are pronounced in reversed order that the wesern fashion. for example 61 is to be composed like "1.alaw" followed by "60.alaw" (in case of alaw codec). is there a way to set this number-composition order to be reversed for the SayNumber application |
| 09:28.10 | Al_WinKiller | hi guys, I have set up mysql db for cdr according to this http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 09:28.19 | Al_WinKiller | but it doesnt save it in mysql |
| 09:28.37 | Al_WinKiller | should I change something in cdr.conf to save cdr in mysql ? |
| 09:28.42 | Al_WinKiller | can someone help me pls ? |
| 09:32.55 | Al_WinKiller | ppl ? some help ? |
| 09:35.32 | *** join/#asterisk Abydos313 (n=abydos31@adsl-76-214-13-12.dsl.lsan03.sbcglobal.net) |
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| 10:22.02 | *** join/#asterisk RoyK (n=roy@91.149.4.251) |
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| 10:48.51 | mkl1525 | Al_WinKiller, does "show modules like mysql" list the module? |
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| 11:03.29 | tzafrir_home | Al_WinKiller, have you install the mysql modules from addons? |
| 11:04.13 | tzafrir_home | badcfe, you probably need to write some code in C |
| 11:04.35 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 11:04.50 | tzafrir_home | Patch main/say.c and maybe some other places |
| 11:10.01 | tzafrir_home | Look at the following extensive patch for Hebrew: http://bugs.digium.com/11662 |
| 11:10.39 | tzafrir_home | You don't need all of it for a sinple SayNumber |
| 11:11.15 | *** join/#asterisk ZX81_ (n=ZX81@202.20.97.211) |
| 11:30.20 | puppet | bad same in norway |
| 11:32.24 | puppet | hhow do i do it so phonecalls on two lines is ulaw/alaw and gsm on the rest?, i mean incoming |
| 11:35.34 | RoyK | puppet: er ikke du svensk? |
| 11:36.19 | RoyK | puppet: btw, afaik say.c is has norwegian fixed already |
| 11:37.03 | tzafrir_home | puppet, also try playing with say.conf, which does not require coding in C |
| 11:37.12 | tzafrir_home | But it will not work for all languages |
| 11:38.11 | RoyK | rewrite say.c to fully support linguistics generically.. |
| 11:38.16 | RoyK | heh |
| 11:40.52 | Pagautas | anybody could help me to configure quintum dx gateway? |
| 11:41.31 | tzafrir_home | RoyK, say.c (and voicemail.c, and one or two other files) have hooks for language-specific functions |
| 11:41.54 | tzafrir_home | Though there's quite a lot of copy&paste between those functions |
| 11:41.56 | Pagautas | call from pbx to sip works perfect |
| 11:42.22 | Pagautas | but i get "Got SIP response 400 "Bad Request" back from" when i call from asterisk to quintum |
| 11:43.44 | puppet | RoyK: jo men jag har ju plugat i norge o vet det ;P |
| 11:49.24 | tzafrir_home | Pagautas, it would help to pastebin relevant configurations |
| 11:49.30 | *** join/#asterisk chronos (n=fooobar@201-43-55-244.dsl.telesp.net.br) |
| 11:50.49 | chronos | someone know what is field regexten on creation of sip friend? |
| 11:56.38 | badcfe | tzafrir_home: yes .. a patch in say.c, so ... should i publish this patch then? where? it will be to handle arabic numbers. |
| 11:57.00 | badcfe | tzafrir_home: and ... how do i make a unified diff that is practical for folks to use with patch? |
| 11:57.22 | tzafrir_home | badcfe, diff -u |
| 11:57.38 | tzafrir_home | or better: svn diff, if you work vs. a copy from the svn |
| 11:57.55 | tzafrir_home | (Subversion) |
| 11:58.06 | badcfe | tzafrir_home: by the way, the patch already contains "flaws" according to arabic syntax, due to decisions taken as part of our proprietary solution ... |
| 11:58.21 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 11:58.21 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:58.28 | badcfe | tzafrir_home: diff in your diff -u is normal diff right, not svn diff .. ? |
| 11:58.46 | tzafrir_home | diff -u produces unified diff |
| 11:59.10 | tzafrir_home | see the man page for diff. This is also the default output format for 'svn diff' |
| 11:59.47 | badcfe | tzafrir_home: since the patch is not strictly following the necessity according to arabic syntax, i guess its not interresting for asterisk main development branch. however it must be released this patch according to license no? |
| 12:00.51 | tzafrir_home | If you distribute Asterisk (and not with a proprietary license - e.g: the business edition), then you must provide all your modifications to whoever you disribute Asterisk to |
| 12:01.06 | tzafrir_home | Or at least clarify to them how to get those modifications |
| 12:01.37 | badcfe | RoyK: i just learned how arabic number composing is structured. actually exactly like the old kind of norwegian except that the plurial form of 2 (two) is different for the rest (for hundredth and thousands) .. |
| 12:02.04 | tzafrir_home | So if you just install it to your clients, you are only required to provide it to them (and they may distribute them to whoever they choose) |
| 12:02.27 | tzafrir_home | That said, in the long run, maintaining patches is not fun |
| 12:02.32 | badcfe | RoyK: in arabic, for 1234 you would say "thousand" "2" "hundred2" "4" "30" |
| 12:02.50 | tzafrir_home | badcfe, there are also gender forms |
| 12:03.25 | badcfe | tzafrir_home: yes. thats part of our proprietary flaws. i ignore gender, cause we always apply it to "minutes" |
| 12:03.27 | tzafrir_home | Some parts are said with the male form, and others - with the female form. At least this is how it is with Hebrew |
| 12:03.59 | tzafrir_home | badcfe, I suggest you commit it as-is. It is a start. And hopefully someone will fix it |
| 12:04.57 | tzafrir_home | There are no differences in the syntax between various dialects of Arabic with respect to saying numbers, right? |
| 12:05.56 | tzafrir_home | What country is this for? |
| 12:05.58 | badcfe | tzafrir_home: okay, but can i just commit? i not a "committer", how do i become one, or where do i get info about that? |
| 12:06.19 | badcfe | tzafrir_home: "ar", but maybe i should use something else for arabic? |
| 12:06.47 | tzafrir_home | no. You submit a bug to bugs.digium.com . It is of severity "feature". prefix the title with [patch] |
| 12:07.10 | badcfe | tzafrir_home: thank you. ill do so. |
| 12:07.18 | tzafrir_home | See http://asterisk.org/developers/bug-guidelines |
| 12:09.30 | tzafrir_home | It would also help testing if you can provide a set of sound files |
| 12:10.32 | tzafrir_home | Even if they are low-quality ones you record on your own. |
| 12:11.20 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
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| 12:23.48 | *** join/#asterisk InHisName (n=Administ@c-71-225-221-149.hsd1.pa.comcast.net) |
| 12:24.35 | InHisName | Can I set asterisk to limit # calls and duration of calls for an extension ? |
| 12:24.44 | InHisName | per day |
| 12:24.59 | *** join/#asterisk sigmounte (n=sigmount@bai59-1-88-172-80-96.fbx.proxad.net) |
| 12:28.14 | tzafrir_home | badcfe, The "names" for languages are ISO-639-1 language codes. See e.g. http://www.loc.gov/standards/iso639-2/php/code_list.php . So yes, you should use "ar" |
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| 12:41.04 | *** join/#asterisk kareena (n=k@unaffiliated/kareena) |
| 12:41.49 | kareena | hi |
| 12:42.05 | kareena | is there is software for taxation for siemens pbx |
| 12:42.07 | kareena | like http://www.telepactechnology.com/fr/produit/eagle.asp |
| 12:43.54 | *** join/#asterisk kareena (n=k@unaffiliated/kareena) |
| 12:44.00 | kareena | hi |
| 12:44.02 | kareena | is there is software for taxation for siemens pbx |
| 12:44.03 | kareena | like http://www.telepactechnology.com/fr/produit/eagle.asp |
| 12:54.10 | *** part/#asterisk Paladine (n=paladine@ns2.scs-live.com) |
| 13:01.03 | *** join/#asterisk squigly (n=bdeluca@203.117.161.222) |
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| 13:03.31 | tzafrir_home | kareena, no need to ask twice. And also: how is this relevant to Asterisk? |
| 13:04.40 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 13:04.42 | k-man_ | hello |
| 13:05.02 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:06.56 | k-man_ | if i am behind a firewall, do i need to do anything so my sip phone can receive calls directly from my vsp? |
| 13:08.00 | mvanbaak | anyone here using openfire ? |
| 13:13.42 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:13.42 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:14.50 | k-man_ | so broadly speaking, how do i set up a dialplan to dial out on my sip line? |
| 13:15.32 | lmadsen | morning all |
| 13:15.44 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583433.dsl.bell.ca) |
| 13:15.44 | lmadsen | k-man_: there's lots of documentation to show you how to do that |
| 13:15.50 | lmadsen | ~book |
| 13:15.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 13:15.58 | k-man_ | ok |
| 13:16.01 | k-man_ | ill look at that |
| 13:16.14 | lmadsen | that's why we write documentation -- so it can be used :) |
| 13:16.16 | k-man_ | in fact, i was already looking at it |
| 13:17.23 | k-man_ | i have set up 2 extensions, and i can succesfully call between the |
| 13:17.43 | k-man_ | however if i call the test echo dial plan i made, it doesn't echo |
| 13:17.51 | k-man_ | it picks up... but no echo |
| 13:18.26 | k-man_ | thats from an spa942 and from zoiper |
| 13:19.12 | k-man_ | should i worry about it? |
| 13:28.05 | tzafrir_home | k-man_, now it is a good time to look at the trace in the CLI |
| 13:28.13 | tzafrir_home | asterisk -r |
| 13:28.17 | tzafrir_home | core set verbose 3 |
| 13:28.26 | tzafrir_home | and see what happens |
| 13:29.39 | mvanbaak | hey lmadsen |
| 13:29.57 | k-man_ | it says it ran the echo test aplication |
| 13:29.58 | lmadsen | mvanbaak: yo |
| 13:30.08 | mvanbaak | lmadsen: you ever tried openfirew ? |
| 13:30.17 | mvanbaak | openfire without the w actually ;) |
| 13:30.18 | lmadsen | never heard of it |
| 13:30.23 | mvanbaak | wildfire |
| 13:30.51 | lmadsen | no idea :) |
| 13:33.35 | *** part/#asterisk maldous (n=user@f28115.upc-f.chello.nl) |
| 13:36.14 | InHisName | Am I able to set asterisk to limit # calls and duration of calls for an extension ? (per day) |
| 13:42.13 | k-man_ | any idea what the error " No audio format found to offer." means when i try and make a sip call? |
| 13:42.28 | fugitivo | maybe wrong codecs |
| 13:43.07 | k-man_ | fugitivo, where do i specify the codecs? |
| 13:43.20 | fugitivo | sip.conf |
| 13:43.48 | k-man_ | right, i have allow=g729 |
| 13:43.53 | lmadsen | InHisName: yep, look at the GROUP() and GROUP_COUNT() functions for limiting number of calls, and look at Dial() flags for the time lengths |
| 13:43.59 | k-man_ | maybe i don't have that codec? |
| 13:44.00 | mvanbaak | do you have a license for g729 ? |
| 13:44.05 | k-man_ | <PROTECTED> |
| 13:44.11 | lmadsen | InHisName: then use func_odbc or AstDB to keep the info in a DB |
| 13:44.13 | mvanbaak | then you cant use it |
| 13:44.17 | fugitivo | k-man_: maybe, try allow=gsm and allow=ulaw |
| 13:44.24 | k-man_ | ok |
| 13:44.27 | k-man_ | ill try those |
| 13:44.45 | fugitivo | ulaw will use more bandwidth |
| 13:44.56 | InHisName | thanks for the tips, lmadsen |
| 13:45.57 | marl | quick question, can the voicemail.conf setting : externnotify : be used in individual contexts, or is it only posible to use it under [general] ? is htere a way to find out if other commands are context or general? |
| 13:48.07 | fugitivo | marl: it's a [general] setting |
| 13:48.25 | fugitivo | marl: http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf |
| 13:49.23 | marl | had been reading that page, just wasnt very sure if it could be used within the contexts, just a bit more programming then to get my program to work then :( LOL |
| 13:49.31 | marl | thanks fugitivo |
| 13:55.35 | k-man_ | so the register => line is for incoming calls? |
| 13:56.01 | lmadsen | marl: just have the script check the context it is in. |
| 13:58.30 | k-man_ | is the => syntax the same as just =? |
| 14:09.11 | *** join/#asterisk curtn (n=curtis@cl-451.trn-01.it.sixxs.net) |
| 14:10.00 | fugitivo | k-man_: no |
| 14:13.17 | Al_WinKiller | can anybody help me wit good manual "asterisk + radius + mysql" ? |
| 14:17.30 | tzafrir_home | raduis and mysql? |
| 14:17.50 | tzafrir_home | look for docs for asterisk+raduis and asterisk+mysql |
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| 14:29.25 | mvanbaak | drmessano: ping |
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| 14:44.03 | mvanbaak | hey oej |
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| 14:44.39 | oej | hello |
| 14:45.02 | mvanbaak | you know where I can get openfire(wildfire) 3.4.2 |
| 14:45.15 | mvanbaak | the latest stable wont connect to asterisk :( |
| 14:45.31 | *** join/#asterisk egypcio (n=egypcio@200.150.132.61) |
| 14:45.32 | Al_WinKiller | no, asterisk + radius, and with support of mysql ( want to save cdr in mysql ) |
| 14:48.45 | jblack | Morning. |
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| 15:12.57 | mvanbaak | hhmm, anyone can tell me how I can test MD5 auth to the manager interface ? |
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| 15:32.58 | teknoprep | anyone know how to turn on SNMP in the cisco phones.. when using SIP ? |
| 15:33.03 | teknoprep | cisco 7940 |
| 15:38.39 | jblack | There's a terminator series? COOL! |
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| 15:45.20 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 16:28.07 | Mw3 | the sarah conor chronicles? |
| 16:31.59 | kyron | ~poe |
| 16:32.00 | jbot | poe is probably Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt. Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices: http://en.wikipedia.org/wiki/Power_over_Ethernet |
| 16:33.16 | kyron | yes, I know that... I want to know about the caveats and known issues of PoE and SIP phones... you lack I.N.T.E.L.L.I.G.E.N.C.E. jbot :P |
| 16:33.56 | [TK]D-Fender | ~jbot |
| 16:33.56 | jbot | i guess jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 16:34.05 | kyron | hihihihi |
| 16:34.17 | kyron | LOL |
| 16:34.35 | [TK]D-Fender | kyron, Now "caveats"? Only 1 : if your switch goes down, your phone isn't powered. |
| 16:34.41 | [TK]D-Fender | kyron, PoE is DESIRABLE |
| 16:34.42 | UnixDog | http://www.metaphorivr.com/index.aspx |
| 16:35.21 | kyron | [TK]D-Fender, I can live with that, if my switch goes down, I have other issues ;) |
| 16:35.52 | kyron | The way I fugre it, a PoE-able switch is less expensive than buying all the powersupplies for the SIP phones ;) |
| 16:36.34 | kyron | also quite less of a hassle for the end user (less wiring, no transformer to take up a complete powerbar, etc.) |
| 16:37.16 | ManxPower | many poe switches can't power all ports at once. |
| 16:37.23 | tzafrir_home | kyron, but you can't plug the phone to a different switch |
| 16:37.36 | [TK]D-Fender | kyron, depends on the phone, and how many, but possibly |
| 16:38.00 | kyron | ahhhhhh... now _THAT_ is what I call caveats! |
| 16:38.23 | kyron | I am thinking around but not more than 20 phones |
| 16:38.39 | ManxPower | know how much PoE power the device takes, know how much power the PoE switch can provide. |
| 16:38.40 | [TK]D-Fender | kyron, Well clearly there is an equipment cost... that isn't a caveat, thats a fact |
| 16:39.52 | [TK]D-Fender | kyron, 20 x IP320 PS ($24ea) = $480. D-Link 1228P 24 port PoE Switch = 364$ |
| 16:41.17 | *** join/#asterisk dimas (n=ds@vbc.elcom.ru) |
| 16:44.35 | dacs | good morning guys |
| 16:44.36 | RoyK | what's the new stuff in trunk, btw? |
| 16:45.33 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 16:45.43 | kyron | [TK]D-Fender, yeah, that clearly gives the switch an advantage over buying 20 PS units |
| 16:47.31 | kyron | [TK]D-Fender, btw, where did you get that price for a 24 port D-Link 1228P (do you know it to be reliable?) |
| 16:47.32 | [TK]D-Fender | kyron, Everything "depends". |
| 16:47.48 | [TK]D-Fender | kyron, http://www.antonline.com/antonline.php?op=inventory&st=DES-1228P |
| 16:47.59 | [TK]D-Fender | kyron, Yes, they are a big place... |
| 16:48.31 | [TK]D-Fender | kyron, And I have customers using them happily, and I myself use its predecessor the DES-1536 |
| 16:49.22 | kyron | let's see if they ship to Canada ;) |
| 16:50.16 | [TK]D-Fender | kyron, This is all just commodity gear. Price it out wherever you like. |
| 16:50.31 | [TK]D-Fender | kyron, NCIX should have decent pricing, etc..... |
| 16:51.22 | kyron | dunnow ncix ...but will look around. ie, a local dealer seems to have nothing under 600$ for 24 ports : http://voipgizmos.com/shopdisplayproducts.asp?id=47&cat=Power+Over+Ethernet |
| 16:51.33 | anonymouz666 | [TK]D-Fender: what you would use it for a big analogic solution? TDM2400P or an Astribank? |
| 16:52.00 | [TK]D-Fender | anonymouz666 : Neither. Mediatrix 1124 or AudioCodes MP-124 |
| 16:52.36 | [TK]D-Fender | anonymouz666, depending I would also include the Linksys SPA-8000 |
| 16:53.42 | *** join/#asterisk _Raptor_ (i=sirasenn@faui08r.informatik.uni-erlangen.de) |
| 16:56.20 | anonymouz666 | [TK]D-Fender: I don't know about Mediatrix or Audiocodes, but I'll search about it. |
| 16:56.42 | _Raptor_ | hello, how can i create an extension of the form _+491234? as it seems asterisk does not like the leading + |
| 16:59.35 | _Raptor_ | ok, it works |
| 17:07.52 | *** join/#asterisk anthm (n=anthm@75-135-78-143.dhcp.mdsn.wi.charter.com) |
| 17:07.52 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:08.36 | tzafrir_home | Hmmm... what should the '+' mean? |
| 17:08.44 | tzafrir_home | Who interperts it? |
| 17:12.28 | mvanbaak | kyron: grab a cisco 3750 with PoE ;) |
| 17:13.20 | kyron | mvanbaak, got 1M$ to spare? |
| 17:13.22 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-198-40.dsl.irvnca.pacbell.net) |
| 17:14.23 | kyron | 170W seems like not much power for 24 ports... (DES-1228P) |
| 17:17.12 | JunK-Y | [TK]D-Fender: mediatrix rocks :) |
| 17:19.38 | kyron | JunK-Y, no bias huh ;) |
| 17:19.44 | kyron | my 1104 are noisy as HELL |
| 17:20.19 | kyron | I am thinking of taking them apart and bolting a 120mm fan on top of em (not going to stack em anyways ;) ) |
| 17:20.51 | jblack | dtmf tones seem to have a habit of getting "stuck" |
| 17:21.14 | JunK-Y | kyron: i prefer a noisy shit which works correctly (and my computers room doest complain at all about it!!!) then a non-functionnal gateway. |
| 17:22.20 | kyron | JunK-Y, yes, obviously, but that means I _have_ a server room and can afford to run the phone wires all the way back to that room...which is not that common for small companies ;) |
| 17:27.38 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 17:27.43 | jblack | Ok, so I have a weird problem. |
| 17:27.59 | kyron | jblack, I see that |
| 17:28.04 | Nugget | if you can't afford a cable pull you've got bigger problems than your phone system. |
| 17:28.27 | jblack | Kitchen calls office. Then, kitchen does #72 (parked call). The transferring seems to work, except what I hear on office, rather than music on hold, is a solid dtmf tone. either # or 2, I can't tell which |
| 17:28.42 | jblack | The dialplan log looks about right though, http://pastebin.com/m699389f4 |
| 17:29.06 | jblack | (that's not the dialplan, but a capture of the log during the call) |
| 17:31.05 | kyron | Nugget, problem is we need to demonstrate the usability of the system before we get a go... no one wants "change" without proof that the alternative is better (this is especially true for small businesses where every penny counts...) |
| 17:31.33 | kyron | _and_ employees are "prima donnas" with an attitude |
| 17:31.52 | jblack | kyron: Sensible on the employees part. Better the devil you know, then the one you don't. |
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| 17:32.54 | kyron | :P |
| 17:33.35 | jblack | Do you have any constantly hungry employees, and are you willing to take a personal risk? |
| 17:33.57 | jblack | If so, offer a pilot project for a handful of people. Promise them a steak dinner if they decide to go back to the old way after 2 weeks. |
| 17:34.48 | kyron | the problem is that there is a big demand such employees...even if they are incompetent and don't know how to use their tools efficiently. So a change to their working environment, especially a critical part such as the phone system, is quite a feat to perform.. |
| 17:35.12 | kyron | I can already hear them complain they don't like the system attendant's voice... |
| 17:36.20 | kyron | jblack, but who will absorb the material cost of "going back"...and one cannot transfer a phone # back and forth within weeks...hell, transferring a phone number takes from 30 to 45 days and you don't know _when_ it will happen... |
| 17:37.12 | kyron | can't do that for a business... but I am thinking of a quick way to "test" the installation.... get a DID and simply transfer the present phone # to test the VoIP solution. If it fails, cancel the transfer. |
| 17:38.15 | kyron | Think that would be the most transparent and seamless approach without major interruption and a good fallback plan in case the test fails (but that doesn't cover the material costs...) |
| 17:38.38 | jblack | Shrug. When I have pushed difficult things through in the past, I did it by taking a personal risk |
| 17:40.35 | kyron | ok, untold truth, the customer is my father and I can't afford to crash his business on a whim ;) But I strongly believe they would benefit from the move (+voicemail,+private extensions,+much more lines,+free inter-office calls,etc.) |
| 17:41.09 | jblack | If you're that sure, then take a personal risk. |
| 17:41.21 | jblack | If you're not, then don't do it. |
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| 17:43.35 | kyron | myeah, always boils down to that. I have to make sure I plan all of this right and buy the right equipment. It would be easy if I had done 1,2,+++ implementations with hardware (experience), but this would be the first time. So 1001 questions tend to come to mind (proper bandwidth+latency, good hardware, redundancy, UPS requirements, etc) |
| 17:44.04 | kyron | VoIP is definately a multidisciplinary field ;) |
| 17:47.35 | *** join/#asterisk Vulkano (n=Vulkano@rgnb-4db1d374.pool.einsundeins.de) |
| 17:47.50 | Vulkano | Servus |
| 17:48.06 | Vulkano | Hat jemand Erfahrung mit 1und1 und Asterisk? |
| 17:48.24 | *** join/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 17:48.54 | *** join/#asterisk AndyGraybeal (n=andy@node124.36.251.72.1dial.com) |
| 17:49.03 | Vulkano | ausgehende Gespräche gehen bei mir nur eingehend funktioniert nix |
| 17:52.06 | *** join/#asterisk m160858 (n=m160858@190.187.16.14) |
| 17:52.19 | m160858 | hi guys |
| 17:52.37 | m160858 | i'm testing queues on asterisk 1.4.17 |
| 17:52.57 | m160858 | but i've a little problem |
| 17:53.38 | m160858 | my agent has configure with password in the agents.conf |
| 17:54.06 | Maxous | Afternoon everyone. |
| 17:54.26 | m160858 | but when i try to login, show me up password incorrect |
| 17:54.41 | m160858 | any idea? |
| 17:54.42 | *** part/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 17:55.24 | TJNII | In an agi, if I'm using WAIT FOR DIGIT to get input from the user, will calline EXEC BACKGROUND (file) work as expected where it will not block WAIT FOR DIGIT so the user can interrupt playback by pressing a key, or will it block like EXEC PLAYBACK (file)? |
| 17:55.52 | m160858 | please? |
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| 17:56.35 | *** join/#asterisk Maxous (n=Maxous@76.97.3.24) |
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| 17:58.19 | *** part/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 17:59.37 | *** join/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 18:01.15 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 18:01.37 | dacs | quick question: my x-lite can call my ATA exten, but when i try to call my x-lite from my ATA extention i get fast busy |
| 18:01.44 | tzafrir_home | m160858, what do you mean by " show me up password incorrect"? |
| 18:02.24 | m160858 | Parsing '/etc/asterisk/sip_notify.conf': Found |
| 18:02.24 | m160858 | <PROTECTED> |
| 18:02.24 | m160858 | <PROTECTED> |
| 18:02.24 | m160858 | <PROTECTED> |
| 18:02.24 | m160858 | [Jan 12 06:53:30] WARNING[9773]: file.c:643 ast_readaudio_callback: Failed to write frame |
| 18:02.26 | m160858 | <PROTECTED> |
| 18:02.30 | tzafrir_home | dacs, can you call other extensions (e.g: echo test) from there? |
| 18:02.32 | tzafrir_home | ~pb |
| 18:02.32 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:03.06 | tzafrir_home | m160858, are you sure the password was actually sent? |
| 18:03.10 | dacs | tzafrir_home: i don't know how set up echo test |
| 18:04.06 | tzafrir_home | dacs, please pastebin a CLI trace of calling from the ATA to the softphone |
| 18:04.12 | *** join/#asterisk curtn (n=curtis@cl-451.trn-01.it.sixxs.net) |
| 18:04.31 | jblack | hmmm. maybe 1.4.10 has a bug with that |
| 18:04.36 | jblack | the dtmf sticking |
| 18:05.14 | tzafrir_home | m160858, can the same SIP phone use any type of IVR? e.g: the demo, or voicemail? |
| 18:05.32 | tzafrir_home | Maybe the problem is that DTMF digits are not properly detected |
| 18:06.10 | jblack | Yeah |
| 18:06.25 | dacs | tzafrir_home: http://pastebin.ca/852245 |
| 18:06.30 | jblack | but something is doing the equivilant of holding down a button |
| 18:06.59 | curtn | how can I change the SIP user on a cisco gateway ? |
| 18:07.19 | jblack | 1.4.15 is in hardy. I'm going to move up to that |
| 18:07.19 | curtn | (all my incoming calls fall in the defaut context) |
| 18:08.17 | curtn | "Found no matching peer or user for ..." |
| 18:08.22 | tzafrir_home | dacs, could you please disable sip debug, but set verbosity to a hi enough value? |
| 18:08.26 | tzafrir_home | core set verbose 3 |
| 18:08.34 | tzafrir_home | sip set debug off |
| 18:08.51 | m160858 | maybe, but the asterisk accept the agent number ... neither should do it ... right? |
| 18:09.37 | m160858 | i'm using asterisk-gui .... bad choice, i think |
| 18:10.15 | TJNII | ~topic |
| 18:10.47 | tzafrir_home | m160858, if you can ask asterisk questions, you can ask them here |
| 18:14.08 | m160858 | i was that |
| 18:14.13 | m160858 | maybe, but the asterisk accept the agent number ... neither should do it ... right? |
| 18:14.32 | tzafrir_home | dacs, you get an error for "extension not found". So you have missing parts in the dialplan to fill |
| 18:15.07 | dacs | tzafrir_home: but when i call from exten 500 it ring exten 600 |
| 18:15.34 | tzafrir_home | You don't call from an "extension". You call from a channel |
| 18:15.46 | tzafrir_home | The channel has an initial context set to it |
| 18:16.06 | tzafrir_home | In that case, the context can be seen in the settings of the sip user |
| 18:16.24 | tzafrir_home | try: sip show users |
| 18:16.32 | dacs | tzafrir_home: i am confused |
| 18:16.33 | tzafrir_home | note: this also show passwords |
| 18:16.55 | dacs | it shows both of them |
| 18:17.28 | tzafrir_home | for the user of the ATA: to what context does it go? |
| 18:18.19 | dacs | Username Secret Accountcode Def.Context ACL NAT |
| 18:18.21 | dacs | phone1 888 internal No RFC3581 |
| 18:18.32 | dacs | sorry |
| 18:18.46 | tzafrir_home | so the context is "internal" |
| 18:18.54 | dacs | no no |
| 18:19.01 | tzafrir_home | What is in that context? |
| 18:19.06 | tzafrir_home | dialplan show internal |
| 18:20.25 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 18:21.00 | jblack | Heh. Bug fix logs always make bugs look disastrous. |
| 18:21.23 | dacs | tzafrir_home: in pm |
| 18:22.07 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 18:23.07 | dacs | brb |
| 18:24.35 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 18:31.50 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 18:32.20 | puppet | how do i do it so phonecalls on two lines is ulaw/alaw and gsm on the rest?, i mean incoming |
| 18:32.51 | mvanbaak | puppet: what tech ? |
| 18:32.56 | tzafrir_home | puppet, that's what "allow" and "disallow" are for |
| 18:33.24 | puppet | tzafrir_home: well you cant use allow/dissalow on one registry can you? |
| 18:33.39 | puppet | the allow dissalow is on top of the file, and affects all registrys dont it? |
| 18:33.45 | ZaVoid | puppet: dissallow all |
| 18:33.50 | ZaVoid | then allow stuff you want |
| 18:34.04 | puppet | ewll yeah, but i want alaw/ulaw for TWO registry rest i want gsm |
| 18:34.20 | ZaVoid | for each device? |
| 18:34.21 | tzafrir_home | so use specific peers / users |
| 18:34.23 | ZaVoid | specfici? |
| 18:34.24 | jblack | amazing. |
| 18:34.36 | ZaVoid | thanks jblack i am pretty damn amazing ;) |
| 18:34.36 | jblack | No problem any more. |
| 18:34.42 | jblack | Yeah, you're amazing |
| 18:35.02 | puppet | ZaVoid: two numbers are for faxm, and they need incoming ulaw/alaw |
| 18:35.10 | puppet | ZaVoid: rest is voice and can use gsm for incoming |
| 18:35.16 | ZaVoid | different sip.conf entries right? |
| 18:35.37 | puppet | can i have a register line, then new allow/disallow for next registry? |
| 18:35.40 | puppet | does that work? |
| 18:35.51 | *** join/#asterisk FlatFoot (n=chatzill@80.88.218.4) |
| 18:36.56 | TJNII | puppet: You can put allow/disallow in individual channel entries. It doesn't have to be global. |
| 18:37.13 | puppet | TJNII: chan_sip im talking about right now |
| 18:37.15 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 18:37.15 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:37.39 | puppet | TJNII: and two sip registry should be alaw/ulaw rest sips gsm, incoming im talking about now, outgoing is no problem since i set that in sip_buddies |
| 18:39.22 | TJNII | puppet: Okay, If I understand corectly you have one or two devices that require u/alaw and the rest gsm. I assume you have a [entry] in your sip.conf for them? |
| 18:39.38 | *** part/#asterisk dimas (n=ds@vbc.elcom.ru) |
| 18:39.44 | puppet | TJNII: yeah, but it didnt go after that allow line, somehow it picked gsm anyway |
| 18:40.32 | puppet | trying agin now |
| 18:40.51 | ZaVoid | show me your sip.conf |
| 18:40.56 | TJNII | Somebody else help me with the terminology here, what do you guys refer to the [entry]\nconfig\nconfig entries in a config file as? I forget |
| 18:41.03 | puppet | zav: 83.140.41.46 08500023XX 14503d25339 00103/00000 gsm No Tx: ACK |
| 18:41.05 | *** part/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 18:41.24 | puppet | ZaVoid: but in the allow line in the sip_buddies it says ulaw;alaw;ilbc |
| 18:41.26 | *** join/#asterisk gerhard7 (n=gerhard@195-241-250-146.dial.ip.tiscali.nl) |
| 18:41.33 | puppet | ZaVoid: and disallow all |
| 18:41.43 | tzafrir_home | TJNII, context, or section, or whatever |
| 18:41.52 | TJNII | Thanks |
| 18:42.05 | ZaVoid | puppet show it to me please |
| 18:42.30 | TJNII | puppet: You put allow alaw and disallow gsm in the sip.conf context of the device and not under the register statement, correct? |
| 18:42.59 | puppet | ZaVoid: its in realtime most of it but ill copypate groundconfig |
| 18:43.04 | puppet | TJNII: ill pastebin groudnconfig |
| 18:43.13 | jameswf-home | ~pb |
| 18:43.14 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:43.35 | jameswf-home | that should say or penutbutter |
| 18:44.20 | *** join/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 18:44.27 | dacs | tzafrir_home: how can i fix it |
| 18:45.06 | jameswf-home | pixie dust |
| 18:45.06 | tzafrir_home | set context=internal for the ATA as well |
| 18:45.14 | puppet | ZaVoid, TJNII http://pastebin.com/d3e36fd34 |
| 18:45.24 | puppet | the parts that ahve to do with codecs and the order i do registr |
| 18:45.40 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 18:45.46 | TJNII | puppet: Which ones need a/ulaw |
| 18:46.10 | jblack | I think that should say "3 or more". |
| 18:46.29 | TJNII | puppet: Is that your entire sip.conf? |
| 18:46.37 | puppet | TJNII: not entire, but rest is in realtime |
| 18:47.00 | puppet | TJNII: the middle at illu needs ulaw/alaw and the last at clearminds |
| 18:47.27 | TJNII | Well, the problem I see is that you have no contexts for those devices where you tell * they have to use alaw/ulaw |
| 18:47.37 | ZaVoid | the devices are in realtime? |
| 18:47.45 | puppet | yeah |
| 18:48.11 | ZaVoid | so in your db... what do you have under the allow and disallow tables |
| 18:48.13 | ZaVoid | fields |
| 18:48.18 | ZaVoid | disallow should be all |
| 18:48.26 | ZaVoid | allow should be alaw;ulaw |
| 18:48.33 | puppet | ZaVoid: but in the allow line in the sip_buddies it says ulaw;alaw;ilbc ZaVoid: and disallow all |
| 18:48.34 | ZaVoid | at least that way for pgsql format |
| 18:48.44 | ZaVoid | and still picking gsm? |
| 18:48.53 | ZaVoid | you sure its hitting realtime entry? |
| 18:49.28 | puppet | havent checked the sql query, just to extracheck the name in the realtime should it be the name of the context? or can it be anything? |
| 18:49.34 | jameswf-home | tzafrir_home: are you representing xorcom at the IT confrence |
| 18:49.42 | puppet | could be that in that case caues i have named them a bit diff, with comp name too |
| 18:50.07 | dacs | tzafrir_home: i fix it now , thank you |
| 18:50.35 | tzafrir_home | jameswf-home, what conference? |
| 18:50.45 | *** part/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 18:51.05 | jameswf-home | it expo in Florida thought I saw your companys name on the list |
| 18:51.20 | jameswf-home | if = internet telephony |
| 18:51.58 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 18:52.52 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 18:55.12 | dacs | tzafrir_home: can you help me test this |
| 18:56.21 | tzafrir_home | jameswf-home, no. I'm not going to be there |
| 18:56.41 | dacs | ok now after you help me fix my problem , another one showed up, i config my FWD DID, and i can call my FWD DID from my x-lite (works fine) but when i try to call it from my cellular its not ringing |
| 18:56.42 | drmessano | dacs: have you actually made a call yet with Asterisk? |
| 18:56.58 | dacs | drmessano: trying too :) |
| 18:57.02 | drmessano | lol |
| 18:57.03 | drmessano | omg |
| 18:57.10 | drmessano | It's been 3 weeks |
| 18:57.13 | dacs | drmessano: am able to call internaly |
| 18:57.32 | dacs | drmessano: i know man but trying to learn |
| 18:57.43 | drmessano | Sure you wouldnt rather use Trixbox? lol |
| 18:57.51 | dacs | drmessano: nope |
| 18:58.17 | drmessano | I'm just giving you a hard time.. keep on |
| 18:58.35 | *** join/#asterisk atomicd (n=atomicd@adsl-69-226-17-248.dsl.irvnca.pacbell.net) |
| 18:58.38 | dacs | drmessano: i am studying for CCNA , plus work, plus family plus reading * book |
| 18:58.43 | puppet | ehm i think something is wrong with the did, cause when i call the 70 number, and i check the channel it says it isincomign on the 75 |
| 18:58.44 | dacs | you know how it goes |
| 18:58.52 | *** part/#asterisk atomicd (n=atomicd@adsl-69-226-17-248.dsl.irvnca.pacbell.net) |
| 18:58.53 | drmessano | yeah |
| 18:59.06 | drmessano | Im getting ready to work on my CCNA |
| 18:59.15 | dacs | drmessano: REALLY |
| 18:59.30 | drmessano | Yep |
| 18:59.32 | jameswf-home | creatively certified nurses assistant? |
| 18:59.53 | drmessano | Certified, Can't Network Anything |
| 19:00.35 | drmessano | I'm one of those idiots who spends years doing IT and doesn't get certs to back up what he knows |
| 19:00.45 | drmessano | So I am in the process of shelling out cash so I can get the paper |
| 19:01.16 | dacs | drmessano: makes two of us |
| 19:01.19 | jameswf-home | yeah gotta choose wisely on certs.. when looking at resumes I avoid MCP and MCSE like the plague... |
| 19:01.43 | jameswf-home | I see that and Im like idiots |
| 19:01.46 | jameswf-home | lol |
| 19:01.57 | dacs | jameswf-home: MCSE is nothing Cisco is the way |
| 19:02.07 | drmessano | I'm staying away from the MS Stuff.. I can partially accept needing an A+, Network+, CCNA for a job.. But I think employers realize an MCSE is like having a bicycle license |
| 19:02.22 | Qwell | A+? |
| 19:02.25 | Qwell | ... |
| 19:02.38 | jameswf-home | Cisco = overpriced linksys or is linksys an properly priced cisco i forget |
| 19:02.40 | drmessano | Some jobs won't even consider you without it.. |
| 19:02.45 | drmessano | Which is insane |
| 19:02.53 | drmessano | Q: What is a motherboard |
| 19:02.58 | drmessano | Q: What is a CPU |
| 19:03.03 | jameswf-home | I dont like A+ either a monkey can plug in wires |
| 19:03.05 | drmessano | Q: Why does the mouse make a ding |
| 19:03.15 | dacs | drmessano: you know that most of telecom companys are now merging to Cisco equipment |
| 19:03.27 | dacs | for IPBH, and IPSHO |
| 19:03.58 | jameswf-home | we are making good money on the telcos upgrades so they can comply with the patriot act :) |
| 19:04.11 | dacs | ^^IPBH=IP Back Hall , IPSHO= IP Soft HandOff |
| 19:04.15 | drmessano | Right now the big push is Networking skills.. Network hardware is too expensive and too essential to have the same dumbasses adminning it that didnt know what they were doing 10 years ago either |
| 19:04.52 | jameswf-home | I would consider netwrk+ or security+ but never A_ |
| 19:05.05 | jameswf-home | s/a_/a+/ |
| 19:05.16 | dacs | drmessano: when i saw how much my company paid for Cisco router they bought resentaly!!!1 i was shocked |
| 19:05.32 | drmessano | Cisco switches are no different |
| 19:05.35 | drmessano | They work an all |
| 19:05.36 | dacs | and i said to my self i must get Cisco Certified |
| 19:05.38 | drmessano | They work and all |
| 19:05.41 | jameswf-home | what can a sisco do that linux cant.... |
| 19:05.50 | dacs | drmessano: $$$$$$ for routers |
| 19:06.10 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 19:06.45 | drmessano | Hell, we just updated some circuits to bonded T1s, and had to shell out for a 2800 because our 2600 couldnt handle the WIC capacity |
| 19:06.57 | drmessano | NEW Cisco just to add a couple lines |
| 19:06.59 | jameswf-home | buying a cisco or any other such product is like using a propriatary pbx... no real need to spend all that money but if your rich |
| 19:07.06 | drmessano | Yep |
| 19:07.10 | drmessano | I completely agree |
| 19:07.27 | *** join/#asterisk Maliuta (n=nikolai@59.167.214.92) |
| 19:08.16 | drmessano | Funny thing is |
| 19:08.24 | dacs | drmessano: we just added 4 7613 Cisco routers |
| 19:08.58 | drmessano | If I tell corporate I need switches, they JUMP on us getting them.. moreso than any other piece of equipment.. |
| 19:09.07 | drmessano | Obviosuly they can argue with you needing more drops |
| 19:09.36 | drmessano | But I needed new switches and had them in 2 days.. done.. handled.. Cisco 2960s |
| 19:09.42 | dacs | drmessano: they will rather spend money in Marketing , than IT |
| 19:09.55 | drmessano | Other than setting the IP, don't do shit-else with em |
| 19:09.56 | dacs | drmessano: they will say you are the IT guy make it work |
| 19:10.09 | denon | drmessano: they're probably scared you go to CompUSSR and buy some linksys switches if they dont |
| 19:10.23 | drmessano | Ive actually done that |
| 19:10.26 | drmessano | But got out of it |
| 19:10.30 | dacs | lol |
| 19:10.45 | dacs | tzafrir_home: are you here |
| 19:10.47 | drmessano | Realized they loved Cisco so much that I dont even try to save them money anymore |
| 19:10.50 | jameswf-home | thats the issue... why trixbox is so popular not only is it cheap but they can fire the IT guy and let the secretary manage it.... bastards |
| 19:11.01 | tzafrir_home | dacs, yes |
| 19:11.43 | dacs | tzafrir_home: so, i was saying that i can call my FWD DID from x-lite but not from external phone, like my cell |
| 19:12.28 | denon | drmessano: cisco makes a nice switch .. a network full of linksys will get very quirky very fast |
| 19:12.48 | drmessano | Depends on the size |
| 19:13.00 | tzafrir_home | "an external phone"? Through what device? or provider? |
| 19:13.04 | denon | well, over a few dozen users |
| 19:13.14 | drmessano | I wouldn't go over 72 ports |
| 19:13.22 | denon | linksys gig-e isn't exactly gig-e :) |
| 19:13.26 | drmessano | Beyond that you need a diff network segment anyway |
| 19:14.00 | jameswf-home | nothing makes me more giddy than a giant colission domain |
| 19:14.05 | drmessano | lol |
| 19:14.40 | drmessano | My former assistant built the network at one of my locations, right... took me almost 2 years to go back and rebuild it.... |
| 19:14.50 | drmessano | This is how we basically had it set up.. |
| 19:15.24 | jameswf-home | how many hubs can you uplink :) |
| 19:15.32 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 19:15.48 | drmessano | Users ----> 3 24 port switches (mixed) <---- Single Cat 5 ----> Switch <--- All the servers in a single rack |
| 19:16.08 | drmessano | bottleneck anyone? |
| 19:16.40 | jameswf-home | ooooh 10 MB hubs like sfo users that would be sweet |
| 19:16.56 | jameswf-home | s/sfo/250/ |
| 19:17.16 | jameswf-home | wow brain and fingers not friends |
| 19:17.27 | drmessano | oh |
| 19:17.31 | drmessano | ALmost forgot the best part |
| 19:17.46 | drmessano | switches were daisychained, not starred |
| 19:17.52 | *** join/#asterisk kiscokid (n=kiscokid@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 19:18.10 | jameswf-home | I think topology is overrated |
| 19:18.37 | jameswf-home | save money wire cat 3 |
| 19:18.54 | jameswf-home | everyone on hpna |
| 19:19.07 | dacs | tzafrir_home: my cellphone |
| 19:19.21 | drmessano | I had users on the second user switch running reports off on of the servers.. Took 32 minutes.. When I moved them to the first user switch, one hop up.. Dropped to 10 minutes |
| 19:19.47 | drmessano | I didnt know we had as large a problem until then |
| 19:20.05 | dacs | tzafrir_home: i have IPKALL which assign a phone number to DID |
| 19:20.21 | drmessano | Want a cool IPKALL trick? |
| 19:20.34 | dacs | drmessano: shoot |
| 19:20.44 | tzafrir_home | dacs, so what happens when a call comes in? To what context does it go? |
| 19:20.51 | drmessano | set the server to your server name, set the extension to the DID itself |
| 19:21.05 | drmessano | then set an inbound route for that DID |
| 19:21.08 | J4k3 | ethernet is the product of people who did way too much lsd. |
| 19:21.12 | tzafrir_home | drmessano, first let's teach him to walk |
| 19:21.14 | dacs | drmessano: hold on please |
| 19:21.27 | dacs | tzafrir_home: thats right |
| 19:21.33 | drmessano | Well, thats the best way to route inbound calls for IPKALL |
| 19:22.11 | tzafrir_home | drmessano, again, in 'sip show users', to what context do calls from FWD go? |
| 19:22.26 | *** part/#asterisk kiscokid (n=kiscokid@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 19:22.36 | tzafrir_home | and don't paste the full line |
| 19:22.44 | tzafrir_home | It has your FWD secret... |
| 19:23.11 | dacs | tzafrir_home: http://pastebin.ca/852320 |
| 19:23.35 | drmessano | oh god |
| 19:23.39 | drmessano | Hang on |
| 19:23.47 | drmessano | Why dont you test with FWD SIP first? |
| 19:24.02 | drmessano | IAX is worse than flaky with them.. you dont have a working variable there |
| 19:24.42 | Mw3 | hm, my telco has put some annoying voice mail on my line (isdn2). they say i should deactivate it by dialing #62#. can i do that with asterisk? i think this is some kind of vertical service code, which cant be dialled with Dial() |
| 19:24.48 | tzafrir_home | IAX works much nicer over NAT and such |
| 19:24.59 | drmessano | Youre missing what I am telling you |
| 19:25.17 | drmessano | FWD IAX doesn't work half the time.. LOTS of user issues |
| 19:25.30 | dacs | drmessano: you are spinning my head |
| 19:25.57 | J4k3 | Mw3: tell the telco to remove it. if they refuse, offer to cancel. |
| 19:26.01 | dacs | ^^ thats what you are doing |
| 19:26.07 | tzafrir_home | well, callwithus.com are nice for starting up, and have IAX as well |
| 19:26.14 | drmessano | If he's NEW, get it working with SIP |
| 19:26.56 | drmessano | No, dacs I am not |
| 19:26.56 | J4k3 | Mw3: thats the only way I could get our local ILEC to get the pay-per-use 3 way and their ghetto-assed voicemail off my lines. |
| 19:26.57 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:26.57 | drmessano | I am trying to tell you FWD IAX is not a 100% working system you can test a newb again |
| 19:26.57 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 19:26.57 | drmessano | You wont ever know if youre wrong or if its broken |
| 19:26.57 | Mw3 | J4k3: well, they are willing to remove it if i cant, but it takes a week (because they need to assign a task to some techical guy) |
| 19:26.57 | drmessano | Try SIP |
| 19:26.58 | dmz | anyone use #asterisk-users anymore? :) |
| 19:27.05 | UnixDog | FWX IAX has issues |
| 19:27.09 | Mw3 | J4k3: so it would be better if i could remove it somehow with this damned code |
| 19:27.10 | UnixDog | use SIP |
| 19:27.12 | drmessano | Thank you, UnixDog |
| 19:27.13 | drmessano | Jesus |
| 19:27.24 | J4k3 | Mw3: actually, deactivating it will likely activate billing. |
| 19:27.50 | J4k3 | Mw3: tell them they obviously didn't need a service order to add it, so you don't need one to remove it. |
| 19:27.50 | UnixDog | drmessano, hass issues but we deal with him i day at a time |
| 19:27.52 | dacs | but when i call 884909 it works |
| 19:27.53 | tzafrir_home | BTW: what's the deal with ipkall? |
| 19:28.02 | J4k3 | give them about a 4 hour window, thats being nice. |
| 19:28.10 | drmessano | dacs: Are you routing your IPKALL number to FWD? |
| 19:28.19 | dacs | drmessano: yes |
| 19:28.31 | tzafrir_home | Do they make some money from this? Just being nice? |
| 19:28.31 | drmessano | dacs, I can help you get that going, EASY, without FWDD |
| 19:28.35 | drmessano | They do |
| 19:28.39 | dmz | anyone here use app-conference & *1.4? |
| 19:28.40 | drmessano | They make money off all incoming calls |
| 19:28.54 | drmessano | They have an FAQ on Voxilla somewhere |
| 19:29.01 | UnixDog | no |
| 19:29.16 | drmessano | Explains their 30 day rule on unused numbers, etc |
| 19:29.19 | UnixDog | I am currently porting app_econfrence |
| 19:29.29 | drmessano | dacs |
| 19:29.37 | dacs | drmessano: here |
| 19:29.45 | UnixDog | much better conf system |
| 19:30.09 | drmessano | Do you have a static IP or external DNS name for the PBX? |
| 19:30.29 | dacs | drmessano: no |
| 19:30.37 | tzafrir_home | there we go. The fun of externip / externhost |
| 19:30.42 | tzafrir_home | Just use IAX |
| 19:30.43 | UnixDog | the drmessano headache line |
| 19:31.18 | drmessano | tzafrir_home, then good luck.. FWD IAX doesnt work, and you'll confuse him more by trying to get a non working provider up |
| 19:31.38 | drmessano | At least use SIP if youre going to get FWD going |
| 19:31.38 | tzafrir_home | So don't use FWD. There are a number of IAX providers out there |
| 19:32.00 | UnixDog | what are the other off shoots of asterisk |
| 19:32.10 | UnixDog | I know theres a few |
| 19:32.18 | drmessano | He wants to route IPKALL to his FWD URI |
| 19:32.25 | tzafrir_home | FWD aren't that good. Not plenty of codecs to choose from. Good luck to you if g711 is too much a bandwidth for you |
| 19:33.26 | UnixDog | you would think they would allow gsm |
| 19:35.51 | UnixDog | what are the other opensource pbx names |
| 19:37.15 | tzafrir_home | callweaver, yate, freeswitch bayonee, sipX, openser |
| 19:37.17 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 19:39.24 | *** join/#asterisk ScrwLoose (n=screwloo@82-42-233-19.cable.ubr10.live.blueyonder.co.uk) |
| 19:39.34 | ScrwLoose | hi |
| 19:40.09 | UnixDog | ok back to drivers |
| 19:40.17 | UnixDog | found what I was loking for |
| 19:40.21 | UnixDog | bbiab |
| 19:40.23 | *** join/#asterisk tc3driver-nii (n=huh@rrcs-24-199-16-118.west.biz.rr.com) |
| 19:40.44 | UnixDog | xpp is in need of fixiing |
| 19:43.39 | drmessano | FWD does allow gsm |
| 19:43.59 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:44.52 | tzafrir_home | UnixDog, where? |
| 19:44.59 | *** join/#asterisk julianjm (n=julianjm@107.Red-88-2-190.staticIP.rima-tde.net) |
| 19:45.09 | tzafrir_home | ah, ok |
| 19:46.33 | J4k3 | heh gsm |
| 19:46.54 | J4k3 | is that a human voice or a dying goat? |
| 19:48.40 | J4k3 | what annoys me is all this voip crap is still wrapped around a pstn world. thats about the only place where services like skype can thumb their nose at conventional voip |
| 19:49.06 | J4k3 | skype sounds *really damned good* with a decent internet connection (and a g711's worth of bandwidth) |
| 19:50.40 | jblack | This is odd. |
| 19:51.39 | jblack | After parking someone and dialing the parking lot, blind transfer (with #) is open |
| 19:51.48 | jblack | I dont' have blind transfer turned on |
| 19:52.25 | ScrwLoose | hey anyone good with iaxcomm ? |
| 19:53.07 | UnixDog | i thought iaxcomm died |
| 19:53.21 | ScrwLoose | heh |
| 19:53.32 | ScrwLoose | What alternatives are there? I tried linphone but that wont start |
| 19:53.41 | ScrwLoose | I need a later version 1.6 |
| 19:54.11 | UnixDog | twinkle |
| 19:54.14 | UnixDog | kiax |
| 19:54.18 | UnixDog | kphone |
| 19:54.26 | ScrwLoose | heh that wont start lol i dont' use kde |
| 19:54.38 | UnixDog | twinkle |
| 19:54.45 | jblack | sounds to me like you're having a locked audio device problem to me. |
| 19:54.46 | UnixDog | should work |
| 19:54.50 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 19:55.06 | ScrwLoose | well probably a distro issu |
| 19:55.07 | ScrwLoose | e |
| 19:57.50 | jameswf-home | kiax sucks and is outdated |
| 19:57.57 | jameswf-home | twinkle is ok |
| 19:58.12 | jameswf-home | I use mozphone verry simple |
| 19:58.31 | ScrwLoose | hmm |
| 19:58.55 | jameswf-home | I am not a fan of fluf |
| 19:59.03 | jameswf-home | s/fluf/fluff |
| 19:59.12 | TJNII | #festival |
| 19:59.14 | TJNII | oops |
| 19:59.33 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 19:59.38 | ScrwLoose | col that looks like it will work |
| 20:03.49 | ScrwLoose | but twinkle is a sip app |
| 20:06.35 | jameswf-home | ScrwLoose: mozphone does iax |
| 20:07.10 | Daviey | Hmm, i'm running both 1.4 and trunk. 1.4 doesn't seem to clear calls when the remote party hangs up - but trunk does using phone SPA942. Is this known? Google aint sharing much :( |
| 20:07.37 | Daviey | not normally an issue, but is with page! |
| 20:08.30 | ScrwLoose | it just so happens the page is down |
| 20:09.17 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 20:14.08 | Daviey | Anybody here using SPA942's? |
| 20:15.09 | tzafrir_home | jameswf-home, any idea how complex it would be to set up mozphone on Linux? dependencies and such? |
| 20:15.36 | tzafrir_home | I'm in need for an IAX phone in addition for twinkle (sip) |
| 20:16.05 | tzafrir_home | and my candidates are so far kiax, iaxcomm and yate-gui |
| 20:17.02 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:17.12 | jameswf-home | mozphone installs as a firefox extension so its painless |
| 20:17.13 | WilliamK | hiya tzafrir and oej |
| 20:17.48 | oej | Hej |
| 20:17.50 | oej | :-) |
| 20:18.10 | WilliamK | how goes? |
| 20:19.46 | tzafrir_home | hmm.... running firefox just for a softphone. Not fun |
| 20:19.47 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 20:26.58 | tzafrir_home | jameswf-home, well, right now I can't even get to the homepage |
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| 21:23.50 | *** join/#asterisk jwh (i=jwh@scarlett.lon.rewt.org.uk) |
| 21:25.35 | InHisName | where can I read about using group() and group_count() functions ? |
| 21:25.37 | Daviey | Hi, i have a problem.. asterisk 1.4 (svn) doesn't seem to locally clear the call (on the handset) on linksys spa942's if the calling party hangs up.. but works fine in trunk! Any ideas? |
| 21:26.26 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 21:26.43 | *** part/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 21:27.30 | magic_hat | hi all. I just installed ubuntu on a new server, along with the build-essential package. * is not letting me compile, though. I try to run make install and it tells me to run configure, which I've already done. Help? |
| 21:27.45 | jblack | daviey: I've been seeing something similiar on sip calls as well. Calls not auto-terminating. |
| 21:27.56 | jblack | That doesn't help you, but perhaps misery loves company. |
| 21:28.02 | Daviey | :( |
| 21:28.12 | Daviey | not a problem with normal calls - but a killer for page! |
| 21:28.45 | file | update to latest SVN. |
| 21:28.57 | Daviey | file: it is 1.4 svn |
| 21:29.05 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 21:29.20 | file | the 1.4 branch can change every hour during the week |
| 21:29.39 | file | so make sure you are running the latest |
| 21:29.40 | Daviey | file: owww, will update - this is 24hrs old |
| 21:31.29 | Daviey | file: just compiling now |
| 21:32.12 | Daviey | maybe i should have tried the tarball :/ |
| 21:32.32 | *** join/#asterisk G-nerd (n=AskMe@dhcp-077-249-041-129.chello.nl) |
| 21:32.45 | G-nerd | hello guys |
| 21:33.34 | magic_hat | anyone got a list of dependencies I need to get * to compile? |
| 21:33.41 | G-nerd | I have a GOOD NEWS! I bought Reily's Asterisk book! |
| 21:35.37 | Daviey | file: Arg! well what do you know, that fixed it >:/ |
| 21:35.59 | Daviey | file: wasted 2 days on that.. thinking it was me or the newish linksys firmware |
| 21:36.14 | Daviey | file: thanks! I owe you a beer |
| 21:36.25 | Daviey | jblack: ^ |
| 21:37.23 | *** join/#asterisk neturallyspeakin (n=jjohnson@pool-72-91-128-56.tampfl.fios.verizon.net) |
| 21:37.58 | jblack | daviey: Me? I didn't help you. Just comiserated. |
| 21:38.20 | Daviey | jblack: just letting you know updating worked |
| 21:38.26 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 21:38.27 | Mercestes | ~book |
| 21:38.28 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 21:38.30 | jblack | Ahh. Thank you. |
| 21:39.02 | InHisName | does that mean you are non comittal, neturallyspeakin ? |
| 21:39.10 | jblack | I'll probably wait for a new version to hit hardy. |
| 21:39.30 | neturallyspeakin | not at all |
| 21:43.09 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 21:43.25 | magic_hat | anyone got some help for me? * is not letting me compile, though. I try to run make install and it tells me to run configure, which I've already done. |
| 21:43.36 | jwh | hm |
| 21:44.19 | tzafrir_home | magic_hat, maybe ./configure gave you an error? |
| 21:44.25 | jwh | hey guys, i'm using asterisk 1.2, on a pbx which does h323<->iax2, after about 3000 calls or so, it stops dead and starts eating all the available cpu, anyone know why this happens? |
| 21:45.01 | nephfl | hello, I need to set up a system for IVR polling, I need a dialer, I am trying to avoid writing as much as possible without paying for programs, can anyone recommend an autodialer or full on ivr polling software |
| 21:45.08 | magic_hat | hmm.... termcap support not found |
| 21:45.30 | tzafrir_home | aptitude install libncurses-dev |
| 21:46.21 | Mercestes | jwh: Did you try upgrading?: |
| 21:46.27 | Mercestes | 1.2 is a little old. |
| 21:46.38 | jwh | Mercestes: not yet, worth a shot? |
| 21:47.26 | tzafrir_home | magic_hat, check the script from http://bugs.digium.com/view.php?id=10523 |
| 21:47.35 | Mercestes | depending on which 1.2 you are running, there are some bug fixes along those lines. Is asterisk chaining multiple threads or is it a single thread that is eating all your CPU? |
| 21:47.49 | jwh | Mercestes: single thread seemingly |
| 21:48.11 | jwh | they're all short calls (10 seconds max) |
| 21:48.22 | jwh | extremely busy |
| 21:48.31 | magic_hat | eek... I just installed the package to get all the dependencies. I'll try the later version after that. |
| 21:48.41 | Mercestes | Hrm. Usually I see asterisk fail to respond withina certain amount of time so it chains off a new thread (which eats more CPU) until it starts cascading threads. |
| 21:48.58 | jwh | hm |
| 21:49.07 | Mercestes | So, I would definately atleast score a new box and try 1.4 adn see how it works for you. |
| 21:49.18 | Mercestes | I can say that h.323 is not exactly the most supported feature of asterisk. |
| 21:49.23 | jwh | quite |
| 21:49.29 | Mercestes | but I can't say your problem is h.323 related or not. |
| 21:49.29 | jwh | awkward customer, but also the buggest |
| 21:49.49 | jwh | I will try 1.4 first |
| 21:50.43 | nephfl | anyone with suggestions, google is not helping me at all |
| 21:51.30 | J4k3 | jwh: be sure to charge accordingly :) |
| 21:52.19 | jwh | J4k3: oh they pay $$$ |
| 21:52.31 | jwh | they can have whatever they want ;) |
| 21:53.00 | J4k3 | right on |
| 21:53.09 | J4k3 | I'm about to double my rates |
| 21:53.15 | J4k3 | for technical services |
| 21:53.20 | jwh | hehe |
| 21:53.33 | J4k3 | I've discovered everybody else is charging a lot more than me |
| 21:53.38 | J4k3 | in the area |
| 21:53.44 | J4k3 | so its like... no wonder I get no respek |
| 21:53.50 | jwh | hehe |
| 21:53.51 | jwh | quite |
| 21:53.57 | jwh | this customer pays silly rates |
| 21:54.10 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:54.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:55.31 | jwh | the most notable thing is |
| 21:55.32 | jwh | <PROTECTED> |
| 21:55.32 | Daviey | J4k3: It's crazy - think of the highest number you feel comfortable charging then double it |
| 21:55.38 | jwh | console is full of that when it dies |
| 21:55.47 | jwh | so it could be a timing issue? |
| 21:57.18 | Daviey | J4k3: partly joking btw |
| 22:03.53 | G-nerd | hi guys, which channel is for programming Linux in C or C++? |
| 22:05.23 | jblack | hmm. looks like fwd services are down. I can't reach 511, 612, 613.. heck. any of them |
| 22:07.06 | drmessano | 613 works |
| 22:07.10 | drmessano | Everything else is toast |
| 22:07.57 | mvanbaak | drmessano: as of today openfire works with asterisk-trunk :) |
| 22:08.11 | jblack | perhaps you got lucky with 613. I can't reach it. |
| 22:08.17 | drmessano | you were using Trunk? |
| 22:08.24 | mvanbaak | drmessano: I am |
| 22:08.26 | drmessano | Ah |
| 22:08.34 | drmessano | Crap |
| 22:08.38 | drmessano | Had I known that |
| 22:08.43 | drmessano | Manager 1.1 issues |
| 22:09.06 | mvanbaak | nah, it was something in the manager MD5 auth |
| 22:09.13 | drmessano | Hmm |
| 22:09.15 | drmessano | ok |
| 22:09.49 | mvanbaak | srt found it and wrote a patch for it. Corydon76 committed a fix based on his problem description |
| 22:10.47 | drmessano | So |
| 22:10.55 | drmessano | Asterisk patch or Asterisk-IM patch |
| 22:10.57 | drmessano | ? |
| 22:12.00 | *** join/#asterisk theunabonger (i=Bhongwan@lunacity.themoon.org) |
| 22:12.17 | *** part/#asterisk theunabonger (i=Bhongwan@lunacity.themoon.org) |
| 22:12.25 | *** join/#asterisk nortex (n=chatzill@64.136.92.71) |
| 22:13.51 | nortex | Does anyone know if there is a way from the dialplan to get a users email address out of their voicemail settings? |
| 22:15.07 | russellb | nortex: nope, not a built in way .... the only good way would be to write an agi that opens the file and finds it |
| 22:16.24 | nortex | Alright, I know you can do it by putting voicemail.conf in realtime and using the realtime application, but I hoped for something by way of variables. Thanks Russell |
| 22:16.54 | *** join/#asterisk AndyGraybeal (n=andy@node216.34.251.72.1dial.com) |
| 22:17.34 | nortex | russellb: Don't work to hard this weekend. |
| 22:18.00 | russellb | nortex: ha, thanks :) |
| 22:18.04 | russellb | i'm not very good at that ... |
| 22:19.01 | *** join/#asterisk RoyK (n=roy@91.149.4.251) |
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| 22:21.40 | *** part/#asterisk curtn (n=curtis@cl-451.trn-01.it.sixxs.net) |
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| 22:24.39 | *** join/#asterisk RoyK (n=roy@91.149.4.251) |
| 22:25.33 | jblack | [TK]D-Fender: Ping |
| 22:26.42 | *** join/#asterisk anthm (n=anthm@75-135-78-143.dhcp.mdsn.wi.charter.com) |
| 22:26.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:27.22 | [TK]D-Fender | <PROTECTED> |
| 22:27.44 | jblack | [tk]d-fender: Do you have a google checkout address, by any chance? |
| 22:28.11 | [TK]D-Fender | jblack, nope... |
| 22:28.31 | jblack | aww. ok |
| 22:28.48 | [TK]D-Fender | jblack, Well I DO have a google address....... http://www.google.com ;) |
| 22:28.57 | [TK]D-Fender | jblack, Go check it out! |
| 22:29.00 | [TK]D-Fender | jblack, ;) |
| 22:29.39 | drmessano | Google checkout? |
| 22:29.55 | jblack | drmessano: Google's version of paypal |
| 22:29.56 | drmessano | ~Google checkout |
| 22:30.10 | drmessano | Google Checkout: Another forgotten Google app |
| 22:30.32 | drmessano | I know |
| 22:30.35 | *** join/#asterisk mog (n=mog@216.207.245.1) |
| 22:30.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:30.40 | drmessano | But Google Checkout is like binpaste |
| 22:30.46 | drmessano | "Not pastebin, but close" |
| 22:30.47 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 22:32.11 | jwh | Mercestes: hey |
| 22:32.16 | jwh | upgraded to 1.4, same thing |
| 22:32.49 | jwh | dies quicker too |
| 22:33.04 | *** join/#asterisk Jam0r (i=Jamie@87.127.190.82) |
| 22:34.07 | jwh | ah, I get useful errors now |
| 22:34.08 | jwh | [Jan 12 22:30:58] NOTICE[53182]: chan_iax2.c:6689 socket_read: Out of idle IAX2 threads for I/O, pausing! |
| 22:34.11 | jwh | [Jan 12 22:30:58] NOTICE[53182]: chan_iax2.c:959 __schedule_action: Out of idle IAX2 threads for scheduling! |
| 22:34.41 | jwh | any way to increase the number of threads? |
| 22:34.47 | jwh | as its only hitting 50 or so currently |
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| 22:39.39 | *** join/#asterisk RoyK (n=roy@91.149.4.251) |
| 22:42.03 | mvanbaak | isn't there some var in chan_iax.c ? |
| 22:42.11 | jwh | ah |
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| 22:45.05 | Jam0r | its defined in there yeah, but locking up about 30k calls short of the limit |
| 22:45.34 | jwh | yeah |
| 22:47.43 | mvanbaak | drmessano: asterisk patch |
| 22:47.59 | mvanbaak | drmessano: http://bugs.digium.com/view.php?id=11749 |
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| 22:57.13 | *** join/#asterisk marc7 (n=marc@S01060018f877b2e0.gv.shawcable.net) |
| 22:57.42 | magic_hat | hey everyone. I'm getting a file not found error when the system goes to play a background sound. The file does in fact exist... ideas? |
| 22:58.08 | marc7 | [root@lucent:~]# zttest // Opened pseudo zap interface, measuring accuracy... -200.002335% -199.994522% -200.001862% -199.999451% -200.000717% |
| 22:58.11 | marc7 | come on... that can't be good |
| 22:58.42 | *** join/#asterisk nDuff (n=ccd@user-387ocuv.cable.mindspring.com) |
| 23:00.00 | nDuff | Anyone in Austin have a PRI with spare capacity? I'm looking for somewhere to host my phone server for a few weeks on very short notice; we don't typically use more than a few channels at any given time. |
| 23:01.22 | mvanbaak | WHAHAHAHAHAHAHAHAHAHA |
| 23:01.24 | mvanbaak | A telephone company cut off an FBI international wiretap after the agency failed to pay its bill on time, according to a U.S. government audit released on Thursday. |
| 23:01.39 | *** join/#asterisk adker (n=chatzill@74-33-205-192.br1.glv.ny.frontiernet.net) |
| 23:02.01 | _ShrikE | yeah, some of those taps were FISA even!! |
| 23:02.25 | mvanbaak | indeed ! |
| 23:02.31 | mvanbaak | brilliant !!!! |
| 23:03.11 | mvanbaak | yeah |
| 23:03.32 | mvanbaak | let them tap me, they wont pay so it's not going to be an issue ;) |
| 23:03.40 | mvanbaak | http://www.reuters.com/article/newsOne/idUSEIC07119120080110 |
| 23:04.17 | mvanbaak | http://www.foxnews.com/story/0,2933,321847,00.html |
| 23:04.27 | mvanbaak | those are the links to the story |
| 23:04.39 | mvanbaak | Poor supervision of the program also allowed one agent to steal $25,000, the audit said. |
| 23:04.47 | mvanbaak | holy crap, I want to work there ! |
| 23:05.21 | magic_hat | file.c:563 ast_openstream_full: File cdngreeting.gsm does not exist in any format .... anyone? |
| 23:05.56 | mvanbaak | magic_hat: what's your dialplan line ? |
| 23:06.18 | magic_hat | exten => s,4,BackGround(cdngreeting.gsm) |
| 23:06.40 | puppet | remove .gsm ? |
| 23:06.43 | nDuff | magic_hat: presuming it exists, leave off the .gsm; asterisk will look for the extension itself. |
| 23:06.51 | mvanbaak | use: exten => s,4,BackGround(cdngreeting) |
| 23:07.30 | magic_hat | yeah, i added the extension and the path to see if that would help. it doesn't work with BackGround(cdngreeting) either |
| 23:07.52 | mvanbaak | yup, my ibook is bricked |
| 23:07.54 | magic_hat | this was working on my old box... just switched everything over to a new server and now it's barfing. |
| 23:07.55 | mvanbaak | *cry* |
| 23:07.57 | tzafrir_home | marc7, what Zaptel device do you use? or is it ztdummy? |
| 23:08.28 | marc7 | tzafrir_home: it's ztdummy, and we're trying to use High Resolution Timer support in 2.6.23.8 |
| 23:08.48 | marc7 | after building zaptel, it says "High Resolution Timer started, good to go" in the dmesg |
| 23:08.59 | marc7 | and both zaptel and ztdummy are listed in an "lsmod" |
| 23:11.01 | marc7 | i'm almost at my wit's end with timing support with ztdummy.. i'm disabling kernel support for IRQ balancing and upping my timer frequency from 250Hz to 1000Hz |
| 23:12.29 | tzafrir_home | marc7, echo 3 >/sys/modules/ztdummy/parameters/debug |
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| 23:12.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:12.52 | tzafrir_home | Then you should get a kernel message every 5000 ticks |
| 23:13.05 | tzafrir_home | This should be 5 seconds |
| 23:13.55 | marc7 | I see the "ztdummy: 5000 ticks from hrtimer" output in dmesg every 5 seconds |
| 23:14.12 | tzafrir_home | really? not 2.5 seconds? strange |
| 23:14.46 | marc7 | anything else I'm supposed to gleam from that? nothing except that same message repeated over and over in dmesg |
| 23:14.48 | *** join/#asterisk catharina (n=ask@78-21-204-113.access.telenet.be) |
| 23:14.52 | tzafrir_home | What version of zaptel is it? cat /sys/modules/zaptel/version |
| 23:15.17 | marc7 | 1.4.7.1, just grabbed it out of svn a moment ago |
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| 23:16.59 | tzafrir_home | Again: are you sure that you get those messages once per 5 seconds? and not once per 2.5 seconds? |
| 23:18.03 | *** join/#asterisk marcan (i=1337@host215-248.cvf.fit.edu) |
| 23:18.14 | marc7 | if I look at /var/log/kernel, it's every 20 seconds |
| 23:18.27 | marc7 | i was previously using dmesg, but the timestamps are every 20 seconds |
| 23:19.08 | *** part/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 23:20.19 | tzafrir_home | hmmm... what do you see on zttest -v -c 3 |
| 23:21.19 | marc7 | Results after 0 passes // Best: 0.000 -- Worst: 100.000 -- Average: 100.000000, Difference: 100.000000 |
| 23:21.51 | *** part/#asterisk catharina (n=ask@78-21-204-113.access.telenet.be) |
| 23:24.44 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 23:25.00 | tzafrir_home | hmmm.. 0 passes? just drop the -c, and try 2 or three lines |
| 23:26.16 | marc7 | 8192 zaptel samples in 32768.203 system clock sample intervals (400.002%) |
| 23:26.16 | marc7 | 8192 zaptel samples in 32767.646 system clock sample intervals (399.996%) |
| 23:26.22 | marc7 | that's from just zttest -v |
| 23:26.35 | marc7 | this is just with the HPET support... |
| 23:27.20 | marc7 | having *no* timer support (USB support isn't compiled into my kernel) performed better than this... 99.0% average which isn't great... but better than this :) |
| 23:34.33 | tzafrir_home | marc7, any chance you could file a bug report with the relevant config details? |
| 23:35.39 | marc7 | i'd love to! bugs.digium.com? and *which* config details? my kernel's /proc/config.gz? what output from the zaptel compile directory? |
| 23:36.39 | marc7 | zaptel's timing (when I have no dedicated digium hardware for that purpose) is my single gripe about the awesomeness of asterisk |
| 23:36.52 | marc7 | that and how roundrobin was deprecated. what were they thinking?! |
| 23:37.04 | Qwell | marc7: rrmemory |
| 23:37.39 | marc7 | but I don't *want* it to resume where it last left off! I want person A always to be called first... person B to be called if person A is on the phone. person C to be called if A is on the phone and B doesn't answer |
| 23:38.03 | mvanbaak | Qwell: DON'T !!!!!!!! |
| 23:38.16 | mvanbaak | you have to commit cfwd.diff before you die ! |
| 23:38.21 | Qwell | marc7: it's not my argument.. |
| 23:38.56 | marc7 | fair enough, I know what roundrobin was deprecated in favor *for*, I just think that was a silly idea when I actually wanted what roundrobin did. |
| 23:39.22 | russellb | i agree that's silly if there is no suitable replacement for the same behavior |
| 23:39.45 | Qwell | well, I think it was decided that rrmemory was what people "actually" wanted. but that apparently isn't the case |
| 23:39.55 | russellb | well, perhaps that's what most people want |
| 23:40.01 | mvanbaak | <--- updating patches that are left alone |
| 23:40.04 | russellb | but the other still gets used .. |
| 23:40.07 | mvanbaak | does that make me sick ? |
| 23:40.24 | Qwell | russellb: so maybe it's something we should revisit at some point.. |
| 23:40.36 | russellb | probably. |
| 23:40.45 | Qwell | this isn't the first time I've heard this |
| 23:40.53 | russellb | i don't remember that discussion when it got deprecated |
| 23:40.53 | *** join/#asterisk piper69 (n=haiger@unaffiliated/piper69) |
| 23:41.23 | _ShrikE | I would like to share the contents of the asterisk database between multiple servers. Can multiple asterisk servers use the same AstDB? |
| 23:41.34 | mvanbaak | _ShrikE: no |
| 23:41.34 | Qwell | _ShrikE: no |
| 23:41.35 | russellb | _ShrikE: no |
| 23:41.41 | russellb | wow. |
| 23:41.42 | Qwell | in surround sound |
| 23:41.48 | mvanbaak | multicast reply ! |
| 23:42.10 | _ShrikE | :) |
| 23:42.19 | Qwell | _ShrikE: doing so would be a *disaster* |
| 23:42.24 | russellb | _ShrikE: i would recommend using func_odbc if you want to share dialplan accessible database stuff. |
| 23:42.28 | mvanbaak | E_OUTOFWINE |
| 23:43.09 | mvanbaak | Qwell: can you commit 10740 please ? |
| 23:43.44 | Qwell | will you comment it, so I can look at it again on Monday? |
| 23:43.48 | jblack | _ShrikE: no (Bouncing off a far-away mountain) |
| 23:43.49 | Qwell | I had some issues with it, iirc |
| 23:43.51 | tzafrir_home | marc7, yes, the details you mentioned |
| 23:44.06 | mvanbaak | blitzrage had no issues, me had no issues |
| 23:44.16 | Qwell | mine were hypothetical issues |
| 23:44.28 | mvanbaak | I'll upload a patch that puts the option in configs/voicemail.conf.sample |
| 23:46.40 | jblack | Man. Callwithus _rocks_. |
| 23:46.57 | jwh | 16k thread limit, still locking up periodically, guess its not releasing/reusing them |
| 23:47.50 | mvanbaak | Qwell: commented, and uploaded new patch |
| 23:49.27 | mvanbaak | russellb: any update on 11116 ? |
| 23:49.31 | mvanbaak | oops |
| 23:52.01 | russellb | it's the weekend |
| 23:52.05 | russellb | i refuse to look at the bug tracker |
| 23:52.09 | russellb | unless it's like ... critical |
| 23:52.11 | russellb | :) |
| 23:54.53 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 23:56.50 | mvanbaak | lol |
| 23:56.50 | mvanbaak | ok |
| 23:56.54 | mvanbaak | I'll stop then |
| 23:57.23 | mvanbaak | hhmm |
| 23:57.48 | mvanbaak | I guess a module for talking to a specific CRM application wont make it into the core of asterisk right ? |
| 23:58.35 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 23:59.17 | lmadsen | _ShrikE: use DUNDi and call the DB() function in the mapping on the remote server to get the data back |
| 23:59.43 | lmadsen | M10740 |
| 23:59.56 | lmadsen | haha... not here :) |