| 00:00.04 | ManxPower | bjames, never heard of it. |
| 00:00.10 | kamileon | no that exten line is in extensions.conf , it seemed like i was saying in default context exten s |
| 00:00.12 | ManxPower | But I doubt you need TDMoE with it. |
| 00:00.20 | kamileon | i was trying to clarify |
| 00:00.37 | bjames | it a T1/E1 to TDMoE |
| 00:00.48 | bjames | of course I need TDMoE |
| 00:01.02 | *** part/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 00:01.16 | ManxPower | bjames, and they support Asterisk's version of TDMoE? |
| 00:01.51 | ManxPower | TDMoE really hasn't been well supported since IAX2 w/trunking came along. |
| 00:02.05 | wunderkin | yes it uses tdmoe and its supposted to work with asterisk but i havent used it |
| 00:02.14 | bjames | it says to use CVS head as of 8/2005 |
| 00:02.47 | wunderkin | holy co |
| 00:02.53 | bjames | the Zaptel driver alone is working with the Redfone, but when I start * the server crashes |
| 00:03.09 | ManxPower | bjames, sounds like time for a bug report |
| 00:03.42 | bjames | yeah, I'm using * 1.0.10 |
| 00:03.43 | wundaboy | how do i set it up so that my polycom doesnt connect to an ftp server for its config file? It just boots and says cannot load <MAC>.cfg and reboots, how do i fix this? |
| 00:04.48 | bryan2 | kamileon: That worked great. Thanks again. |
| 00:04.57 | *** join/#asterisk Litex (i=tilex@equinox.alluvium.com) |
| 00:05.01 | *** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu) |
| 00:06.03 | mistral | wundaboy: is the channel named #polycom :D |
| 00:06.36 | wundaboy | mistral: i just figured someone would know... |
| 00:08.27 | Spida | or where should I go (read) for that? |
| 00:09.18 | kamileon | bryan2 : now if i can just get *my* box working |
| 00:09.28 | airdog | adibar: thx again, lots of stuff in there. I'll have to investigate longer. In the meantime, have to rush off. Bye |
| 00:10.06 | wunderkin | i havent had very good luck with tdmoe yet but i havent tried very hard yet |
| 00:10.07 | adibar | airdog: welcome, cya |
| 00:10.38 | j0n | does anyone have sip presence working with ael? |
| 00:13.08 | SwK[Work] | anyone know of a service for pushing CNAM on to the SS7 network? |
| 00:13.26 | rob0 | kamileon: pastebin your dialplan |
| 00:15.04 | [av]bani | grrr, moh sux |
| 00:15.53 | [av]bani | this makes no sense at all |
| 00:17.23 | kamileon | http://pastebin.ca/40430 |
| 00:17.33 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 00:18.04 | kamileon | brb |
| 00:18.51 | *** join/#asterisk jets (n=jetsnoc@meowwwww.pmt.coop) |
| 00:19.03 | Qwell | jets: y0 y0 |
| 00:19.22 | jets | Qwell! how are ya? |
| 00:19.31 | Qwell | jets: pretty good, you? |
| 00:19.33 | rob0 | line 10, shouldn't "FXO" be replaced by a variable (and that variable declared in the global section)? |
| 00:20.04 | [av]bani | why would moh stutter? |
| 00:20.11 | kamileon | yes it should be 5 |
| 00:20.13 | jets | I can't complain.. Just fleshing out an updated resume :) |
| 00:20.19 | Qwell | uh oh |
| 00:20.22 | kamileon | but still, the pattern doesnt match |
| 00:20.38 | jets | oh just for some contracting and other opportunities. |
| 00:20.40 | kuku5 | is a 96 channel pri considered a t3 or a fractional t3 |
| 00:20.56 | Qwell | kuku5: a T3 is 600 some channels |
| 00:21.00 | jets | hrm it would be 3 pri's riding a fractional ds3 i would imagine? |
| 00:21.00 | Qwell | 28 T1's iirc |
| 00:21.08 | file | Qwell: !!!??? |
| 00:21.09 | [av]bani | i dont get this at all :( |
| 00:21.15 | Qwell | file: I said some |
| 00:21.27 | tehdely | can asterisk pull SIP and IAX configuration from MySQL? |
| 00:21.30 | file | Qwell: boss status? |
| 00:21.31 | Qwell | tehdely: sure |
| 00:21.35 | tehdely | sweeeet 8) |
| 00:21.36 | Qwell | file: flying |
| 00:22.00 | file | Qwell: translation? |
| 00:22.01 | kamileon | rob0: changed the ZAP to 5 and same issue |
| 00:22.05 | Qwell | file: couldn't talk to him today |
| 00:22.18 | rob0 | do the extensions 201-204 work? |
| 00:22.43 | kuku5 | Qwell: how would one connect a 96 channel pri through 1 ethernet cable ? |
| 00:22.49 | Qwell | kuku5: umm |
| 00:22.54 | Qwell | one wouldn't? |
| 00:23.23 | kuku5 | ah |
| 00:23.31 | kuku5 | then how can a provider do it |
| 00:23.35 | kamileon | rob0: no, brb |
| 00:23.36 | Qwell | they don't? |
| 00:23.42 | kuku5 | hm |
| 00:23.42 | Qwell | or tdmoe or something? |
| 00:23.44 | kuku5 | :) ok - thanks |
| 00:24.05 | jets | Qwell: Did your company bring on some *? |
| 00:24.07 | Qwell | I guess tdmoe is possible. I know 0 about that though |
| 00:24.11 | Qwell | jets: we're working on it |
| 00:24.25 | Qwell | jets: got a test box and a few web apps going right now |
| 00:24.31 | jets | Excellent! |
| 00:24.40 | Qwell | probably gonna pilot about 40 users |
| 00:25.05 | *** join/#asterisk fiber0pti (n=John@206-169-194-79.gen.twtelecom.net) |
| 00:25.23 | fiber0pti | how do I set the outgoing callerid coming from a specific extension? |
| 00:25.59 | jets | in sip.conf you can specify it for the actual device. |
| 00:26.18 | *** join/#asterisk elg (n=fugalh@hfugal.NMSU.Edu) |
| 00:26.24 | Spida | can I get help here for getting my fritz pci to work with mISDN, too? |
| 00:26.30 | jets | under the extension it is just called=Brian McManus <2084347146> |
| 00:26.35 | Spida | or where should I go (read) for that? |
| 00:26.54 | Qwell | Spida: You can try |
| 00:29.46 | Spida | I have a "AVM Fritz!PCI v2.0 ISDN" installed misdn cvs from today and enabled |
| 00:29.46 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:30.16 | Spida | ISDN_CAPI and CONFIG_MISDN_AVM_FRITZ |
| 00:31.00 | fiber0pti | jets: But is it for outgoing non sip calls in the sip.conf callerid is used for? |
| 00:31.11 | *** join/#asterisk Umaro (n=umaro@68.142.142.105) |
| 00:31.25 | Spida | the card is found by the driver (according to dmesg), but (a) I get an error in dmesg not possible to autoload mISDN_l1 please try to load manually |
| 00:31.28 | Umaro | Hey guys.. in your opinion, what's the best card to use purely as a hardware timer for *? |
| 00:31.29 | Spida | mISDN: INTERNAL ERROR in drivers/isdn/hardware/mISDN/stack.c:596 |
| 00:31.47 | Qwell | Umaro: if it's ONLY for timing...probably an x100p |
| 00:32.30 | Spida | and (b) can't do anything from userspace with it (/dev/capi* doesn't exist and /proc/capi doesn't show any devices) |
| 00:33.14 | Spida | I am pretty sure the error is on my side (the last time I used isdn was in linux-2.2) |
| 00:34.11 | mog_work | any card should work fine Umaro |
| 00:34.32 | Spida | any ideas? |
| 00:35.05 | *** join/#asterisk bweschke (n=bweschke@68.156.43.202) |
| 00:35.19 | Umaro | Qwell: so are they all the same, timer wise? |
| 00:35.25 | Qwell | Umaro: yes |
| 00:35.46 | mog_work | however getting support for an x100p outside of this channel is dificult |
| 00:36.08 | mog_work | as digium only supports the x100ps sold by digium over a year ago |
| 00:36.08 | ManxPower | getting support for X100P clones on this channel can be hard too |
| 00:36.11 | *** join/#asterisk TooMe (n=in-ter-e@65.116.137.10) |
| 00:36.11 | Umaro | ok. I've been trying ztdummy (2.6.15.2) and zttest keeps giving me 99.92% scores |
| 00:36.15 | mog_work | and very few of the oem support them |
| 00:38.10 | TooMe | i've been have a bit of trouble getting /bin/sh -c (aka System() ) to run cp or mv |
| 00:38.15 | Umaro | it seems odd to me that with 2.6.13+'s RTC changes, the kernel RTC driver still isn't good enough, it seems |
| 00:38.49 | mog_work | what problem toome? |
| 00:39.32 | Umaro | does anyone have any tips at getting a slightly higher zttest score with ztdummy, or other software based timers? |
| 00:39.44 | TooMe | well in the dialplan, it flat out doesn't work...when i run it from the prompt it gives me "missing file arguement" and when i run it without /bin/sh -c it goes through fine |
| 00:40.11 | mog_work | turn off serial umaro |
| 00:40.12 | mog_work | dma |
| 00:40.16 | mog_work | turn on dma |
| 00:40.21 | Umaro | turn off serial? |
| 00:40.22 | mog_work | and dont run xwindows |
| 00:40.23 | Umaro | dma is on |
| 00:40.27 | Umaro | xwindows isn't running |
| 00:41.13 | Umaro | turning off serial helps, though? |
| 00:41.13 | Umaro | never heard that |
| 00:41.13 | mog_work | yes |
| 00:41.13 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:41.13 | mog_work | serial console eats interrupts |
| 00:41.13 | mog_work | sorry turning off serial consoles helps |
| 00:41.13 | ManxPower | I thought ztdummy used RTC on 2.6. |
| 00:41.13 | mog_work | serial ports themselves dont do anyting |
| 00:41.15 | ManxPower | I suppose I could be wrong. It's happened before. |
| 00:41.15 | mog_work | yes it does ManxPower |
| 00:41.33 | ManxPower | I don't think I've ever used ztdummy |
| 00:42.10 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 00:42.10 | Umaro | mog_work: serial console is off by default in 2.6.15.2. hmm. |
| 00:42.26 | Umaro | If this switch wasn't in india, I'd already have a digium card in it by now |
| 00:42.45 | Umaro | but india customs will want to dissect it for a couple weeks before they let it through |
| 00:43.17 | *** part/#asterisk Litex (i=tilex@equinox.alluvium.com) |
| 00:43.35 | TooMe | do you have any friends that work at dell? have one of them ship it |
| 00:44.07 | mog_work | just say it is a network audio adapter ^_^ |
| 00:44.09 | Spida | TooMe: ah, the "do not touch or redirect" thing? |
| 00:44.38 | TooMe | Spida: the we employ 15k++ of your countries top talent thing ;) |
| 00:45.11 | Umaro | mog_work: do you use ztdummy? |
| 00:46.28 | mog_work | i do on one of my boxes that has no hw |
| 00:46.32 | mog_work | err no pci slots |
| 00:46.49 | Umaro | mog_work: do you do meetme and IAX2 trunking on it? |
| 00:46.55 | mog_work | yeah |
| 00:46.58 | mog_work | everything works |
| 00:47.15 | mog_work | i dont do any trunking |
| 00:47.21 | mog_work | but i have meetmed off of it |
| 00:47.24 | [av]bani | hmm some kind of 1.2.4 bug ... |
| 00:48.02 | TooMe | would i offend anyone if i cut/paste the system string i'm trying to get to work? |
| 00:48.06 | CodeGuru | hey guys, what is the difference between Asterisk and Asterisk@Home ? |
| 00:48.34 | mog_work | yes and no CodeGuru |
| 00:48.42 | mog_work | you wont find anyone in here reccomending it |
| 00:48.46 | mog_work | they have their own channel though |
| 00:48.50 | mog_work | <PROTECTED> |
| 00:48.54 | shido6 | /dcc send ManxPower |
| 00:48.55 | shido6 | /dcc |
| 00:49.01 | shido6 | grr gaim... |
| 00:49.04 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 00:50.10 | CaT[tm] | damn. the voiceone setup failed. grumble. |
| 00:51.15 | *** join/#asterisk jyukes (n=jameshot@pool-141-150-181-246.atc.east.verizon.net) |
| 00:51.19 | litage | when a call goes softphone -> Asterisk -> softphone, only * needs g729 licenses. where are the g729 licenses needed when the call goes softphone -> SER -> softphone, or softphone -> SER -> Asterisk -> Softphone ? |
| 00:51.37 | CodeGuru | any expert here would help me ? i feel lost in this telephony space ! |
| 00:51.51 | mog_work | asterisk needs g729 everytime you convert from g729 to something else |
| 00:52.01 | mog_work | so if both softphones have g729 you just pass the call along |
| 00:52.54 | CodeGuru | mog_work: could you help me find a suitable solution for my problem ? |
| 00:53.08 | *** join/#asterisk Garak_ (n=garak@209.5.171.170) |
| 00:53.09 | mog_work | whats the prob bob |
| 00:54.08 | CodeGuru | well, im a software developer, i was asked to develope the sales Dept. phone system |
| 00:54.31 | malverian[work] | Whoa... |
| 00:54.43 | malverian[work] | Feb 6 19:54:34 WARNING[1564]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 00:54.44 | mog_work | okies |
| 00:54.49 | malverian[work] | Getting spammed about 15 per second with those.. |
| 00:54.51 | mog_work | weird malcolmd |
| 00:54.55 | malverian[work] | Anyway I can tell where they're coming from? |
| 00:54.56 | mog_work | weird malverian[work] |
| 00:55.07 | mog_work | yeah you are trying to convert gsm to slin |
| 00:55.10 | mog_work | but its not working |
| 00:55.12 | mog_work | which is odd |
| 00:55.20 | mog_work | do you have the codecs loaded? |
| 00:55.22 | malverian[work] | I don't use gsm nor slin |
| 00:55.32 | malverian[work] | I use mulaw for everything. |
| 00:55.34 | CodeGuru | i can develop an sip/h323 client but i need a server to take calls from pstn and convert them to voip and vice versa, also recording, caller id, forwarding, ....... you know the rest |
| 00:55.53 | CaT[tm] | argh |
| 00:55.55 | CodeGuru | so i was thinking Asterisk, but i know nothing about linux or unix |
| 00:56.02 | mog_work | oh well something is trying to speak gsm to you |
| 00:57.19 | CodeGuru | so i basically need help choosing the suitable hardware, installing, configuring the server. |
| 00:57.29 | mog_work | okies |
| 00:57.35 | mog_work | you guys need a t1? |
| 00:57.40 | CodeGuru | could we take this in private ? |
| 00:57.48 | malverian[work] | mog_work, Figured it out.. |
| 00:57.53 | mog_work | what was it? |
| 00:57.55 | mog_work | ohhhh |
| 00:57.58 | mog_work | was it a file |
| 00:58.01 | mog_work | being played back |
| 00:58.05 | hypnox | CodeGuru sounds like you should recommend that your employers hire an asterisk consultant |
| 00:58.08 | malverian[work] | Somehow my intercom (alsa/default) never hungup |
| 00:58.10 | mog_work | and you didnt have gsm stuff up? |
| 00:58.13 | mog_work | ahh |
| 00:58.28 | CodeGuru | are you interested ;) ? |
| 00:58.45 | Qwell | CodeGuru: there are several consultants here |
| 00:58.46 | malverian[work] | I had gsm stuff up. |
| 00:58.46 | litage | mog_work: if a softphone or ip phone supports g729, does that mean that it has a valid g729 license installed on it? |
| 00:58.55 | malverian[work] | Just a weird fluke i guess.. |
| 00:59.00 | Qwell | CodeGuru: If you're interested in paying somebody, there are several places to get ahold of some |
| 00:59.03 | mog_work | right it has its own g729 license |
| 00:59.12 | CodeGuru | like ? |
| 00:59.26 | Qwell | CodeGuru: here, the forums, the asterisk-biz mailing list, the wiki |
| 00:59.59 | CodeGuru | i want ab Asterisk Consoltant, any1 interested ? |
| 01:00.35 | CodeGuru | looks like no one is interested in getting payed :D ! |
| 01:01.03 | Qwell | CodeGuru: msg me |
| 01:01.33 | mog_work | ill take money |
| 01:01.41 | mog_work | but only through digium ^_^ |
| 01:03.34 | litage | mog_work: softphones that support g729 probably won't be free, right? |
| 01:03.46 | mog_work | none that i know of |
| 01:04.12 | mog_work | as it costs $$ for licenses |
| 01:04.31 | *** join/#asterisk psi_force (n=mark@marksnb.eng.unimelb.edu.au) |
| 01:04.51 | psi_force | hi all |
| 01:05.06 | mog_work | that is a nifty nick |
| 01:05.36 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 01:07.07 | psi_force | I have a problem with accountcode. In the sip.conf, every user has "accountcode=SIP-xxx" however if i look the cdr records one person who uses us as a sip trunk has managed to clear his accountcode. How is this possible? |
| 01:07.13 | *** join/#asterisk Spida (i=Spida@p508A2759.dip0.t-ipconnect.de) |
| 01:10.50 | psi_force | any ideas? |
| 01:12.34 | *** join/#asterisk austinnichols10 (n=austinni@dsl-10-169.cofs.net) |
| 01:19.32 | g4m | other than ztdummy (which seems to kill schedualing on my debian box) how else can i get timing to work for my meetme rooms? |
| 01:20.02 | Qwell | g4m: buy a zap card |
| 01:23.05 | g4m | well looks like i have to get zaptel working then |
| 01:23.08 | g4m | err ztdummy |
| 01:23.16 | fafnir | Well it looks like I'm on top |
| 01:23.52 | CaT[tm] | poo. all that installing and I just now realise that voiceone.it doesn't do voicemail. sigh. |
| 01:24.14 | fafnir | oh |
| 01:24.23 | fafnir | i once spent the whole day taking out my starter |
| 01:24.40 | fafnir | only to realize that i had been given an alternator by the junk yard |
| 01:24.50 | justinu | lol |
| 01:29.06 | *** part/#asterisk Garak_ (n=garak@209.5.171.170) |
| 01:29.12 | psi_force | so does anyone have any experience with the accountcode variable? |
| 01:30.04 | *** join/#asterisk Garak_ (n=garak@209.5.171.170) |
| 01:30.22 | litage | when would you have Asterisk act as a sip client? |
| 01:31.49 | *** join/#asterisk palomiux (n=lecaus@200.30.160.186) |
| 01:31.58 | palomiux | Hi there |
| 01:32.15 | palomiux | I have a question about asterisk, can anyone give me a little help? |
| 01:36.13 | palomiux | Can anyone tell me how this channel works? |
| 01:36.29 | *** join/#asterisk QbY (n=Kelvin@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 01:36.58 | QbY | anyone seen weird behavior out of 1.2.1 ?? as in just crashing, and when you reload it takes 2 minutes to do so? |
| 01:37.12 | QbY | scratch that, its still loading.. so >4 minutes |
| 01:40.24 | QbY | > 6 minutes (loading dialplan now) |
| 01:40.51 | g4m | anyone know what this might mean: zaptel: no version for "struct_module" found: kernel tainted.? |
| 01:41.05 | QbY | hey.. i just got that same error.. but ipip |
| 01:41.30 | litage | do you set the packetization (Eg: 20ms) on the enduser device (Eg: softphone), on *, or on SER? |
| 01:41.40 | *** join/#asterisk xtrvd (n=j@d209-121-36-44.bchsia.telus.net) |
| 01:43.38 | palomiux | I have a question about asterisk, can anyone give me a little help? |
| 01:43.56 | Qwell | palomiux: You ask a question and wait for an answer |
| 01:44.38 | _Sam-- | more like, you hope for answer :) |
| 01:44.48 | palomiux | :) |
| 01:44.59 | palomiux | anyone knows about asterisk home? |
| 01:45.09 | palomiux | what about asterisk live? |
| 01:46.26 | _Sam-- | i think there are *@~ ?s than i ever remember |
| 01:46.30 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 01:47.46 | xtrvd | Here's a quick question for practically anybody: How easy is it to install Zaptel drivers for a TDM404B digium card? |
| 01:47.58 | _Sam-- | make make install modprobe |
| 01:48.30 | xtrvd | That difficult eh? =) |
| 01:48.39 | _Sam-- | easier said than done, always :) |
| 01:48.41 | litage | what are the differences between SER and openSER? |
| 01:48.52 | _Sam-- | installing it is the easy part...configuring it...that is the harder part :) |
| 01:48.54 | mog_work | 4 leters |
| 01:48.55 | Qwell | litage: open, give or take |
| 01:49.13 | litage | Qwell: not sure what you mean...they're both FOSS, no? |
| 01:50.06 | Qwell | litage: the word "open", or as mog_work said, 4 letters |
| 01:51.04 | litage | if there's no difference, why are they 2 separate projects? |
| 01:51.14 | Qwell | read the "About OpenSER" link |
| 01:51.18 | mog_work | one group was upset with the other |
| 01:53.10 | litage | so at the moment, they're the same, but later on they may diverge? |
| 01:54.21 | _Sam-- | at what number of users does something like ser start be valuable? |
| 01:54.39 | litage | _Sam--: 150-200= |
| 01:54.43 | litage | s/=/+/ |
| 01:56.27 | JamesDotCom | litage: openser basically took a branch of ser, and started working on it their own way, both have continued development and now have a few differences |
| 01:56.37 | JamesDotCom | both do the same thing |
| 01:56.55 | JamesDotCom | the syntax is a little different, modules are all pretty much the same, with a few differences |
| 01:57.05 | JamesDotCom | imo, openser is easier for a beginner these days |
| 01:57.16 | *** join/#asterisk tuxinator_linux (n=tuxinato@m110e36d0.tmodns.net) |
| 01:57.22 | JamesDotCom | and onsip.org has one of the best tutorials for ser/openser |
| 01:57.41 | litage | JamesDotCom: but they both have 99% of the same functionality? |
| 01:57.49 | JamesDotCom | exactly |
| 01:58.18 | CaT[tm] | bleh. back to AMP for me. |
| 01:58.37 | JamesDotCom | _Sam--: a sip proxy is valuable whenever you're using sip, not just to do with the amount of users |
| 01:58.45 | QbY | CaT[tm]: No.. Why do you wanna do that? |
| 01:58.47 | [av]bani | wow... ubuntu is jacked.... |
| 01:58.51 | JamesDotCom | unfortunately, most dont understand how to configure them |
| 01:58.56 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:59.13 | JamesDotCom | or bother to read a RFC 3261 |
| 01:59.14 | *** join/#asterisk riddlebox (n=blah@24-171-11-166.dhcp.stls.mo.charter.com) |
| 01:59.15 | CaT[tm] | need a gui interface for others to use. can't find one complete enough. :/ |
| 01:59.45 | QbY | CaT[tm]: That's why I left AMP.. So many things I wanted to do and couldn't.. |
| 02:00.15 | *** join/#asterisk relyuhcs (n=Bosco@c-69-247-188-78.hsd1.al.comcast.net) |
| 02:00.33 | Ariel_ | QbY, so did you make your own. Or did you find one you like... GUI that is. |
| 02:00.36 | CaT[tm] | unfortunately it seems good enough (used it from AAH before). the only other thing that looked good was voiceone but there's no voicemail support in it. |
| 02:00.38 | tuxinator_linux | [av]bani, wants jacked with ubuntu? |
| 02:00.46 | _Sam-- | JamesDotCom : in a small office setting of 20 SIP , what would be the advantage of using a proxy? |
| 02:00.56 | _Sam-- | er..20 SIP clients / phones |
| 02:00.56 | tuxinator_linux | s/wants/whats |
| 02:01.10 | QbY | Ariel_: I didn't make my own.. I have written a few scripts for what we do.. |
| 02:01.23 | QbY | i do most of my stuff my hand.. |
| 02:01.38 | Ariel_ | ahh same here |
| 02:01.46 | tuxinator_linux | jbot, are you sleeping? |
| 02:02.00 | Ariel_ | ~weather ktmb |
| 02:02.13 | Ariel_ | looks like he is working |
| 02:02.20 | tuxinator_linux | hmm, doesn't do seen and spelling |
| 02:02.26 | JamesDotCom | _Sam--: the only one i need ever mention, it means asterisk doesn't need to touch everything |
| 02:02.38 | QbY | Ariel_: Granted, I loved AMP.. But I was needing some other stuff that AMP just couldn't do.. And when I'd modify, then switch to AMP, all of my stuff was killed.. And having to remember which file I could touch and which one I couldn't with AMP.. Got too confusing.. So I just went to VI |
| 02:02.41 | JamesDotCom | and can stick to being a pbx |
| 02:02.48 | tuxinator_linux | Ariel_, I think jbot likes you, maybe even has a crush on you |
| 02:02.49 | g4m | my zttest scores are 99.97 yet i still seem to be getting choppy meetme rooms? any ideas? |
| 02:03.07 | _Sam-- | but what is the ADVANTAGE of that in a small deployment...my asterisk box is available to do that for the 20 clients it handles |
| 02:03.17 | Ariel_ | ~docs |
| 02:03.18 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 02:03.30 | tuxinator_linux | ~docs |
| 02:03.31 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 02:03.39 | tuxinator_linux | there we go, oh well |
| 02:04.21 | JamesDotCom | _Sam--: it means you're not relying on asterisk for handling sip, whether it's the signalling, the media or both |
| 02:04.58 | _Sam-- | i dont see the advantage for small installations though...why do i care if asterisk is handling the signalling or both for 20 clients? |
| 02:05.00 | _Sam-- | thats all |
| 02:05.13 | Ariel_ | stun server is more of what is needed if you have nat issues |
| 02:06.01 | Ariel_ | _Sam--, are your user are mostly inside your network switches or out on the internet |
| 02:06.18 | JamesDotCom | _Sam--: resources, stability, rfc compliance, you dont even need the asterisk machine there |
| 02:07.04 | _Sam-- | i dont need it there now...but i dont want them have to rely on hosted PBX |
| 02:07.05 | FuriousGeorge | hey all |
| 02:07.16 | _Sam-- | Ariel_: in this case, these clients are all on 1 internal net |
| 02:07.24 | Ariel_ | JamesDotCom, for small setups you really don't need a proxy.. Belive me. |
| 02:07.37 | Ariel_ | _Sam--, then you don't need a proxy. |
| 02:07.41 | JamesDotCom | haha |
| 02:07.43 | JamesDotCom | that's fine |
| 02:07.46 | JamesDotCom | you do it your way |
| 02:07.48 | JamesDotCom | i'll do it mine |
| 02:07.52 | FuriousGeorge | i got this curious issu where oneof my voicemail boxes' greetings have disappeared and been replaced by the temp default (allison), and none of my mailboxes have the ability to change their voicemail recording in advanced options |
| 02:08.07 | _Sam-- | im not saying i need it one way or another...im just trying to understand |
| 02:08.15 | _Sam-- | at what point it becomes valuable |
| 02:08.22 | _Sam-- | and which people use it and why |
| 02:08.37 | _Sam-- | thats all..im not saying your way of doing things isnt right or good for you |
| 02:08.41 | [av]bani | oh dear sam being argumentative again? :) |
| 02:08.42 | JamesDotCom | it's valuable because it's a sip proxy, instead of some all-encompassing pbx |
| 02:08.50 | JamesDotCom | i was mostly saying that to ariel |
| 02:08.55 | Flyboy-SR22 | Hey Everyone |
| 02:08.59 | relyuhcs | Hey |
| 02:09.00 | _Sam-- | its all good, i just like learning |
| 02:09.10 | Ariel_ | _Sam--, it depends on the setup. I don't use proxy unless there are issues with b/w and locations. But I would not worried with anything less them 75 or 100 users. |
| 02:09.29 | litage | JamesDotCom: onsip.org says that SER doesn't know when a call is taking place. if that's true, how does SER generate CDRs? |
| 02:10.13 | JamesDotCom | yeah, that's cool... i found that as i read the sip rfc more often, i started disliking asterisk more for my purposes |
| 02:10.28 | _Sam-- | and when you hit 75-100 users what becomes the main issue that makes a proxy become more efficient/better/etc...the registrations? |
| 02:10.28 | Ariel_ | JamesDotCom, when you start adding others in the path with smaller installation it's wasteful. Asterisk can handle most sub 100 user setups just fine. |
| 02:10.42 | JamesDotCom | litage: there's ways to do it based off the sip messages, but i do it where it matters, from the gateways |
| 02:10.51 | Ariel_ | not a proxy but stun if your if your doing nat |
| 02:11.09 | *** part/#asterisk relyuhcs (n=Bosco@c-69-247-188-78.hsd1.al.comcast.net) |
| 02:11.15 | palomiux | anyone knows about asterisk home? |
| 02:11.18 | palomiux | what about asterisk live? |
| 02:11.38 | _Sam-- | palomiux: most of the people here just use 'pure' asterisk (i think) |
| 02:11.40 | Flyboy-SR22 | Got a question - I have been unis * for over a year and I am looking for a GREAT web interface - I know there is alist of them on the wiki, but I would like someopinions on what people have been using and what they think about it.. |
| 02:11.47 | Ariel_ | asterisk@home works well. there is a location here for them #amportal |
| 02:12.01 | _Sam-- | Flyboy-SR22 : i dont think one exists, at least not in the wild for free, that ive found. |
| 02:12.09 | _Sam-- | alot of people seem to be working on that though |
| 02:12.22 | Flyboy-SR22 | :-) _Sam-- I don't mind paying for one that is good |
| 02:12.30 | palomiux | Ariel: What about its features against "pure" asterisk? |
| 02:12.32 | Flyboy-SR22 | :-) _Sam-- Finding one is a differnet matter |
| 02:12.38 | Ariel_ | for the ones out there the better one is amp but you have limitations with all of them |
| 02:12.53 | _Sam-- | what are your main needs for a GUI...what do you need to configure? |
| 02:12.54 | Ariel_ | palomiux, it's pure asterisk with a gui |
| 02:13.05 | palomiux | cool! |
| 02:13.14 | Ariel_ | in my view if you can edit with vi you can do your own custom.conf files |
| 02:13.20 | Flyboy-SR22 | seems the people that have them bundle them with their hardware and software (Fonality and SwitchVox being too examples) |
| 02:13.25 | Ariel_ | I have amp running with lots of customers just fine |
| 02:13.46 | litage | JamesDotCom: you don't use your ser(s) to terminate calls? |
| 02:13.51 | palomiux | Ariel, do you know about an asterisk that has a Flash interface? |
| 02:13.53 | Ariel_ | fonality has a nice setup. But I don't like that you have to use there portal for everything... that is at there site |
| 02:13.58 | _Sam-- | i dont think ser CAN terminate calls |
| 02:13.59 | _Sam-- | its a proxy |
| 02:14.00 | Flyboy-SR22 | I am trying to get * systems into smaller businesss and when they hear Linex they run screaming the other direction |
| 02:14.02 | JamesDotCom | litage: how do you mean? |
| 02:14.06 | JamesDotCom | exactly |
| 02:14.09 | JamesDotCom | it doesnt touch any media |
| 02:14.13 | JamesDotCom | just signalling |
| 02:14.25 | Ariel_ | palomiux, a@h has flash operator panel with it's install |
| 02:14.28 | Ariel_ | so does amp |
| 02:14.36 | palomiux | what is "amp"? |
| 02:14.46 | palomiux | sorry, newbie at asterisk |
| 02:14.50 | JamesDotCom | i have audiocodes isdn gateways terminating calls |
| 02:14.52 | Ariel_ | amp = Asterisk Management Portal |
| 02:15.15 | *** join/#asterisk robin_sz (n=yeah@host-212-18-247-190.static.mailbox.co.uk) |
| 02:15.15 | _Sam-- | where is Sao Tome and Principe? :) |
| 02:15.19 | robin_sz | meep? |
| 02:15.27 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 02:15.30 | litage | JamesDotCom: a sip phone registers with ser, then places a call. ser forwards/terminates the call to the appropriate destination |
| 02:15.32 | Ariel_ | I have used SER as a proxy and it works. But it's for large setups not very useful for normal use. |
| 02:15.59 | Ariel_ | normal / small biz pbx |
| 02:16.03 | _Sam-- | JamesDotCom: using canreinvite = yes whenever possible doesnt get close the same effect as ser for a small deployment? |
| 02:16.18 | robin_sz | sigh ... upgrading that GXP2000 firmware .. big mistake |
| 02:16.25 | _Sam-- | no way...good choice. |
| 02:16.32 | _Sam-- | i just upgraded about 25 today |
| 02:16.47 | robin_sz | _Sam--: you WILL regret that |
| 02:16.53 | _Sam-- | no way |
| 02:17.01 | _Sam-- | my desk phone has been upgraded since sat |
| 02:17.04 | _Sam-- | its fine, better than ever. |
| 02:17.15 | _Sam-- | what is your issue? |
| 02:17.19 | robin_sz | I have 2 here ... displays go blank .. need power off once a day |
| 02:17.21 | palomiux | Ariel: amp is an add on for asterisk? |
| 02:17.25 | *** join/#asterisk juice (i=1000@209.33.108.78) |
| 02:17.29 | _Sam-- | did you get the new firmware from 2 days ago? |
| 02:17.34 | robin_sz | yip |
| 02:17.39 | _Sam-- | < i read about that problem>...but havent had it myself. |
| 02:17.43 | _Sam-- | on 0 of 25 phones |
| 02:17.47 | Ariel_ | palomiux, yes it's an addon but not support by asterisk people |
| 02:18.00 | robin_sz | and MWI does not work |
| 02:18.10 | _Sam-- | does for me...i came into today to message |
| 02:18.14 | _Sam-- | and stutter tone |
| 02:18.19 | palomiux | Ariel: I see, what do you recommend me? |
| 02:18.45 | Flyboy-SR22 | robin_sz, what is MWI |
| 02:18.50 | Ariel_ | palomiux, if your starting I would say look at Asterisk@home and start learning the setup yourself after it's done |
| 02:18.51 | robin_sz | im not seeing SUBSCRIBE requests at all |
| 02:18.57 | _Sam-- | message waiting indicator |
| 02:19.01 | Flyboy-SR22 | ah |
| 02:19.03 | Flyboy-SR22 | thanks |
| 02:19.05 | litage | JamesDotCom: is serweb compatible with openser/ |
| 02:19.26 | palomiux | Ariel? |
| 02:19.30 | _Sam-- | how many gxps you have? |
| 02:19.33 | robin_sz | the blank displays is a PITA .. is ok for a day or so .. then dead, sometimes only lasts an hour |
| 02:19.35 | robin_sz | 2 |
| 02:19.50 | _Sam-- | there was something regarding the older MAC addresses |
| 02:19.52 | _Sam-- | did you see that? |
| 02:19.54 | palomiux | Ariel: I see, what do you recommend me? |
| 02:19.56 | robin_sz | nope |
| 02:20.05 | robin_sz | can I get a new MAV address? |
| 02:20.10 | robin_sz | MAC |
| 02:20.12 | _Sam-- | posting you a PM |
| 02:20.27 | _Sam-- | i bet you have the old mac |
| 02:20.32 | _Sam-- | check your phone status for the mac number |
| 02:20.49 | _Sam-- | sounds like yours is diff problem...wondering thoug |
| 02:21.14 | _Sam-- | i checked all my mac's before i upgraded |
| 02:22.06 | robin_sz | this is completely blank screen ... |
| 02:22.28 | _Sam-- | yeah i see the notes on the tiki page |
| 02:22.44 | robin_sz | I also have a problem when upgrading, that it continually downloads, installs and reboots .. I have to remove the http server directory (or rename it anyway) to stop it ... |
| 02:22.55 | _Sam-- | yeah thats normal |
| 02:22.58 | _Sam-- | BUG FIXED IN 1.0.2.7 ALPHA. NEXT BETA RELEASE SHOULD NOT HAVE THIS BUG. - thetatag |
| 02:23.10 | _Sam-- | bani and myself reported it a while ago |
| 02:23.18 | _Sam-- | its only on apache httpd |
| 02:23.20 | _Sam-- | bani figured that out |
| 02:23.26 | robin_sz | coo |
| 02:23.37 | robin_sz | "only" on apache httpd ... |
| 02:23.38 | palomiux | Guys, what is best? amp or asterisk@home_?? |
| 02:23.42 | robin_sz | like theres any other ;) |
| 02:23.44 | _Sam-- | yeah who runs apache? :) |
| 02:24.07 | trelane | palomiux, reread the features of A@H? |
| 02:24.07 | _Sam-- | there's been alot of action on this page, might be good for you: |
| 02:24.08 | _Sam-- | http://www.voip-info.org/tiki-index.php?page=GXP-2000 |
| 02:24.09 | robin_sz | you shold try my Zyxel wifi phone .. thats a laugh |
| 02:25.35 | _Sam-- | you should check your mac on your phone(s) anyway |
| 02:25.39 | _Sam-- | can you do it ? |
| 02:25.48 | _Sam-- | someone says the screen blnaking may be related |
| 02:26.38 | robin_sz | ummm .. maybe .. wait .. |
| 02:26.56 | robin_sz | its at home, im at work .. but maybe my vpn will let me .. |
| 02:27.10 | _Sam-- | when did you get it? |
| 02:27.55 | robin_sz | <PROTECTED> |
| 02:27.58 | robin_sz | yay |
| 02:28.07 | _Sam-- | lucky you |
| 02:28.13 | _Sam-- | you're screwed |
| 02:28.14 | _Sam-- | This display problems occur on all GXP2000 devices with MAC addresses 000b82-03xxxx |
| 02:28.32 | robin_sz | what do I win? |
| 02:28.42 | _Sam-- | well it sounds like they are aware of the problem |
| 02:28.49 | _Sam-- | i bet you will win some working firmware in a few days |
| 02:28.57 | _Sam-- | especially since you cant downgrade :) |
| 02:28.59 | robin_sz | I downgraded to 1.0.2.3 |
| 02:29.11 | _Sam-- | that is ok? |
| 02:29.15 | robin_sz | mmm.... |
| 02:29.17 | _Sam-- | supposedly that still has the problem |
| 02:29.18 | robin_sz | not really |
| 02:29.22 | _Sam-- | but it doesnt occur as frequently |
| 02:29.32 | robin_sz | going to try 1.0.2.6 again I think .. |
| 02:29.49 | robin_sz | hey, they put the warning back at the top of the page after I moaned :) |
| 02:30.17 | _Sam-- | you got a warning added to the page, i got a feature i wanted added to the firmware..its a real win/win :) |
| 02:30.18 | *** join/#asterisk ThaZZa_Work (n=me@203.80.44.200) |
| 02:30.32 | robin_sz | feature? |
| 02:30.37 | robin_sz | dancing girls? |
| 02:30.53 | _Sam-- | my request for this feature: |
| 02:30.54 | _Sam-- | Added allow disable miss-call features as per-account setting, changing this setting takes immediate effect without reboot. |
| 02:31.02 | robin_sz | the "whit on black" menus doesnt really work |
| 02:31.04 | _Sam-- | the missed call log was driving my guys nuts |
| 02:31.44 | robin_sz | time to go home ... the laser has finished |
| 02:32.08 | _Sam-- | have fun, and good luck |
| 02:32.15 | robin_sz | thanks. |
| 02:33.15 | Flyboy-SR22 | Does A@H support ACD does anyone know..? |
| 02:33.28 | Flyboy-SR22 | or is more a home type of system as opposed to the business system |
| 02:33.43 | Dr-Linux | anybody have example for this ? |
| 02:33.45 | Dr-Linux | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VMAuthenticate |
| 02:36.13 | _Sam-- | Dr-Linux: you're not all done yet? :) |
| 02:36.21 | austinnichols10 | flyboy-sr22: latest version of aah supports acd based on several distribution methods |
| 02:36.43 | ravsi | what is it called when you have 1 main number which people call and you hand those calls off to back-up numbers to allow other calls on the main # |
| 02:36.52 | litage | is serweb compatible with openser? |
| 02:37.00 | Flyboy-SR22 | austinnichols10, Thanks - I am going to download the live CD and give it a try :-) |
| 02:37.20 | austinnichols10 | flyboy-sr22: check out mundy.org/blog before getting started - excellent tutorial |
| 02:37.30 | *** join/#asterisk Majestik (n=Majestik@S0106000024c058cc.ed.shawcable.net) |
| 02:37.30 | RoyK | are anyone working on SOAP for asterisk? |
| 02:37.40 | Flyboy-SR22 | austinnichols10, Thanks - I will look at that now.. |
| 02:37.57 | austinnichols10 | royk: there's some xmlrpc stuff out there |
| 02:38.03 | Dr-Linux | _Sam--: i'm stopped doing AGI stuff yet |
| 02:38.30 | RoyK | austinnichols10: iirc soap is an xmlrpc implementation |
| 02:39.36 | *** join/#asterisk ThaZZa_Work (n=me@203.80.44.200) |
| 02:40.21 | ThaZZa_Work | ~seen thazza |
| 02:40.27 | jbot | thazza <n=thazza@229.9.233.220.exetel.com.au> was last seen on IRC in channel #asterisk, 15h 3s ago, saying: 'mut: Much better to put caster suger mixed with cooking oil.'. |
| 02:41.29 | _Sam-- | why do people choose *@home? just because its easy to install? |
| 02:42.29 | austinnichols10 | sam: cuts way down on setup time and has all of the gui stuff to make it an appliance |
| 02:42.42 | *** join/#asterisk juanjoc (n=jcomella@222-32-235-201.fibertel.com.ar) |
| 02:42.58 | austinnichols10 | sam: but the downside is that a lot of people probably don't go much deeper than gui config |
| 02:43.15 | [av]bani | _Sam--: i have come to the conclusion only 'tards use a@h. |
| 02:43.39 | _Sam-- | basically, if you didnt know anything about linux or how to install it...then i guess @home make sense |
| 02:43.52 | austinnichols10 | sam: I found it a great way to cut down on the inital learning curve. I'm starting to move past what you can do with AMP |
| 02:43.54 | _Sam-- | or if you just needed a really quick pbx? im still not even sure |
| 02:44.08 | [av]bani | _Sam--: its not even good at that. there are better distros |
| 02:44.16 | _Sam-- | i have my own base setup in a file system for future installs |
| 02:44.19 | *** join/#asterisk rmorris (n=rmorris@d221-85-117.commercial.cgocable.net) |
| 02:44.25 | _Sam-- | i can just unzip it, make it bootable, and its good to go |
| 02:44.27 | austinnichols10 | which distro is better at this point? |
| 02:44.36 | [av]bani | anything? a@h is pretty crappy |
| 02:44.40 | *** join/#asterisk T42X (n=T42X@193.219.62.88) |
| 02:44.47 | rmorris | Anyone use a Sipura-3000 for outgoing calls from Asterisk? |
| 02:44.53 | [av]bani | astlinux maybe |
| 02:45.01 | _Sam-- | what linux is @home installed with? |
| 02:45.04 | _Sam-- | er what distro |
| 02:45.04 | austinnichols10 | astlinux = no gui |
| 02:45.11 | austinnichols10 | centos 4 |
| 02:45.30 | [av]bani | amp != gui |
| 02:45.39 | palomiux | ?? |
| 02:45.40 | _Sam-- | austinnichols10: why did you originally choose @home? |
| 02:45.45 | austinnichols10 | no gui will eliminate 95% of the people coming from win |
| 02:45.49 | _Sam-- | the configuration isnt any easier than regular asterisk |
| 02:45.59 | _Sam-- | and you could have still used amp |
| 02:46.08 | austinnichols10 | mainly because I wanted to get a box up and running so I could check it out without having to spend a week doing RTFMs |
| 02:46.19 | _Sam-- | fair enough |
| 02:46.24 | _Sam-- | im just trying to understand |
| 02:46.31 | _Sam-- | you were pretty well versed with linux before that? |
| 02:46.32 | [av]bani | no, youre just trying to be argumentative :) |
| 02:46.38 | _Sam-- | but just not with *? |
| 02:46.44 | austinnichols10 | I bought a sipura 841 and slapped the cd in a spare box in the office and started playing around to see how the system worked and what it was capable of |
| 02:46.46 | _Sam-- | (i always am..argumentative that is) |
| 02:47.06 | _Sam-- | but i still havent thought of my killer xml app |
| 02:47.08 | _Sam-- | :) |
| 02:47.15 | austinnichols10 | I *used* to be versed with linux but it had been quite a while. |
| 02:47.42 | austinnichols10 | So I could move around the file system, edit files, etc. When it came to compiling I would need to go back and look at instructions. |
| 02:47.50 | _Sam-- | austinnichols10: and now you find yourself already against the limits of what you can do with the gui? like you want to do more things, but either the gui cant do it or you dont know how? |
| 02:48.18 | [av]bani | amp is really limiting... |
| 02:48.32 | [av]bani | its like a fisher price "my first pbx" with working horn |
| 02:48.43 | austinnichols10 | Not exactly. I can handle all of the day-to-day stuff at the office. I ended up going into the config files to add my callback (ringy), a tie to my house system, etc. |
| 02:49.01 | austinnichols10 | I think the next box I'll build will run astLinux |
| 02:49.13 | _Sam-- | i tried that, wasnt that impressed |
| 02:49.23 | _Sam-- | make your own custom setup |
| 02:49.42 | _Sam-- | to install what @home provides, isnt that hard |
| 02:49.43 | austinnichols10 | the other nice thing about having AMP is that I can delegate responsibility to non-linux users at the office and that way I don't have to do absolutely everything for them |
| 02:49.52 | _Sam-- | 1) install linux...2) install asterisk 3) install AMP |
| 02:49.54 | [av]bani | it isnt hard to install amp |
| 02:49.57 | file | _Sam--: I did what you asked |
| 02:50.02 | [av]bani | it practically installs itself |
| 02:50.10 | _Sam-- | file: THANKS. |
| 02:50.36 | _Sam-- | granted, @home comes with some other agi and stuffs |
| 02:51.06 | austinnichols10 | I agree that installing linux isn't bad anymore. When you get over to installing/configuring apache I find it a bit harder. Integrating php takes time if you haven't done it before. |
| 02:51.30 | _Sam-- | thats an honest answer, i take it for granted since i do it all day long. |
| 02:51.44 | austinnichols10 | Now that I'm more familiar I would be much more comfortable in setting up my own build |
| 02:51.47 | _Sam-- | but who wants to spend an hour or hours learning how to configure and install apache....me personally i can do it in 3 minutes |
| 02:51.56 | austinnichols10 | exactly |
| 02:52.07 | _Sam-- | but thats fair..and i can understand why you would pick that |
| 02:52.18 | austinnichols10 | In the first place I just wanted to find out a bit about asterisk |
| 02:52.19 | _Sam-- | what do you like about astlinux? |
| 02:52.26 | austinnichols10 | embedded |
| 02:52.39 | austinnichols10 | I don't really want to have another full-blown box at the house |
| 02:52.40 | _Sam-- | well not embedded...but maybe solid state |
| 02:52.53 | _Sam-- | i guess you could embed it, if youhad an embedded device |
| 02:53.04 | _Sam-- | i built my own solid state version |
| 02:53.04 | austinnichols10 | sorry, yes - technically not embedded |
| 02:53.09 | _Sam-- | i started with astlinux |
| 02:53.10 | _Sam-- | didnt like it |
| 02:53.13 | _Sam-- | made my own on compact flash |
| 02:53.26 | _Sam-- | i think mines better, but you'd have to know linux to use it |
| 02:53.43 | austinnichols10 | I've also thought about just installing it on my linksys but from what I've read it's not really all that practical |
| 02:55.00 | austinnichols10 | I think there's a great opportunity for the community with AAH. I ended up finding a few people who offer paid support for AAH. I ended up buying a couple of hours before I set up my first system (my office) just so that I knew someone had my back in case of a problem. |
| 02:55.20 | [av]bani | yay http bug fixed |
| 02:55.26 | _Sam-- | people got your back here...my first asterisk installation, i came here, and paid someone to do ti |
| 02:55.27 | _Sam-- | it |
| 02:55.31 | _Sam-- | zoa as a matter of fact |
| 02:55.40 | austinnichols10 | Now I do just about everything myself and then get the rest from IRC / formus, etc. |
| 02:55.56 | _Sam-- | if you want pay, i think you can find alot of qualified talent here as well :) |
| 02:56.17 | austinnichols10 | yes and no regarding here. you can't just come in and ask a noob question and not expect to get slapped around a bit |
| 02:56.32 | _Sam-- | especialy if its about @home |
| 02:56.36 | austinnichols10 | right |
| 02:56.37 | _Sam-- | "They have their own channel" |
| 02:56.45 | austinnichols10 | how's that? |
| 02:56.54 | [av]bani | all of the asterisk distros suck more or less though, i've been contemplating my own |
| 02:57.02 | _Sam-- | bani im in |
| 02:57.07 | austinnichols10 | I see people get referred to amp all the time, but I think that's kind of worthless |
| 02:57.08 | _Sam-- | lets do it..i have a good base already :) |
| 02:57.12 | Qwell | [av]bani: tried astlinux? |
| 02:57.31 | austinnichols10 | I would love to see an alternative to AAH. There's a bunch of stuff that could be improved. |
| 02:57.36 | [av]bani | Qwell: not yet, it's next on my list |
| 02:57.41 | [av]bani | a@h is made of poo though |
| 02:57.42 | _Sam-- | its junk |
| 02:57.45 | xachen | heh |
| 02:57.48 | xachen | all * distros suck |
| 02:57.52 | _Sam-- | astlinux was good if you only had 100megs to work with |
| 02:57.57 | _Sam-- | but now 1G cards are the norm |
| 02:57.58 | _Sam-- | and larger |
| 02:58.03 | [av]bani | well, what else do you need for * ? |
| 02:58.10 | xachen | The only way your going to set something like * perfect is if you do it manually |
| 02:58.39 | _Sam-- | for installs, when i take a server to a client..i like to have X |
| 02:58.41 | _Sam-- | with some tools |
| 02:58.47 | _Sam-- | that way i am self contained |
| 02:58.48 | [av]bani | ugh |
| 02:58.58 | _Sam-- | you dont have to run it...if you dont need it |
| 02:59.02 | [av]bani | whats wrong with console... |
| 02:59.10 | xachen | console is the only way :) |
| 02:59.12 | _Sam-- | what if you need web access |
| 02:59.17 | _Sam-- | like you need to find a bug report |
| 02:59.19 | xachen | lynx :) |
| 02:59.21 | _Sam-- | no way |
| 02:59.25 | Qwell | Then you use another machine |
| 02:59.30 | xachen | yeah |
| 02:59.32 | austinnichols10 | here's the thing, so far aah is the only distro that I've found that is fairly close to being an appliance |
| 02:59.35 | Qwell | asterisk is a pbx, not a desktop env |
| 02:59.35 | _Sam-- | the extra overhead to run X is worth it..i like to be self containned and not displace my clients |
| 02:59.43 | xachen | X is bloated |
| 02:59.46 | xachen | rather it bloats a box |
| 02:59.48 | _Sam-- | so logout when you leave |
| 02:59.50 | austinnichols10 | if there was something better I would definitely be there. |
| 02:59.53 | _Sam-- | it runs like 2 processes |
| 03:00.01 | [av]bani | links |
| 03:00.01 | *** join/#asterisk stormfr (n=StorM@sgc91-2-82-237-76-2.fbx.proxad.net) |
| 03:00.03 | _Sam-- | i like having tools |
| 03:00.10 | Qwell | _Sam--: use another machine |
| 03:00.14 | _Sam-- | like having gaim, mozilla, my mp3 player etc :) |
| 03:00.22 | [av]bani | mp3 player... at customer site ... |
| 03:00.23 | xachen | thats just retarded |
| 03:00.24 | _Sam-- | i dont want to...the one im working on is fine |
| 03:00.33 | xachen | Your there to work |
| 03:00.36 | [av]bani | _Sam-- just likes to play :) |
| 03:00.41 | [av]bani | fess up :) |
| 03:00.44 | xachen | not enjoy yourself and talk to your friend that lives down the block |
| 03:00.46 | _Sam-- | im there to setup the box, if it aint setup already |
| 03:00.59 | _Sam-- | they arent paying me by the hour, what do they care...i made the deal to install a PBX |
| 03:01.06 | _Sam-- | so when i go, i like to be able to install the pbx ! |
| 03:01.16 | _Sam-- | and do what i need to do without a million computers and displacing others |
| 03:01.22 | xachen | yeah |
| 03:01.24 | stormfr | hello, i have daily chan_sip stop responding by said "grab the lock". Is there a way to identify where the lock ? (realtime mysql + addons/head or 1.2.x) |
| 03:01.29 | xachen | that extra laptop is going to set you off... |
| 03:01.40 | _Sam-- | why do i need the laptop? |
| 03:01.48 | _Sam-- | the server has all the tools i need |
| 03:01.53 | Qwell | servers don't run X |
| 03:01.56 | Qwell | That's all there is to it |
| 03:01.59 | _Sam-- | they dont when i leave |
| 03:02.07 | Qwell | They shouldn't ever |
| 03:02.16 | xachen | indeed |
| 03:02.28 | xachen | and if a person can't run a console and in installing a PBX like * |
| 03:02.29 | xachen | well.... |
| 03:02.30 | [av]bani | i love how apple thinks itunes belongs on every headless server |
| 03:02.32 | xachen | yeah |
| 03:02.53 | [av]bani | i keep removing it and apple keeps reinstalling it without asking me |
| 03:03.06 | [av]bani | i guess steve knows whats best for me |
| 03:04.26 | _Sam-- | Qwell: id like to hear your rationale why a server should NEVER run x |
| 03:04.32 | _Sam-- | because it takes extra ram? |
| 03:04.39 | Qwell | _Sam--: There are MANY reasons. |
| 03:04.42 | Qwell | resources are one |
| 03:04.46 | Qwell | there are also security implications |
| 03:04.54 | xachen | extreme security implications |
| 03:04.55 | _Sam-- | as with any service that allows remote connections |
| 03:04.58 | Qwell | there is no reason a server should ever have X |
| 03:05.09 | Qwell | _Sam--: Would you run telnet on a server? |
| 03:05.22 | _Sam-- | if i needed it, i would...but i dont. |
| 03:05.27 | Qwell | You never need it |
| 03:05.29 | _Sam-- | telnetd? |
| 03:05.30 | [av]bani | historically x has been an endless security hazard :/ |
| 03:05.31 | Qwell | there is always ssh |
| 03:05.32 | Qwell | yes, telentd |
| 03:05.34 | Qwell | telnetd |
| 03:05.45 | _Sam-- | if i had a client without ssh or something...who knows |
| 03:05.51 | _Sam-- | or was testing something |
| 03:05.56 | _Sam-- | i could make good firewall rules |
| 03:05.59 | xachen | telneted |
| 03:06.00 | _Sam-- | and i wouldnt worry about it really |
| 03:06.01 | xachen | oh god |
| 03:06.05 | xachen | You'll never work for me |
| 03:06.18 | _Sam-- | good, ive never worked for anyone in my life :) |
| 03:06.19 | [av]bani | i dont think hed want to :) |
| 03:07.10 | _Sam-- | but while telnet may be a problem if done wrongly....if you secured your telnetd...and you were confident in your skills what is the problem? |
| 03:07.20 | Qwell | telnet can't be secured |
| 03:07.23 | [av]bani | heh |
| 03:07.35 | Qwell | it's impossible |
| 03:07.38 | _Sam-- | if you cant access it, and it runs from inted not as a daemon... |
| 03:07.40 | _Sam-- | how do you get it? |
| 03:07.44 | [av]bani | well, you could do telnet-tls |
| 03:07.50 | [av]bani | but then why not use ssh instead |
| 03:07.50 | Qwell | [av]bani: That isn't telnet |
| 03:07.58 | Qwell | that's telnet-tls ;] |
| 03:08.00 | [av]bani | Qwell: yes it is, its telnet over tls |
| 03:08.03 | _Sam-- | how would access my telnetd if i restrict access by IP |
| 03:08.04 | litage | onsip.org's getting started document is in sgml format. how read this? |
| 03:08.04 | [av]bani | like pop3 over tls |
| 03:08.18 | Qwell | _Sam--: it isn't just about accessing it |
| 03:08.25 | austinnichols10 | sam: ip spoofing |
| 03:08.25 | _Sam-- | its not running as a daemon |
| 03:08.29 | Qwell | passwords are sent in cleartext |
| 03:08.35 | _Sam-- | and you would have know what IPs were allowed |
| 03:08.39 | _Sam-- | you couldnt spoof it |
| 03:08.39 | Qwell | doesn't matter |
| 03:08.54 | _Sam-- | what doesnt matter...tell me how you are going to hack my telne |
| 03:08.54 | austinnichols10 | if you can sniff the traffic then you could spoof it |
| 03:08.55 | _Sam-- | t |
| 03:09.10 | Qwell | I'll sniff the telnet traffic... |
| 03:09.16 | Qwell | then I'll walk up to the keyboard, and type in the root password |
| 03:09.18 | _Sam-- | sure you will...what, after you get on my box or router? |
| 03:09.22 | xachen | get your password. and since your passwords will be the same.... haha |
| 03:09.30 | Qwell | xachen: that too :P |
| 03:09.37 | _Sam-- | i will give you 1000 dollars if you can. |
| 03:09.40 | _Sam-- | i'll start telnetd right now. |
| 03:09.45 | Qwell | root pw for 1.5 = blah, root pw for 1.6 also = blah :p |
| 03:09.48 | _Sam-- | you give me 1000 if you cant |
| 03:09.53 | Qwell | ssh 1.6, root, blah, telnet 1.5 root, blah |
| 03:10.00 | *** join/#asterisk techie (i=gus@antibala.com) |
| 03:10.07 | _Sam-- | i will put it right now in a paypal escrow |
| 03:10.17 | austinnichols10 | I'm running a IPS at the data center and I see brute-force attempts like that all day long |
| 03:10.24 | Qwell | _Sam--: not from here, no |
| 03:10.27 | _Sam-- | i owned an ISP for 10 years |
| 03:10.37 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 03:10.43 | WasPhantom | my condolences to you _Sam-- heh |
| 03:10.44 | _Sam-- | xachen same offer goes to too |
| 03:10.50 | WasPhantom | congratulations on getting out ;-) |
| 03:11.06 | xachen | I thought I'd just laugh when you do get cracked |
| 03:11.28 | _Sam-- | been 10 years...it has happened. |
| 03:11.29 | _Sam-- | once. |
| 03:11.41 | _Sam-- | live and learn. |
| 03:12.35 | WasPhantom | I'm out in 6 weeks, after 9 years |
| 03:12.58 | _Sam-- | i am not claiming that my servers are invincible (i do think they are!) but i dont think if i setup my firewalls right and my telnet/inetd right...you would have 0 chance of getting a damn thing. |
| 03:13.11 | _Sam-- | because you wouldnt be able to sniff my box or get to my telnet |
| 03:13.35 | _Sam-- | that is all |
| 03:13.41 | shido6 | USD? |
| 03:13.43 | _Sam-- | maybe you could, maybe you couldnt |
| 03:13.50 | austinnichols10 | sam: as long as that traffic doesn't pass outside of your control, yeah |
| 03:14.19 | *** join/#asterisk Cadu20 (n=Cadu83@201-3-232-23.fnsce7004.dsl.brasiltelecom.net.br) |
| 03:14.22 | Qwell | if, however, somebody was on your LAN already...telnet is pie |
| 03:14.39 | Qwell | be it a remote attacker, or a malicious employee |
| 03:14.46 | _Sam-- | if they are on my lan, my telnet will be the last or close to last of my worries :) |
| 03:14.56 | Qwell | Not if they're a trusted user |
| 03:15.02 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 03:15.05 | _Sam-- | for the record, i dont run telnet :) |
| 03:15.12 | Qwell | nor was that the point |
| 03:15.17 | _Sam-- | understood |
| 03:15.23 | Qwell | the point was that running X on a server is as bad as running telnet |
| 03:15.32 | Qwell | It just isn't something you do |
| 03:15.34 | _Sam-- | only if people remotely auth? |
| 03:15.37 | Qwell | no |
| 03:15.46 | Qwell | even locally, there are very large issues |
| 03:15.54 | Cadu20 | Hi, i don´t know if this was previously asked but... there is a way to change CANREINVITE on the fly?... i mean.. based on dialed number prefix..? |
| 03:16.02 | _Sam-- | thank you for your input,..i am here to learn. |
| 03:16.15 | _Sam-- | i will do some more learning about it |
| 03:16.42 | Qwell | if people access it from the outside, OR if it runs a realtime application, it shouldn't run X |
| 03:16.51 | _Sam-- | if a * is running X on a 192 network secured from the outside by firewalls, how would someone from the outside or inside do anything? |
| 03:17.06 | Qwell | _Sam--: The security aspect isn't the only one |
| 03:17.14 | Qwell | there is a multitude of reasons not to run X on a server |
| 03:17.29 | _Sam-- | i know i ask alot of questions, its not because i am confrontational, but because im curious and want to learn and know |
| 03:17.31 | Qwell | I'll get you the top 3 reasons right now |
| 03:17.56 | tainted- | i run X in a vmware virtual machine on my windows me box |
| 03:18.04 | Qwell | tainted-: sadistic SOB |
| 03:18.08 | _Sam-- | lol |
| 03:18.16 | dudes | Nothing wrong with X |
| 03:18.21 | dudes | on a desktop ... |
| 03:18.34 | tainted- | i use unixODB to connect to my MS Access farm |
| 03:18.48 | pauldy | just don't install it on a P2 400 and expect asterisk to produce smooth audio |
| 03:19.11 | dudes | * does alright on a XP 2800 /w X running =) |
| 03:19.24 | Cadu20 | thus.. whats the point on not using vi?... it´s so damn cool.. lol |
| 03:19.26 | Qwell | oh great |
| 03:19.28 | palomiux | Bye guys |
| 03:19.31 | palomiux | thanks for the info |
| 03:19.33 | Qwell | now I got two other people debating on why it's bad :p |
| 03:19.55 | Qwell | the 3 reasons were summed up as 1) security, 2) bloat, 3) security |
| 03:19.56 | Abydos313 | i run x on my servers. i have to. oracle needs it |
| 03:19.58 | _Sam-- | i could strip it out of my stuff super easy...i only need it for the development work |
| 03:20.08 | _Sam-- | but i like having it for installations |
| 03:20.15 | _Sam-- | in case problems come up |
| 03:20.22 | Cadu20 | About the reinvite thing... any input from you guys? |
| 03:20.32 | Qwell | _Sam--: okay, I got a real #3 |
| 03:20.35 | Qwell | _Sam--: <bougyman> you lose your geek card if you put X on your servers. |
| 03:20.40 | _Sam-- | lol |
| 03:20.44 | *** part/#asterisk palomiux (n=lecaus@200.30.160.186) |
| 03:21.02 | _Sam-- | are there any xservers that are more secure than any others? |
| 03:21.24 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 03:21.50 | Qwell | _Sam--: one that doesn't listen on tcp is a start (xorg CAN be set to not do so) |
| 03:21.53 | _Sam-- | bloat is no longer a valid reason, either with the cost of disk space so cheap |
| 03:22.17 | _Sam-- | and same for ram |
| 03:22.24 | _Sam-- | the cost of resources is cheap |
| 03:22.36 | _Sam-- | so bloat while a factor in the past, in my mind is no longer a valid factor |
| 03:22.37 | Cadu20 | _Sam--: but still, when you need to update your system.. you´ll be caught on dozens of dependencies issues... |
| 03:23.04 | litage | do you set the packetization (Eg: 20ms) on the enduser device (Eg: softphone), on *, or on SER? |
| 03:24.15 | _Sam-- | either way, thank you for taking the time to give some pointers. |
| 03:24.29 | Cadu20 | no way to change canreinvite on the fly?... |
| 03:24.44 | Cadu20 | anyone?.. . no?... :) thanks then.. |
| 03:24.49 | FuriousGeorge | whoops, i can in fact still reset my greetings, i was just looking in the wrong place. i still dont get why its using the temp mailbox instead of the messages we recorded |
| 03:24.50 | Flyboy-SR22 | with A@H can you manage fairley complet extensions.conf files..? FOr example, I have a lot of different contexts that allow and deny certain phone from dialing international, etc... |
| 03:24.50 | FuriousGeorge | -- Executing VoiceMail("Zap/5-1", "u0") in new stack |
| 03:24.50 | FuriousGeorge | <PROTECTED> |
| 03:25.10 | Qwell | FuriousGeorge: temp is always played |
| 03:25.22 | FuriousGeorge | i looked in the dir and the file that corresponds tho the message we recorded is there |
| 03:26.09 | FuriousGeorge | Qwell: ok but its allison not what we recorded. the next line is : |
| 03:26.10 | FuriousGeorge | -- Playing 'vm-intro' (language 'en') |
| 03:26.39 | *** join/#asterisk PrivalODC (i=user69@Kitchener-HSE-ppp3571800.sympatico.ca) |
| 03:26.54 | Majestik | My asterisk stoped taking incoming calls over IAX, and I can't seem to find anything in the logs indicating why.. I'm getting traffic from my provider's server just fine, |
| 03:27.04 | FuriousGeorge | and thres a 56 meg wav in the dir which has got to be the greeting we recorded, im gonna scp it over here and verify |
| 03:27.06 | _Sam-- | iax2 show registry |
| 03:27.22 | _Sam-- | make sure you are registered to your iax provider first |
| 03:27.27 | PrivalODC | Hi, I'm trying to test a PRI setup. I connected the 2 PRI on my sangoma A102 together using a PRI xover, but I always get a congestion. Any hints? |
| 03:28.00 | Qwell | FuriousGeorge: what is the filename? |
| 03:28.20 | Majestik | _Sam--: Yup, registered just fine.. I can still make outbound calls just fine, and I can call exten to exten from port to port on my pap2 |
| 03:28.40 | _Sam-- | Majestik: what do you see on your console / cli? |
| 03:29.17 | Majestik | _Sam--: A bunck of POKE/PONG/ACKs.. and a few REGAUTH and REGREQ messages.. want the details? |
| 03:29.32 | _Sam-- | on the asterisk console? |
| 03:29.38 | Majestik | yeah |
| 03:29.43 | Majestik | thast's with iax2 debug |
| 03:30.25 | _Sam-- | what is your server version? |
| 03:30.34 | Majestik | 1.2.1 |
| 03:30.48 | _Sam-- | and restarting the server has no effect? |
| 03:30.56 | Majestik | not a thing |
| 03:31.13 | _Sam-- | i ran 1.2.1 w/ iax trunks for a while...without a hiccup |
| 03:31.15 | Majestik | And.. I honestly didn't touch anything between it working, and not working :) |
| 03:31.24 | Majestik | So did I until this afteroon :) |
| 03:31.32 | _Sam-- | maybe its worth a shot at upgrading to 1.2.4? |
| 03:32.02 | _Sam-- | no firewall rules or anything different? |
| 03:32.06 | Majestik | Not a thing |
| 03:32.14 | _Sam-- | the box is behind nat/firewall? |
| 03:32.32 | Majestik | It is.. but it was working just fine. |
| 03:32.42 | Majestik | And, I can see the packets with tcpdump too.. |
| 03:32.50 | PrivalODC | Ok, for those interested, I had to use the 1st channel of both PRI only (channel 1 and 25)... |
| 03:33.01 | _Sam-- | i thought iax is udp |
| 03:33.18 | _Sam-- | not that you cant see udp with tcpdump |
| 03:33.19 | Majestik | Yup |
| 03:33.23 | _Sam-- | just caught me off gaurd :) |
| 03:33.23 | Majestik | :) |
| 03:33.50 | _Sam-- | it sounds like you've done some poking around...probably more than i can help you with! |
| 03:34.04 | _Sam-- | other than recommending to try an upgrade, i dont have anything useful to add for ya! |
| 03:34.26 | _Sam-- | what is the iax provider? |
| 03:34.27 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 03:34.30 | Majestik | I'm thinking I'd best get a hold of my service provider :) |
| 03:34.33 | Majestik | Thinktel.ca |
| 03:34.53 | *** join/#asterisk mwgbc (n=junkmail@c-67-188-17-12.hsd1.ca.comcast.net) |
| 03:34.58 | _Sam-- | its pretty easy to upgrade to 1.2.4 if you decide to try |
| 03:35.10 | FuriousGeorge | qwell, i got unavail.wav and unavail.WAV and a corresponding gsm file, same thing for busy.... i cant seem to open em with anything, but i notice its actually 556k not 56mb so i guess they must have deleted the greeting they recorded somehow or something |
| 03:35.11 | brockj49464 | Is there a way to get history of peer quality? |
| 03:35.23 | FuriousGeorge | ill just tellem they gotta record it again and they must have broke it somehow :) |
| 03:35.27 | Majestik | K, thanks _Sam-- |
| 03:35.28 | PrivalODC | Is there a way to know shich zaptel version is installed? |
| 03:35.47 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 03:35.50 | _Sam-- | Majestik: i didnk think there was a compelling reason to upgrade from 1.2.1 to 1.2.4...until i read the changelog |
| 03:35.57 | _Sam-- | there's really been a lot of fixes |
| 03:36.11 | Majestik | Yeah.. I'm thinking I'm going to get a hold of the provider first.. |
| 03:37.22 | mwgbc | Help Please. I have been running an asterisk dialer for a while now. I have been making calls just fine until a about an hour ago. The server running asterisk is a dedicated server based in florida I just started getting straight "call failed to go through reason 1" I dont think anything has changed. Any ideas where to look? |
| 03:37.52 | Qwell | mwgbc: your provider |
| 03:38.10 | mwgbc | Qwell: provider is voipjet.com |
| 03:38.16 | Qwell | call them up |
| 03:38.35 | Abydos313 | did your .25 cents run out? heh |
| 03:38.39 | Math` | lol |
| 03:38.46 | Math` | your probably out of funds yeah |
| 03:38.49 | mwgbc | Qwell: I pinged their servers fine. and I have $125.00 left on my account |
| 03:38.54 | Qwell | call anyways |
| 03:38.57 | Qwell | ping means nothing |
| 03:39.01 | Qwell | SIP isn't ICMP |
| 03:39.03 | I-MOD | maybe their shit just broke |
| 03:39.11 | mwgbc | Qwell: thanks, I'll do that. |
| 03:39.11 | [av]bani | bleah |
| 03:39.12 | dudes | callfiles? |
| 03:39.15 | file | try to make an actual call... so you get a better reason... |
| 03:39.22 | file | instead of using a call file |
| 03:39.47 | mwgbc | file: No, when I try to make a call it just times out. |
| 03:40.05 | file | how do you know it times out? what do you get? have you pastebinned it? have you done an iax2 debug? |
| 03:40.26 | file | The moral of this story is, provide as much info as you can to faciliate the diagnosis procedure |
| 03:41.32 | mwgbc | file: I'll check with the provider first and get back with what help I need after that. Thanks |
| 03:45.31 | *** part/#asterisk QbY (n=Kelvin@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 03:46.18 | PrivalODC | Anyone has a link or document that gives the differences betwen the different echo canceler? |
| 03:47.25 | znoG | to create a menu in Asterisk with lots of if/then etc, is it best to use AEL? i need to dial ext 1, if ext 1 doesn't pick up within 20 seconds, ring ext 2, if ext 2 doesnt pick up try 3, if 3 doesn't pick up go to voicemail. |
| 03:47.49 | znoG | pretty tedious to do with standard dialplan syntax, I thought AEL would be more appropriate |
| 03:48.11 | Flyboy-SR22 | AEL..? |
| 03:48.16 | Flyboy-SR22 | (sorry) |
| 03:48.30 | Qwell | ~ael |
| 03:48.32 | jbot | hmm... ael is Asterisk Extension Language |
| 03:48.36 | *** join/#asterisk bmg505 (n=leon@dsl-146-44-226.telkomadsl.co.za) |
| 03:48.37 | Flyboy-SR22 | ah |
| 03:48.39 | Flyboy-SR22 | thanks |
| 03:48.40 | Flyboy-SR22 | :-) |
| 03:48.47 | znoG | what do you think Qwell ? |
| 03:48.47 | Flyboy-SR22 | more TLA to learn and remember |
| 03:49.03 | _Sam-- | does ael give much for functionality than you get with a regular dialplan? |
| 03:49.08 | Math` | ael is great |
| 03:49.09 | *** join/#asterisk Majestik (n=Majestik@S0106000024c058cc.ed.shawcable.net) |
| 03:49.12 | Qwell | jbot: no, ael is Asterisk Extension Language - a dialplan language with 'c like' syntax? |
| 03:49.14 | jbot | Qwell: okay |
| 03:49.30 | Math` | same functionality, different syntax, better readability |
| 03:49.34 | _Sam-- | more for functionality = much MORE functionality |
| 03:49.41 | _Sam-- | i see |
| 03:49.46 | znoG | looks like AEL2 can include? |
| 03:49.53 | Math` | ael2? |
| 03:50.00 | Qwell | znoG: nope |
| 03:50.05 | Qwell | oh, 2 |
| 03:50.06 | znoG | <PROTECTED> |
| 03:50.06 | znoG | <PROTECTED> |
| 03:50.07 | znoG | <PROTECTED> |
| 03:50.09 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 03:50.14 | Math` | znoG: er thats not new |
| 03:50.16 | Qwell | yeah, murf's AEL can |
| 03:50.25 | Math` | thats a context include |
| 03:50.34 | znoG | yeah, i thought standard AEL couldn't do that |
| 03:50.39 | znoG | somebody was complaining about that here the other day |
| 03:50.40 | Qwell | it can't do #include |
| 03:50.47 | Majestik | _Sam--: Just figured out one more detail... I can make a call from my asterisk to my phone number, which makes a channel to the provider, and back, and that works just fine.. it's definitly upstream :) |
| 03:50.47 | Qwell | ie; include other files |
| 03:50.48 | Math` | of course it can do context inclusions |
| 03:51.03 | Math` | yeah I need to run a "commit.sh" scripts that replaces #includes |
| 03:51.04 | znoG | Qwell: oh i see |
| 03:51.09 | _Sam-- | why couldnt it #include/ |
| 03:51.10 | _Sam-- | ? |
| 03:51.19 | znoG | ok so AEL is probably the way to go |
| 03:51.19 | Qwell | _Sam--: nobody added that functionality |
| 03:51.26 | Math` | because it doesnt use the same config file reading functions as other asterisk files |
| 03:51.31 | Math` | and nobody implemented it |
| 03:51.37 | Math` | maybe I should implement it |
| 03:51.46 | Qwell | Math`: no need, really |
| 03:51.46 | litage | do you set the packetization (Eg: 20ms) on the enduser device (Eg: softphone), on *, or on SER? |
| 03:51.50 | Qwell | if AEL2 goes in...AEL is dead |
| 03:52.02 | znoG | ... why? |
| 03:52.28 | Math` | Qwell: AEL2? is that on mantis? |
| 03:52.31 | *** join/#asterisk Assid (n=assid@203.115.64.14) |
| 03:52.33 | _Sam-- | most people use either AEL OR plain extensions.conf, not both? or most use both? |
| 03:52.39 | Math` | I just use ael |
| 03:52.51 | znoG | i'll be trying out AEL tomorrow |
| 03:53.01 | Math` | nvm reading wiki |
| 03:53.35 | *** join/#asterisk dca[laptop] (n=dca[lapt@c-67-166-23-243.hsd1.co.comcast.net) |
| 03:53.51 | _Sam-- | whats up DCA |
| 03:53.59 | dca[laptop] | oh dear |
| 03:54.01 | dca[laptop] | not much |
| 03:54.04 | PrivalODC | Anyone wkno the differences between the different echo cancelers? |
| 03:54.07 | _Sam-- | heh...my isp is dca.net |
| 03:54.19 | [av]bani | * needs lua |
| 03:54.29 | dca[laptop] | heh, does that mean i own you? |
| 03:54.30 | _Sam-- | [av]bani: you never talked to the teliax guy? |
| 03:54.49 | _Sam-- | after i told you to email him |
| 03:54.54 | [av]bani | _Sam--: not yet, should i? |
| 03:54.59 | _Sam-- | i bet he would listen. |
| 03:55.03 | Dandan | ~vbp |
| 03:55.04 | PrivalODC | Ok, to rephrase, anyone saw BIG improvement switchink from MARK2 to KB1? And is MG2 a lot better than KB1? |
| 03:55.07 | Dandan | ~vpb |
| 03:55.25 | [av]bani | _Sam--: i'm waiting for them to tend to my open ticket, without having to prod them to action |
| 03:55.40 | _Sam-- | its the same one you've been waiting on for like 1 week? |
| 03:55.45 | *** join/#asterisk tehdely (n=delysiid@home.teambarry.org) |
| 03:55.46 | [av]bani | yes |
| 03:55.51 | tehdely | YES! |
| 03:56.19 | _Sam-- | you didnt switch your incoming over to junction networks yet? |
| 03:56.28 | [av]bani | no, they dont have local DIDs |
| 03:56.29 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 03:56.32 | [av]bani | teliax does |
| 03:57.26 | Assid | hrmm. anyone seen any quality difference between voipjet and teliax ? |
| 03:57.45 | _Sam-- | Assid: have you tried either? |
| 03:57.48 | [av]bani | voipjet seems to only do termination? |
| 03:57.48 | dca[laptop] | [av]bani: what are you trying to get teliax to do? |
| 03:57.56 | [av]bani | dca[laptop]: accept my money |
| 03:57.56 | Math` | [av]bani: thats correct |
| 03:57.59 | dca[laptop] | lol |
| 03:58.04 | dca[laptop] | oh dear, you ask so much |
| 03:58.23 | [av]bani | after that i'll poke them about stuttering issues |
| 03:58.30 | dca[laptop] | i know some of the teliax guys, shall i cattle prod them for you? |
| 03:58.31 | Assid | _Sam--: using voipjet right now.. signed up for teliax.. waiting for the credit card amt for verification |
| 03:58.47 | [av]bani | dca[laptop]: well, it shouldnt be necessary to cattle prod them to take my money should it? |
| 03:59.06 | Qwell | [av]bani: he is a teliax guy...cattle prod away |
| 03:59.07 | Assid | [av]bani: you gota do this whole verify the amount thing.. |
| 03:59.15 | [av]bani | im beginng to wonder if their trouble ticket system has any meaning |
| 03:59.17 | dca[laptop] | heh, no, but what do you mena they won't take the money? |
| 03:59.32 | [av]bani | dca[laptop]: my ticket remains open and untouched re: billing issues |
| 03:59.40 | [av]bani | dca[laptop]: error 23, and nobody responded |
| 04:00.03 | [av]bani | i want to add more to my account, but i guess teliax doesnt like my money |
| 04:00.11 | Math` | I'd do |
| 04:00.14 | [av]bani | though nobody else seems to have issues with it :/ |
| 04:00.23 | Math` | whats your local area code |
| 04:00.52 | [av]bani | Math`: ?? |
| 04:01.14 | Math` | with teliax, in which area are your DIDs |
| 04:01.41 | [av]bani | 541-226 |
| 04:01.58 | Abydos313 | so voipjet sucks? |
| 04:02.06 | Math` | voipjet doesnt suck |
| 04:02.08 | JunK-Y | brookshire: alive? |
| 04:02.26 | Assid | _Sam--: ..? |
| 04:02.41 | Assid | so anyone tried both? |
| 04:02.50 | _Sam-- | i use teliax...the service works well for me. |
| 04:03.05 | _Sam-- | alot of it is dependent on what network/backbone you are on and how far from teliax you are |
| 04:03.12 | [av]bani | i love heisenbugs |
| 04:03.13 | *** part/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 04:03.33 | *** join/#asterisk newl (n=newlook@203-59-210-244.dyn.iinet.net.au) |
| 04:03.53 | [av]bani | _Sam--: 23ms |
| 04:04.12 | Assid | yeah i know.. was just wondering if there was any quality difference provider wise |
| 04:04.21 | _Sam-- | ok alot of is dependent on HOW your network traffic gets to teliax not just latency :) |
| 04:04.30 | _Sam-- | and what routes it takes, apparently. |
| 04:04.44 | [av]bani | well, apparently both my routes to teliax suck :/ |
| 04:05.07 | [av]bani | att->pnap->rockynet and qwest->savvis->rockynet |
| 04:05.15 | _Sam-- | i think its rockynet personally |
| 04:05.22 | [av]bani | oddly enough i hear the other end just fine |
| 04:05.24 | file | are they a little... rocky? |
| 04:05.28 | file | hahahahaha |
| 04:05.30 | [av]bani | and when teliax originates, theres no stuttering |
| 04:05.31 | _Sam-- | hahah |
| 04:05.36 | [av]bani | only when i terminate, there is stutter |
| 04:06.38 | Assid | hrmm |
| 04:06.42 | Assid | i cant ping voip-co2.teliax.com |
| 04:06.44 | _Sam-- | interesting...it looks like teliax turned of ICMP |
| 04:06.48 | _Sam-- | on voip-co2.teliax.com |
| 04:06.59 | [av]bani | prolly coz all the ddos from #asterisk :) |
| 04:07.06 | Assid | hehe |
| 04:07.08 | _Sam-- | either that or cause the packet loss was showing |
| 04:07.17 | file | lol |
| 04:07.22 | [av]bani | i never had any packetlos... |
| 04:07.25 | file | that's a rather drastic approach to hiding it |
| 04:07.32 | _Sam-- | today i reported 2% packet loss to that host |
| 04:07.36 | _Sam-- | and it was AT that host |
| 04:07.42 | _Sam-- | maybe i was wrong |
| 04:07.43 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 04:07.50 | _Sam-- | the hsot right before it was dropping ICMP |
| 04:08.00 | _Sam-- | but the voip-co2.teliax host was dropping packets |
| 04:08.00 | file | _Sam--: you have been having weird network issues |
| 04:08.10 | file | ala switch-04 |
| 04:08.11 | _Sam-- | im watching the mtr right now |
| 04:08.30 | Assid | yeah voipjet seems a whole lot closer to me than the network of co2 of teliax |
| 04:08.38 | _Sam-- | but if the router before their host drops ICMP..it doesnt mean that their host should have any packet loss |
| 04:09.04 | _Sam-- | is that true or not true? |
| 04:09.25 | Assid | _Sam--: true |
| 04:09.46 | I-MOD | dropped ICMP packets == packet loss in the literal sense, but it shouldnt affect sip |
| 04:10.18 | _Sam-- | yeah but if a router between myself and host im pinging MTR'ing to drops packets...the host im tracing to / pinging shouldnt drop those packets? |
| 04:10.27 | trixter | its also hard to compare icmp vs udp anymore, people put up filters for icmp a lot more now than they used to |
| 04:10.31 | _Sam-- | they will make it all the way |
| 04:10.34 | *** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder) |
| 04:10.38 | xbmodder_lappy | ibot |
| 04:10.51 | trixter | so unless you know for a fact that icmp isnt filtered by any router inbetween you and the target you shouldnt rely on it 100% |
| 04:11.20 | _Sam-- | but if i know the host im pinging repsonds to pings...it shoulld repsond 100% of the time, regardless of whats in the middle? |
| 04:11.28 | trixter | traceroute works by udp packets sent to a high numbered port with a low ttl and a time exceeded in transit returned |
| 04:11.30 | Assid | not really |
| 04:11.34 | trixter | some sites have for decades filtered that |
| 04:11.55 | trixter | _Sam--: what if something in the midde does QoS and puts icmp so low that its dropped? |
| 04:11.57 | Assid | i have a few boxes setup for dropping 50% icmp |
| 04:12.05 | trixter | that doesnt equate to network performance for other protocols |
| 04:12.07 | Assid | so that people dont wanna try and kill it |
| 04:12.13 | trixter | and the forward path may not be the same as the reverse |
| 04:12.20 | [av]bani | most routers prioritize icmp lowest by default |
| 04:12.26 | [av]bani | linux rate limits icmp |
| 04:12.36 | *** part/#asterisk mwgbc (n=junkmail@c-67-188-17-12.hsd1.ca.comcast.net) |
| 04:12.40 | _Sam-- | i watch and see routers drop ...but my pings /mtr still makes it to the final host with 0 loss |
| 04:12.47 | _Sam-- | that is what im saying |
| 04:12.47 | Assid | 50% icmp... if the limit of 4 simultanous pings are reached |
| 04:12.53 | Assid | hrmm |
| 04:12.55 | [av]bani | which reminds me, asterisk doesnt handle port unreachable... |
| 04:12.59 | [av]bani | i should fix that |
| 04:12.59 | Assid | anwyasy.. brb |
| 04:13.05 | _Sam-- | like i see 10% packet loss at a router int he middle...but i get 0% on the host /mtr im pinging / tracing |
| 04:13.18 | _Sam-- | i guess you ahve to use MTR to see what i mean |
| 04:13.38 | [av]bani | _Sam--: that just means the route rin teh middle drops 10% icmp or udp when directed at it |
| 04:13.48 | [av]bani | 0% on the endpoint means there is no loss |
| 04:13.50 | _Sam-- | right, but the packet loss at the final destination is still valid |
| 04:14.02 | _Sam-- | so if my endpoint had 1% packet loss, its still 1% packet loss |
| 04:14.06 | _Sam-- | not because of something in the middle |
| 04:14.08 | _Sam-- | when i use MTR |
| 04:14.09 | [av]bani | yes |
| 04:14.30 | _Sam-- | that is all im saying...routers in between can drop icmp all they want |
| 04:14.33 | _Sam-- | i am still seeing the real loss |
| 04:14.35 | _Sam-- | when i use mtr |
| 04:14.40 | [av]bani | they can drop anything they want, means nothing |
| 04:14.45 | [av]bani | drop 100% of packets directed at them |
| 04:14.56 | _Sam-- | that is what i thought |
| 04:15.06 | [av]bani | as long as they forward 100% |
| 04:16.23 | _Sam-- | bani do you know if mtr uses icmp ? |
| 04:16.28 | _Sam-- | i guess it has to |
| 04:16.32 | _Sam-- | unless it uses traceroute type |
| 04:16.48 | _Sam-- | i think it does use icmp |
| 04:16.54 | tainted- | does asterisk require mysql |
| 04:17.03 | [av]bani | no |
| 04:17.03 | _Sam-- | it doesnt require it, no |
| 04:17.06 | tainted- | i'm trying to slim the box as much as possible |
| 04:17.21 | _Sam-- | its nice to have it for cdr records |
| 04:17.27 | _Sam-- | or another sql maybe |
| 04:17.46 | tainted- | yea |
| 04:17.49 | tainted- | i use cdr_tds |
| 04:17.52 | _Sam-- | guess you could just as easliy make it insert them in another place |
| 04:18.38 | [av]bani | write out cdr to csv... |
| 04:19.03 | _Sam-- | or just have it connect to a remote sql |
| 04:19.09 | [av]bani | yeah |
| 04:19.29 | _Sam-- | mysql5 is a diskhog |
| 04:19.38 | _Sam-- | i think on my cf card it takes a lot of space |
| 04:20.52 | _Sam-- | eh not as bad as i thought |
| 04:22.25 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 04:27.26 | De_Mon | tds? |
| 04:28.13 | De_Mon | _Sam-- sqlite would work just as good |
| 04:28.22 | [av]bani | what distro you put on your cf? |
| 04:34.16 | *** join/#asterisk CANO-1982 (i=alejandr@201.255.48.248) |
| 04:34.32 | *** part/#asterisk CANO-1982 (i=alejandr@201.255.48.248) |
| 04:34.37 | [av]bani | heh the chinese grandstream guys english is better than the snom guys |
| 04:42.14 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 04:43.45 | *** join/#asterisk Jizzbug (i=derekm@63-254-64-44.ip.mcleodusa.net) |
| 04:44.00 | tronix | [av]bani: i've seen the DSL distro (damn small linux) -- looks interesting, would fit. don't know about maintability |
| 04:44.12 | tronix | I suspect some other distro might do better on maintenance angle. |
| 04:44.28 | tronix | I've seen Debian fit in 60-95 MB |
| 04:46.11 | *** join/#asterisk brookshire[bar] (n=matt@68.62.203.242) |
| 04:47.29 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 04:47.47 | [av]bani | what i want is a distro which has a buildroot so you can put together your own custom build |
| 04:49.36 | [av]bani | openwrt comes close, but its not x86 yet |
| 04:55.15 | *** join/#asterisk benjk (n=benjamin@66-215-63-81.dhcp.atsc.ca.charter.com) |
| 04:56.17 | benjk | anybody here who recently upgraded a Cisco 7960 from SIP 2.0 to 3.0 ? |
| 04:57.29 | Qwell | why 3.0? |
| 04:57.33 | Qwell | Isn't the latest like...7? |
| 04:58.41 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 05:00.44 | Nugget | yeah, but you have to pass through v3 to get to v7. |
| 05:00.51 | *** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net) |
| 05:01.08 | Mavvie | -- Channel 0/4, span 4 received AOC-E charging 166821040 units |
| 05:01.25 | benjk | well, apparently Cisco doesn't allow you to go from 2.0 directly to 7.5 |
| 05:01.42 | benjk | and it even appears is if you can't even go from 2.0 directly to 3.0 |
| 05:01.52 | benjk | s/is/as |
| 05:02.47 | Qwell | You can, with the universal app loader |
| 05:02.59 | mog_home | ? |
| 05:03.00 | Qwell | 2 > 7.x works fine |
| 05:03.07 | Qwell | worked fine for me |
| 05:03.18 | mzo | how nice are the 7960 phones? Is there any different with the 7940s? |
| 05:03.25 | Qwell | mzo: 4 extra lines |
| 05:03.34 | Qwell | one extra softkey too? I don't recall |
| 05:03.41 | benjk | well, it simply doesn't work |
| 05:03.46 | Qwell | "doesn't work"? |
| 05:03.59 | benjk | the phones were on SCCP 3.1 |
| 05:04.14 | benjk | and I followed Cisco's documentation by the letter |
| 05:04.14 | Qwell | Have you read the firmware upgrade matrix? |
| 05:04.20 | Qwell | worked just fine for me, multiple times |
| 05:04.29 | benjk | nothing worked until I used the SIP 2.0 firmware |
| 05:04.43 | mzo | oh, wow, that is kind of nice? |
| 05:04.54 | mzo | so i could conference 3 or 4 people in at once? |
| 05:05.05 | Qwell | mzo: You could do that with the 7940 |
| 05:05.19 | Qwell | with sccp at least |
| 05:05.26 | Qwell | I can have 100 calls on one line appearance |
| 05:05.40 | mzo | i'll have to try it at work. I've never messed with my phone much |
| 05:05.50 | benjk | now, I changed content of the OS79XX.TXT on my TFTP server to the SIP 3.0 image and also the SIPDefault.cnf parameter, but the phone doesn't even make any attempt to request the image |
| 05:06.16 | Qwell | benjk: Can your phone hit the internet? |
| 05:06.32 | benjk | what do you mean? |
| 05:06.37 | Qwell | for the tftp |
| 05:06.49 | benjk | no the TFTP server is on the LAN |
| 05:07.22 | benjk | and its not just one phone, its a whole barrage of 'em |
| 05:07.45 | benjk | Oh well, I guess I will have to try the 2.1 firmware now |
| 05:08.01 | Qwell | go back to the 3.1 sccp if you can... |
| 05:08.10 | Qwell | That'll upgrade straight to 7.x |
| 05:08.11 | benjk | next time a customer wants Cisco phones, it will be 500 USD per phone for configuration |
| 05:08.28 | Qwell | because you don't know how to configure them? Seems a bit unfair to me. :) |
| 05:08.30 | benjk | No SCCP 3.1 did not upgrade straight to 7.X |
| 05:08.43 | Qwell | does here |
| 05:08.44 | benjk | for any X between 0 and 5 |
| 05:08.50 | Qwell | http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm |
| 05:08.54 | Qwell | You used that site? |
| 05:08.58 | benjk | yes |
| 05:09.08 | Qwell | http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm#wp1053742 |
| 05:09.22 | Qwell | procedure C |
| 05:09.39 | Qwell | I can hook it up through ssh |
| 05:11.08 | benjk | I have tried most of this since yesterday morning |
| 05:11.17 | Qwell | I'm a pro :P |
| 05:11.25 | benjk | the only thing that worked was from SCCP 3.1 to SIP 2.0 |
| 05:11.38 | mzo | i'd love to buy a 7940 or something (a phone) in the same class and same general fit and finish. |
| 05:11.43 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 05:12.16 | benjk | the software and admin facilities are a pile of dog poo though |
| 05:12.27 | benjk | even the Chinese can do better |
| 05:13.28 | trixter | hey benjk |
| 05:13.29 | Qwell | benjk: let me know if you want me to take a look at your tftp stuff |
| 05:13.37 | Qwell | I'll be around |
| 05:14.12 | benjk | well, it's very simple, the phone's firmware is P0S30200 |
| 05:14.26 | [av]bani | <3 cisco |
| 05:14.37 | Qwell | SIP 2 only understands 8.3 |
| 05:14.43 | benjk | the content of OS79XX.TXT is now P0S3-03-0-00 |
| 05:15.02 | benjk | so, I need to rename the image file then? |
| 05:15.07 | trixter | who would prefix firmware with a model number POS ? |
| 05:15.11 | trixter | seems they are saying something |
| 05:15.14 | Qwell | yes |
| 05:15.23 | benjk | yeah Cisco == PoS |
| 05:15.25 | Qwell | take out the dashes, remove a 0 (it doesn't really matter which one) |
| 05:15.38 | Qwell | rename the file, and set the OS79XX.txt to the same name |
| 05:15.42 | file | Qwel! |
| 05:15.51 | Qwell | file: I'm 8.3 compliant already! |
| 05:16.05 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 05:16.07 | benjk | and the image name parameter in SIPDefault.cnf too I presume |
| 05:16.08 | Qwell | I'm a .ell file |
| 05:16.18 | Qwell | benjk: it probably won't even get that far |
| 05:18.35 | benjk | ok, rebooting ... |
| 05:18.50 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 05:19.26 | benjk | nope |
| 05:19.44 | benjk | the phone will not even request the file |
| 05:19.47 | benjk | what a piece of crap |
| 05:19.47 | Qwell | pastebin me the tftp logs |
| 05:19.55 | Qwell | What's being NAK'd? |
| 05:20.02 | benjk | nothing in the logs |
| 05:20.08 | Qwell | You got verbose? |
| 05:20.13 | benjk | the phone doesn't request anything other than the cnf files |
| 05:20.21 | Qwell | What config? |
| 05:20.33 | Qwell | config(s) |
| 05:21.41 | benjk | SIPDefault.cnf and SIP_mac-addr.cnf |
| 05:21.47 | Qwell | rename those, let them NAK |
| 05:21.55 | Qwell | What else gets requested after that? |
| 05:21.59 | benjk | nothing |
| 05:22.07 | Qwell | you need to rename them first :p |
| 05:23.24 | Qwell | On some images, if the SIP configs are NAK'd, it'll request an sccp config (which is an easy upgrade) |
| 05:24.21 | benjk | I have just spent 2 days trying to get away from SCCP |
| 05:24.33 | Qwell | yes, and you can still use SIP |
| 05:24.36 | *** join/#asterisk tronix (n=dsf@mappy.catbert.org) |
| 05:26.18 | tainted- | strange.. asterisk compiles find but refuses to start up |
| 05:26.20 | brookshire[bar] | junk-y!! |
| 05:26.22 | tainted- | i get no errors |
| 05:26.29 | JunK-Y | yay! |
| 05:26.33 | tronix | tainted-: did you start with -vvvvc? |
| 05:26.50 | tainted- | yea |
| 05:27.17 | tronix | and there's nothing interesting in /var/log/asterisk/messages ? |
| 05:27.37 | tronix | does -vvvvc startup give any errors? |
| 05:27.43 | brookshire[bar] | drunk-y! |
| 05:27.57 | JunK-Y | mouhahhahaha haha |
| 05:27.57 | tronix | because if it works, it'll put you in a *CLI> prompt. if it doesn't, it'll complain about something |
| 05:28.02 | benjk | I renamed the .cnf files, and it still doesn't change a thing |
| 05:28.11 | Qwell | benjk: Does it try to request any extra files now? |
| 05:28.14 | benjk | the phone won't request the 3.0 firmware |
| 05:28.17 | Qwell | You were far better off with sccp... |
| 05:28.22 | JunK-Y | brook: http://www.asterisk.org/blog/4 |
| 05:28.38 | benjk | SCCP firmware is a piece of crap because it doesn't allow me to do manual changes to the config |
| 05:28.44 | tainted- | <PROTECTED> |
| 05:28.47 | tainted- | is the last line i get |
| 05:28.53 | benjk | I want manual settings as a choice |
| 05:29.11 | benjk | if I don't get that choice, then the product is a pile of fascist junk |
| 05:29.21 | brookshire[bar] | junk: you should put up photos on your blog |
| 05:29.23 | Qwell | yes, and you can go to SIP once you've upgraded |
| 05:29.35 | Qwell | I can switch between SIP and SCCP in less than 60 seconds |
| 05:29.43 | Qwell | back and forth, by changing one line |
| 05:29.47 | benjk | well not on those phones |
| 05:30.27 | benjk | I have already had 3 different Cisco savvy folks give me instructions and they couldn't get me any further than SIP 2.0 |
| 05:30.35 | Qwell | newbs |
| 05:30.54 | benjk | anyway, trying 2.1 now |
| 05:31.05 | Qwell | give me read access to /var/log/messages and write access to /tftproot/, and I'll have it fixed in minutes :p |
| 05:31.12 | file | Qwell IS ELITE! |
| 05:31.22 | benjk | this LAN is not connected to the net |
| 05:31.28 | brookshire[bar] | he's so elite he's ultraviolet ;) |
| 05:31.40 | file | brookshire[bar]: I ought to smack you for that |
| 05:31.43 | Qwell | eh..your loss :p |
| 05:32.31 | tainted- | is there an error log i can chat? |
| 05:32.41 | tainted- | s/chat/check/ |
| 05:33.18 | tronix | there's /var/log/asterisk/messages (or whereever it is on your setup) |
| 05:33.41 | tronix | but might have to strace -f -o /tmp/foo asterisk <any startup options> or similar |
| 05:34.00 | tronix | and then look at /tmp/foo around the line it prints out chan_phone.so message |
| 05:34.19 | tronix | did this ever work before, or is this a new setup? |
| 05:34.35 | tainted- | tronix i don't have a messages in that folder |
| 05:34.38 | tainted- | hmm |
| 05:34.46 | tainted- | only event_log and queue_log |
| 05:35.00 | tronix | it's probably not getting far enough in startup to write out messages. unfortunate. |
| 05:35.32 | tronix | strace will likely give insight if you can decode how to read it, or pastebin output around that area |
| 05:36.10 | tronix | I've used it in the past to figure out problems loading a codec was due to a perms problem on the directory. so it's indispensable. |
| 05:36.29 | mzo | i heard this 'rumor' taht you can integrate google talk with asterisk, is that actually true? |
| 05:36.37 | mog_home | not today |
| 05:36.42 | Qwell | mog_home: tomorrow?! |
| 05:36.45 | mog_home | there is a guy that has propitery method |
| 05:36.46 | mog_home | sssh |
| 05:36.50 | mog_home | dont ruin my suprise |
| 05:36.51 | Qwell | aww |
| 05:36.52 | Qwell | k |
| 05:36.57 | Qwell | ooo, tomorrow is GSK day |
| 05:37.01 | mog_home | gsk? |
| 05:37.01 | Qwell | GSX |
| 05:37.05 | Qwell | vmware |
| 05:37.05 | mog_home | gsx? |
| 05:37.05 | mzo | mog_home, for google talk? |
| 05:37.12 | Qwell | ssshhh |
| 05:37.16 | Qwell | also secret |
| 05:37.23 | mzo | argh, no one ever tells me sekrets |
| 05:37.23 | Qwell | birdie told me |
| 05:38.18 | mog_home | people in asterisk are very secretive |
| 05:38.23 | mog_home | and many oss projects |
| 05:38.36 | mog_home | developers tend to not want to make promises they cant keep |
| 05:38.39 | file | I have to be secretive! |
| 05:38.39 | mzo | classified! |
| 05:38.41 | mog_home | and have users bugging them |
| 05:38.55 | mzo | but really, iis it true about google talk and asterisk ? |
| 05:39.05 | mog_home | that may or not be true |
| 05:39.05 | file | super secret project 3.141592653! |
| 05:39.16 | Qwell | pie project? |
| 05:39.16 | mog_home | and i may or may not have a paypal account that accepts donations ^_^ |
| 05:39.18 | mzo | some of you have promising careers in elected office. :P |
| 05:39.22 | file | I like pie |
| 05:39.23 | Corydon76-home | It's not true. |
| 05:39.26 | mog_home | pi is good |
| 05:39.37 | mog_home | aww /me frowns |
| 05:39.46 | mog_home | i havae been working hard Corydon-w but i am a bit of a nub |
| 05:39.47 | file | who wants the 14? |
| 05:40.06 | Corydon76-home | file: Knuth beat you to that |
| 05:40.57 | Corydon76-home | Just like someone joked about Morse code on the mailing list... and by the next day, it was in trunk... |
| 05:41.19 | file | you just have to perk someone's interest |
| 05:41.33 | mzo | i'll take the .14 |
| 05:41.44 | file | mzo: mog has the . |
| 05:41.52 | file | you can have 141 though |
| 05:41.55 | mog_home | damn spiffy |
| 05:42.13 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:42.28 | file | mog_home: Matttttttttttt |
| 05:42.30 | Corydon76-home | As long as I have the 6535, I'm fine |
| 05:42.36 | mog_home | jossssssssssssssh |
| 05:43.06 | mzo | oh, okay, ty! |
| 05:44.36 | mog_home | mzo you get message? |
| 05:45.04 | Qwell | I guess I'm stuck with 926? :( |
| 05:45.14 | Qwell | erm, 592 :( :( |
| 05:46.10 | Corydon76-home | Qwell: the nice thing about irrationality is that any sequence is available, somewhere. |
| 05:46.55 | *** join/#asterisk _-_ (n=nabudoco@206.135.48.98) |
| 05:47.13 | *** part/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 05:47.35 | Qwell | Corydon-w: even 1.2? |
| 05:47.39 | Qwell | <tab> |
| 05:48.13 | Corydon76-home | If by . you mean any digit, yes |
| 05:48.18 | Qwell | I mean . |
| 05:48.57 | Corydon76-home | decimal points don't count |
| 05:49.08 | Qwell | too irrational? |
| 05:51.23 | Qwell | benjk must've given up |
| 05:51.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-112-192.red.bezeqint.net) |
| 05:52.07 | benjk | I never give up |
| 05:52.38 | benjk | however, upgrading from 2.0 to 2.1 doesn't work either |
| 05:52.51 | Qwell | I could have it done in < 10 minutes :P |
| 05:53.22 | Qwell | You need to go to sccp |
| 05:53.43 | benjk | I have been on SCCP |
| 05:53.44 | ravsi | What is it called when you have 1 main number and you hand off calls to other lines in order to answer the main line agian? |
| 05:53.48 | Qwell | yes, go back to sccp |
| 05:53.53 | Qwell | 3.1 or whatever it was, if you can |
| 05:54.00 | benjk | and then? |
| 05:54.06 | Qwell | and then go to the latest sccp |
| 05:54.10 | Qwell | which will have the UAL |
| 05:54.17 | Qwell | which can easily go back and forth between sccp and sip |
| 05:54.17 | benjk | tried that already |
| 05:54.58 | benjk | and it could be so easy |
| 05:55.04 | Qwell | it is easy ;] |
| 05:55.27 | benjk | all the phone would need to do is self assign a link-local address and advertise itself |
| 05:56.04 | benjk | then click on the service in your browser and enter the location of the firmware and click upgrade |
| 05:57.20 | Qwell | Can you get it back to 3.1? |
| 05:57.25 | Qwell | sccp |
| 05:57.25 | benjk | alternatively, the server with the images could advertise them and the phone would pick them up, give you a list to choose from, push select and done |
| 05:57.28 | benjk | no |
| 05:57.48 | benjk | the phone basically ignores whatever is in OS79XX.TXT |
| 05:57.57 | Qwell | Did you put it in the sip config? |
| 05:58.01 | benjk | a real piece of shit |
| 05:58.14 | benjk | hence the P0S prefix of those files |
| 05:58.33 | benjk | I will suggest to Cisco they add an 'R' in front of that |
| 05:58.42 | benjk | R for real piece of shit |
| 05:59.01 | benjk | yes, I added it to SIPDefault.cnf too |
| 06:01.50 | Qwell | and you're sure it isn't NAK'ing? |
| 06:03.27 | ManxPower | ravsi, a hunt group |
| 06:03.29 | benjk | I am not sure of anything anymore |
| 06:03.37 | Qwell | benjk: You can't temp put it on the internet? |
| 06:03.43 | ravsi | a hunt group , ManxPower |
| 06:03.52 | ravsi | Manxpower: thanks! |
| 06:03.59 | benjk | I am in a remote village with 6 houses in the mountains |
| 06:04.07 | benjk | this LAN is not connected to the internet |
| 06:04.27 | benjk | I am on my notebook which is hooked up to a wireless link |
| 06:04.28 | CaT[tm] | hmmm. I'd rather be in a remote village in the mountains right about now. |
| 06:04.33 | Qwell | ssh tunnel? |
| 06:04.38 | benjk | but the LAN here has no connectivity |
| 06:04.45 | Qwell | surely the laptop has ethernet also? |
| 06:04.57 | benjk | yes it has but behind 2 NATs |
| 06:05.13 | benjk | and I have no control over the access points |
| 06:05.56 | benjk | I mean what's the big deal? |
| 06:06.18 | benjk | Cisco says the phone should pick up on what is in OS79XX.TXT and it simply doesn't |
| 06:06.25 | Qwell | only some images |
| 06:07.02 | file[laptop] | I do believe I'm going to sleep |
| 06:07.02 | benjk | well, if the phone is on SIP 2.0 and the image in OS79XX.TXT is 2.1, why the heck wouldn't it pick up on that? |
| 06:07.19 | Qwell | is it looking for OS79XX.TXT? |
| 06:10.16 | *** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 06:11.59 | benjk | it may not be, but if so, then Cisco are liars |
| 06:12.09 | Qwell | It doesn't always look for that |
| 06:12.26 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.63.82.Dial1.Chicago1.Level3.net) |
| 06:13.12 | benjk | and why would it not? |
| 06:13.17 | *** join/#asterisk afu888 (n=jkr888@c-69-248-26-95.hsd1.nj.comcast.net) |
| 06:14.00 | benjk | Cisco documentation states very clearly that it will compare the name of the image in OS79XX.TXT with its own installed image, for that to be true, it would have to look at it |
| 06:14.17 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 06:17.19 | Qwell | It just doesn't...don't question it |
| 06:17.25 | Qwell | check the logs and see what it actually looks for |
| 06:17.31 | stormfr | hello, i have daily chan_sip stop responding by said "grab the lock". Is there a way to identify where the lock is ? (using realtime mysql + addons, head or 1.2.x) |
| 06:24.47 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 06:25.08 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 06:26.23 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 06:34.42 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-46.west.biz.rr.com) |
| 06:36.11 | *** join/#asterisk daguerro (i=daguerro@lss-67-68.ee.ITB.ac.id) |
| 06:39.05 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-234.claranet.co.uk) |
| 06:40.11 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 06:41.25 | *** join/#asterisk rt (n=markv@c-67-180-32-90.hsd1.ca.comcast.net) |
| 06:41.41 | rt | sigh. My PAP2 reset, downloaded new firmware, and now is pretty effectively locked me out. |
| 06:46.28 | rt | should i get a sipura 3000 or a clipcomm cg-200? |
| 06:47.21 | benjk | Qwell, if Cisco states in their documentation that it does and it doesn't then I take issue with that |
| 06:47.43 | Qwell | Which firmware release notes are you reading? |
| 06:48.15 | Qwell | there is a new one for every revision |
| 07:02.07 | *** join/#asterisk JunK-Y (n=junky@Toronto-HSE-ppp3742560.sympatico.ca) |
| 07:03.13 | *** join/#asterisk st3v (n=spj@netblock-66-218-41-231.dslextreme.com) |
| 07:04.38 | st3v | every time I connect to the directory, it says "No directory entries match your search". It doesn't prompt me to enter the first 3 letters of the last name. I switched to asterisk@home and it started doing that. |
| 07:04.51 | st3v | I have extensions and voicemail configured |
| 07:10.27 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 07:10.54 | mog_home | hey brimstone is #anime back? |
| 07:11.03 | brimstone | yes |
| 07:11.07 | brimstone | comcast went down |
| 07:11.34 | mog_home | yay |
| 07:11.39 | brimstone | yay for comcast! |
| 07:15.04 | rob0 | Mine went down too. |
| 07:16.06 | *** join/#asterisk Micc (n=dotirc@c-24-16-228-130.hsd1.wa.comcast.net) |
| 07:16.29 | Micc | I have an interesting asterisk problem, is anyone around tonight that might be able to help me? |
| 07:19.50 | Micc | Ok, so here is the problem. I have a linksys pap2-na. I've setup a number of the pap2-na's for my friend and he uses them without problem. When I dial his extension there are no problems. But when I dial an outside number that goes out the PRI it is choppy on the outbound. |
| 07:20.57 | Micc | It doesn't make any sense because when I'm doing a call to another asterisk sip line it works fine. So I know the firewall is good. |
| 07:21.45 | Micc | the link is the same. pap2-na -> cable modem -> router -> asterisk -> router -> cable modem -> pap2-na. |
| 07:22.16 | Micc | So when its pap2-na -> cable modem -> router -> asterisk -> PRI it has issues. How can there be a difference? |
| 07:22.58 | brimstone | which card is providing PRI Micc ? |
| 07:23.05 | brimstone | a digium card by chance? |
| 07:23.07 | Micc | digium. |
| 07:23.12 | Micc | yes. |
| 07:23.24 | brimstone | did they fix the IRQ problem yet mog_home ? |
| 07:23.31 | brimstone | i'd check it's IRQ Micc |
| 07:23.33 | Micc | Its just a single span. |
| 07:23.38 | brimstone | make sure it's by itself |
| 07:23.44 | Micc | How do I do that? |
| 07:23.51 | brimstone | lspci -vb |
| 07:23.56 | brimstone | look for Unknown Controller |
| 07:23.59 | brimstone | or Tiger Jet |
| 07:24.02 | brimstone | or something |
| 07:24.09 | brimstone | maybe jsut d161 would find it |
| 07:25.23 | *** join/#asterisk iarebad (n=bad@203-219-93-126.static.tpgi.com.au) |
| 07:25.35 | iarebad | hello |
| 07:25.39 | iarebad | can someone tell me what asterisk is all about? |
| 07:25.43 | brimstone | hi iarebad |
| 07:25.48 | brimstone | it's a free pbx! |
| 07:25.50 | iarebad | im new to VOIP scene and i read about asterisk a lot |
| 07:26.25 | iarebad | so basically, if i setup asterisk at home on a PC, i can call anywhere around the world on landlines/mobs for free? |
| 07:26.41 | brimstone | termination isn't free, but the pbx part is |
| 07:26.51 | Micc | ok, its sharing the IRQ with a bunch of stuff. |
| 07:26.53 | Micc | IRQ 5 |
| 07:27.00 | brimstone | interesting sites to read, asterisk.org, freeworlddialup.net |
| 07:27.04 | iarebad | ok sorry, what is termination? |
| 07:27.10 | mog_home | ? |
| 07:27.11 | brimstone | you can get really, really cheap termination to iarebad |
| 07:27.12 | iarebad | anywhere i can read about all this |
| 07:27.20 | brimstone | voip-info.org too iarebad |
| 07:27.24 | brimstone | lots-o-info there |
| 07:27.47 | iarebad | ok thanks .. will start off there |
| 07:27.48 | iarebad | cheers. |
| 07:27.53 | brimstone | take luck! |
| 07:28.38 | *** join/#asterisk xtr-II (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 07:31.43 | *** join/#asterisk lalito (n=erg@201.135.215.14) |
| 07:36.12 | benjk | you're not going to believe this ... |
| 07:36.19 | brimstone | i just mine |
| 07:36.20 | brimstone | *might |
| 07:36.29 | benjk | those stupid Cisco phones |
| 07:36.43 | benjk | it's upgrading now |
| 07:36.56 | benjk | but you have to cripple the SIPDefault.cnf file |
| 07:37.31 | benjk | and not even leave a CRLF at the end of the line with the image name |
| 07:38.37 | *** join/#asterisk g0mb0 (n=g0mb0@external.micom.mng.net) |
| 07:43.33 | iarebad | hypothetical question: what do u need to be a VOIP provider? |
| 07:43.59 | JamesDotCom | more support than an irc channel |
| 07:44.20 | newl | knowhow, bandwidth, hardware, and operating capitol in no particular order. :) |
| 07:44.26 | iarebad | lol |
| 07:44.31 | brimstone | and a cool logo |
| 07:44.36 | brimstone | or a catchy name |
| 07:44.43 | *** join/#asterisk tzafrir_laptop (n=tzafrir@85-64-243-145.barak-online.net) |
| 07:44.43 | newl | ^^ what he said. |
| 07:44.58 | brimstone | ones ending in "com" "tel" "net" are popular |
| 07:45.03 | mzo | yeah comtel.net |
| 07:45.07 | mzo | or netcom.tel |
| 07:45.08 | JamesDotCom | hahaha |
| 07:45.12 | mzo | or something including all three of those words |
| 07:45.22 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:45.22 | JamesDotCom | i have none of them in my name |
| 07:45.24 | mzo | comtel.net IS a voip provider... |
| 07:45.36 | brimstone | lmao too funny mzo |
| 07:45.37 | mzo | netcom.tel is open :P |
| 07:45.49 | mzo | aww telnet.com is spam |
| 07:46.11 | mzo | newcom.tel! |
| 07:46.46 | iarebad | ok apart from knowhow, bandwith, hardware, which all need a capital .. how much approx is that? |
| 07:46.48 | iarebad | LOL |
| 07:46.56 | Qwell | <iarebad> hypothetical question: what do u need to be a VOIP provider? |
| 07:47.01 | Qwell | You need to know...THE ANSWER TO THAT QUESTION |
| 07:47.04 | Qwell | :) |
| 07:47.22 | Abydos313 | any of you guys try vonage? i was wondering if i bought an account here in the states with local area code if the same phone could call from out of the country to local areas from where the area code is? |
| 07:47.34 | iarebad | Qwell, lol, what if i have the spare CAPITAL, do i still need to know the answer? |
| 07:47.35 | JamesDotCom | iarebad: what country you in, and do you have a job for me? :P |
| 07:47.45 | Qwell | iarebad: You need to hire somebody who does. :p |
| 07:48.11 | iarebad | no doubt .. and so i need to know what a regular VOIP investent might be .. then i can start some research |
| 07:48.31 | Qwell | iarebad: depends on what you want at first |
| 07:52.37 | *** join/#asterisk e3g (i=e3g@u15157627.onlinehome-server.com) |
| 07:55.00 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 07:55.40 | benjk | And another Cisco braindamage |
| 07:55.58 | benjk | it says "Upgrading software" but it doesn't actually upgrade |
| 07:56.13 | Qwell | from what to what? |
| 07:56.22 | benjk | same version than before once it's done "upgrading software" |
| 07:56.27 | benjk | from 2.1 to 3.0 |
| 07:56.30 | Qwell | sip? |
| 07:56.52 | benjk | of course |
| 07:56.59 | tuxinator_linux | Qwell, do you sleep? |
| 07:57.03 | Qwell | check for NAK's.. If it's missing any of the files, it'll abort |
| 07:57.06 | Qwell | tuxinator_linux: of course not |
| 07:57.30 | tuxinator_linux | What time zone are you in? |
| 07:57.35 | Qwell | Whichever I choose |
| 07:57.46 | Qwell | I'm omnipresent |
| 07:57.54 | tuxinator_linux | :-) |
| 07:57.56 | benjk | W350 Invalid proxy_backup |
| 07:58.11 | benjk | W351 Invalid proxy_emergency |
| 07:58.33 | Qwell | I don't think those are related to the upgrade |
| 07:58.42 | benjk | W231 Upgrade P0S30300.bin ... Bad header |
| 07:58.53 | Qwell | better |
| 07:58.56 | *** join/#asterisk angom_h (n=angom@red-corp-200.76.252.151.telnor.net) |
| 07:59.02 | benjk | presumably the name changes was not such a good idea |
| 07:59.07 | Qwell | name change is fine |
| 07:59.12 | benjk | s/changes/change |
| 07:59.25 | benjk | so what's the thing with the bad header? |
| 07:59.27 | Qwell | Did you change all of the files? |
| 07:59.32 | Qwell | s/change/rename/ |
| 07:59.48 | benjk | no, only the 3.0 binary |
| 07:59.57 | benjk | there isn't any more for the 3.0 |
| 07:59.58 | Qwell | rename them all, the .bin, .sbn, .load |
| 07:59.59 | Qwell | oh |
| 08:00.02 | Qwell | You sure? |
| 08:00.15 | benjk | sorry but for version 3.0 there is only one binary |
| 08:00.37 | benjk | P0S3-03-0-00.bin |
| 08:00.52 | benjk | which I renamed to P0S30300.bin |
| 08:01.46 | Qwell | If you rename a file it looks for, does it NAK? |
| 08:01.53 | Qwell | in the tftp logs |
| 08:02.06 | benjk | I don't see much in the tftp logs |
| 08:02.20 | Qwell | up the verbosity |
| 08:02.20 | benjk | and I can't figure out how to make it anymore verbose, given up on that |
| 08:02.22 | tronix | could run tcpdump or ethereal for 'port tftp' |
| 08:02.43 | benjk | yeah, I know that much but the tftpd on OSX is a piece of crap |
| 08:02.46 | tronix | ahhh |
| 08:02.49 | benjk | it's got -d for debug verbosity, which doesn't work |
| 08:02.52 | tronix | ick |
| 08:02.58 | tronix | that's disappointing. |
| 08:03.17 | benjk | and it's got -l for logging every successful download and that doesn't work all too well either |
| 08:03.24 | Qwell | -vv? |
| 08:03.26 | tronix | ethereal does work on os x tho |
| 08:03.29 | benjk | no -v |
| 08:03.36 | Qwell | What tftpd? |
| 08:04.00 | benjk | well, this is clearly related to the file being perceived bad |
| 08:04.12 | benjk | and for 3.0 there is only one file |
| 08:04.24 | benjk | just like there is only one for all the v 2.x |
| 08:04.35 | Qwell | until you can see the tftp logs...you can't assume anything |
| 08:04.58 | benjk | yes I can assume this because at this low version Cisco didn |
| 08:05.02 | tronix | the theory is good, but debugging will require observing actions... and if logging works like crap, gotta try a different tack |
| 08:05.03 | benjk | 't have multiple files |
| 08:05.33 | benjk | Cisco did not have multiple boot files at this version level of firmware |
| 08:06.09 | benjk | consequently it is folly to assume that a software from 2001 would expect additional files in 2005 just because they were added some time in 2003 |
| 08:06.17 | tronix | http://www.pch.net/resources/discussion/inoc-dba/archive/2004-June/001078.html |
| 08:06.24 | tronix | it may actually want the dashes in there |
| 08:06.25 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:06.38 | tronix | I have a vague recollection of problems with my upgrade to SIP 7.4 firmware from SCCP 3.1 |
| 08:06.43 | benjk | yeah, that's what I thought, the name change is probably a bad thing |
| 08:06.45 | tronix | until I did the dashes and all |
| 08:07.05 | benjk | but Qwell said that v2 did only understand 8.3 file name s |
| 08:07.11 | tronix | ahh right. hm. |
| 08:07.19 | benjk | maybe I have to go up to v 2.3 |
| 08:07.25 | tronix | there *was* a workaround mentioned for v2... |
| 08:07.30 | tronix | was on one of the voip-info pages |
| 08:07.33 | Qwell | You need to get up to sccp > 6.3 |
| 08:07.51 | e3g | $cardno = $AGI->get_data("astcc-accountnum2"); <---in ASTCC this line get the card number but doesnt give enough time difference between calling card digits...any help to increase the time to wait for digits? |
| 08:11.10 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 08:11.29 | *** join/#asterisk xtr (i=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 08:12.10 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 08:12.25 | *** join/#asterisk eivindtr (n=wingnut-@cC3012269.inet.catch.no) |
| 08:12.33 | tuxinator_linux | Here's a quicky, how long (on average) does it take Cisco, once they receive your order, to provide access to the files (so I can SIP my 7960)? |
| 08:14.25 | tuxinator_linux | anoying question, I know |
| 08:14.43 | Qwell | tuxinator_linux: a week or so maybe |
| 08:14.48 | Qwell | less if you're lucky |
| 08:14.52 | tuxinator_linux | thanks |
| 08:14.56 | Qwell | sip sucks anyhow :p |
| 08:15.24 | tuxinator_linux | I have a used phone and no files, so I need to put something on it |
| 08:15.58 | tuxinator_linux | is asterisk doing better at skinny? |
| 08:16.46 | Qwell | I use the third-party chan_sccp |
| 08:17.22 | benjk | upgrade from 2.3 to 3.0 works but not with the name change |
| 08:17.47 | benjk | it wants the dashes in there |
| 08:18.47 | benjk | upgrading from 2.1 to 3.0 does not work |
| 08:19.10 | benjk | you really have to do the whole lot incrementally on minor versions |
| 08:19.46 | Qwell | I didn't |
| 08:22.29 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:25.47 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-25-222.cybersurf.com) |
| 08:26.06 | benjk | from 2.0 you have to |
| 08:26.18 | benjk | maybe it gets better once we're at 3.0 |
| 08:26.28 | Qwell | You were at 3.1 |
| 08:26.35 | benjk | SCCP 3.1 |
| 08:26.42 | Qwell | so? |
| 08:26.48 | benjk | which didn't accepts anything but 2.0 |
| 08:26.51 | Qwell | sccp will go straight to 7 |
| 08:26.57 | benjk | well it didn't |
| 08:27.05 | Qwell | Because you did it wrong :P |
| 08:27.18 | benjk | because Cisco have a fucked up process |
| 08:27.31 | Qwell | nah, you just have to read it |
| 08:27.34 | benjk | if five different people can't figure it out |
| 08:27.47 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 08:27.47 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:27.47 | Qwell | I would have had it in minutes, if I had access to the box |
| 08:27.50 | benjk | over the course of some 20 something hours or so |
| 08:28.15 | benjk | that only speaks for the cumbersome nature of their processes |
| 08:28.29 | benjk | there's no point in arguing otherwise |
| 08:32.06 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 08:34.08 | reza | anyone here use nufone? |
| 08:34.18 | Qwell | reza: yeah |
| 08:34.58 | reza | i'm new to this; got nufone setup and been using a normal phone via the tdm400 fxs |
| 08:35.16 | reza | but the like gets choppy every so often |
| 08:35.38 | reza | i thought it might be the tdm400, but i just got a sip ip phone, and i have the same problem |
| 08:35.43 | reza | could it be nufone? |
| 08:35.48 | benjk | well, 3.0 to 4.0 is a no go |
| 08:36.06 | benjk | it's incremental again |
| 08:36.34 | benjk | somebody at Cisco ought to be tortured for this |
| 08:36.51 | mzo | hahaha, how do you know they weren't? |
| 08:37.20 | reza | anyone else use another cheap aix service i can try to see if it's nufone or something else |
| 08:41.51 | benjk | if so, they ought to be tortured again and more painfully |
| 08:41.57 | jaike | voicepulse |
| 08:42.00 | jaike | ease to setup |
| 08:42.02 | jaike | easy |
| 08:42.06 | benjk | if need be even beheaded |
| 08:42.21 | *** join/#asterisk Ariek48 (n=ariek@141.252.216.37) |
| 08:42.24 | Qwell | benjk: hire a professional |
| 08:42.40 | benjk | bullcrap |
| 08:42.49 | benjk | there is no excuse for such a messy process |
| 08:42.57 | Qwell | it's a one step upgrade... |
| 08:43.04 | benjk | besides, do you think I am doing this for fun or what? |
| 08:43.07 | jaike | reza: might also be your internet connection...voip is packet loss sensitive |
| 08:43.12 | reza | jaike - was that directed to me? |
| 08:43.12 | benjk | its not |
| 08:43.16 | Qwell | oh, you ARE the professional, eh? |
| 08:43.17 | benjk | not on those phones |
| 08:43.28 | benjk | five professionals have run out out ideas |
| 08:43.39 | Qwell | because you can't give them the info they need |
| 08:43.44 | benjk | following the Cisco documentation by the letter |
| 08:43.44 | jaike | reza: yup...try voicepulse |
| 08:43.53 | reza | they only seem to have unlimited plans |
| 08:44.02 | reza | i just want to buy a couple bucks worth of service to try them out |
| 08:44.05 | Qwell | I've done it multiple times with multiple phones..never had any problems |
| 08:44.06 | benjk | and I can tell you that I have already found a number of things that the documentation claimed that are simply not true |
| 08:44.17 | Qwell | benjk: stop looking at the 6.x release notes |
| 08:44.44 | benjk | I looked at the document how to convert from SCCP to SIP |
| 08:45.35 | tronix | benjk: okay.. cisco sucks. now what? what will you do next? |
| 08:45.35 | reza | are there any patches for asterisk to help with choppy audio? |
| 08:45.35 | Qwell | well, whatever |
| 08:45.46 | reza | or anything i can do to tweak the server for better performance? |
| 08:45.49 | Qwell | just let me know when you're done dicking around, and are ready for a real professional to help you |
| 08:45.56 | mzo | i'm waiting for someone to say 'pay me and i'll fix it' :P |
| 08:46.11 | Qwell | mzo: I offered to do it for free |
| 08:49.31 | jaike | reza: jitterbuffer...but that cant help much. how high is the ping time between your * server and nufone? |
| 08:49.42 | mzo | ha, bad bad |
| 08:51.57 | benjk | I think I had explained the situation with net access |
| 08:53.53 | *** join/#asterisk cuco (n=diego@85-64-243-145.barak-online.net) |
| 08:55.26 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:57.29 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 08:58.48 | benjk | oh, well, 3.2 is the last in the 3.x line but it won't upgrade to 4.0 |
| 08:58.59 | mzo | i'm sure it will but you're doing it wrong :P |
| 08:59.05 | benjk | ah yea |
| 08:59.05 | Qwell | indeed |
| 08:59.24 | benjk | well, Cisco's documentation lists three steps |
| 09:00.15 | benjk | 1) edit OS79XX.TXT to contain the image file name without the .bin ending, then 2) enter the image name likewise in SIPDefault.cnf and 3) reboot the phone |
| 09:03.39 | *** join/#asterisk Ariek48 (n=ariek@141.252.216.37) |
| 09:07.39 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 09:08.39 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:13.27 | Qwell | bed |
| 09:13.36 | Qwell | benjk: if you can't figure it out...suck it up and call cisco |
| 09:14.25 | benjk | I always figure things out, but that's not the point |
| 09:14.44 | benjk | the point is that IT today is a bunch of idiots doing idiotic things |
| 09:14.52 | benjk | while it was well organised in my day |
| 09:15.20 | benjk | and Cisco used to be a cheap crap brand that no self respecting IT should would buy |
| 09:15.25 | Qwell | I think you've just proven your own point, with yourself as the example |
| 09:15.28 | Qwell | but yes, bed |
| 09:15.31 | benjk | real IT shops bought DEC |
| 09:16.19 | benjk | 4.0 firmware on Cisco's website is borked |
| 09:16.44 | benjk | alternative image I just got from a friend works |
| 09:16.46 | mzo | hahaha, 'my day' |
| 09:16.56 | mzo | this conversation is pointless (but entertaining while i'm at work) |
| 09:17.26 | benjk | that's right: in my day |
| 09:18.06 | mzo | if you're old enough to say 'in my day' it's time to consider retiring. DEC hasn't been significant in IT since the early 1990s |
| 09:18.53 | benjk | well, their stuff simply worked |
| 09:19.03 | mzo | no 'it' stuff 'simply worked' |
| 09:19.04 | benjk | and documentation was high quality |
| 09:19.09 | mzo | if you want 'it worked' buy a mac |
| 09:19.16 | benjk | I already got a Mac |
| 09:19.22 | mzo | everything else has their own quirks |
| 09:19.32 | benjk | but unfortunately they don't make IP phones in Cupertino |
| 09:20.10 | mzo | and they never will, so i guess you better call Cisco. :P |
| 09:20.19 | benjk | quirks are acceptable as long as the documentation shows how to work around them |
| 09:20.45 | benjk | about what? fixing their broken links and documentation? |
| 09:20.48 | *** join/#asterisk voipme (n=root@193.120.103.128) |
| 09:27.04 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 09:27.04 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:29.24 | benjk | oh well, the 6.3 image times out |
| 09:29.58 | benjk | who know, cisco.com may have closed for today |
| 09:31.02 | voipme | morning all |
| 09:31.26 | voipme | anyone have $HANGUPCAUSE working successfully on a euroisdn pri |
| 09:31.47 | voipme | all i vere get is 0 returned ona bt or eircom PRI in teh uk or ireland |
| 09:32.10 | voipme | if i d channel debug on teh same line with a cisco box i see teh correct cause |
| 09:36.07 | reza | ok, here's a simple question, how do i add a prefix automatically to an exten? |
| 09:36.07 | reza | exten => _NXXXXXX,1,Dial(sip/1415${EXTEN}) |
| 09:36.12 | reza | that's not working so well |
| 09:36.39 | Zeeek | reza that's not the extension it's the phone's id |
| 09:37.14 | reza | how do i modify the ${EXTEN} then? SetVar? |
| 09:37.22 | reza | there has to be an easy way of doing this |
| 09:37.37 | Zeeek | explain what you are trying to do exactly |
| 09:38.29 | reza | if i dial 1234567, i want to change the extension by prepending "1415" to the string, and recurse |
| 09:38.36 | reza | there's a rule that matches 14151234567 |
| 09:39.02 | reza | exten => _NXXXXXX,1,Dial(default/1415${EXTEN}) |
| 09:39.02 | reza | exten => _1NXXNXXXXXX,1,SetCallerID(8888888888) |
| 09:39.02 | reza | exten => _1NXXNXXXXXX,2,Dial,(IAX2/thereza@NuFone/${EXTEN}) |
| 09:39.05 | Zeeek | If you really have a good reason for recusion, use Local channel |
| 09:39.29 | reza | is there an easier way to do it? |
| 09:39.38 | Zeeek | Local is easy |
| 09:39.45 | Zeeek | Local/12345 |
| 09:39.59 | Zeeek | replace SIP with local |
| 09:42.32 | reza | excellent |
| 09:42.34 | reza | that seemed ot work |
| 09:44.02 | *** join/#asterisk stas (n=hartger@ip51cf10fa.direct-adsl.nl) |
| 09:44.19 | reza | if i want an extension to be '#0', will '_#0' match, or do i have to do some other thing? it doesn't seem happy with it currently |
| 09:45.30 | reza | nevermind, i have some other problem |
| 09:48.27 | reza | i'm doing something wrong... |
| 09:48.31 | reza | [internal] |
| 09:48.31 | reza | exten => _#0,1,Dial,Sip/FrontDesk|60 |
| 09:48.35 | reza | [sip] |
| 09:48.35 | reza | exten => _#.,1,Dial(internal/${EXTEN}) |
| 09:48.51 | reza | when i dial #0 from sip's context I get 'no channel type registered for internal' |
| 09:48.54 | reza | ? |
| 09:49.58 | Zeeek | you made up internal? |
| 09:51.24 | reza | yeah |
| 09:51.37 | pb__ | I guess you meant Dial(Local/${EXTEN}@internal) or something |
| 09:51.54 | reza | so what does local do exactly? |
| 09:51.57 | reza | uses the same context? |
| 09:52.12 | reza | or uses asterisk and not another device? |
| 09:52.53 | Zeeek | read this: http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels |
| 09:53.07 | reza | thanks |
| 09:53.13 | reza | pb- it worked |
| 09:53.47 | pb__ | very good |
| 09:54.18 | reza | can put SetCallerID in [general] or does it have to be associated with every outgoing call? |
| 09:55.48 | *** join/#asterisk morne (n=chatzill@mail.marang.net) |
| 09:56.30 | morne | Hey everyone .. was wondering if there could be a helping hand with a teething problem I have |
| 09:57.02 | morne | I'm runnning FC3 with Asterisk v 1.0.10, on a 2.4Ghz Intel machine 512Mb memory .. |
| 09:57.49 | morne | It seems that everytime I load the wcfxo modules, the CPU itilization drops with about 20-30% and the idle state of the box/server never goes beyond 60% |
| 09:58.12 | morne | I unnload it, and the box is happy at 97% idle time |
| 09:58.21 | morne | Annyone had a problem like this before ? |
| 09:58.46 | benjk | 7.5 on all the 7960s, upgrade path was 2.0, 2.1, 2.2, 2.3, 3.0, 3.2, 4.0, 6.3, 7.5 |
| 09:58.50 | benjk | hilarious |
| 09:59.19 | jaike | morne: any reason why you cant upgrade to 1.2.4? |
| 09:59.59 | morne | yeah ... I have not tested this in the"real" world yet .. |
| 10:00.17 | morne | We load the standardimage on all our servers ... |
| 10:01.01 | morne | I have not seen this problem on any other of our servers out ther e |
| 10:01.26 | morne | I have two TDM400 digium 4 port analogue cards that drive the 8Premicell units .. |
| 10:01.42 | morne | I first thought it might be a card that mightbe faulty .. |
| 10:02.00 | morne | but the results stay the same on BOTH the cards, even if I istall one at a time to test |
| 10:02.21 | morne | As sson as I load the module .. ZAP .. CPU soots to 50% |
| 10:02.59 | *** join/#asterisk dZen|n| (n=AbasCatu@cpe.atm2-0-1081053.0x50a4f886.arcnxx10.customer.tele.dk) |
| 10:03.07 | dZen|n| | hello |
| 10:03.08 | morne | the kernel is 2.6.13-15.7-default |
| 10:03.47 | dZen|n| | I really need an asterisk freak on debian |
| 10:03.48 | dZen|n| | ???? |
| 10:04.42 | pb__ | morne: those cards do generate a lot of interrupts, but you shou;ldn't be seeing 20% load |
| 10:05.14 | morne | I thought it might be interupt issues ... but that utilization on the CPU is plain weird ... |
| 10:05.48 | tzafrir_laptop | dZen|n|, not sure I'm a freak, but I'm known to use Debian |
| 10:06.13 | dZen|n| | well i can't compile zaptel or asterisk |
| 10:06.19 | dZen|n| | I got error 1 |
| 10:06.46 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 10:06.57 | dZen|n| | I read that I should turn off ppp in kernel |
| 10:07.06 | dZen|n| | but I don't know how to do that |
| 10:07.18 | tzafrir_laptop | dZen|n|, what debian? what kernel? what zaptel exactly? (zaptel-source of asterisk.org tarball) |
| 10:07.31 | dZen|n| | kernel |
| 10:07.35 | dZen|n| | 2.4.27 |
| 10:07.35 | tzafrir_laptop | never had to turn off ppp here |
| 10:07.49 | dZen|n| | how do i see the version of zaptel ? |
| 10:08.01 | tzafrir_laptop | dZen|n|, what exactly do you try to build |
| 10:08.32 | dZen|n| | well I wanna run pbx box on lan |
| 10:08.44 | dZen|n| | so i can call my employers |
| 10:08.46 | dZen|n| | ?? |
| 10:10.25 | dZen|n| | tzafrir_laptop: so on which linux did you installed it ? |
| 10:11.31 | jaike | 2.6 kernel would be advisable |
| 10:11.55 | dZen|n| | jaike, I had 2.6 but changed to 2.4 |
| 10:11.57 | dZen|n| | :d |
| 10:12.11 | dZen|n| | I followed this instuction |
| 10:12.12 | dZen|n| | http://users.pandora.be/Asterisk-PBX/InstallAsterisk.htm |
| 10:12.17 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:13.23 | *** join/#asterisk IronHelixz (n=irc@ool-45785cfe.dyn.optonline.net) |
| 10:14.25 | tzafrir_laptop | dZen|n|, hmmm, now I can recommend http://xorcom.com/rapid ... |
| 10:14.44 | tzafrir_laptop | Which is basically a Debian with Asterisk |
| 10:15.05 | tzafrir_laptop | Disclaimer: I'm one of the authors |
| 10:16.10 | dZen|n| | ohh nice |
| 10:16.20 | dZen|n| | can i download this cd ? |
| 10:16.24 | dZen|n| | image |
| 10:16.47 | CaT[tm] | does it play well with a normal debian install? |
| 10:18.09 | dZen|n| | tzafrir_laptop: is this a image with debian and asterisk on it ? |
| 10:18.25 | tzafrir_laptop | Yes |
| 10:18.29 | dZen|n| | nice |
| 10:18.35 | dZen|n| | debian net_install = |
| 10:18.37 | tzafrir_laptop | A minimal Sarge |
| 10:18.46 | dZen|n| | ok nice |
| 10:19.26 | CaT[tm] | do you plan on updating the asterisk version? |
| 10:19.47 | dZen|n| | CaT[tm]: who do you ask |
| 10:20.02 | CaT[tm] | sorry. Tzafrir :) |
| 10:20.14 | reza | i'm confusing myself abou tthe zapata.conf file |
| 10:20.40 | reza | if i have 3 fxs ports on a tdm400, then do i need to allocate 3 channels (i.e. channel => 1-3) |
| 10:21.57 | reza | i think i broke something; i used to get a dialtone on the outgoing channels, now nothing |
| 10:22.22 | dZen|n| | lol |
| 10:22.27 | dZen|n| | shit happens |
| 10:22.28 | dZen|n| | :d |
| 10:22.32 | reza | grumble |
| 10:22.56 | reza | what file defines how an fsx should work? |
| 10:24.57 | tzafrir_laptop | CaT[tm], working on it as we speak |
| 10:25.43 | tzafrir_laptop | CaT[tm], yes, it should. Generally, try one of the following two sources with Sarge: |
| 10:26.02 | tzafrir_laptop | deb http://rapid.dotsrc.org/rapid sarge main |
| 10:26.11 | tzafrir_laptop | deb http://rapid.dotsrc.org/ unstable/ |
| 10:26.23 | tzafrir_laptop | (deb-src works as well) |
| 10:28.53 | tuxinator_linux | Rez, read the docs? |
| 10:28.57 | tuxinator_linux | ~docs |
| 10:28.58 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 10:29.35 | Zeeek | docs are not to be read; they exist to be complained about only |
| 10:29.46 | tuxinator_linux | Zeeek, seems that way |
| 10:30.04 | Zeeek | and then only after complaining about the support here ;) |
| 10:30.06 | tuxinator_linux | Am I the only goofball that read them? |
| 10:30.38 | dudes | doubt it |
| 10:30.40 | Zeeek | I've drifted into ignorance, but when I began, I read everything I could get my hands on |
| 10:30.49 | tuxinator_linux | as did I |
| 10:30.54 | CaT[tm] | tzafrir: cool, because installing amp in a way that doesn't futz everything is, well, ever so entertaining. |
| 10:30.57 | Zeeek | especially the docs that became the O'Reilly book |
| 10:31.28 | tuxinator_linux | I had to read it, know the authors, felt obligated |
| 10:31.50 | Zeeek | I proofed it a hundred times |
| 10:31.54 | jaike | zeeek: i enjoyed reading the oreilly book. learned a lot |
| 10:32.11 | Zeeek | and the IRQ listing is from my asterisk box - I'm so proud :) |
| 10:32.31 | *** join/#asterisk draga (i=draga@host209-235.pool8253.interbusiness.it) |
| 10:32.33 | jaike | which reminds me..gotta return it to the office |
| 10:32.53 | draga | hello everybody |
| 10:32.55 | tuxinator_linux | Zeeek is the proud parent of a IRQ listing, congrats ! |
| 10:33.48 | tuxinator_linux | I need to stop chatting so far past my bed time, start talking silly |
| 10:34.05 | Zeeek | haha - it's 11:34 AM here |
| 10:34.08 | Zeeek | go to bed |
| 10:34.57 | tuxinator_linux | 2:30 AM here in SoCal |
| 10:35.16 | Zeeek | socal? Wherebouts? |
| 10:35.29 | Zeeek | city, zip and geo coordinates? |
| 10:35.56 | tuxinator_linux | Yorba Linda, 92886 29887, geo....let me see |
| 10:36.22 | Zeeek | That's near San DIego? |
| 10:36.31 | tuxinator_linux | pretty close |
| 10:36.37 | tuxinator_linux | Take day trips there |
| 10:36.46 | Zeeek | so I'm not as senile as they think! |
| 10:36.55 | Zeeek | I lived in SoCal for 10 years |
| 10:37.01 | Zeeek | never near SD tho |
| 10:37.06 | tuxinator_linux | oh ya? |
| 10:37.19 | tuxinator_linux | I grew up here, moved away for 15 years, and now I'm back |
| 10:37.30 | Zeeek | Orange County, East LA, West Hollywood |
| 10:37.39 | Zeeek | Manhattan Beach |
| 10:37.46 | Zeeek | and even ... the Valley |
| 10:37.59 | tuxinator_linux | Which valley? |
| 10:38.12 | tuxinator_linux | the yucky LA valley? |
| 10:38.21 | tuxinator_linux | or the yucky Riverside Valley |
| 10:38.27 | Zeeek | In fact I was remembering the other day that calling Orange COunty from L.A. was way more expensive as it is now to call the US from Europe (using voIP) |
| 10:38.34 | trixter | Wireless VoIPon the horizon http://www.itwales.com/799581.htm This year could well see wireless VoIP (voice over internet protocol) telephony emerge on the communications scene, according to a new report by IDC |
| 10:38.46 | Zeeek | San Fernando Valley (when you say the valley, that's the one) |
| 10:39.00 | trixter | depends on where you are |
| 10:39.17 | tuxinator_linux | I'm still learning the lingo, my highschool years were in arizona |
| 10:39.18 | Zeeek | I lived in the San Joachin also |
| 10:39.19 | trixter | its like 'the city' in northern NJ its NY city, on the san francisco penninsula its san francisco |
| 10:39.30 | trixter | in sacramento 'the valley' means the sacramento valley |
| 10:39.46 | Zeeek | but in L.A. the valley is The Valley |
| 10:40.07 | trixter | yeah there, it depends, I didnt catch the first part of that so I didnt realize that you were being context sensitive already |
| 10:40.19 | tuxinator_linux | the only reference to "The Valley" is in "Clueless" (Silverstone, pretty hot) |
| 10:40.27 | *** join/#asterisk fulgas (n=fulgas@209.8.233.12) |
| 10:40.41 | trixter | and it meant that same place when I lived in LA in the 70s when the sky was always orange and news reports would warn you not to let kids and pets outside |
| 10:40.59 | trixter | tuxinator_linux: valley girls are girls generally from that area that talk and act a certain way |
| 10:41.06 | trixter | like oh my god! |
| 10:41.16 | trixter | and they were rampant when I lived there |
| 10:41.25 | dudes | hate fuckable bitches |
| 10:41.27 | tuxinator_linux | Entertaining/Anoying girls |
| 10:41.36 | Zeeek | you can get a "Valley Girl" voice from Cepestral |
| 10:41.50 | tuxinator_linux | Zeeek, You may appriciate subsonicradio.com |
| 10:41.59 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:42.04 | tuxinator_linux | Zeeek, or hate it |
| 10:42.07 | Zeeek | heh |
| 10:42.13 | Zeeek | is it RealMedia? |
| 10:42.17 | tuxinator_linux | MP3 |
| 10:42.20 | Zeeek | k |
| 10:42.36 | tuxinator_linux | Mplayer streams it fine for me |
| 10:42.41 | Zeeek | as soon as I figure out why the table isn't being created in my db, I'll go take a listen |
| 10:43.13 | tuxinator_linux | ~cepestral |
| 10:43.19 | tuxinator_linux | what is cepestral? |
| 10:43.33 | Zeeek | a voice software you can use with asterisk |
| 10:43.37 | tuxinator_linux | ahh |
| 10:43.41 | Zeeek | I can never remember their site URL |
| 10:43.42 | *** join/#asterisk newl (n=newlook@203-59-210-244.dyn.iinet.net.au) |
| 10:43.57 | Zeeek | Wait, maybe it was the other people that had the Valley Girl voice |
| 10:44.05 | tuxinator_linux | Zeeek, so, are you in UK or Australia? |
| 10:44.12 | Zeeek | Yo can use the demo on the web to make it say any dirty stuff you want |
| 10:44.20 | Zeeek | no I'm in France |
| 10:44.22 | tuxinator_linux | must be funny |
| 10:44.35 | tuxinator_linux | France, SoCal to France, bit of a change |
| 10:44.40 | Zeeek | Ya like you have it says "I got yer zaptel right here!" |
| 10:44.55 | Zeeek | Born in Minnesota - quite a change indeed! |
| 10:44.55 | CoKane | Hy guys I installed the 4port BRI card from Junghanns |
| 10:45.13 | CoKane | looking for help in setting it up for incoming calls |
| 10:45.27 | tuxinator_linux | ~docs |
| 10:45.29 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 10:45.46 | *** join/#asterisk Ariek48 (n=ariek@e37-216.dot1x.nhl.nl) |
| 10:46.01 | tuxinator_linux | CoKane, if you have read the docs, then ask your question |
| 10:46.20 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 10:46.20 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 10:47.59 | CoKane | ok I have this is my zaptel.conf |
| 10:48.07 | CoKane | loadzone=uk |
| 10:48.07 | CoKane | defaultzone=uk |
| 10:48.07 | CoKane | # qozap span definitions |
| 10:48.07 | CoKane | # most of the values should be bogus because we are not really zaptel |
| 10:48.08 | CoKane | span=1,1,3,ccs,ami |
| 10:48.08 | CoKane | span=2,2,3,ccs,ami |
| 10:48.10 | CoKane | span=3,0,3,ccs,ami |
| 10:48.12 | CoKane | span=4,0,3,ccs,ami |
| 10:48.14 | CoKane | bchan=1,2 |
| 10:48.16 | CoKane | dchan=3 |
| 10:48.18 | CoKane | bchan=4,5 |
| 10:48.20 | CoKane | dchan=6 |
| 10:48.22 | trixter | ~pb |
| 10:48.24 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 10:48.24 | CoKane | bchan=7,8 |
| 10:48.24 | tuxinator_linux | ~pastebin |
| 10:48.26 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 10:48.26 | CoKane | dchan=9 |
| 10:48.26 | CoKane | bchan=10,11 |
| 10:48.28 | CoKane | dchan=12 |
| 10:48.39 | tuxinator_linux | CoKane, use a pastebin |
| 10:48.55 | CoKane | sorry guys, will do |
| 10:49.46 | RoyK | ~pb |
| 10:49.48 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 10:50.11 | tuxinator_linux | Morning RoyK, a little late for a ~pb ;-) |
| 10:50.27 | CoKane | ok here is my Zapata.conf |
| 10:50.28 | CoKane | http://pastebin.com/543030 |
| 10:51.20 | tuxinator_linux | I don't have experience with BRI's, but someone here may be able to help |
| 10:53.04 | tuxinator_linux | eyes are too heavy, Night guys |
| 10:54.04 | reza | ok, i really broke something here. what do i have to do to generate a dialtone on a fxs card when i pick up the handset? |
| 10:54.06 | reza | it used to work |
| 10:54.37 | CoKane | ok to save confusion I wrote a doc while installing the BRI card, |
| 10:54.39 | CoKane | http://pastebin.com/543034 |
| 10:54.53 | tuxinator_linux | RoyK, are you exposing me to the world? |
| 10:55.10 | dZen|n| | tzafrir_laptop: ok I am about to burn the image iso, would it install everything from cd or do I need to make some moves ? |
| 10:57.14 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 10:58.24 | Money5ack | argl... why did asterisk answers sip messages always on the first ethernet interface and not on the interface that the phone sends the request to ?! |
| 11:00.01 | stas | <PROTECTED> |
| 11:01.18 | Money5ack | no routing... the asterisk has 3 public ips. one normal ip and two ip-aliases but all on the same subnet and the same interface |
| 11:01.58 | stas | how is your default route then? |
| 11:02.03 | stas | on which interface? |
| 11:02.11 | Money5ack | mom |
| 11:03.30 | Money5ack | www.xxx.yyy.zzz 0.0.0.0 255.255.255.192 U 0 0 0 eth0 |
| 11:03.33 | Money5ack | 0.0.0.0 www.xxx.yyy.zzz 0.0.0.0 UG 0 0 0 eth0 |
| 11:03.36 | Money5ack | 0.0.0.0 www.xxx.yyy.zzz 0.0.0.0 UG 0 0 0 eth0 |
| 11:03.41 | Money5ack | that is the routing table |
| 11:04.24 | trixter | I have a routing table |
| 11:04.31 | trixter | I eat dinner on it from time to time |
| 11:04.38 | Money5ack | :P |
| 11:05.02 | stas | hehe :-) |
| 11:05.38 | Money5ack | so my interfaces are named eth0, eth0:1 and eth0:2 as ips for example 192.168.1.1, 192.168.1.10 and 192.168.1.20 |
| 11:05.39 | stas | so it is normal behavior that your asterisk server sends on the default IP address thats your default route |
| 11:06.03 | stas | if you have a SIP request on 192.168.1.20 |
| 11:06.21 | stas | and your default is on eth0 |
| 11:06.24 | mut | umm |
| 11:06.29 | mut | for the zapata config |
| 11:06.48 | stas | but maybe you have a setting that you specify hard the outgoing ip ? |
| 11:06.51 | mut | rxgain and txgain aren't per channel are they? |
| 11:06.53 | stas | in asterisk |
| 11:07.12 | stas | wait i loook :-) |
| 11:07.22 | mut | yea stas |
| 11:07.26 | Money5ack | the setting externalip or somethink is there.. |
| 11:07.29 | Money5ack | but this is for nat |
| 11:07.48 | Money5ack | and i need this setting as user/peer-setting.. |
| 11:08.36 | Money5ack | because some clients will register on ip 192.168.1.10 and others on 192.168.1.20 |
| 11:09.18 | stas | hmm |
| 11:09.43 | *** join/#asterisk crich1999 (n=crich@p54BF99B1.dip0.t-ipconnect.de) |
| 11:09.53 | stas | i wil look for an solutions first lunch :-) |
| 11:10.16 | reza | ~dialtone |
| 11:10.52 | Money5ack | hum... |
| 11:11.00 | reza | ok, giving up for the day |
| 11:11.02 | *** join/#asterisk donnib (n=aaa@0x555281d0.adsl.cybercity.dk) |
| 11:11.10 | Money5ack | but... *think* |
| 11:11.20 | donnib | anyone now how i can debug a No such host exist message ? |
| 11:11.35 | Money5ack | my phone is on another ip-subnet... |
| 11:11.53 | Money5ack | and routing goes thru eth0 |
| 11:11.54 | Money5ack | hmm |
| 11:13.10 | donnib | anyone ? |
| 11:13.15 | Money5ack | ah... no ... that isn't the failure... because there is a softphone in the same subnet that can't register too... |
| 11:13.18 | Money5ack | damn.. |
| 11:13.38 | Money5ack | ngrep |
| 11:13.44 | dZen|n| | see sometimes it's good to speak/chat with your self hehehe |
| 11:14.02 | Money5ack | donnib: ngrep -s 1500 -W byline port 5060 |
| 11:15.05 | donnib | i am running AsteriskWin32 |
| 11:15.10 | donnib | can't do that |
| 11:15.16 | donnib | i mean run ngrep |
| 11:15.22 | Money5ack | hmm |
| 11:15.30 | dZen|n| | ok I have an stupid q, do i need a ip phone to use asterisk or can i use my headsets ? |
| 11:15.36 | donnib | i only have the CLI |
| 11:15.53 | Money5ack | donnib: sip debug |
| 11:15.55 | reza | you can use a headset |
| 11:16.34 | dZen|n| | how about calls, how do i make them ? |
| 11:16.46 | dZen|n| | Money5ack: donnib is on windows |
| 11:17.53 | Money5ack | dzen: i know. didn't "sip debug" work under win32 ? |
| 11:19.04 | mut | fuckin a |
| 11:19.10 | donnib | sip debug is on |
| 11:19.13 | mut | my portmaster didn't come up when i rebooted it |
| 11:19.30 | donnib | still getting chan:sip.c:4070 sip_reg_timeout |
| 11:19.34 | mut | this thing ahd better come back up |
| 11:19.35 | donnib | no such host. |
| 11:19.37 | mut | i sure as hell hope i don't have to drive downstate |
| 11:19.48 | donnib | i know that the host is correct |
| 11:20.01 | *** join/#asterisk madounet (n=mad_net@juv34-2-82-226-155-19.fbx.proxad.net) |
| 11:31.12 | reza | ok, i will paypal someone $5 if they can tell me what i did to break my config - i cant get a dialtone out of an tdm400 fxs card - i can call the port from a sip phone, but i hear no audio |
| 11:31.41 | mut | you're behind nat |
| 11:31.45 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 11:31.48 | mut | paypal me plz |
| 11:31.51 | reza | it's all on a lan |
| 11:31.53 | reza | :P |
| 11:32.04 | dZen|n| | extension number ? < which number is that ? |
| 11:32.35 | reza | it's the number in the extensions.conf file; you configure it |
| 11:32.51 | dZen|n| | ooooh |
| 11:32.52 | dZen|n| | :P |
| 11:33.02 | reza | are you asking me? |
| 11:33.24 | dZen|n| | who me ? |
| 11:33.27 | reza | nevermind |
| 11:33.30 | dZen|n| | ok |
| 11:33.37 | dZen|n| | i was asking everybody hehe |
| 11:33.38 | dZen|n| | :d |
| 11:33.49 | dZen|n| | is there a guide how to use this ? |
| 11:33.54 | dZen|n| | on net |
| 11:34.05 | dZen|n| | insted of I ask you I could read |
| 11:34.19 | reza | google for voip wiki |
| 11:34.30 | reza | that's the site that i've been using to try to figure this shit out |
| 11:34.35 | reza | drivng myself nuts |
| 11:34.41 | reza | it worked just fine |
| 11:34.52 | reza | i teaked the hardware a bit, moved it to a different room, it broke |
| 11:35.00 | reza | i've reverted the hardware and it still doesn't work |
| 11:35.17 | reza | i'm upgrading the software to the latest version to see if that helps, though i doubt it would |
| 11:35.26 | dZen|n| | hahaha |
| 11:35.32 | dZen|n| | well thtas some shit |
| 11:35.51 | dZen|n| | it allways goes down when it just work perfect |
| 11:35.53 | dZen|n| | :d |
| 11:35.57 | reza | i just dont get it |
| 11:36.26 | reza | it's as if the hardware can't recognize that i've picked up the phone |
| 11:36.28 | reza | hmm |
| 11:36.38 | dZen|n| | well it can be a small problem or a big... |
| 11:36.54 | dZen|n| | you always need to have a plan on how to troubleshoot |
| 11:36.55 | dZen|n| | :d |
| 11:37.11 | dZen|n| | always start with hardware and end with software |
| 11:37.35 | dZen|n| | in details |
| 11:38.09 | tzafrir_laptop | ~voip-info |
| 11:38.10 | jbot | it has been said that voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 11:38.48 | dZen|n| | tzafrir_laptop: you are to slow for google hehhehe |
| 11:39.09 | *** join/#asterisk vinsik (n=vinsik@gw-ff.verkkokauppa.com) |
| 11:39.59 | dZen|n| | tzafrir_laptop: this cd you and your group maded is damn good, never saw an easyer way to use somthing on linux |
| 11:40.18 | tzafrir_laptop | thanks |
| 11:40.25 | dZen|n| | i have tryied with suse 9.3 and I just got the same error as in all other versions |
| 11:40.47 | dZen|n| | thx very much |
| 11:43.08 | *** join/#asterisk NassoR (n=root@step.funcitec.rct-sc.br) |
| 11:44.24 | fenlander | hi, is anyone working on a chan_jingle to interface with google talk? |
| 11:44.55 | trixter | doesnt google use jabber? |
| 11:45.24 | fenlander | yes - with jingle for voice iirc |
| 11:46.19 | fenlander | the specs have been available for a while - I was wonderig if anyone has started work on it |
| 11:46.27 | dZen|n| | damn I think I founded a bug in mirc.. |
| 11:46.40 | dZen|n| | but i don't how to explain :d |
| 11:46.56 | dZen|n| | but this is a huge bug |
| 11:47.25 | *** part/#asterisk madounet (n=mad_net@juv34-2-82-226-155-19.fbx.proxad.net) |
| 11:48.53 | dZen|n| | hmm is debian.org and freenode.org the same irc server ? |
| 11:49.27 | RaYmAn-Bx | considering irc.debian.org is just a cname to chat.freenode.org, I suspect the answer is yes :P |
| 11:49.36 | RaYmAn-Bx | freenode.net actually |
| 11:49.57 | dZen|n| | damn then there is no bug :D |
| 11:50.49 | trixter | there is no spoon |
| 11:51.05 | CaT[tm] | indeed for I am eating steak. GO THE FORK! |
| 11:53.38 | stas | <PROTECTED> |
| 11:53.39 | stas | cleared, reason 7 (Remote user stopped calling [65 - Bearer capability not implemented]) |
| 11:54.43 | jaike | very few h323 using people here |
| 11:55.17 | stas | hmm |
| 11:56.24 | tzafrir_laptop | dZen|n|, irc.debian.org , not debian.org |
| 11:57.17 | dZen|n| | well i know |
| 12:00.58 | *** join/#asterisk EriSan (n=erisan@151.8.109.74) |
| 12:01.52 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 12:09.35 | *** join/#asterisk morne (n=chatzill@dsl-146-76-159.telkomadsl.co.za) |
| 12:15.09 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 12:23.12 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 12:32.17 | *** join/#asterisk zotz (n=zotz@24.244.133.10) |
| 12:36.56 | *** join/#asterisk bigjb (n=bigjb@195.60.10.114) |
| 12:45.02 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 12:46.41 | *** part/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 12:46.48 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 12:47.18 | mut | anyone know how to get caller-id name to pass thru a cisco as5350? |
| 12:47.22 | mut | it's not coming through |
| 12:50.54 | *** part/#asterisk rene- (i=rene@dsl-201-128-115-222.prod-infinitum.com.mx) |
| 12:51.36 | voipme | mut: works with qsig |
| 12:51.45 | voipme | what setup you using |
| 12:51.49 | voipme | pri? |
| 12:51.56 | voipme | e1 or t1 |
| 12:52.03 | mut | t1 |
| 12:52.32 | voipme | euro based im afraid, not much good with t1 |
| 12:53.01 | voipme | does your isdn debug on the cisco show you receiving or sending it? |
| 12:53.17 | mut | yea |
| 12:53.25 | mut | if i hookup my te405p |
| 12:53.28 | mut | i get name |
| 12:54.55 | voipme | whats your cisco config? |
| 12:55.54 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 12:57.30 | mut | what should it be? |
| 12:58.09 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 13:00.51 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:06.15 | hypnox | can you call macros from macros? |
| 13:07.00 | kaldemar | sure you can. |
| 13:10.30 | *** join/#asterisk bartpbx (n=bartpbx@proxy.prodyna.com) |
| 13:14.25 | mut | O_o |
| 13:14.44 | Morex | Anybody out there have experience using Asterisk with Avaya? |
| 13:15.01 | *** join/#asterisk vattern (n=vattern@dsl-146-148-171.telkomadsl.co.za) |
| 13:16.47 | bartpbx | hello |
| 13:17.18 | bartpbx | I want to start using dundi. How wants to peer with me in a test setup? |
| 13:17.43 | mut | do any hardware video phones exist? |
| 13:17.50 | mut | sip/voip |
| 13:18.20 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:18.57 | mut | like the grandstream gxv-3000 |
| 13:19.06 | mut | is it even worthwhile |
| 13:24.00 | *** join/#asterisk meljack (n=esculli@host172138.metrored.net.ar) |
| 13:24.50 | meljack | hi... |
| 13:25.20 | meljack | anyboly uses the parameter "callprogress" in the file zapata.conf? |
| 13:25.52 | bartpbx | noone using dundi here? |
| 13:27.12 | *** join/#asterisk madounet (n=mad_net@juv34-2-82-226-155-19.fbx.proxad.net) |
| 13:27.22 | *** part/#asterisk madounet (n=mad_net@juv34-2-82-226-155-19.fbx.proxad.net) |
| 13:31.39 | meljack | anybody uses the parameter "callprogress" in the file zapata.conf? |
| 13:32.17 | bartpbx | no |
| 13:32.20 | fugitivo | no |
| 13:33.00 | mut | no |
| 13:33.11 | mut | will you stop asking now? |
| 13:33.15 | mut | ..... |
| 13:33.17 | mut | will you stop asking now? |
| 13:34.29 | meljack | no.. |
| 13:35.55 | meljack | when I call to an external number (over a zap channel), I don’t receive any event when the target answer, Who can help me? |
| 13:37.35 | *** join/#asterisk AlexLee (n=AlexLee@221.221.173.42) |
| 13:37.44 | AlexLee | ok.. |
| 13:37.50 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 13:38.27 | meljack | when I call to an external number (over a zap channel), I don’t receive any event when the target answer, Who can help me? |
| 13:38.38 | brimstone | dude, chill |
| 13:38.53 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 13:39.10 | bartpbx | meljack, please stop this. I there where anyone how could help you he would answer you! |
| 13:40.14 | jaike | theyre all asleep |
| 13:40.39 | meljack | ok, can you help me bartpbx? |
| 13:40.50 | bartpbx | no. I cant |
| 13:41.05 | bartpbx | if I could i would have answered you |
| 13:41.11 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 13:41.17 | meljack | ok, thanks |
| 13:41.48 | meljack | when is more better ask? |
| 13:41.58 | iDunno | repeating the question, however, is probably going to stop anyone that might no the answer from answering you too - it's considered damned rude ;) |
| 13:42.05 | jaike | when they away is better more ask |
| 13:42.07 | iDunno | s/no/know/; |
| 13:42.09 | jaike | awake |
| 13:43.29 | *** join/#asterisk [Atlas] (n=whois@216.190.144.90) |
| 13:44.18 | meljack | jajaja |
| 13:44.33 | bartpbx | german? |
| 13:45.03 | [Atlas] | astrican? |
| 13:49.34 | znoG | what exactly is a "catch block" in ael? |
| 13:49.39 | znoG | <PROTECTED> |
| 13:49.40 | znoG | <PROTECTED> |
| 13:49.40 | znoG | <PROTECTED> |
| 13:49.40 | znoG | <PROTECTED> |
| 13:49.45 | znoG | ie. i don't understand what "catch a" does |
| 13:50.18 | znoG | oh, maybe "a" is for "asterisk" (as in the * key) |
| 13:50.53 | meljack | where is this code? |
| 13:51.15 | znoG | in the wiki |
| 13:51.46 | meljack | what is the application? |
| 13:51.53 | brookshire | Macros |
| 13:51.53 | brookshire | ------------------------- |
| 13:51.53 | brookshire | A macro is defined in its own block like this. The arguments to the macro are |
| 13:51.53 | brookshire | specified with the name of the macro. They are then reffered to by that same |
| 13:51.53 | brookshire | name. A catch block can be specified to catch special extensions. |
| 13:52.20 | brookshire | so i guess it's a macro to catch an extension |
| 13:52.23 | brookshire | lol |
| 13:52.24 | znoG | yes I read' that bit, to catch special extension |
| 13:52.38 | znoG | but what special extensions... maybe a is for "asterisk" |
| 13:53.54 | brookshire | oh |
| 13:53.58 | *** join/#asterisk AlexLee (n=AlexLee@221.221.173.42) |
| 13:54.00 | meljack | what special extensions do you catch? |
| 13:55.02 | znoG | exactly, what I want to find out |
| 13:55.13 | znoG | i couldn't find much in the way of documentation for AEL, too |
| 13:55.44 | meljack | sorry but... what is AEL? |
| 13:55.57 | brookshire | it's like dialplan 2.0, lol |
| 13:56.09 | jaike | extensions reloaded |
| 13:57.50 | brookshire | hah.. all i can find is 'h' means hangup |
| 14:00.29 | znoG | http://www.voip-info.org/wiki/index.php?page=Asterisk+standard+extensions |
| 14:00.30 | znoG | there we go |
| 14:00.32 | brookshire | a must mean answer |
| 14:01.00 | brookshire | or not! |
| 14:02.53 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 14:04.01 | znoG | would be nice if there was a way to check the status of SIP extensions before dialing them, but i guess the SIP protocol wasn't designed like that |
| 14:05.18 | znoG | actually it looks like you can with some SIP phones, like the SNOM |
| 14:09.44 | iCEBrkr | yo yo yo |
| 14:10.14 | vinsik | I cant send fax to my asterisk from a local PSTN Fax. |
| 14:10.15 | vinsik | -- Executing RxFAX("SIP/xxxx-081aaa88", "/home/asterisk/html/fax/_1139320878.1.tiff") in new stack |
| 14:10.15 | vinsik | Feb 7 16:02:19 NOTICE[8984]: chan_sip.c:12090 do_monitor: Disconnecting call 'SIP/xxxx-081aaa88' for lack of RTP activity in 61 seconds |
| 14:10.15 | vinsik | Anybody knows what is wrong? |
| 14:10.36 | vinsik | using version Asterisk SVN-trunk-r9157M |
| 14:11.12 | vinsik | with T38 support... |
| 14:12.04 | vinsik | i guess nobody sends faxes :) |
| 14:12.27 | *** part/#asterisk bartpbx (n=bartpbx@proxy.prodyna.com) |
| 14:12.57 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 14:13.32 | AlexLee | hi, I am from china, I hope wirte extenstion No. from other service or program into asterisk server, but I do not know how to config or programing, anyone can help????? |
| 14:13.48 | AlexLee | help!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 14:13.51 | AlexLee | hi, I am from china, I hope wirte extenstion No. from other service or program into asterisk server, but I do not know how to config or programing, anyone can help????? |
| 14:15.07 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 14:15.11 | vattern | maybe use the mysql realtime utils and populate your db from your other source? |
| 14:15.57 | AlexLee | oh... |
| 14:16.03 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 14:16.10 | AlexLee | any utils? |
| 14:16.45 | AlexLee | would you please show me, vattern? |
| 14:17.09 | AlexLee | hi, I am from china, I hope wirte extenstion No. from other service or program into asterisk server, but I do not know how to config or programing, anyone can help????? |
| 14:17.18 | vinsik | alexlee: uhh.. www.voip-info.org |
| 14:17.24 | znoG | AlexLee: STOP REPEATING!!!!!!!!!!!!!!!! |
| 14:17.56 | *** join/#asterisk sch19 (n=sch19@adsl-2-114-222.mia.bellsouth.net) |
| 14:18.08 | AlexLee | ok, i saw but there is no help document for my reference on www.voip-info.org |
| 14:18.39 | vattern | AlexLee: I do not know how either, but thats how I would do it .. now please RT Fine Manual |
| 14:19.05 | AlexLee | Fine Manual? |
| 14:19.17 | znoG | s/ine/ucking |
| 14:19.17 | AlexLee | where should i download, please? |
| 14:19.18 | iDunno | yeah - it's the polite version :) |
| 14:19.30 | *** join/#asterisk vivekj1 (i=1076@203.199.110.93) |
| 14:20.03 | _Sam-- | i havent seen that version! |
| 14:20.25 | vattern | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions |
| 14:20.29 | AlexLee | vattern? would you show me what utils to use? |
| 14:20.56 | vattern | as I said .. I do not know .. |
| 14:20.57 | vinsik | AlexLee: first you need to calm down. |
| 14:21.06 | vinsik | AlexLee: cant be that important :) |
| 14:21.06 | AlexLee | ok, thanks a lot... |
| 14:21.11 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:21.22 | AlexLee | oh... |
| 14:21.47 | vinsik | AlexLee: then try to explain what r u trying to do? |
| 14:21.48 | AlexLee | Vinsik, what do you mean? i am confused... |
| 14:21.53 | vinsik | normally |
| 14:22.07 | vinsik | i cant understand what are u trying to do. |
| 14:22.31 | AlexLee | ok.. Vinsik, i can explain... |
| 14:23.04 | AlexLee | i have website.. |
| 14:23.51 | AlexLee | i hope i can provide a extension No. to users after they register... |
| 14:24.46 | AlexLee | and.. they can login the Voip client to make call |
| 14:24.53 | AlexLee | but... |
| 14:25.18 | vinsik | AlexLee: are u using any kind of database on your webserver? e.g. MySQL, etc? |
| 14:25.34 | AlexLee | but the database of my website and the database of asteriske are divided |
| 14:25.52 | vinsik | AlexLee: different machines? |
| 14:26.06 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:27.10 | AlexLee | mysql |
| 14:27.27 | AlexLee | yes..different machines... |
| 14:27.44 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 14:28.20 | vinsik | AlexLee: i would do like this: Build a united database that can accessed by your asterisk and webserver... Made an extension rule that uses addon mysql module to get sip user by regexten from database. |
| 14:30.10 | *** join/#asterisk holmeh (i=holm@blackedge.org) |
| 14:31.51 | AlexLee | my Sql |
| 14:32.26 | *** join/#asterisk ZX81 (n=ZX81@213-140-22-78.fastres.net) |
| 14:32.54 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 14:33.04 | shmaltz | list is down? or is it gmail? |
| 14:33.12 | ZX81 | not down for me |
| 14:33.13 | ZX81 | :) |
| 14:33.20 | AlexLee | but how make asterisk to access to the united database? |
| 14:33.48 | vinsik | alexlee: install asterisk-addons |
| 14:34.08 | vinsik | alexlee: and real Reatime configuration from www.voip-info.org |
| 14:34.12 | vinsik | read even |
| 14:34.33 | ZX81 | ~ping |
| 14:34.34 | jbot | pong |
| 14:34.42 | ZX81 | ~adn |
| 14:34.45 | jbot | extra, extra, read all about it, adn is the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://www.sineapps.com/rssfeed.php for RSS |
| 14:34.58 | ZX81 | ~pong |
| 14:35.00 | jbot | | . |
| 14:35.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:35.38 | *** join/#asterisk SparFux (n=player@tor/session/x-372385c7b4822796) |
| 14:35.57 | AlexLee | vinsik, would you login to my server to check whether i can do like you said |
| 14:36.27 | *** join/#asterisk upsite (n=upsite@wls.swh.uni-halle.de) |
| 14:37.16 | SparFux | I want asterisk to *immediately* enter "h" extension once the peer has hungup the call. But it seems to first terminate the current extension before executing "h". What can I do? |
| 14:38.35 | *** join/#asterisk Luke-Jr (n=luke-jr@CPE-24-31-249-53.kc.res.rr.com) |
| 14:38.46 | upsite | hey guys i got a problem, i switched to the non-root installation and now my queues are not plaing the voicepromts |
| 14:38.59 | upsite | is there maybe a problem with the permission handling? |
| 14:39.02 | *** join/#asterisk Aughey (n=jha@ns1.washucsc.org) |
| 14:39.32 | SparFux | upsite: probably. What about the paths to the sound files? |
| 14:39.32 | SparFux | Are they world readable? |
| 14:39.49 | *** join/#asterisk crich_ (n=crich@p54BFC163.dip0.t-ipconnect.de) |
| 14:40.05 | upsite | if i just copy the sounds dir from the * install it is working |
| 14:40.11 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 14:40.22 | upsite | but if i set the permissions to the user my* is running its nonworking |
| 14:40.31 | _Sam-- | <PROTECTED> |
| 14:41.18 | upsite | vm ? |
| 14:41.39 | _Sam-- | <PROTECTED> |
| 14:41.54 | upsite | im taling bout the sounds from the queue ? |
| 14:42.08 | upsite | !talking |
| 14:42.14 | _Sam-- | oh hah...i thought you said voicemail |
| 14:42.36 | upsite | hehe no that one is working fine |
| 14:42.36 | _Sam-- | do you know what dir the sound files are in for your install? |
| 14:43.00 | upsite | normally i place them in /home/asterisk/var/lib/asterisk/sounds |
| 14:43.16 | upsite | but this is completly nonworking |
| 14:43.41 | _Sam-- | you just did make install from source? or how did you isntall the new version |
| 14:43.42 | _Sam-- | and what is it |
| 14:43.48 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 14:43.50 | upsite | in ma *.conf i set astvarlibdir to /home/asterisk/var/lib/asterisk |
| 14:44.02 | _Sam-- | hey rich |
| 14:44.14 | _Sam-- | john send you here to learn up? :) |
| 14:44.22 | rculp | :) |
| 14:44.27 | rculp | just to ask about the grandstream update |
| 14:44.29 | upsite | make INSTALL_PREFIX=/home/asterisk |
| 14:44.36 | _Sam-- | what about rich, it works well |
| 14:44.39 | _Sam-- | i told him to do it! |
| 14:44.42 | _Sam-- | bastard wont listen |
| 14:44.44 | rculp | to see if it fixes speakerphone as well |
| 14:44.47 | upsite | and then i chmod the whole * dir to asterisk.asterisk |
| 14:44.49 | znoG | hehe bkw's interview is funny |
| 14:45.07 | _Sam-- | asterisk cant read your /home |
| 14:45.09 | _Sam-- | most likely |
| 14:45.21 | _Sam-- | er within there, it cant get in there... |
| 14:45.31 | Nugget | znoG: url? |
| 14:45.34 | _Sam-- | are you sure its all owned by asterisk? |
| 14:45.38 | upsite | yes |
| 14:45.50 | rculp | because the issue we have with grandstreams is not that they don't work well, but that they sound horrible on speaker |
| 14:45.51 | rculp | :) |
| 14:46.05 | rculp | no ec on them |
| 14:46.09 | upsite | for bugtracking i changed the astvarlibdir du /var/lib/asterisk |
| 14:46.23 | upsite | and copied the sounds dir form the install to this location |
| 14:46.24 | _Sam-- | you have sounds there? |
| 14:46.26 | upsite | and it works |
| 14:46.28 | _Sam-- | that should work |
| 14:46.29 | upsite | yeh |
| 14:46.44 | upsite | but when i chown this folder to * its not working |
| 14:47.22 | _Sam-- | rculp: if you grandstreams dont have the low mac address that gets the display bug, i dont think there is a single downside to upgrading. |
| 14:47.37 | _Sam-- | i dont know about speaker...just because i dont use it in this environment myself |
| 14:47.45 | rculp | Sam: that's our main gripe |
| 14:47.54 | rculp | just how it sounds in speakerphone |
| 14:48.03 | rculp | because the mic picks up the speaker |
| 14:48.12 | _Sam-- | there is some stuff for that i think |
| 14:48.17 | vinsik | heh |
| 14:48.20 | _Sam-- | did you check the change logs? |
| 14:48.28 | rculp | looking at them now |
| 14:48.31 | vinsik | tailf /var/log/asterisk/messages |
| 14:48.56 | vinsik | it usually says whats wrong |
| 14:49.00 | _Sam-- | like i said..e.ven it doesnt fix your speaker...there is no downside though. |
| 14:49.06 | _Sam-- | and everything else works like 10x better. |
| 14:49.24 | _Sam-- | i think this has to do with speaker/mic...AGC change |
| 14:49.24 | rculp | right |
| 14:49.49 | rculp | I really think it's just a hardware issue on the gxp |
| 14:49.55 | rculp | that no software will fix |
| 14:50.00 | rculp | as a handset |
| 14:50.04 | rculp | we've not had any problems |
| 14:50.18 | upsite | sam any ideas ? |
| 14:50.21 | rculp | we're just going to get a polycom conference phone |
| 14:50.26 | _Sam-- | i would disagree....the speaker and the mic are far enough apart for it work well |
| 14:50.36 | _Sam-- | and i think they could adjust the mic sensitivity if that was the problem |
| 14:50.46 | _Sam-- | and do it automatically...hence agc...automatic gain control |
| 14:51.14 | _Sam-- | but i do think you could probably buy better speaker phones! |
| 14:51.25 | rculp | and that's our plan :) |
| 14:51.31 | _Sam-- | i also think you should at least try the new version. |
| 14:51.31 | rculp | we just need one |
| 14:51.40 | rculp | I'm definately planning on upgrading |
| 14:51.44 | rculp | at an off hour |
| 14:51.59 | _Sam-- | it takes like 5 minutes ya pussy |
| 14:52.09 | upsite | :P |
| 14:52.23 | rculp | there's always a line in use during the day |
| 14:52.25 | rculp | so |
| 14:52.31 | _Sam-- | sorry i couldnt help it |
| 14:52.31 | rculp | instead of getting yelled at |
| 14:52.35 | rculp | by sales staff |
| 14:52.38 | rculp | I will wait |
| 14:52.39 | rculp | ya prick |
| 14:52.40 | *** part/#asterisk SparFux (n=player@tor/session/x-372385c7b4822796) |
| 14:52.44 | _Sam-- | lol good stuff! |
| 14:52.45 | rculp | :p |
| 14:53.01 | vinsik | :D |
| 14:53.20 | _Sam-- | you ahvent upgraded any firmware at all since you got those things? |
| 14:53.22 | rculp | only responding to you like that since I kinda know you ;) |
| 14:53.27 | vinsik | rculp: hehe.. i have been posponding upgrade because of that same reason for a week now :D |
| 14:53.27 | rculp | I have once |
| 14:53.30 | Ikarus | Hrm, BRIStuff doesn't compile (both 0.2.0 and 0.3.0), 0.3.0 exploding on q921.c |
| 14:53.34 | _Sam-- | so you have .13 on them? |
| 14:53.36 | rculp | vinsik: heh |
| 14:53.56 | vinsik | never remember in the evening |
| 14:54.07 | _Sam-- | crich_: with your cavtel pri how many DIDs you have? |
| 14:54.15 | _Sam-- | er rich = rculp, damn nick completion |
| 14:54.29 | rculp | 20 |
| 14:54.37 | _Sam-- | rculp: why doesnt your IP reverse? |
| 14:54.51 | _Sam-- | can you set the outgoing caller ID on your calls to any of the DIDs of your PRI? |
| 14:54.59 | pjz | anyone else have a problem with polycom500s taking a couple seconds to adapt when you make an outgoing call, such that you miss the first couple of seconds of audio? |
| 14:55.00 | _Sam-- | this was the underlying problem of MY cavtel pri |
| 14:55.05 | rculp | because I chose not to reverse mine |
| 14:55.13 | Ikarus | Anyone have a suggestion (other then don't use BRIStuff) |
| 14:55.13 | crich_ | hehe |
| 14:55.14 | rculp | and nope, if they supported a NI2 |
| 14:55.20 | rculp | connection it would work |
| 14:55.28 | rculp | but I'm forced to connect via ni1 |
| 14:55.36 | rculp | which does not support all caller id functions |
| 14:55.36 | _Sam-- | i see...so i wasnt crazy |
| 14:55.42 | _Sam-- | they tended to think i was |
| 14:55.50 | rculp | I tended to agree.... :) |
| 14:56.04 | _Sam-- | that was the whole reason i got 20 dids |
| 14:56.08 | _Sam-- | 1 for each sales guy |
| 14:56.22 | _Sam-- | but they couldnt set the outgoing caller id so people would call them back |
| 14:56.44 | rculp | well I am able to set the primary outgoing # |
| 14:56.46 | rculp | but not the name |
| 14:56.49 | rculp | so when we called |
| 14:56.55 | rculp | it would say unknown |
| 14:56.57 | rculp | but show the # |
| 14:57.05 | shmaltz | am I the only one? anybody else having problems with the list on gmail? |
| 14:57.07 | _Sam-- | hmmm...that sounds familiar.... |
| 14:57.15 | _Sam-- | our primary would show up as unknown |
| 14:57.17 | rculp | so I just passed the callerid back to them for now |
| 14:57.21 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 14:57.21 | _Sam-- | and we could only set it to the primary |
| 14:57.35 | _Sam-- | it was a mess to say the least |
| 14:57.46 | _Sam-- | but i feel a bit better knowing that i wasnt crazy |
| 14:58.03 | rculp | you may still be crazy, just not about callerid |
| 14:58.14 | rculp | all kidding aside |
| 14:58.22 | rculp | it was a point of frustration for me |
| 14:58.23 | _Sam-- | no, that is an honest assessment! |
| 14:58.34 | rculp | when I couldn't get callerid working |
| 14:58.36 | rculp | but |
| 14:58.44 | rculp | for some reason the wildcard |
| 14:58.45 | rculp | 110 |
| 14:58.49 | rculp | won't work in ni2 |
| 14:58.54 | rculp | on their switch |
| 14:59.57 | _Sam-- | with all the connectivity you guys have its a sin you use a PRI and not a remote gateway |
| 15:00.03 | _Sam-- | you could save a few hundred every month |
| 15:00.53 | rculp | the nice thing about using a pri is just in case there's a hardware problem with our router |
| 15:00.58 | _Sam-- | i run our biz off a remote gateway...and we are 10x more phone dependent and reliant |
| 15:01.06 | _Sam-- | i do 10mil a year in mail order /phone biz |
| 15:01.11 | rculp | we don't go down |
| 15:01.15 | _Sam-- | what? you are not confident and your hardware? |
| 15:01.19 | rculp | I am |
| 15:01.29 | AlexLee | Help!!!!!!!!!!!, I am in Beijing China, I use asterisk@home, The problem is now, that I have a website, and hope provide VOip service to the register users on this website after they register on my website, so I hope the User Id and password for users of the website is the same one with Asterisk server, to do this so, i have write or add extension No. from other application or pragram into asterisk, anyone can show me how to config or program? |
| 15:01.32 | rculp | there are pluses and minuses to both |
| 15:01.36 | rculp | and for now |
| 15:01.39 | rculp | we're exploring the pri |
| 15:01.44 | rculp | and we may eventually drop that |
| 15:01.49 | rculp | before the term is up |
| 15:01.55 | rculp | and go over bandwidth |
| 15:01.59 | _Sam-- | if you had BGP there... |
| 15:02.06 | _Sam-- | and got off that craptel circuit :) |
| 15:02.11 | _Sam-- | then you could use a reliable remote gatewayu |
| 15:02.19 | _Sam-- | sorry i tell john the same thing |
| 15:02.27 | rculp | :) |
| 15:02.29 | _Sam-- | i watched that circuit this weekend |
| 15:02.31 | _Sam-- | it was crap |
| 15:02.36 | rculp | but not going to debate our business model in here |
| 15:02.44 | rculp | or defend myself |
| 15:02.47 | rculp | to you in here |
| 15:02.51 | _Sam-- | i hear ya, and im not criticizing you! |
| 15:02.52 | rculp | no one else needs to see it |
| 15:03.02 | rculp | understood |
| 15:03.08 | _Sam-- | all i do is bust balls...im all talk! |
| 15:03.28 | _Sam-- | and i am by no means insinuating you dont know what you're doing. |
| 15:03.29 | rculp | you'll never cause me to lose my cool |
| 15:03.35 | _Sam-- | or that what you're doing is no good! |
| 15:03.39 | rculp | :) |
| 15:04.06 | _Sam-- | how is the dedicated server biz going? |
| 15:04.12 | _Sam-- | you have those racks full? |
| 15:05.05 | AlexLee | anyone can help? |
| 15:06.01 | _Sam-- | rich honestly i hope you dont take what i say personally...it is more said in ball busting than anything else with a small bit of truth usually :) |
| 15:06.52 | *** join/#asterisk heka (n=Horror@80.80.174.140) |
| 15:07.12 | heka | Hello, what version of asterisk is SVN-trunk-r7230 1.0 or 1.2 ? |
| 15:07.45 | AlexLee | Help!!!!!!!!!!!, I am in Beijing China, I use asterisk@home, The problem is now, that I have a website, and hope provide VOip service to the registered users on this website after they register on my website, so I hope the User Id and password for users of the website is the same one with Asterisk server, to do this so, i have write or add extension No. from other application or pragram into Asterisk database, anyone can show me how to config or prog |
| 15:07.45 | AlexLee | ram? |
| 15:07.46 | synthetiq | 1.2 |
| 15:07.59 | rculp | it's all good sam |
| 15:08.29 | _Sam-- | john said you guys had to redo your IVR? |
| 15:08.44 | _Sam-- | "it was too confusing to customers" or something ...but im not sure if he was talking about your incoming voice menu |
| 15:08.45 | heka | synthetiq: was that to me? |
| 15:09.38 | file | it was to the moon :D |
| 15:09.59 | mut | anyone know how to get callerid name to work passthru pri->sip on a cisco as5350 12.3 IOS |
| 15:10.10 | rculp | I think it's that customers don't listen |
| 15:10.19 | rculp | and they're used to calling |
| 15:10.23 | rculp | and getting a live person |
| 15:10.24 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:10.27 | rculp | to direct their calls |
| 15:10.31 | _Sam-- | yeah we're in that boat here |
| 15:10.33 | *** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au) |
| 15:10.50 | _Sam-- | morning dca |
| 15:11.35 | AlexLee | Help!!!!!!!!!!!, I am in Beijing China, I use asterisk@home, The problem is now, that I have a website, and hope provide VOip service to the registered users on this website after they register on my website, so I hope the User Id and password for users of the website is the same one with Asterisk server, to do this so, i have write or add extension No. from other application or pragram into Asterisk database, anyone can show me how to config or prog |
| 15:11.35 | AlexLee | ram? I am in Beijing china, it is night now here , I have to go home, if someone can help me please leave message to me or email to me, my email is alexlii@yahoo.com |
| 15:12.57 | thazza | AlexLee: Research.. Just like everyone else learns. |
| 15:13.34 | mut | school of hard knocks |
| 15:13.37 | mut | thats how i learn |
| 15:13.51 | AlexLee | thazza, any document to refer? |
| 15:13.59 | thazza | AlexLee: And secondly, i wouldn't be using asterisk@home if you want to do that kind of stuff. Yet if you insist on using A@H. |
| 15:14.07 | thazza | ~amp |
| 15:14.10 | jbot | rumour has it, amp is NOT supported here! people using it should join #amportal |
| 15:14.47 | file | jbot: botsnack |
| 15:14.47 | jbot | aw, gee, file |
| 15:15.01 | AlexLee | no, i can change Asterisk version, but which version should i use? |
| 15:15.05 | thazza | AlexLee: A great place to start.. Read these. |
| 15:15.07 | thazza | ~docs |
| 15:15.09 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 15:15.32 | thazza | AlexLee: Best to use the most up2date version.. I would recommend 1.2.4 |
| 15:15.33 | AlexLee | read what? |
| 15:15.46 | _Sam-- | i think alexlee just wants someone to do it for him |
| 15:16.02 | thazza | AlexLee: All those links.. They are the locations for the documents on learning asterisk.. |
| 15:16.06 | AlexLee | sam, i do not mean that.. |
| 15:16.30 | thazza | AlexLee: If you just want someone to make it for you.. Then i hope you are looking to pay someone. |
| 15:16.35 | AlexLee | i do not meant that but really need help on how to setup |
| 15:17.17 | thazza | The best thing about * (asterisk) is that it is free. And so configurable.. yet freedom always comes with a price.. You need to learn. |
| 15:17.19 | AlexLee | thazza, I not mean that , but tell me which version can satify my need....please |
| 15:17.49 | AlexLee | this is my first time to use Xchat and first time to use Asterisk.... |
| 15:17.51 | thazza | AlexLee: Asterisk 1.2.4 would be a start.. and i would suggest, poss looking at the AGI scripting. |
| 15:18.00 | AlexLee | this is my first time to use Xchat and first time to use Asterisk.... |
| 15:18.30 | file | Asterisk is a toolkit, you can do tons of stuff - but you have to make it do what you want |
| 15:18.38 | file | there's no "right" solution for you |
| 15:18.45 | thazza | AlexLee: If it is your first * time.. I suggest taking a couple of months holiday from your work.. Getting your head into asterisk. and then you will be right to make what you want. |
| 15:18.48 | file | so you have to learn what's available, what's not, make what you need, configure what exists, etc |
| 15:19.31 | thazza | AlexLee: Unless you are looking for a billing solution. and there are quite a few of these around. using asterisk. so you do have to compile, and so setup config files. |
| 15:20.08 | thazza | file: Can you continue and take it from here. Just finished rebuilding mates computer.. its 2:20am.. I am tired. and have work in morning. :-( |
| 15:20.38 | thazza | s/ywans/yawns |
| 15:20.40 | file | thazza: I think we've outlined/made the point |
| 15:20.44 | AlexLee | Best to use the most up2date version? |
| 15:21.21 | thazza | AlexLee: Best to use up2date stable versions.. Not good to use CVS on a live system. |
| 15:21.42 | thazza | AlexLee: Sorry have to go to bed now.. Read those Doc's. and probally have a look at: |
| 15:21.45 | AlexLee | thazza, thanks.... |
| 15:21.46 | thazza | ~thebook |
| 15:21.47 | jbot | from memory, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 15:21.48 | AlexLee | ok |
| 15:21.49 | AlexLee | see u |
| 15:21.56 | Damin | AlexLee: If you are using Asterisk@Home and thinking that you want to run a business off of it, I don't think you are ready to make an informed decision on what you should be doing yet. ;) |
| 15:22.31 | rculp | so, is there anyone with experience with the gxp2000 phones and speakerphone use on them? |
| 15:22.38 | *** join/#asterisk fugitivo (n=ajf@201.255.178.118) |
| 15:23.26 | AlexLee | this is my time for me to use asterisk.. i do not know which version i should use.. |
| 15:23.37 | AlexLee | this is my time for me to use asterisk.. i do not know which version i should use.. |
| 15:24.15 | AlexLee | <Damin> this is my time for me to use asterisk.. i do not know which version i should use..any suggesions? |
| 15:24.32 | synthetiq | any version less than 1.0 |
| 15:25.15 | AlexLee | Damin, this is my first time to use asterisk.. i do not know which version i should use..any suggestions? |
| 15:25.57 | thazza | AlexLee: Yes everyone heard this.... A small suggestion as i Go.. Never a good idea to repeat yourself over and over.. People love to help people who research and try to find the answers themselves.. yet people that just keep asking same questions, usally end up leaving unheard. |
| 15:26.05 | jaike | alexlee: 1.2.4. and just do it. you wont learn just by asking |
| 15:27.38 | thazza | AlexLee: Once again.. if you want to learn. get a spare box.. and READ.. |
| 15:27.40 | thazza | ~docs |
| 15:27.42 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 15:27.43 | thazza | ~thebook |
| 15:27.44 | jbot | thebook is, like, Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 15:27.56 | voipme | alexlee: get astlinux to start with |
| 15:27.57 | thazza | Nite all. |
| 15:27.58 | AlexLee | thazza, thanks... I just want a little more suggestions from differece experienced person... |
| 15:28.05 | voipme | it even comes in a vm ready to go |
| 15:28.09 | voipme | to play with |
| 15:28.24 | voipme | if it's a business, compile yourself on your distro of choice |
| 15:28.33 | *** join/#asterisk zzxxcc (n=zzxxcc@221.232.5.87) |
| 15:28.40 | mut | so no one has used caller-id name before? |
| 15:29.04 | mut | on a cisco AS router |
| 15:31.05 | *** join/#asterisk coppice (n=chatzill@251.204.17.210.dyn.pacific.net.hk) |
| 15:32.39 | *** join/#asterisk azzie (n=az@azzie.net) |
| 15:32.44 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-85.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 15:36.00 | mdave | I'm trying to find a flat monthly DID in US/MI/HOLLAND |
| 15:36.16 | mdave | calls delivered over SIP or IAX |
| 15:36.39 | mdave | poked around a lot, nothing solid |
| 15:37.05 | *** join/#asterisk vivekjj (i=1076@203.199.110.93) |
| 15:37.30 | vivekjj | anyone knows dtmf inband packets being dropped |
| 15:39.53 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfmtk.dialup.mindspring.com) |
| 15:41.52 | *** join/#asterisk lthnnpwr (n=alias@alias.lt) |
| 15:42.16 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 15:42.26 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 15:44.17 | *** part/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 15:44.49 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 15:44.58 | vattern | I am experiecing some difficulty in connecting to FWD via IAX .. |
| 15:45.10 | vattern | I keep getting Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ |
| 15:45.11 | vattern | <PROTECTED> |
| 15:45.11 | vattern | <PROTECTED> |
| 15:45.11 | vattern | <PROTECTED> |
| 15:45.16 | *** join/#asterisk DeeJay[2] (n=bleh@office.abi.ca) |
| 15:45.19 | DeeJay[2] | Hi! |
| 15:45.55 | DeeJay[2] | I have a TE210P and a TE410P in the same server which runs Debian with kernel 2.4.30 and Asterisk 1.2.4 and zaptel 1.2.3. |
| 15:46.01 | vattern | could someone please point me , I am using AMP to set up the trunk, and have followed the instructions, but somewhere I am getting lost .. |
| 15:46.13 | DeeJay[2] | But when a customer receives a call while already on the line, it freezes his Zap channel |
| 15:46.17 | DeeJay[2] | until I restart asterisk. |
| 15:46.29 | *** join/#asterisk Cadu20 (n=Cadu20@200.102.53.174) |
| 15:46.36 | fourcheeze | can Meetme config be done with realtime? |
| 15:46.42 | DeeJay[2] | and I get this error message when typing "show channel": Feb 7 10:20:09 WARNING[15034]: channel.c:787 channel_find_locked: Avoided deadlock for '0x82f53e8', 10 retries! |
| 15:46.54 | DeeJay[2] | It happens for all version of 1.2 |
| 15:47.04 | Cadu20 | Hi... dont know if asked previously... does anyone knows if tehre is a way to set CANREINVITE on the fly, based on the prefix dialed? |
| 15:47.16 | *** part/#asterisk rculp (n=rculp@66.173.240.20) |
| 15:47.27 | Cadu20 | I want to decide when to use or not REINVITE based on the dialed number... |
| 15:47.30 | DeeJay[2] | We hoped that from 1.2.0 to 1.2.4 it could be fixed but it seems to still happen.. |
| 15:49.18 | swb | hello all, I have a problem with macro variables |
| 15:50.17 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) |
| 15:50.37 | swb | I start a macro passing it two variables that are stoed in ${ARG1} and ${ARG2}, this works fine however, at some point in the macro, I use Gosub to execute a subroiutine, and it when it returns from that the macro variables have been wiped |
| 15:50.46 | swb | any ideas how I could save them somehow? |
| 15:51.36 | Luke-Jr | Is there a simple way to allow comments on CDRs? |
| 15:51.44 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 15:52.28 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 15:52.47 | queuetue | What kinds of toolkits exist for building SIP clients? I'd like to make a tiny daemon that will notify me of incoming CID data. |
| 15:53.11 | Luke-Jr | how about a module to work with XMPP for CID notification and/or command input? ;) |
| 15:54.19 | gaupe | queuetue: look at the sip-daemon in linphone |
| 15:55.43 | queuetue | gaupe: Have you ever built linphone on OSX (or any BSD?) It's not in ports, so I'm skeptical... |
| 15:56.21 | gaupe | nope, linux only - linphone has been build for freebsd - seen it on mailinglists |
| 15:57.06 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 16:01.57 | iCEBrkr | queuetue: you mean like this? |
| 16:02.00 | iCEBrkr | queuetue: http://www.cyberdyne.org/~icebrkr/cpg142/thumbnails.php?album=59 |
| 16:02.17 | iCEBrkr | queuetue: page down to the bottom.. http://www.cyberdyne.org/~icebrkr/?page_id=5 |
| 16:03.59 | mdave | asterisk and sipbroker rock |
| 16:04.07 | mdave | well voip in general, too |
| 16:04.09 | queuetue | iCEBrkr: Not too much like that, since it a) seems to be on windows and b) look slike it would be hard to wire into an existing notification system. |
| 16:04.49 | vivekjj | anyone knows dtmf inband packets being dropped |
| 16:05.03 | iCEBrkr | queuetue: Yea, it's for windows.. But it's as simple as it gets.. |
| 16:05.56 | *** part/#asterisk jaike (n=a@203.131.137.76) |
| 16:06.03 | iCEBrkr | queuetue: and you don't mess with SIP for notification type stuff.. You use the manage port. |
| 16:06.15 | queuetue | iCEBrkr: Is there source? |
| 16:06.20 | vattern | Anybody here have their * hooked up with FWD ? |
| 16:06.27 | iCEBrkr | queuetue: If you used SIP, you'd have to register the notifier AND your phone-- and you can't register the same client twice. |
| 16:06.37 | iCEBrkr | queuetue: Source isn't available. |
| 16:06.45 | iCEBrkr | vattern: I'm sure a lot of people do. |
| 16:06.53 | vattern | I am still getting CAUSE : Registration Refused |
| 16:06.53 | vattern | <PROTECTED> |
| 16:06.57 | queuetue | iCEBrkr: Ah, I thought the same client could be registered multiple times. |
| 16:07.01 | vattern | if i turn on iax2 debug |
| 16:07.35 | *** join/#asterisk arbius (n=arbius@c-67-173-45-34.hsd1.il.comcast.net) |
| 16:10.15 | iCEBrkr | queuetue: Not with SIP |
| 16:10.30 | iCEBrkr | queuetue: There's a notifier writen in Java for Linux. |
| 16:10.35 | *** join/#asterisk GerbilNut (i=GerbilNu@65.88.144.41) |
| 16:10.43 | iCEBrkr | I forget what it's called, but the author hangs out here once in awhile |
| 16:11.43 | iCEBrkr | Oh here, I think it's ADM |
| 16:11.49 | iCEBrkr | queuetue: http://www.voip-info.org/wiki/view/ADM+-+Asterisk+Desktop+Manager |
| 16:12.47 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 16:13.38 | Aughey | [TK]D-Fender: I got the Sangamon cards yesterday |
| 16:15.02 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 16:15.11 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.80.122.Dial1.Chicago1.Level3.net) |
| 16:15.52 | *** join/#asterisk djMax (n=chatzill@artsalliancelabs.com) |
| 16:16.28 | djMax | somebody mentioned that voicemail filenames were date-based, but I don't see that behavior. anybody know which is right? |
| 16:20.01 | *** join/#asterisk saftsack (n=saftsack@p54A7CA87.dip.t-dialin.net) |
| 16:20.03 | saftsack | hi |
| 16:20.20 | saftsack | if i send a fax with hylafax over isdn both channels are used. |
| 16:20.24 | saftsack | why? |
| 16:20.34 | iDunno | make it quicker. |
| 16:20.43 | saftsack | ? |
| 16:20.50 | [TK]D-Fender | Aughey : Which on? |
| 16:21.00 | saftsack | hylafax -> modem -> fxs-card -> misdn -> line |
| 16:21.31 | djMax | is the latest zaptel meant to actually compile? |
| 16:22.30 | mzo | it's a trick release, it's supposed to mess with your head :) |
| 16:22.31 | djMax | I'm getting ZAPTEL_VERSION undeclared |
| 16:22.52 | *** join/#asterisk RoyKa (n=roy@80.239.107.70) |
| 16:23.14 | djMax | and indeed grep doesn't find ZAPTEL_VERSION anywhere |
| 16:24.14 | GerbilNut | Anyone in here have experience with a Snom 360 and setting it up to show if another Extension is on a call? |
| 16:26.47 | *** join/#asterisk sch19 (n=sch19@adsl-2-114-222.mia.bellsouth.net) |
| 16:27.13 | sch19 | good morning |
| 16:27.22 | Aughey | [TK]D-Fender: I got the 6 FXO port card. 3 modules and a blank. I haven't installed it yet |
| 16:27.28 | djMax | apparently I needed to call "make version.h"... |
| 16:27.59 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 16:28.02 | *** part/#asterisk RoyKa (n=roy@80.239.107.70) |
| 16:28.46 | Aughey | Does anyone have a good dialplan for allowing US domestic calling and blocking toll numbers? |
| 16:28.57 | Luke-Jr | hm... double Monitor seems to break |
| 16:29.08 | sch19 | is there not a good sample on the wiki somewhere? |
| 16:29.10 | Umaro | hey guys.. anyone know where I can get a X100P (or some other kind of digium card) near pune, india? |
| 16:29.25 | mike240se | Umaro: ebay |
| 16:29.36 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
| 16:30.04 | Umaro | mike240se: ebay.in doesn't have any listed |
| 16:30.12 | Aughey | I did some searching, but haven't found anything yet |
| 16:30.21 | mike240se | there are a couple of international sellers on ebay.com |
| 16:30.23 | sch19 | they almost always have a couple buy-it-now's on ebay for the x100p i thought |
| 16:31.02 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 16:31.02 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:31.10 | mike240se | Umaro: or you can do what i did one day, go to every computer store in the area opening up boxes that say "intel hardware modem" on them and seeing if they are the right chip set |
| 16:31.20 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:31.36 | Umaro | mike240se: lol |
| 16:31.47 | [TK]D-Fender | Aughey : That's really easy stuff and yuo can figure it out fast just by looking at the sample extensions.conf file |
| 16:31.56 | Luke-Jr | mike240se: Intel rebrands Digium stuff? |
| 16:32.21 | mike240se | Luke-Jr: for the x100p, ye |
| 16:32.23 | mike240se | yes |
| 16:32.24 | Beirdo | ummm |
| 16:32.25 | Beirdo | no |
| 16:32.26 | mogorman | no |
| 16:32.34 | mogorman | x100p is a winmodem |
| 16:32.37 | Beirdo | Digium used an Intel chipset IIRC |
| 16:33.24 | mogorman | that digium wrote drivers for |
| 16:33.24 | mogorman | a long time ago |
| 16:33.24 | Beirdo | right :) |
| 16:33.24 | mike240se | i should say it will work the same |
| 16:33.25 | mike240se | if you get the right one |
| 16:33.27 | Aughey | I can easily filter out 900 numbers and others, but the AsteriskTFOT talks about 809 being a bad area code too. I just don't know what other gotchas are there |
| 16:33.30 | mike240se | actually meant to say digium uses intel stuff |
| 16:33.34 | mike240se | i thought thats what you asked |
| 16:34.43 | mike240se | although that would be nice for digium if intel bought and rebranded their stuff |
| 16:34.46 | [TK]D-Fender | Aughey : Yeah I guess if there are a bunch of "unknowns" that'd make it harder.... not sure where to go for that. |
| 16:35.19 | sch19 | maybe the phone book |
| 16:35.57 | sch19 | assuming yu have one, i guess |
| 16:36.19 | sch19 | *you |
| 16:36.33 | *** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 16:36.57 | sch19 | does anyone here use ael ? |
| 16:37.08 | Luke-Jr | can Asterisk modify a CDR after it is logged (the call has ended) |
| 16:37.10 | Luke-Jr | ? |
| 16:37.50 | sch19 | can it? or will it w/o coersion ? :P |
| 16:38.55 | sch19 | I don't know either way, just seems like the question should be expanded to under any circumstances.. |
| 16:40.44 | jaiger | Luke-Jr, what are you trying to accomplish? |
| 16:40.58 | hardwire | a sexy singing career |
| 16:41.14 | jaiger | I have my CDR logged to Postgres and from there can modify at will |
| 16:41.16 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 16:41.30 | Luke-Jr | jaiger: Asterisk IMs CID info to me when there's a call, and I want to reply with a comment to log in CDR |
| 16:41.35 | Luke-Jr | jaiger: but I want the ability to comment on the call *after* it ends |
| 16:41.53 | mike240se | woah that seems like a lot of work |
| 16:41.59 | mike240se | an IM for every call? |
| 16:42.01 | hardwire | Luke-Jr: just respond after you hang up? |
| 16:42.16 | Luke-Jr | hardwire: ... the asterisk end needs the ability to change the CDR |
| 16:42.23 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool138-85.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 16:42.23 | hardwire | hardly. |
| 16:42.26 | Luke-Jr | mike240se: every call to me, yes |
| 16:42.36 | mike240se | Luke-Jr: oh |
| 16:42.41 | Luke-Jr | the comment goes in the CDR... |
| 16:42.46 | mike240se | Luke-Jr: you could write a script i suppose |
| 16:42.48 | hardwire | in an sql database |
| 16:42.57 | hardwire | which could be done from any authorized script |
| 16:43.03 | hardwire | reading the log to your IM server |
| 16:43.25 | hardwire | nope |
| 16:43.33 | hardwire | that would require editing a line 50 calls back |
| 16:43.37 | hardwire | on a live log |
| 16:43.39 | mike240se | Luke-Jr: it would be alot more complex to write one for csv |
| 16:43.48 | Luke-Jr | hardwire: s/would/could |
| 16:44.01 | hardwire | if you like your fs barfing.. go right ahead |
| 16:44.07 | Beirdo | hehe |
| 16:44.09 | mike240se | ha |
| 16:44.10 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
| 16:44.20 | Beirdo | as long as it does so in the toilet... |
| 16:44.21 | jaiger | Luke-Jr, I recommend you use sql and I'm sure a script could be made to do what you need |
| 16:44.29 | hardwire | or if you like queueing your CDR until each write.. you would easily get your IM for your most recent calls a few minutes lagged |
| 16:44.30 | Luke-Jr | in theory, why couldn't the original be blanked and the new one appended? |
| 16:44.58 | mike240se | Luke-Jr: you will be opening the file, which will make asterisk unhappy i assume |
| 16:45.04 | jaiger | I'm not familiar with the IM integration to make recommendations |
| 16:45.09 | hardwire | Luke-Jr: do you have mad skillz with this sort of thing? |
| 16:45.11 | Luke-Jr | mike240se: I want Asterisk to handle the interface itself |
| 16:45.26 | Luke-Jr | hardwire: which part of it? |
| 16:45.33 | jaiger | Luke-Jr, sql would be much easier to implement I'm sure |
| 16:45.33 | mike240se | Luke-Jr: post a request for a patch i guess |
| 16:45.35 | hardwire | the editing a log file in the middle part |
| 16:45.42 | Luke-Jr | jaiger: probably, but I like CSVs... |
| 16:45.43 | *** part/#asterisk Aughey (n=jha@ns1.washucsc.org) |
| 16:45.50 | hardwire | I recommend just use sqlite and dump to csv via a select if you want it |
| 16:45.56 | jaiger | Luke-Jr, you can always dump your db to csv if you wish |
| 16:45.57 | Luke-Jr | hardwire: CDR CSV isn't a log per se |
| 16:45.59 | hardwire | or any sql |
| 16:46.04 | Luke-Jr | hardwire: the times are in the fields, so they don't need to be in order |
| 16:46.05 | mike240se | Luke-Jr: its quite easy to write a script with sql |
| 16:46.09 | iCEBrkr | Why the hell would you want to append a comment (instant message) to the CDR? |
| 16:46.16 | hardwire | Luke-Jr: no but every log gets opened - appended - closed |
| 16:46.16 | hardwire | isn' |
| 16:46.24 | hardwire | t that what the csv call log does? |
| 16:46.39 | hardwire | now you have to queue your writes. |
| 16:46.41 | jaiger | Luke-Jr, did you get those Linksys ATAs way back when? |
| 16:46.44 | Luke-Jr | iCEBrkr: better call log details |
| 16:46.50 | hardwire | modify a line.. extend the contents one chunk at a time to the end |
| 16:46.51 | Luke-Jr | jaiger: got one, yes |
| 16:47.09 | iCEBrkr | Luke-Jr: Huh, how much more detail do you need? Phone number, time date, start, stop, |
| 16:47.12 | iCEBrkr | duration |
| 16:47.13 | iCEBrkr | blah |
| 16:47.15 | Luke-Jr | hardwire: replace the old line w/ spaces and append the modified one to the end |
| 16:47.16 | hardwire | Luke-Jr: a real solution is having csv-plain and csv-comment |
| 16:47.22 | Luke-Jr | iCEBrkr: content comment |
| 16:47.29 | hardwire | Luke-Jr: that is so qbasic |
| 16:47.31 | iCEBrkr | Sounds like a waste of time to me. |
| 16:47.32 | Luke-Jr | iCEBrkr: eg, "job opening" or such |
| 16:47.34 | jaiger | I think the comment on a call isn't a bad idea, just shoe-horning CSV is |
| 16:47.52 | hardwire | Luke-Jr: I would make a module for what you want. |
| 16:47.59 | Luke-Jr | jaiger: you have a better solution other than SQL? =p |
| 16:48.03 | hardwire | that finds the record in the regular csv |
| 16:48.09 | iCEBrkr | Really, it sounds like he needs notepad to keep track of these calls. |
| 16:48.15 | Damin | iCEBrkr: Care to do some cdr hacking on cdr_odbc? :) |
| 16:48.15 | hardwire | then makes another log with the comments for each line.. commented or not. |
| 16:48.15 | jaiger | Luke-Jr, no sql gives you what you need. why do you resist? |
| 16:48.17 | Damin | iCEBrkr: Looking to try and get it to support multiple datasources.. |
| 16:48.32 | iCEBrkr | Damin: *Groan* |
| 16:48.36 | hardwire | jaiger: I would if I didn't want to run an sql server on a 266mhz geode |
| 16:48.37 | Damin | iCEBrkr: Want to try and have it write to MySQL and MS-SQL at the same time. ;) |
| 16:48.45 | *** join/#asterisk Jizzbug (n=derekm@199.227.154.26) |
| 16:48.47 | Jizzbug | j #asterisk-dev |
| 16:48.53 | jaiger | Luke-Jr, or as recommended, make another CSV file with your call-id + comment in it |
| 16:48.57 | Jizzbug | er, sry |
| 16:48.58 | hardwire | Jizzbug: fail. |
| 16:48.59 | Luke-Jr | jaiger: portability |
| 16:49.10 | jaiger | Luke-Jr, what about portability? |
| 16:49.22 | Damin | iCEBrkr: My other option is to simple load cdr_odbc as cdr_odbc2 w/ a different config file.. :) |
| 16:49.25 | Luke-Jr | jaiger: I don't like restricting people to what CDR module they can use |
| 16:50.46 | Luke-Jr | so my idea would be to have CSV CDRs work in a maybe-ugly way, but at least work; and if someone complains about the ugliness, tell them to use SQL then |
| 16:50.48 | jaiger | Luke-Jr, that's not a CDR feature IMO it's an extension and the comment data doesn't really belong in the CDR table. even if developing w/ sql I'd recommend another table |
| 16:50.54 | iCEBrkr | Damin: Off the top of my head, I was thinking it should just read in all the destination info and start making connections |
| 16:51.16 | Luke-Jr | jaiger: comment is just as on-topic as CIDName for CDR |
| 16:51.27 | jaiger | Luke-Jr, or simply append another cdr line to the CSV. one that's a duplicate of the original plus your comment |
| 16:51.36 | mut | i've got a problem here... i have an asterisk server with a te405p in it, i'm trying to make a call out the te405p pri to an adtran 550 pri to a cisco as5350 back to the same asterisk server via sip, i'm calling from one extension to another, the problem is this, i call from a verizon land line and it comes in on the second pri in the cisco then to the asterisk box and my phone rings |
| 16:51.52 | Damin | iCEBrkr: It actually looks like it would be pretty trivial to just hack the source to have cdr_odbc2 and have it loaded as a separate module.. |
| 16:51.53 | Luke-Jr | jaiger: that's what I was saying... except that I'd also be blanking the original line |
| 16:51.55 | mut | now when i dial from one exten through the big long route, * -> adtran -> cisco -> * |
| 16:51.59 | mut | i get a fast busy |
| 16:52.06 | mut | the call makes it all the way to the cisco |
| 16:52.22 | mut | then the cisco tells me |
| 16:52.22 | mut | ICause i = 0x8095 - Call rejected |
| 16:52.33 | hardwire | 0wn3d |
| 16:52.37 | mut | i have asterisk verbose 5 on |
| 16:52.38 | Luke-Jr | jaiger: In my case, it will be very rare for a second call to come in prior to the first one's being commented in every case |
| 16:52.41 | mut | and i don't see the call hit it at all |
| 16:52.49 | jaiger | I wouldn't bother blanking the line. let your post-processing scripts clean up duplicates |
| 16:52.52 | Luke-Jr | jaiger: In a case with more frequent calls, I imagine someone would be using a SQL CDR |
| 16:53.39 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 16:53.43 | Damin | iCEBrkr: There is a single static definition for the config file: static char *config = "cdr_odbc.conf"; |
| 16:54.08 | Damin | iCEBrkr: Probably some simple hacks to get it to register as a different module name.. |
| 16:54.13 | iCEBrkr | Damin: Yea, it should read the different [] |
| 16:54.13 | bweschke | Damin: how would you handle multiple sources? each source gets a cdr write? |
| 16:54.27 | iCEBrkr | Damin: it's a bit more complex than that.. I just looked at the code |
| 16:54.37 | iCEBrkr | Damin: Lots of loops and shit need to be wrapped around all the functions |
| 16:54.40 | Damin | bweschke: Yeah.. |
| 16:54.45 | mut | ideas input anything? |
| 16:54.57 | hardwire | no ideas here |
| 16:55.06 | Damin | iCEBrkr: not if you copy cdr_odbc.c to cdr_odbc2.c and then modify things to suite; :) |
| 16:55.34 | hardwire | http://mobotix.com/mx_english/mx_produkte.htm |
| 16:55.41 | hardwire | voip/video over ip network cameras |
| 16:55.51 | hardwire | speaker mic and two cameras per unit |
| 16:56.06 | hardwire | its just sexy |
| 16:56.52 | Luke-Jr | jaiger: and obviously if there is no CDRs after the one being commented on, the comment can just be appended easily |
| 16:57.44 | mut | anyone know why an adtran strips callerid? |
| 17:01.16 | *** join/#asterisk stevie_d1111 (n=stephen@213.166.6.123) |
| 17:02.53 | jaiger | mut, what adtran are you talking about? I have an adtran TA750 and get callerid through FXO no problem |
| 17:03.11 | mut | 550 |
| 17:03.39 | lo_tech | mut: 'pri debug span XX' and check for IE... will tell you about the presentation and the callerid... as well as help debug the call from the cisco (though it looks like the call is not getting through the cisco to the *) |
| 17:03.54 | mut | lo_tech: yea |
| 17:04.00 | mut | debug isdn q931 |
| 17:04.15 | mut | the call is getting to the cisco |
| 17:04.26 | mut | and then says rejected |
| 17:04.29 | mut | maybe.. |
| 17:04.29 | mut | sec |
| 17:05.10 | lo_tech | mut: yeah, but it looks like the cisco is rejecting the call... so you wouldn't see an inbound on the 2nd *.. pri debug will at least let you see what you are getting and you can walk it back form there. |
| 17:05.22 | lo_tech | s/form/from |
| 17:06.56 | *** join/#asterisk Tired_ (n=tired@S010600095b4654ab.gv.shawcable.net) |
| 17:07.19 | *** part/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 17:07.26 | Tired_ | Hello. I have an idea, and I'm wondering if Asterisk would be an appropriate tool to use. |
| 17:07.52 | hardwire | http://www.engadget.com/2006/02/07/a4-techs-talky-voip-keyboard/ |
| 17:07.53 | hardwire | hehe |
| 17:08.03 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 17:08.11 | lo_tech | mut: also, check the outbound call on the pri to the adtran... you might not be sending the CID you expect, the presentation might not be allowing, or your not using the same pridialplan |
| 17:08.12 | Tired_ | I want to set up a voice mail system, where people can call and leave a voice message to a box that they type in on the phone that they are calling from, so homeless people can have a number to get jobs with. |
| 17:08.37 | Tired_ | Can Asterisk do that kinda thing? |
| 17:08.40 | lo_tech | Tired_: surely |
| 17:08.53 | Tired_ | sweet. i'll start a-googling then :) |
| 17:09.15 | vivekjj | does anyone know asterisk dropping dtmf inband packets |
| 17:09.16 | *** join/#asterisk Nemesis760 (n=nemesis@71.36.28.33) |
| 17:09.20 | *** join/#asterisk lahaine (n=qzxcd@2.67.119-80.rev.gaoland.net) |
| 17:09.23 | lahaine | oy ppl |
| 17:09.29 | hardwire | Tired_: I wanted to do something similar |
| 17:09.31 | Tired_ | is there a wiki for asterisk anywheres? |
| 17:09.49 | lo_tech | http://www.voip-info.org? |
| 17:09.53 | lahaine | voip-info.org ? |
| 17:09.53 | hardwire | Tired_: but they keep getting booted away from pay phones |
| 17:09.56 | Tired_ | hardwire -> I'd be glad to keep you informed on the project if you like. |
| 17:09.57 | justinu | ~docs |
| 17:10.01 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:10.09 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 17:10.12 | Flauto | hi people |
| 17:10.15 | Flauto | good morning |
| 17:10.16 | lo_tech | Tired_: booted? |
| 17:10.17 | hardwire | Tired_: I wanted to do one for road rage.. |
| 17:10.20 | Flauto | ihave a question |
| 17:10.30 | hardwire | and that idea turned into a myspace kinda idea for leaving messages to a state plate id. |
| 17:10.57 | hardwire | "please say plate state" "please say plate number" |
| 17:11.00 | justinu | hardwire: thought about that kinda thing many times |
| 17:11.02 | hardwire | then they could register to a plate |
| 17:11.05 | hardwire | online |
| 17:11.05 | justinu | some kind of driver complaint system |
| 17:11.05 | Tired_ | this is just so they can have that important phone number for their resumes, somewhere that answers and takes messages. |
| 17:11.10 | hardwire | and retrieve mp3's of their recordings |
| 17:11.14 | *** join/#asterisk apardo (n=apardo@87.218.44.2) |
| 17:11.17 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 17:11.28 | hardwire | and then of course "is this a positive.. or negative comment" |
| 17:11.33 | hardwire | you could start rating people by state |
| 17:11.37 | Flauto | i wanted to use sip to dial a romote spa adapter, and i opened the port 5060 on the remote side for the spa, and the call can go through but no voice transmission. is there anyone can tell me why? |
| 17:11.46 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 17:11.46 | hardwire | they could publish some of their messages |
| 17:11.48 | justinu | hardwire: you can't make it anonymous tho... |
| 17:11.49 | *** join/#asterisk rva (n=Miranda@200.206.141.250) |
| 17:11.56 | hardwire | justinu: sure you can |
| 17:12.06 | hardwire | anybody can call in on a plate |
| 17:12.11 | Tired_ | hmm. i wonder if the voice input from a vopice mail could be run through speech recognition and transcribed to email..... |
| 17:12.13 | hypnox | Tired_ good idea, asterisk does that pretty much out of the box, you just have to set up the numbers and corresponding voicemail boxes |
| 17:12.16 | justinu | then you're going to have one asshole ruin the whole thing |
| 17:12.20 | hardwire | but you have to go to the site to register your plate to see if anybody called in |
| 17:12.30 | lo_tech | Tired_: yes, but not without additional software |
| 17:12.33 | hardwire | justinu: thats exactly the point |
| 17:12.37 | hardwire | anybody can call in |
| 17:13.00 | Nugget | http://lnk.nu/cafepress.com/84g <-- cute |
| 17:13.02 | justinu | you just became the worst driver in internet history, simply because you failed to signal a lane change, and it pissed me off. |
| 17:13.02 | hardwire | its not supposed to be usefull :) |
| 17:13.04 | Tired_ | lol, i have a grant...additional software is fine as long as it's f/oss and has docs somewhere. :) |
| 17:13.21 | lahaine | is there somebody knowing the meaning of a Via header field with a branch=0 field ? |
| 17:13.21 | hardwire | its supposed to be fun and a good way to fluff off anger anonymously |
| 17:13.42 | lahaine | (i know it's not related to asterisk but if a SIP guru can demistify this to me ;) ) |
| 17:13.44 | hardwire | justinu: yes. wouldn't that be fun :) |
| 17:13.45 | lo_tech | Tired_: speech-to-text is not exactly 100%... accents, dialects, etc. make for some interesting conversions to text |
| 17:13.50 | Tired_ | that is a good idea...a line to call to vent |
| 17:13.58 | Nemesis760 | hardwire: Sounds pretty cool. |
| 17:14.05 | hardwire | Tired_: or to leave your number with the sexy woman driving next to you |
| 17:14.18 | hardwire | "nice boobs.. lets get some pizza.. call me" |
| 17:14.33 | lo_tech | hardwire: lol, seek help |
| 17:14.38 | hardwire | lo_tech: I am |
| 17:14.40 | hardwire | its not working |
| 17:14.57 | lo_tech | yet |
| 17:15.02 | Tired_ | lo_tech -> I was thinking the primary means for clients to access voicemails would be to read them in emails sent to webmail, then listen on the phone if the speech to text is too garbled...we'll only have a couple lines, for maybe 1000 clients. |
| 17:15.18 | jaiger | lo_tech, he's been leaving his number but no one returns his call |
| 17:15.24 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 17:15.28 | Tired_ | so hopefully most will use the email option to help keep the lines avaialable |
| 17:15.40 | vivekjj | wisdom: do u know dtmf inband packets being dropped from asterisk |
| 17:15.54 | _Sam-- | i could be wrong but i think you will have a really hard time doing accurate speech to text |
| 17:16.03 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:16.12 | Nemesis760 | Tired: Why not just forward an .mp3 of the message? |
| 17:16.23 | Flauto | any can help |
| 17:16.24 | Flauto | please |
| 17:16.29 | Tired_ | i hope i can make it work. i am very much a newbie at this, but the project is so exciting.... |
| 17:16.38 | lo_tech | Tired_: reading emails (text-to-speech) is a trivial thing, but the other way is problematic |
| 17:16.40 | _Sam-- | true...if they are checking email anyway...why not just attach the message |
| 17:16.57 | Tired_ | Nemesis760 -> this is for homeless clients...nowhere to forward it to, and they can't listen to it on Hotmail at the library. |
| 17:17.12 | *** join/#asterisk loick (n=loick@APuteaux-151-1-88-77.w86-205.abo.wanadoo.fr) |
| 17:17.12 | _Sam-- | pretty much anything that could mail with POP3 could play a wav/mp3 file attachment |
| 17:17.31 | _Sam-- | er check MAIL with pop3 |
| 17:17.33 | Tired_ | no speakers/sound cards there |
| 17:17.41 | _Sam-- | where? |
| 17:17.47 | Tired_ | public library |
| 17:17.59 | _Sam-- | it wont work but you cant waste some time |
| 17:18.07 | jaiger | _Sam--, no privacy there either |
| 17:18.11 | _Sam-- | things that do speech recognition based on YOUR OWN voice barely work |
| 17:18.13 | Nemesis760 | Tired: gotchya... was just being Dr. Obvious. ;) |
| 17:18.21 | rva | hi guys...do iax2 show peers really work? i have 2 clients...that are registered the same way in the server and one apeers as UNREACHABLE and the other OK |
| 17:18.24 | _Sam-- | let alone trying to interpret 1000s of unique voices |
| 17:18.28 | adibar | apropos Speech2text... could some integrate Sphinx to asterisk ? |
| 17:18.32 | Tired_ | lol |
| 17:18.51 | hardwire | you wanna hear drunk? |
| 17:19.02 | hardwire | hard to do tect-to-speach on messages? |
| 17:19.06 | jaiger | adibar, I was thinking of doing it after lunch today |
| 17:19.09 | hardwire | oh shit I cannot spell |
| 17:19.26 | Tired_ | _Sam-- -> I think you convinced me. I'll try to get it working with a phone-only interface to start, and then try the speech thing when the basic system works. |
| 17:19.48 | adibar | jaiger: Good luck... If you suceed, let me know how to do it... I couldn't manage it... |
| 17:19.52 | _Sam-- | i am not saying its not a good idea...i had someone ask me about it just the other day |
| 17:19.58 | lo_tech | drunk? I'm still trying to translate Pacific-Rim English software docs... no time for the drunk-to-english translation... babelfish have that one yet? |
| 17:19.59 | _Sam-- | but i dont think you could reliably implement it |
| 17:20.01 | _Sam-- | i could be wrong. |
| 17:20.02 | jaiger | adibar, joking. that's a big project |
| 17:20.03 | Tired_ | is asterisk easily configurable to serve people messages from their own voicemail box over the telephone? |
| 17:20.09 | hardwire | Tired_: gimme updates |
| 17:20.20 | Tired_ | ok. pm me your email |
| 17:20.21 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-85.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 17:20.33 | rob0 | Tired_: voicemailmain() |
| 17:20.58 | adibar | jaiger: For me it seemed more a life-time decision than a project ;-) |
| 17:21.17 | Tired_ | awesome. sounds like this is exactly what I need for this. |
| 17:21.33 | _Sam-- | you could also setup the web based voicemail checker |
| 17:21.40 | _Sam-- | vmail.cgi |
| 17:21.41 | *** part/#asterisk vivekjj (i=1076@203.199.110.93) |
| 17:21.45 | Nemesis760 | adibar: http://www.voip-info.org/wiki-Sphinx |
| 17:21.53 | _Sam-- | but the message will be played from wav/GSM/mp3 etc |
| 17:21.54 | lo_tech | Tired_: very easy to config... just the speech-to-email part is a chore |
| 17:21.57 | Nemesis760 | References an example I put together. |
| 17:22.10 | shmaltz | anybody else having any problems with the list? i'm using gmail |
| 17:22.10 | adibar | jaiger: Everything is ready and installed @ my place... but I have no idea how to integrate ;-( |
| 17:22.31 | jaiger | Tired_, a DTMF-only version of your system should be easy to build. voicemail comes free with asterisk |
| 17:22.34 | Nemesis760 | adibar: It's only a jumping off point. Would require a lot of work to get a large enough dictionary for transactiption. |
| 17:22.53 | Tired_ | excellent. well, thanks for all the advice...i'm sure I'll be back when i get confused, and I'll keep you posted on how it deploys... |
| 17:23.21 | jaiger | adibar, I haven't used sphinx yet. I downloaded it but didn't have time to devote to learning curve |
| 17:23.47 | adibar | jaiger: I know that page as hell, but it was not really helpfull... Poor adibar was more confused than before ;-) |
| 17:23.50 | Tired_ | if it works well here, maybe the civic minded among you could recommend it to your own local community councils. |
| 17:23.58 | Tired_ | :) |
| 17:24.08 | adibar | jaiger: That's the point. |
| 17:24.08 | *** join/#asterisk docelm0 (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 17:24.16 | Nemesis760 | transaciption.... er. transcription that is. |
| 17:24.36 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 17:24.46 | jaiger | adibar, maybe when my kids are in college. I'll see you here in 20 years |
| 17:24.54 | Tired_ | i'm going to leave the transcription feature for later... |
| 17:24.57 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 17:25.07 | mut | http://pastebin.ca/40577 |
| 17:25.11 | mut | my cisco debug of the call |
| 17:25.20 | mut | it hits the correct dial peer |
| 17:25.32 | *** part/#asterisk Tired_ (n=tired@S010600095b4654ab.gv.shawcable.net) |
| 17:25.34 | mut | and gets all the way down to sending the call out via sip, which is where the debug starts |
| 17:26.01 | adibar | jaiger: till than you can get speech2tetx as tiny lil executable with selfinstalling capability ;-) |
| 17:26.25 | Falle | Have anyone got some decent ringtones(ringX.bin) for the GXP 2000 that they can share? |
| 17:27.39 | mut | http://pastebin.ca/40578 |
| 17:27.44 | mut | thats the debug of me calling |
| 17:27.49 | mut | and then ther eis a hangup |
| 17:28.07 | adibar | Falle: I've got the song Paranoia from Art Of Noise ;-) |
| 17:28.43 | Falle | adibar: hmm, dont know what that is but it sounds dangourus ;) |
| 17:28.56 | jbalcomb | [TK]D-Fender iCEBrkr got my two Cisco 7940Gs |
| 17:29.06 | *** part/#asterisk techie (i=gus@antibala.com) |
| 17:29.08 | Nemesis760 | Does anyone here now how to gracefully busy out B-Channels of a PRI? I have multiple servers, with many T1s. Calls hunt ascending and I need to be able to coax calls onto the next server if the first is overloaded. |
| 17:29.10 | adibar | Falle: One mom, I prepare it for U |
| 17:29.28 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:29.43 | Falle | adibar: ok, thanks :) |
| 17:29.58 | [TK]D-Fender | jbalcomb : Great. Had a chance to give them all a serious test-drive? |
| 17:30.55 | jbalcomb | [TK]D-Fender have to convert them from SCCP to SIP first. |
| 17:31.07 | Qwell | meh,sip |
| 17:31.10 | Qwell | bbl |
| 17:31.40 | jbalcomb | anyone have 'P0S30100.bin' that I could get without having to register with cisco? |
| 17:31.51 | jbalcomb | its the SIP image for the Cisco 7940G |
| 17:31.55 | Nemesis760 | $50 via PayPal for the first complete answer. |
| 17:32.06 | adibar | Falle: U can get it from: http://linux.ch/gxp/ I don't remember which one it is, so I placed all three ;-) |
| 17:32.23 | Falle | adibar: great :) |
| 17:33.00 | adibar | Falle: from the size I would think it's #2 |
| 17:33.01 | docelm0 | jbalcomb, I do hold on.. Lemme go get it |
| 17:33.13 | docelm0 | or whats your email? I will email it to you |
| 17:33.21 | *** part/#asterisk arbius (n=arbius@c-67-173-45-34.hsd1.il.comcast.net) |
| 17:34.05 | synthetiq | when doing an IF fucntion evaluation, for it to work you would have to do it in Set or NoOP? |
| 17:34.08 | Falle | adibar: downloaded them now, rebooting phone |
| 17:34.24 | jbalcomb | docelm0 thanks. jbalcomb@imtco.com |
| 17:34.35 | jbalcomb | Falle you dont have to reboot the phone for that |
| 17:34.48 | docelm0 | What version IOS is it? |
| 17:35.19 | adibar | Falle: How's the weather in Sweden ? ;-) |
| 17:35.31 | lo_tech | Nemesis760: well, there is 'zap destroy channel' but DONT USE IT unless you know what it does... there is no 'busyout/release channel xx' equivalent afaik |
| 17:35.52 | Falle | adibar: frezing :/ |
| 17:37.01 | Nemesis760 | lo_tech... yeah... zap destroy != graceful. Any chance there's a way to do it outside of asterisk? I'm using Sangoma A104s. |
| 17:37.04 | lo_tech | Nemesis760: we use macro dial to test DIALRESULT... if we get a certain result, it forces a second (or tertiary) dial to a different trunk (that may be a local zap or a iax2 to another server, depending) |
| 17:37.18 | tainted_ | is there an error log stored someplace for asterisk? 1.2.4 compiled cleanly but doesn't load successfully and doesn't have any error messages |
| 17:37.25 | *** join/#asterisk fulgas (n=fulgas@209.8.233.242) |
| 17:37.50 | tainted_ | [chan_phone.so] => (Linux Telephony API Support) is the last line i get before it returns to shell prompt |
| 17:38.03 | jbalcomb | docelm0 ? |
| 17:38.46 | *** join/#asterisk seelen_ (n=_seele@200.124.172.72) |
| 17:38.51 | Falle | adibar: somehow it dont wanna download the tones with the new firmware anymore. I'll have to look in to that later :/ Thanks anyway |
| 17:38.54 | lo_tech | Nemesis760: well, I'm connected to equipment that allows me to do the equivalent of a 'stop trunk XX gracefully' ... and I doubt you're connected anywhere near like we are... |
| 17:39.00 | Flauto | hi people, how to dial a remote sipura adapter which is not registered to my asterisk? |
| 17:39.06 | Nemesis760 | lo_tech: All our traffic is inbound on these PRIs. I know I could bridge them over to another box (even TDMoE) but I'd really like to just have the call diverted by the switch. If box 1 is overloaded, it shouldn't be bridging a bunch of calls. |
| 17:39.24 | Flauto | i was trying xxxxxx@ipaddress or the sipura |
| 17:39.34 | Flauto | the phone is ringing on the other side |
| 17:39.38 | Flauto | but no voice |
| 17:39.50 | adibar | Falle: Strange, I used them with the old and now with the beta-firmware and it allways worked. |
| 17:40.01 | seelen_ | hey i wanna know how to listen a call that's being established in another extension different from my own.. without being discovered... in other words, i wanna spy on someone's call, is that possible with asterisk.. how?? |
| 17:40.03 | *** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it) |
| 17:40.50 | lo_tech | Nemesis760: if you have a number of telco-inbound trunks then how are you getting full circuits that would go to another box? (i.e., if the calls are inbound... how could you possibly get the call delivered to another server on the inbound leg?) |
| 17:40.51 | Nemesis760 | lo_tech: I've got 4 DS3s coming in to Adtran MX2800 mux's... then on to the Sangomas... Do you know if the Mux can do this? |
| 17:41.20 | seelen_ | hey i wanna know how to listen a call that's being established in another extension different from my own.. without being discovered.. in other words i wanna spy on someone's call, is that possible with asterisk? |
| 17:41.43 | lo_tech | we can with our DDM-1000, no clue on the syntax of the Adtran |
| 17:41.47 | Damin | seelen_: app_chanspy or app_monitor |
| 17:41.54 | lo_tech | afk, brief |
| 17:41.56 | Nemesis760 | lo_tech: all our PRIs are in the same trunk group. |
| 17:41.59 | mut | the heck does Feb 7 12:40:04 WARNING[8945]: chan_sip.c:9527 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"Scott" <sip:9896851099@65.111.222.5>;tag=as1d154c26' mean |
| 17:42.05 | Nemesis760 | Well.. split up over 2. |
| 17:42.15 | seelen_ | Damin, is that included in Asterisk? explain me how to make it work please. |
| 17:42.28 | Damin | seelen_: TO make it work, you should read the docs. |
| 17:42.44 | seelen_ | Damin, Which ones?? please |
| 17:42.57 | Damin | seelen_: Check out the book "Asterisk, the Future of Telephony" from O'Reilly. That is a great book.. |
| 17:43.04 | file | VERY great book |
| 17:43.29 | jbalcomb | Most AWESOME great book EVER! |
| 17:43.29 | Damin | seelen_: What documentation have you read so far? What investigation have you done to this point? |
| 17:43.47 | Flauto | hello? |
| 17:43.59 | Flauto | please help with my issue |
| 17:44.08 | seelen_ | Damin, I have just been requested (my boss), if thats a feature that asterisk supports. |
| 17:44.14 | file | Flauto: if someone wants to help, they'll try to help |
| 17:44.27 | Damin | seelen_: Then he is paying you to figure it out, right? |
| 17:45.14 | Falle | adibar: the phone only tries to download the cfg<MAC> from the webserver. nothing else. Suppose i have to make one of those from a templatefile then :/ |
| 17:45.21 | Nemesis760 | lo_tech: I'll look into that. Perhaps I can use expect scripts to talk to the Mux. If this works out, I'd like to give you credit. Do you have an email tied to PayPal? |
| 17:46.37 | seelen_ | Hey, how can i configure an E1 card, lines? |
| 17:46.49 | adibar | Falle: I did it half a year ago... But I don't remember anymore if I had to tweak also something inside the web-gui of the GXP |
| 17:47.32 | Flauto | as i understand, it should be pretty simple to call a remote ata adapter |
| 17:47.41 | Flauto | but it does not work |
| 17:47.48 | Flauto | the phone would ring but no voice |
| 17:47.51 | file | is it behind NAT? |
| 17:47.54 | file | a firewall? |
| 17:47.57 | Flauto | file |
| 17:48.12 | Flauto | i did forward ports from 5060 to 5063 |
| 17:48.19 | Flauto | it does file |
| 17:48.21 | file | that doesn't matter |
| 17:48.30 | Flauto | it is behind a router |
| 17:48.32 | file | you have to tell Asterisk to ignore that essentially |
| 17:48.36 | file | use nat=yes and canreinvite=no |
| 17:48.45 | Flauto | file |
| 17:48.50 | file | the IP address and port for audio is carried in the SIP signalling, so it's trying to go to your internal IP address |
| 17:48.55 | Flauto | let me tell you more of my situation |
| 17:49.00 | Flauto | i am using asterisk on my side |
| 17:49.04 | adibar | Falle: As far as I remember I also had to check the "custom ring tone" option inside of one of the lines. |
| 17:49.13 | Flauto | the sip dapter is at my sister's place in new jersey |
| 17:49.19 | file | just do what I said :) |
| 17:49.26 | file | and when it works, you'll realize you wasted time explaining it |
| 17:49.31 | file | because your situation is one that tons of people have |
| 17:50.01 | Flauto | she does not have an asterisk on her side |
| 17:50.14 | Flauto | so i need to set her adapter to nat=yes |
| 17:50.30 | Flauto | canreinvite=no? |
| 17:50.30 | file | http://www.joshua-colp.com/?page_id=5 |
| 17:50.52 | Flauto | file, thanks for your help |
| 17:51.02 | Damin | seelen_: Well, for configuring an E-1, I'd reccomend looking at the Chapter in "Asterisk The Future of Telephony" from O'Reilly that specifically discusses the configuration of Zap devices. Alternatively, you could look on google... |
| 17:51.07 | file | so many people have had your problem, I wrote a page about it! |
| 17:52.03 | Flauto | file |
| 17:52.09 | Falle | adibar: i solved it.. i had dissabled the "automatic firmware checking" after the 1.0.2.3 beta whent into the reboot loop erlier |
| 17:52.12 | Falle | :) |
| 17:52.20 | Flauto | the thing is my sister's sip adapter is not a part of my asterisk |
| 17:52.27 | Flauto | she is using her own broadvoice |
| 17:52.34 | file | then put nat=yes and canreinvite=no in your general part |
| 17:52.37 | file | or make a peer entry |
| 17:52.39 | jbalcomb | FYI: +10 karma points for docelm0 |
| 17:52.42 | adibar | Falle: Do U like the sound ? |
| 17:52.43 | Flauto | so i it has it already |
| 17:53.01 | Flauto | but since her adapter is not registered as a part of my asterisk |
| 17:53.02 | Damin | jbalcomb: Oh yea? What'd he do? |
| 17:53.04 | file | just. try. it. |
| 17:53.15 | *** join/#asterisk ToTo (n=ToTo@host191-157.pool872.interbusiness.it) |
| 17:53.25 | Flauto | i have it on my general part in sip.conf |
| 17:53.43 | file | is your Asterisk behind NAT too? |
| 17:54.15 | jbalcomb | Damin He got me the SIP firmware image for my new Cisco 7940Gs |
| 17:54.24 | seelen_ | Damin, Thanks you've been of much help |
| 17:54.31 | jbalcomb | Damin you have to have 'access' to Cisco to get it. |
| 17:54.35 | adibar | Any Pattern-Matching-God online ? |
| 17:54.45 | hardwire | i feel depressed |
| 17:54.45 | Damin | seelen_: Always a pleasure! |
| 17:54.50 | file | adibar: just ask a question and you'll hopefully get an answer |
| 17:54.59 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 17:55.00 | file | s/a question/a better more complete question |
| 17:55.02 | Falle | adibar: sure, its very nice :) Dont like the annoying noise the phone makes when restarting the loop though.. but i suppose thats just my phone? |
| 17:55.11 | hardwire | file: I am depressed! |
| 17:55.17 | file | hardwire: when aren't you? |
| 17:55.24 | hardwire | don't you sympathize! |
| 17:55.30 | file | hardwire: not really, not for you :P |
| 17:55.31 | Damin | jbalcomb: Well, I'd suggest that you not broadcast that too loudly lest he lose his ability to access CCO! ;) Your not supposed to be re-distributing their firmware, although they don't much seem to care about it. |
| 17:55.37 | hardwire | file: now thats just mean. |
| 17:55.49 | hardwire | *sigh* |
| 17:55.53 | file | Damin can vouch |
| 17:55.53 | hardwire | I feel like eddie |
| 17:55.58 | Flauto | file, here is my sip.conf general http://pastebin.ca/40581 |
| 17:56.01 | jbalcomb | Damin I'll keep it hush hush for sure. |
| 17:56.10 | Flauto | file, it is working from my end |
| 17:56.12 | hardwire | ok.. brb.. I am going to go lift my spirits with some vodka! |
| 17:56.14 | adibar | file: why does such a construct not work: _123X./1234,s... |
| 17:56.15 | Flauto | i think it is my sister's side |
| 17:56.17 | file | Flauto: externhost |
| 17:56.22 | file | not externip |
| 17:56.29 | Flauto | okay |
| 17:56.31 | Flauto | let me try |
| 17:56.31 | file | and you need nat=yes |
| 17:56.33 | file | and canreinvite=no |
| 17:56.39 | rob0 | file was never mean to me! I feel cheated!! |
| 17:56.44 | file | rob0: ummm |
| 17:56.49 | Flauto | okay |
| 17:56.53 | adibar | Falle: I made it also because the default was anoying and loud as hell ;-) |
| 17:56.53 | rob0 | ouch!! |
| 17:57.35 | file | adibar: what exactly do you want it to do |
| 17:58.53 | Flauto | file, even though my sister's spa is not registered to my asterisk, i am trying to call xxxxxxxxxx@heripaddress |
| 17:59.15 | file | it uses the options in general if it's not using a peer entry |
| 17:59.17 | *** join/#asterisk techie (i=gus@antibala.com) |
| 17:59.25 | Flauto | the xxxxxxxxxx is her authid with broadvoice |
| 17:59.28 | file | now in an ideal world you'd read what I said before and used a peer entry, but since we don't live in an ideal world that's okay |
| 17:59.32 | Flauto | that would work |
| 17:59.39 | kippi | does A2Billing give you all the calls you have made, incoming calls etc? |
| 17:59.54 | Flauto | peer entry |
| 17:59.55 | Flauto | okay |
| 18:00.05 | Flauto | okay |
| 18:00.07 | Flauto | i got it |
| 18:00.08 | *** join/#asterisk j4m3s_ (n=j4m3s@146.229.185.15) |
| 18:00.08 | Flauto | thanks |
| 18:00.11 | Flauto | i am so stupid |
| 18:00.13 | Flauto | sorry |
| 18:00.18 | areski | kippi, those that pass through a2billing |
| 18:00.27 | Ahrimanes | kippi: hm only if you pass incomming calls through a2billing |
| 18:00.31 | Ahrimanes | areski: :) |
| 18:00.38 | kippi | is that easy to setup? |
| 18:00.39 | areski | Ahrimanes, :) |
| 18:00.48 | Flauto | i thought that her adapter is not registered with my asterisk there would had nothing to do with my asterisk |
| 18:00.51 | adibar | file: I've got ISDN with 10 MSNs on all of them I can and want to reveive SMS. So before I distribute the individual numbers to theyr owners I want to check if the call comes from the SMS-central and if yes it goes to another sub-routine. Currently I do it ten times. but I want to make it in one check. |
| 18:01.00 | Flauto | yes, i can set a peer entry |
| 18:01.01 | areski | Ahrimanes, I will stop to answer qst |
| 18:01.06 | areski | Ahrimanes, I leave it to u :D |
| 18:01.11 | Ahrimanes | areski: haha |
| 18:01.18 | Flauto | this way |
| 18:01.26 | Ahrimanes | areski: somehow i think you know a2billing slightly better than me ;) |
| 18:01.26 | file | adibar: okay can you paste that pattern again? cleared my screen. |
| 18:01.28 | Flauto | i dont need to do anythign in general section |
| 18:01.44 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 18:02.16 | areski | Ahrimanes, I know the code, I dont use it :P |
| 18:02.20 | adibar | file: exten => _1234X./0622100000,1,do something |
| 18:02.24 | Ahrimanes | areski: that's true |
| 18:02.32 | Ahrimanes | areski: but neither do i, yet ;) |
| 18:02.46 | file | okay so that'll match the dialed number 1234[0-9] and allow any digits after that to be matched, with a caller id of 0622100000 |
| 18:02.48 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 18:03.10 | adibar | file: should, but does not... |
| 18:03.28 | file | well, make one that it will absolutely match for sure |
| 18:03.34 | file | and grab the callerid and extension dialed in a noop |
| 18:03.38 | areski | Ahrimanes, go & find those blond Danish babes |
| 18:03.40 | file | and use your brain to match it against the pattern |
| 18:03.54 | Ahrimanes | areski: might just have one waiting at home... hope she cooks |
| 18:04.09 | areski | hahaha |
| 18:04.58 | Ahrimanes | areski: see ya man.. gotta go find food |
| 18:05.15 | adibar | file: I could do it with an ExecIf, but I think that is no that elegant like the other solution. The only advantage would be, taht it would work ;-) |
| 18:05.52 | areski | Ahrimanes, see u later |
| 18:06.09 | adibar | file: My main problem is, that I don't understand why it does not work. |
| 18:07.22 | adibar | file: Something like 12345/_543X.,1,do works fine, but why not the other case ? |
| 18:08.55 | file | interesting... |
| 18:09.15 | file | I'd have to look closer |
| 18:09.20 | file | but if it works... use it! |
| 18:10.14 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 18:11.01 | Nemesis760 | lo_tech: Thanks for the suggestions... unfortunately it doesn't appear the Adtran MX2800 can manipulate individual DS0s. |
| 18:11.38 | adibar | file: mom |
| 18:12.40 | jpablo | hey people anyone is ussing chan_oss with recent asterisk ? |
| 18:14.11 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:14.28 | adibar | file: does not work, checked it. |
| 18:14.56 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:16.03 | Flauto | file, you taught me a lesson, thanks |
| 18:16.04 | *** join/#asterisk Hertell (n=Rene@jumbo52.adsl.netsonic.fi) |
| 18:16.20 | Hertell | Good evening everybody! :-) |
| 18:16.20 | Flauto | i never even thought about that i should set a peer entry for this |
| 18:16.20 | lo_tech | Nemesis760: no paypal chief, just help the next guy |
| 18:16.34 | file | Flauto: and now it works? |
| 18:16.43 | Flauto | i always thought it should be the problem from the other end |
| 18:16.49 | Flauto | my sister's end |
| 18:16.53 | Flauto | i have not tried yet |
| 18:16.58 | Flauto | my parents are taking a nap now |
| 18:17.02 | Flauto | i will try later for sure |
| 18:17.11 | Flauto | but i did setup a peer entry |
| 18:17.31 | Hertell | can someone point me to a howto in how to configure a sipora spa 3000 with asterisk? |
| 18:17.36 | file | with nat=yes and canreinvite=no? and fixed your general section with externhost? |
| 18:17.37 | Flauto | now, i see sip show peers |
| 18:17.44 | Flauto | i see the connection |
| 18:17.54 | Flauto | yes, file |
| 18:17.55 | Flauto | i did |
| 18:18.14 | Flauto | i was using nat=yes on my most entries |
| 18:18.15 | adibar | file: back to static numbers it works again :( |
| 18:18.21 | Flauto | now, i can use nat=yes in general |
| 18:18.33 | Flauto | and then, left the ones in every entry |
| 18:18.54 | *** join/#asterisk justinu (n=justin@72.18.13.34) |
| 18:18.58 | hypnox | Hertell read and understand some very basic sip concepts, then look for the corresponding fields in the sipura web config |
| 18:19.21 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfmtk.dialup.mindspring.com) |
| 18:19.44 | scardinal | whenever I dial I get the busy tone.. the phone shows up using sip show users/peers |
| 18:19.51 | scardinal | any suggestions why I get the busy tone?! |
| 18:20.08 | mut | anyone have any tellabs 253 shelfs? |
| 18:20.08 | jpablo | is there a way to load modules in a running * ? |
| 18:20.16 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 18:20.57 | Hertell | hypnox: do you a link to such sip-documents? |
| 18:21.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:21.55 | hypnox | Hertell not at hand, but all sip is generally the same, so just look for general instructions for connecting a sip device to asterisk |
| 18:24.00 | Hertell | hypnox: well, the basics i have somehow there. I have got FWD connected with iax, and I can place a test-call that rings in the phone connected to the spa3000 |
| 18:24.41 | hypnox | sounds like it works |
| 18:24.42 | Hertell | hypnox: but I can't really understand how i for example can call out via the pstn-line.. |
| 18:24.47 | voipme | exit |
| 18:25.54 | hypnox | it's not complicated but you'd be better off reading some general asterisk guides, they will be more useful than irc, unless you have a specific question |
| 18:26.28 | *** join/#asterisk vaewyn (i=freeman@mail.parrishmachine.com) |
| 18:26.37 | vaewyn | howdy all |
| 18:26.50 | vaewyn | anyone try the units from x100p.com? |
| 18:27.11 | Hertell | mostly i'm lost with the basic terms used within *, like eg trunk.. :-( |
| 18:28.26 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
| 18:29.20 | *** join/#asterisk WasPhantom (n=neil@203-86-197-11-lightning.thepacific.net) |
| 18:34.10 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 18:35.19 | hypnox | does anyone here know of a good place to get some test numbers? Im looking for non-UK ones. |
| 18:36.28 | marcus2_ | how do i turn down the gain on my zap t1 card |
| 18:36.30 | docelm0 | hypnox, try DIDX.org they bost they give 2 out free on signup |
| 18:36.49 | hypnox | ta |
| 18:36.55 | docelm0 | ?? |
| 18:37.05 | hypnox | that means thanks :) |
| 18:37.11 | docelm0 | oo k |
| 18:38.07 | wundaboy | im trying to learn asterisk. Ive been reading voip-info and learning alot about asterisk, but the pieces are not falling into place just yet. where do i setup a sip phone? extensions.conf? |
| 18:38.41 | docelm0 | wundaboy, yes and also sip.conf |
| 18:38.55 | docelm0 | at least your doing it the right way and not using A@H or AMP |
| 18:39.16 | wundaboy | do i setup my voip provider in iax.conf? and it would be a peer, correct? |
| 18:39.27 | docelm0 | extensions is used for your dialing plan. Where to route the call and such.. Sip.conf is used for setting up your SIP end points |
| 18:39.43 | docelm0 | yes and depends on who your provider is and if they support IAX |
| 18:40.11 | wundaboy | im using junction networks and my did is setup for iax incoming and either sip or iax outgoing |
| 18:41.56 | docelm0 | then yes.. IAX for incoming and then your choice for IAX/SIP for outbound. |
| 18:44.03 | wundaboy | is it ok to use the sample config files for my server? do they by default contain settings that would not allow me to setup incoming/outgoing on an ip500? |
| 18:44.25 | *** join/#asterisk fiber0pti (n=John@206-169-194-79.gen.twtelecom.net) |
| 18:44.37 | fiber0pti | How do you set callerID for all outgoing non-sip calls |
| 18:44.38 | docelm0 | phone doesnt matter for calling for the most part.. Just features.. |
| 18:44.51 | docelm0 | fiber0pti, same as SIP calls |
| 18:45.11 | docelm0 | set(CallerID(num)=???) |
| 18:45.18 | docelm0 | I believe is the syntax. I would have to check |
| 18:45.20 | fiber0pti | docelm0: I'm attempting to set it via CALLERID(number) in the outgoing context, but it doesn't change |
| 18:45.28 | Damin | docelm0: Hows it going? Things settle down for you yet? |
| 18:45.30 | docelm0 | Thats been depreciated |
| 18:45.35 | fiber0pti | I also see it executing in the cli but the clid still comes up as the old one |
| 18:45.36 | Math` | Set(CALLERID(number)=123456789) |
| 18:45.39 | docelm0 | Damin, god I wish |
| 18:45.50 | docelm0 | thanks Math` I knew I was close |
| 18:45.51 | Damin | docelm0: Going off the deep end yet? :) |
| 18:46.01 | docelm0 | Damin, dude I jumped a month ago |
| 18:46.12 | *** join/#asterisk DrData (n=michael@p54B278FD.dip.t-dialin.net) |
| 18:46.17 | docelm0 | Damin, but making progress should have new code out this weekend hopefully |
| 18:46.24 | Damin | docelm0: Rock! |
| 18:46.46 | *** join/#asterisk saftsack (n=oliver@IP-213188106101.dialin.heagmedianet.de) |
| 18:46.47 | saftsack | hi |
| 18:46.48 | Damin | docelm0: If you want my .01 cents worth, let me know.. |
| 18:46.49 | wundaboy | docelm0: is it ok to use sample config files? |
| 18:46.49 | docelm0 | wundaboy, yes.. you can use sample.. However it depends on where you live and your dialing plan |
| 18:46.52 | fiber0pti | Math`: I've tried that and it doesn't set it on the caller id on another phone that's external to the system |
| 18:46.56 | docelm0 | Damin, thanks.. :) |
| 18:47.02 | DrData | SIP+CAPI: how do I get the callerid through to the called ISDN phone |
| 18:47.04 | wundaboy | docelm0: i live in portland, oregon, usa |
| 18:47.15 | wundaboy | i assume that its fine? |
| 18:47.15 | docelm0 | wundaboy, then NA Dialing Plan |
| 18:47.21 | saftsack | my variable office/voicemail doesnt work. do i have to initalize it anywhere? |
| 18:47.45 | docelm0 | wundaboy, standard for most of us.. You just have to setup and route your outbound based on how you match your extensions |
| 18:48.06 | *** join/#asterisk bjames (n=bjames@67-102-228-17.adsl.lbdsl.net) |
| 18:48.08 | bjames | hi |
| 18:48.14 | docelm0 | lo |
| 18:48.26 | bjames | where do I submit a bug report? |
| 18:48.32 | *** join/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net) |
| 18:48.36 | docelm0 | bugs.digium.com |
| 18:49.05 | sevard | Has anyone made a weather agi script that uses Allison Smith's voice? I have a script that downloads daily from noaa.gov and uses festival for text to speech, but that doesn't use Allison. |
| 18:49.33 | docelm0 | Im not sure allison would approve.. :) |
| 18:49.52 | docelm0 | sevard you could have her make them custom for you.. its fairly inexpensive and she is quick |
| 18:49.58 | docelm0 | 24 hour or less turn around |
| 18:50.16 | sevard | docelm0: there is already a WX directory |
| 18:50.28 | sevard | i was curious if someboyd built a script around those gsm |
| 18:50.28 | docelm0 | ohh |
| 18:50.49 | docelm0 | cant be that hard.. I could probably code one up in no time flat |
| 18:51.06 | sevard | want to? |
| 18:51.21 | docelm0 | Ill add it to my list of GNU asterisk projects.. :) |
| 18:51.30 | docelm0 | probably have it done by this weekend |
| 18:51.45 | docelm0 | send me your scripts and such and I will build one |
| 18:52.37 | sevard | mine is simply a bash script |
| 18:52.59 | docelm0 | Send away.. I will make it work for ya |
| 18:53.11 | docelm0 | Hell I could probably make it where it prompts for the ZIP and reports live :) |
| 18:55.09 | wundaboy | type=peer will allow both incoming and outgoing, correct? |
| 18:55.23 | docelm0 | Nope.. Friend |
| 18:55.25 | docelm0 | will |
| 18:55.45 | wundaboy | gotcha |
| 18:56.45 | docelm0 | user == incoming to PBX, peer == outbound from pbx, friend == a very happy option :) |
| 18:56.45 | wundaboy | what is peercontext? |
| 18:57.12 | wundaboy | the wiki dosent go into it, is it necesary? |
| 18:57.23 | docelm0 | I dont use it |
| 18:57.39 | sevard | dothat would be wsesome |
| 18:57.45 | sevard | docelm0: that would be awesome |
| 18:58.03 | sevard | docelm0: i can c/p my bash script in msg, that work for you? |
| 18:58.20 | sevard | it's only like 20 lines long |
| 18:58.32 | docelm0 | pastebin.ca |
| 18:58.36 | docelm0 | I dont wanna get flooded off |
| 18:58.41 | sevard | k |
| 18:59.05 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 18:59.39 | docelm0 | I may add that script to my PBX at the office.. I took away everyone's access to weather sites.. :) |
| 18:59.44 | docelm0 | Im an ass.. I know it. |
| 18:59.50 | wundaboy | if i setup an iax (context?) do i still need to 'register' it? or does it do that automatically? |
| 19:00.02 | docelm0 | Nope.. Need Register => |
| 19:00.29 | docelm0 | ok need NICCOTINE.. brb |
| 19:00.47 | sevard | docelm0: http://pastebin.ca/40587 |
| 19:01.22 | mut | whats |
| 19:01.22 | mut | Feb 7 14:00:09 WARNING[8945]: chan_sip.c:1208 retrans_pkt: Maximum retries exceeded on transmission 4fb6c544677f5b6177dcc7da0b6ee049@65.111.222.5 for seqno 102 (Critical Request) |
| 19:01.22 | mut | Feb 7 14:00:09 WARNING[8945]: chan_sip.c:1225 retrans_pkt: Hanging up call 4fb6c544677f5b6177dcc7da0b6ee049@65.111.222.5 - no reply to our critical packet. |
| 19:01.22 | mut | <PROTECTED> |
| 19:01.25 | mut | mean? |
| 19:01.32 | mut | i try to call a logged in user |
| 19:01.36 | mut | and it just sits |
| 19:01.38 | mut | and then gives that |
| 19:01.43 | mut | and the call fails |
| 19:01.52 | sevard | docelm0: it was quick and it's not efficient |
| 19:02.38 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 19:04.01 | sevard | docelm0: if you do decide to write the script you should do it so that the txt files are croned wget processes and catched locally, the script shouldn't download and run all on one call, that eats bandwidth and sometimes the ftp server can take 30-40 seconds to respond |
| 19:04.04 | sevard | not ideal |
| 19:04.20 | GerbilNut | where would you put, and what would you put, to allow a phone to transfer a call to another phone |
| 19:04.34 | *** join/#asterisk Eggplant (n=none@dsl-304.cascadeaccess.com) |
| 19:04.39 | Qwell[] | GerbilNut: What type of phone? |
| 19:04.44 | GerbilNut | Snom 360 |
| 19:04.50 | Qwell[] | they have transfer buttons |
| 19:05.00 | GerbilNut | It doesn't seem to be working : / |
| 19:05.26 | GerbilNut | does the user need to be allowed to transfer or should it just work? |
| 19:05.33 | Qwell[] | should just work |
| 19:05.49 | _Sam-- | what about the Tt |
| 19:06.00 | Qwell[] | no |
| 19:06.02 | Qwell[] | never use tT |
| 19:06.08 | _Sam-- | im not saying use both |
| 19:06.11 | _Sam-- | but that could be the cause |
| 19:06.11 | _Sam-- | no t |
| 19:06.13 | _Sam-- | or T |
| 19:06.23 | Qwell[] | Those allow # transfers |
| 19:06.25 | GerbilNut | where would that be? |
| 19:06.32 | Qwell[] | the transfer is done on the phone with SIP |
| 19:06.43 | mut | anyone have any clues |
| 19:06.59 | _Sam-- | GerbilNut: how are you trying to transfer the call? using the button on the phone? |
| 19:07.04 | _Sam-- | <there are a few different ways> |
| 19:07.05 | docelm0 | sevard, I will code it in php and make it pull live from the website.. :) |
| 19:07.39 | docelm0 | But at the same token have it pull locally from a file if its recient |
| 19:07.39 | GerbilNut | yeah |
| 19:07.47 | sevard | docelm0: the only thing is that if you pull it live from the website on call a caller might sit and wait while your php tries to parse with a slow connection |
| 19:08.13 | docelm0 | Well why on gods green earth would someone want to use a dialup w/ asterisk? Thats nuts.. |
| 19:08.14 | docelm0 | :) |
| 19:08.20 | docelm0 | But understandable.. |
| 19:08.42 | GerbilNut | i get a call, answer it, push the transfer button, dial the other extension |
| 19:08.47 | lithi | Does accountcode=101 work in the iax.conf file? Cause I set it for a user then did a NoOp(${ACCOUNTCODE}) and it came up blank. (I did do a shutdown/restart after making iax.conf changes) |
| 19:09.04 | GerbilNut | now, it's ringing the other extension and dropping the call from me so I can talk to the other person and tell them who it is before it transfers |
| 19:09.23 | GerbilNut | i guess it's doing a blind transfer, not a supervised transfer |
| 19:09.24 | saftsack | hi |
| 19:09.33 | saftsack | pickup brings my asterisk to segfaulting :( |
| 19:10.26 | lithi | GerbilNut: Yea you want a 'Attended Transfer (or "consultative transfer")' but how you do that, no idea |
| 19:11.07 | docelm0 | crap.. Im gonna have to use weather.com they use ZIP codes.. Sevard does NOAA have forecasts based on ZIP? |
| 19:11.23 | sevard | docelm0: no, not dialup, NOAA is sometimes bogged down with traffic |
| 19:11.33 | sevard | docelm0: it does it based on zip, i think. |
| 19:11.44 | sevard | the ftp is sorted by state/city though |
| 19:11.57 | sevard | you could do a POST to the search bar on their webpage |
| 19:12.11 | _Sam-- | GerbilNut: have you tried it this (dont know if it will work ...dont have that phone)....call on line you want to transfer....put that call on hold....call the person you want to transfer to on line 2 |
| 19:12.22 | _Sam-- | talk to the person, let them know about the caller on line...then press transfer |
| 19:12.26 | _Sam-- | and transfer the 2 together |
| 19:12.32 | GerbilNut | can try it |
| 19:13.06 | *** join/#asterisk vattern (n=vattern@dsl-146-150-218.telkomadsl.co.za) |
| 19:14.06 | docelm0 | sevard, I dont see it.. |
| 19:14.09 | docelm0 | I will work something |
| 19:14.45 | GerbilNut | _Sam--, that works, alittle complicated for the people here, but it'll work |
| 19:15.13 | _Sam-- | there is probably another way as well |
| 19:16.46 | *** join/#asterisk dchen (n=dchen@66.146.130.82) |
| 19:17.01 | sevard | docelm0: top of the page :) |
| 19:17.09 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 19:17.36 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 19:17.59 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 19:18.11 | docelm0 | top of the page? |
| 19:19.08 | *** join/#asterisk clive- (n=pirch@dsl-165-117-139.telkomadsl.co.za) |
| 19:20.37 | *** join/#asterisk __alex (n=alex@62.206.18.218) |
| 19:21.15 | dchen | anyone got a good success story of how asterisk made you rich? |
| 19:21.18 | iCEBrkr | docelm0: What's up Mr. MIA |
| 19:21.43 | iCEBrkr | dchen: Pass the dooby, man.. |
| 19:21.45 | dchen | by the way, sangoma is right up the street fromw here i work |
| 19:21.49 | dchen | are they good? |
| 19:22.15 | dchen | (for hobby, super-doper answer-machine project, not for work) |
| 19:22.18 | clive- | dchen in canada? |
| 19:22.20 | docelm0 | Busy as hell |
| 19:22.23 | dchen | yep |
| 19:22.23 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-112-192.red.bezeqint.net) |
| 19:22.29 | docelm0 | sevard, top of what page? Send me a URL |
| 19:22.30 | dchen | i am on steeles and woodbine |
| 19:22.57 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) |
| 19:23.08 | clive- | dchen well the guy is orginaly from south africa...and sangoma=witchdoctor,,,,just some useless info for you |
| 19:23.19 | dchen | clive-: huh? |
| 19:23.47 | Hymie | hey all.. does anyone else have problems in MeetMe, with clipping audio from SIP users? |
| 19:23.50 | dchen | what are some fairly hands-off business that i can build with asterisk? |
| 19:24.00 | dchen | pre-paid cards? |
| 19:24.09 | dchen | dating service? |
| 19:24.12 | iCEBrkr | LOL |
| 19:24.22 | iCEBrkr | dchen: You sound like one of those 'Get rich quick' scammers |
| 19:24.25 | docelm0 | dchen, dream it build it |
| 19:24.26 | dchen | i just want a little income, something fun to goof around with |
| 19:24.30 | clive- | no such thing as a "hands-off" business:),,,,,,,trust me |
| 19:24.42 | dchen | ... |
| 19:24.47 | iCEBrkr | clive-: Sure there is, collect the money and then leave the contry. |
| 19:24.49 | iCEBrkr | err country |
| 19:25.03 | clive- | lol...run like hell ,:) |
| 19:25.16 | dchen | actually i am just trying to find some business excuse ot contribute to the asterisk OSS movement |
| 19:25.35 | dchen | if it takes the wife and me to an extra dinner outside a month, that's a nice to have |
| 19:25.48 | _Sam-- | [av]bani you around yet |
| 19:25.57 | clive- | dchen , you will have grey hair like me!..:) |
| 19:26.22 | dchen | if it doesn't make a dime, at least i got to play around with some cool hardware |
| 19:26.29 | dchen | clive-: *winnk* |
| 19:26.35 | dchen | clive-: in toronto? |
| 19:26.45 | *** join/#asterisk Derkommissar (n=Alberto@adsl-144-122-212.mia.bellsouth.net) |
| 19:26.50 | clive- | me,,no I am in south africa |
| 19:26.55 | dchen | sweet |
| 19:26.57 | *** join/#asterisk [hC] (n=hardcore@209.153.195.139) |
| 19:27.03 | lithi | Can you set a call to two diffrent groups? ie Set(GROUP()=1) Set(GROUP()=2) will both group 1 and 2 be +1? |
| 19:27.06 | Derkommissar | i just put the latest svn on my system |
| 19:27.20 | Derkommissar | and my atas are loosing reguistration like crazy |
| 19:27.27 | Derkommissar | i use mostly grandstream |
| 19:27.35 | Derkommissar | is there a reason why? |
| 19:28.32 | *** join/#asterisk vattern (n=vattern@dsl-146-150-218.telkomadsl.co.za) |
| 19:30.35 | *** join/#asterisk StreamR (i=cool@ppp027.111-253-207.mtl.mt.videotron.ca) |
| 19:31.05 | sevard | docelm0: sorry, back. |
| 19:31.20 | *** join/#asterisk hhoffman (n=hhoffman@tor/session/x-83649ecc7bd381a0) |
| 19:31.56 | hhoffman | hi, is it possible to send faxes to asterisk from a DID line provided by a IAX vendor? |
| 19:32.09 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 19:33.37 | StreamR | hi, I would like to know if there is a way to set up asterisk much like a key system? I currently got my asterisk@home to work using the 9 pool key, but is there a way to access a specific trunk line instead of a pool? |
| 19:34.01 | docelm0 | no biggie.. Can you give me the URL for ZIP lookup? |
| 19:34.16 | asterboy | asterisk |
| 19:34.21 | asterboy | ~polycom |
| 19:34.23 | jbot | well, polycom is the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 19:34.45 | docelm0 | Can you say who coded jbot that they are biased for polycom? |
| 19:35.15 | asterboy | polycom good |
| 19:35.30 | nestar | lol |
| 19:35.38 | nestar | jbot is tainted! |
| 19:35.44 | asterboy | check out skype |
| 19:35.48 | asterboy | ~skype |
| 19:35.49 | jbot | rumour has it, skype is um programa de bate-papo via voz, proprietário e fechado, que usa padrões proprietários e fechados, dos mesmos autores do (spyware) kazaa; procure usar alternativas (pelo menos) com padrões abertos/livres, como os do projeto openh323 <http://www.openh323.org/>, speakfreely <http://www.speakfreely.org/> ... |
| 19:36.07 | asterboy | jbot is multilingual |
| 19:36.21 | asterboy | hablo espanola |
| 19:36.21 | StreamR | lol |
| 19:36.51 | asterboy | benjk looooves skype |
| 19:36.52 | Hymie | hey all.. does anyone else have problems in MeetMe, with clipping audio from SIP users? |
| 19:37.22 | StreamR | is there any function to access a specific trunk line? or a way to create one? |
| 19:37.26 | SplasPood | is there something different about the syntax of SIPAddHeader when used in an AEL dialplan? |
| 19:37.34 | asterboy | numbe of users in here has jumped from 300 to 400 |
| 19:37.41 | asterboy | since last on |
| 19:37.51 | _Thor | Hello guys |
| 19:37.54 | iCEBrkr | asterboy: and only 10 of those users actually say anything |
| 19:38.18 | SplasPood | Hrm... figured it out.. it didn't like the space between Alert-Info: Value |
| 19:38.37 | iCEBrkr | SplasPood: Asterisk is pissing me off with this bullshit whitespace shit |
| 19:39.18 | SplasPood | iCE: typical. |
| 19:39.45 | *** join/#asterisk julien[re] (n=mactouch@AStDenis-103-1-11-187.w80-8.abo.wanadoo.fr) |
| 19:39.49 | iCEBrkr | SplasPood: Whitespace between operators and such shouldn't be an issue |
| 19:39.50 | julien[re] | hi there |
| 19:39.55 | sevard | docelm0: did you see what I was saying? |
| 19:39.59 | Meaty | iCEBrkr didn't like the space between iCE: typical. |
| 19:40.02 | Meaty | :o |
| 19:40.12 | SplasPood | iCE: Yea I know.. typical asterisk crap. |
| 19:40.18 | iCEBrkr | SetVar: Test=1 and SetVar: Test = 1 shouldn't be different |
| 19:40.21 | julien[re] | i've got a problem since i purchased and install g729 from digium: init.d and safe_asterisk won't work |
| 19:40.33 | SplasPood | julien: prolly cause asterisk is dying on load |
| 19:40.34 | docelm0 | No |
| 19:40.36 | julien[re] | http://bugs.digium.com/view.php?id=6433 |
| 19:40.50 | docelm0 | I think I got something I found XML feeds |
| 19:40.50 | brookshire | julin: did you call digium support? |
| 19:40.52 | julien[re] | splas: if i launch it on console, it does work |
| 19:41.05 | SplasPood | julien: what output did it give you? |
| 19:41.08 | iCEBrkr | julien[re]: Sounds like it could be persmissions? |
| 19:41.17 | julien[re] | i'm not really fluent in english |
| 19:41.18 | sevard | docelm0: XML feeds are goooooooooooood' |
| 19:41.19 | asterboy | lol...only 10 say anything |
| 19:41.35 | sevard | docelm0: that would be even better, especially if the XML feeds have update hints that you know how to work with |
| 19:42.15 | SplasPood | julien: If you paste the last line of output somewhere... Not sure how to help if you don't understand english, I'm rather uneducated language wise |
| 19:42.20 | brookshire | julien: you can email them, support@digium.com |
| 19:42.29 | julien[re] | ok i'll try email |
| 19:42.31 | sevard | docelm0: plus if you go to http://www.noaa.gov/wx.html at toe top of the page on the left side there is a field to enter zip codes |
| 19:42.49 | julien[re] | the prob is that i've no output |
| 19:42.53 | julien[re] | when i start with init.d |
| 19:43.01 | SplasPood | yes, start it manually |
| 19:43.02 | julien[re] | i just get: |
| 19:43.02 | julien[re] | Asterisk ended with exit status 127 |
| 19:43.03 | SplasPood | from the console |
| 19:43.11 | SplasPood | asterisk -vvvvvvvvvvvvvvvvvvvcg |
| 19:43.13 | julien[re] | it does work from the console |
| 19:43.24 | SplasPood | oh thats semi-interesting.. |
| 19:43.24 | dchen | clive-: did you mention earlier that sangoma hangs in these channels? |
| 19:43.41 | julien[re] | Asterisk Ready. |
| 19:43.45 | julien[re] | from the console |
| 19:43.52 | brookshire | splas: you need a couple more v's |
| 19:43.54 | SplasPood | julien: when init fails to start it, is there anything in... /var/log/asterisk/messages |
| 19:43.54 | brookshire | :) |
| 19:44.03 | SplasPood | brookshire: heh... I just hold v till I get bored :P |
| 19:44.15 | brookshire | that's still not enough |
| 19:44.46 | *** join/#asterisk bigjb (n=bigjb@195.60.10.113) |
| 19:44.47 | julien[re] | nothing in /var/log/asterisk/messages |
| 19:44.58 | julien[re] | and safe_asterisk keeps restarting * |
| 19:45.05 | julien[re] | which exitted statut 127 |
| 19:45.33 | julien[re] | <PROTECTED> |
| 19:45.34 | *** join/#asterisk zigman (i=zigman@irc.zigman.de) |
| 19:45.35 | julien[re] | and so on |
| 19:45.43 | julien[re] | and of course * isnt running |
| 19:45.48 | upsite | @Corydon-w> you are such a funny guy .. if it is a configuration issue on my side i won'T comde to the dev chan |
| 19:46.19 | elg | this is very strange: meetme sounds TERRIBLE (scratchy), definitely seems like a timing issue, but the box has two TDM400P cards each on its own IRQ |
| 19:46.22 | *** join/#asterisk WasPhantom (n=neil@203-86-197-11-lightning.thepacific.net) |
| 19:46.28 | julien[re] | before installing g729 it was fine |
| 19:47.53 | dchen | ~sangoma |
| 19:47.55 | jbot | well, sangoma is a company that makes PRI cards the way Digium should have done it in the first place.... |
| 19:48.20 | docelm0 | sevard, I got the XML code.. I will be building something |
| 19:48.26 | dchen | impressive |
| 19:48.30 | dchen | any sangoma guys here? |
| 19:49.24 | shido6 | any avaya? :) |
| 19:49.28 | SplasPood | julien: somehow safe_asterisk is starting asterisk differently than running it from the console |
| 19:49.37 | julien[re] | guess so |
| 19:49.48 | SplasPood | julien: Maybe the license files cannot be found/read by asterisk |
| 19:49.51 | julien[re] | but the point is that it did work correctly before g729 |
| 19:50.01 | SplasPood | still dunno why it'd be different depending on how it's started |
| 19:50.07 | julien[re] | adeixis-ipbx*CLI> show g729 |
| 19:50.07 | julien[re] | 0/0 encoders/decoders of 4 licensed channels are currently in use |
| 19:50.18 | julien[re] | it is recognized |
| 19:50.25 | julien[re] | (when started from the cli) |
| 19:50.25 | dchen | is this the wrong channel to look for sangoma folks? |
| 19:50.29 | SplasPood | Yea, when you start.. exactly. |
| 19:50.39 | SplasPood | dchen: I'd guess so. |
| 19:51.06 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool138-85.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 19:51.23 | dchen | SplasPood: bummers |
| 19:51.40 | dchen | SplasPood: and it was a really good paper |
| 19:51.52 | julien[re] | it's not a blocking problem, until there's a power failure and i'm not here |
| 19:52.19 | julien[re] | i dont like the idea of having daemon not starting without human intervention |
| 19:52.36 | tzafrir_laptop | upsite, here? |
| 19:52.50 | upsite | yes |
| 19:52.54 | asterboy | I have a Polycom IP 600 that just won't startup properly. |
| 19:53.09 | asterboy | The procedure for my other phones was painless. |
| 19:53.10 | SplasPood | julien: heh, you should be on a UPS :) I'd login and take a look if you wanted |
| 19:53.20 | SplasPood | asterboy: what happens? |
| 19:53.21 | tzafrir_laptop | I wonder what might cause this. |
| 19:53.34 | asterboy | The phone boots up but not with the loaded bootrom and sip app. |
| 19:53.34 | upsite | yeah |
| 19:53.37 | upsite | me too |
| 19:53.43 | upsite | thats why i'm asking |
| 19:53.46 | asterboy | The red light stays on and the phone acts strange. |
| 19:54.00 | asterboy | really delayed. |
| 19:54.01 | SplasPood | aster: do you see it even hitting your tftp? |
| 19:54.08 | asterboy | yes |
| 19:54.14 | tzafrir_laptop | what system is that? what distro? what asterisk? what's the output of 'time ll |wc' |
| 19:54.35 | upsite | its an lfs |
| 19:54.54 | *** join/#asterisk Assid (n=assid@59.183.40.101) |
| 19:55.01 | upsite | tzafrir_laptop 39 306 2429 |
| 19:55.01 | upsite | real 0m0.035s |
| 19:55.01 | upsite | user 0m0.020s |
| 19:55.01 | upsite | sys 0m0.010s |
| 19:55.23 | tzafrir_laptop | not a big directory |
| 19:55.31 | upsite | asterisk 1.2.4 |
| 19:55.44 | asterboy | The sip version on the phone is 1.1.0 and I'm trying to put on 1.5.3 ... doesn't seem to take it off TFTP server |
| 19:55.51 | julien[re] | splaspood, do u have some time to login onto my box? |
| 19:55.56 | julien[re] | (be prepared to a low ping :p) |
| 19:55.59 | SplasPood | julien: 15-20min prolly |
| 19:56.06 | julien[re] | ok no prob |
| 19:56.26 | asterboy | Bootrom is 2.4.1, I want 2.5.3 TFTP server says its delivering, but I don't think its loading properly. |
| 19:56.33 | julien[re] | ping gonna be 350ms since i'm on a sunny island lost in the ocean (but on a submarine cable) |
| 19:56.36 | SplasPood | aster: hrm.. never dealt with such an old revision.. Do you see it requesting ANYTHING.. even files tha... ok nevermind, you just answered it |
| 19:56.57 | asterboy | bootrom.ver is not found though. |
| 19:57.01 | SplasPood | aster: You do know that 3.1.2 bootrom and 1.6.4 SIP is latest |
| 19:57.03 | SplasPood | oh |
| 19:57.08 | SplasPood | that may be related |
| 19:57.08 | asterboy | yes |
| 19:57.14 | SplasPood | the newer firmware doesn't even request that |
| 19:57.27 | docelm0 | ok officially XML is a pain in the ass |
| 19:57.28 | asterboy | yes I know its latest, but I want to stay inside 2.5 bootrom cause you can't go back once your on 3 |
| 19:57.57 | asterboy | How do I force the phone to load the bootrom? |
| 19:58.11 | asterboy | Or better yet, how do I reset the phone to wipe it back to factory. |
| 19:58.21 | dchen | docelm0: yep |
| 19:58.32 | tzafrir_laptop | upsite, so, when you run 'll' the voice stops permanently? or for a while? |
| 19:58.32 | dchen | docelm0: xml == the devil's used napkins |
| 19:58.38 | *** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net) |
| 19:59.18 | upsite | tzafrir_laptop: permantly |
| 19:59.41 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 19:59.53 | tzafrir_laptop | "permanently": until a restart of asterisk? until a reload? until a reboot? |
| 19:59.57 | upsite | i have to delete the /var/lib/asterisk dir and then make install again |
| 20:00.00 | upsite | then it works |
| 20:00.28 | tzafrir_laptop | and others did not work? (restart/reload) |
| 20:00.32 | upsite | nope |
| 20:00.42 | Qwell[] | asterboy: Why not upgrade? |
| 20:00.42 | upsite | im trying a reboot right now |
| 20:01.03 | tzafrir_laptop | and have you tried running 'll' in any other directory? |
| 20:01.17 | asterboy | They say unless you really need to upgrade to 3 don't |
| 20:01.21 | upsite | wait a sec |
| 20:01.57 | asterboy | its at 2.4.1 and I want to upgrade to 2.5.3 |
| 20:02.04 | upsite | reboot doesnt' fix it |
| 20:02.05 | asterboy | but its not loading the bootrom |
| 20:02.32 | asterboy | no reboot does not help |
| 20:03.07 | asterboy | WHere is the bootrom.ver? |
| 20:03.27 | asterboy | what goes in that file is the version number like in sip.ver, I'm guessing |
| 20:06.21 | *** join/#asterisk jets (n=jetsnoc@meowwwww.pmt.coop) |
| 20:06.24 | docelm0 | dchen, well I just got a killer XML feed for wether. I believe I can bring my application to life.. :) |
| 20:06.40 | *** join/#asterisk _Paulo_ (n=paulos@200-168-112-132.dsl.telesp.net.br) |
| 20:06.42 | *** join/#asterisk dijit0 (n=dijit0@adsl-69-110-151-230.dsl.pltn13.pacbell.net) |
| 20:06.44 | Qwell[] | jets: hey |
| 20:07.01 | jsaunders | Any c++ coders wanna help add sip media renegotiation while forwarding rtp to Yate? |
| 20:07.27 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-5.dynamic.qsc.de) |
| 20:07.50 | dijit0 | anyone up for helping me with a weired caller id issue? i called 800 4444444 so it would tell me what number i am caling from, and it says im calling from 9897200700 but my HOME phone says i call from 9999991234 but thats nothing close to teh caller id i have set in asterisk |
| 20:07.53 | dijit0 | neither of those numbers |
| 20:08.06 | tzafrir_laptop | upsite, if a reboot did not solve it, there must be some change to the file system |
| 20:08.48 | tzafrir_laptop | what filesystem do you use? any chance it is corrupted? |
| 20:08.57 | upsite | ext3 |
| 20:09.07 | upsite | <PROTECTED> |
| 20:09.22 | upsite | and as i said its the same on 4 diffrent boxes |
| 20:09.28 | tzafrir_laptop | other things to do: 1. verbose and debug |
| 20:09.32 | dijit0 | come on, someones gotta know whats wrong with the callerid |
| 20:09.47 | *** join/#asterisk Hondo (n=mdobbins@ptech7-231.acdmis.com) |
| 20:09.50 | upsite | verbose is set to 20 and debug is on ..but no messages |
| 20:10.05 | upsite | nothing about "file not found " or "wrong permission2 |
| 20:10.29 | upsite | its just not plaiyng anything only the moh |
| 20:10.33 | tzafrir_laptop | 2. strace (-f) the asterisk process and see if there is any change in the way it accesses sound files before/after you run that command |
| 20:10.36 | Hymie | man, why do kphone and linphone have to suck so much |
| 20:10.37 | Hymie | I mean |
| 20:10.43 | Hymie | couldn't they at least suck just a little less? ;) |
| 20:11.10 | tzafrir_laptop | Hymie, do you use Sarge? |
| 20:11.25 | Hymie | yes, although I am using the testing versions of the above apps anyhowe |
| 20:12.15 | Hymie | all I wanted was a different SIP device to debug the crackling audio in meetme |
| 20:12.24 | Hymie | but oh no ;P |
| 20:12.26 | sevard | docelm0: back again, almost got my car towed, stupid parking bitch, you need anything? |
| 20:12.31 | Hymie | can't have that ;P |
| 20:13.27 | tzafrir_laptop | Testing has iaxcomm, kiax and twinkle |
| 20:14.15 | Hymie | tzafrir_laptop: ok, going to check these beasts |
| 20:14.21 | sevard | docelm0: I just had an idea, you should put in a festival fallback for words that aren't available for Allison |
| 20:14.30 | upsite | tzafrir_laptop asterisk spawns 15 childs which do i have to strace? |
| 20:14.51 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 20:14.53 | Hymie | and if kiax does sip |
| 20:15.13 | tzafrir_laptop | upsite, that's a good question. try them all, and filter for fileacceess |
| 20:15.23 | upsite | ok :) |
| 20:15.36 | tzafrir_laptop | Hymie, no. But you might as well set up an extra iax extension |
| 20:15.45 | tzafrir_laptop | Practically a copy of the SIP setup |
| 20:15.55 | cpm | I know this is a question that gets asked so often, that it borders on stupid. But I am really unhappy with voicepulse, I think they have oversold their bandwidth. Is there a reputable iax peering vendor out there? |
| 20:15.57 | Hymie | tzafrir_laptop: that won't help with debugging an issue that only occurs with SIP in Meetme |
| 20:16.32 | Hymie | tzafrir_laptop: and yes, I have to use sip... unfortunately |
| 20:19.23 | _Sam-- | cpm: it depends on what you need, what backbone you're on, DID needs, etc |
| 20:19.49 | _Sam-- | i teliax for origination and asterlink for termination...both IAX |
| 20:19.58 | _Sam-- | asterlink doesnt do local DID, but they do toll free DID |
| 20:20.05 | *** join/#asterisk snowflyer (n=mdobbins@ptech7-231.acdmis.com) |
| 20:20.18 | file | and I'm always here... since I'm a geek |
| 20:20.32 | _Sam-- | for me and my location, i get better sound quality from asterlink |
| 20:20.45 | _Sam-- | but file's customer service is really lacking! :) |
| 20:20.54 | file | yeah, I'm horrible |
| 20:21.10 | clive- | file=asterlink ? |
| 20:21.10 | file | I'm very rude and don't help at all |
| 20:21.13 | docelm0 | sevard, Im already WAY ahead of you |
| 20:21.17 | file | I'm an employee |
| 20:21.25 | docelm0 | I am working on the PHP now should have something fairly soon to test |
| 20:21.31 | clive- | file whats a toll free number cost? |
| 20:21.35 | docelm0 | And Im using WEATHER.COM :) |
| 20:21.36 | sevard | What does everyone do for a guy whose ISP is blocking outgoing port 5060? |
| 20:21.40 | _Sam-- | 0 + 2c / minute |
| 20:21.44 | _Sam-- | + 14% |
| 20:21.47 | file | $1.95/mth |
| 20:21.49 | docelm0 | sevard, setup VPN |
| 20:21.50 | _Sam-- | + 1.95 sign up |
| 20:21.53 | file | plus what _Sam-- said |
| 20:21.53 | mogorman | iax docelm0 |
| 20:21.53 | sevard | docelm0: noaa is more reliable than weather.com :P |
| 20:21.55 | sevard | <PROTECTED> |
| 20:22.12 | sevard | docelm0: I'm going to use Hamachi, i was just wondering if anyone knew any better situations |
| 20:22.13 | docelm0 | sevard, true.. BUT I can search on ZIP |
| 20:22.33 | clive- | sam, thats a lot of plusses...:).... |
| 20:22.35 | cpm | _Sam--, thanks for the reply. Basically I'm looking for someone doing the same thing as connect.voicepulse.com without the dodgy latency issues that magically crop up, that they keep blaming on your circuit. (wierd, I'm not seeing any latency in my http traffic, , , and I'm not congested, , ??) |
| 20:22.37 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:22.48 | file | all the plusses gave me headache |
| 20:22.48 | _Sam-- | clive-: at first it seems that way...but in the end, the price is about the same as everyone else |
| 20:22.59 | cpm | It was sort a lame rant anyway. |
| 20:23.08 | file | we can also do PSTN failover... and VoIP failvoer |
| 20:23.11 | file | er over |
| 20:23.14 | _Sam-- | cpm...asterlink is ok with me...but i did have the latency problems with teliax |
| 20:23.20 | _Sam-- | but again alot of it is network dependent |
| 20:23.27 | _Sam-- | like which network backbone you are on, the routes, etc |
| 20:23.30 | clive- | thanks for the info |
| 20:24.03 | cpm | I'm on AS7013, and I'm kinda all alone out here in the sticks, until I hit pittsburg |
| 20:24.07 | _Sam-- | clive-: for 5 bucks you can get a pretty good taste of asterlink |
| 20:24.14 | lithi | Why when a softphone is offline does asterisk think its busy? It should be unavailable. CLI reports Unable to create channel of type 'IAX2' (cause 3 - No route to destination). |
| 20:24.16 | harryvv | Anyone know if most of these voip sersellers are just selling there voip service at a steady 19.95 per month when most if not all customers will never aproach the 19 dollars in cdr calls? I think that this is how vonage works. |
| 20:24.25 | sevard | docelm0: you can search on zip at noaa too |
| 20:24.27 | harryvv | file, whats up |
| 20:24.37 | file | I am presently working. |
| 20:24.38 | sevard | i told you, go to noaa.gov/wx.html and wher it says Enter city/state jsut enter a zip code |
| 20:24.38 | sevard | where, just, got my typing is horrid today |
| 20:24.48 | *** join/#asterisk areski (n=areski@28.Red-83-44-66.dynamicIP.rima-tde.net) |
| 20:24.52 | cpm | 7018 rather |
| 20:25.01 | MikeJ[Laptop] | file, that's right!!! :P |
| 20:25.01 | _Sam-- | harryvv: people that buy 'unlimited' services like vonage aand broadvoice may use a lot of minutes |
| 20:25.01 | harryvv | file, I guess you wont ever sell 604 pr 778 dids? |
| 20:25.16 | file | harryvv: talk to MikeJ[Laptop] |
| 20:25.29 | MikeJ[Laptop] | harryvv, what ya need? |
| 20:25.36 | harryvv | sam, you mean over the 19 dollars that the voip retail seller that may be chaged by there wholsaler? |
| 20:25.37 | lithi | harryvv: Its exactly like the webhosting business, they give you more then you really need and hope to hell you dont actually use it. |
| 20:25.44 | *** part/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net) |
| 20:26.07 | _Sam-- | harry people that have to pay other wholesalers alot of times have 'softcaps" |
| 20:26.10 | _Sam-- | teliax comes to mind |
| 20:26.15 | harryvv | Lithi, okay. Is there any industry stats for age group demographics ect on who will use more or less minites? |
| 20:26.17 | lithi | thats true |
| 20:26.23 | _Sam-- | they offer an unlimited service...but cap it at 2500 minutes |
| 20:26.26 | MikeJ[Laptop] | harryvv, pr? |
| 20:26.28 | _Sam-- | then start charging you per minute |
| 20:26.37 | harryvv | I see |
| 20:26.41 | *** join/#asterisk mmejiav (n=mmejiav@200.119.46.92) |
| 20:26.45 | file | "unlimited ... up to" |
| 20:26.45 | lithi | harryvv: I am sure vonage would know, but for that kind of research your looking at $$ |
| 20:26.45 | file | :D |
| 20:26.47 | cpm | Vonage works, near as I can tell, by rolling out dodgy technology, hyping the living heck out of it, and having a very streamlined billing/collection system, and relying on the local isp to cover their ass. |
| 20:26.51 | _Sam-- | broadvoice and teliax i really do think are true unlimited though. |
| 20:27.04 | file | _Sam--: they aren't |
| 20:27.14 | _Sam-- | they have stated caps? |
| 20:27.17 | _Sam-- | or its at their discretion? |
| 20:27.20 | lithi | teliax does |
| 20:27.38 | MikeJ[Laptop] | unlimmited, till you use too much.. then we kick you off the unlimited plan |
| 20:27.39 | harryvv | I mean I can easily sell voip service after I find a carrier to hose my asterisk box but worry about somone or more then one going over the 19 dollar limit that alot of voip carriers are selling at. |
| 20:27.44 | file | teliax specifically says it |
| 20:27.51 | file | their residental is 1500 minutes |
| 20:27.54 | file | corporate is 2500 minutes |
| 20:28.00 | _Sam-- | harryvv: in my mind...there are plenty of resellers...and reselling voip service isnt the golden ticket |
| 20:28.15 | harryvv | sam, then what is? |
| 20:28.17 | _Sam-- | setting up and selling asterisk / service / support, in MY opinion has more potential |
| 20:28.28 | harryvv | okay |
| 20:28.33 | _Sam-- | purely opinion |
| 20:28.40 | _Sam-- | but voip minutes are just a commodity |
| 20:28.44 | lithi | I mean if you take a look at your User Aggreement *EVERY* single company will cap you in one way or another. Basicly they say if your abusing the residental service you may be asked to leave or upgrade to business class (even if its just a ton of calls to grandma) |
| 20:28.49 | _Sam-- | and more and more people are selling them...driving prices down |
| 20:28.49 | sevard | I say, do both. |
| 20:28.57 | hypnox | aye, the big telcos will walk all over the service market |
| 20:29.01 | harryvv | specially when you charge 65 and up per hour for a install or support vs 5 cents per min for service. |
| 20:29.52 | cpm | selling voip service sounds to me like how to spend a whole lot of money to make very little money. |
| 20:29.53 | harryvv | Lithi, any examples of these user agreemnts online? |
| 20:29.57 | lithi | _Sam--: You are right BUT if your doing asterisk and also reselling min that means just a little more montly income and makes things easier to setup. |
| 20:30.01 | _Sam-- | not to mention, when you start providing a 24/7/365 service...who wants to babysit servers all your life |
| 20:30.04 | harryvv | cpm, it really is. |
| 20:30.19 | _Sam-- | lithi/: i resell the voip service to my clients who i install |
| 20:30.27 | _Sam-- | but i dont actively try to find people to buy voip service through me |
| 20:30.28 | lithi | harryvv: Just look at any of them but yea one sec ill find one and quote it |
| 20:30.38 | harryvv | sam, I know. thats what the telcom people do. I met a guy who retired from telus. He spent his 20 years in one small building. |
| 20:30.39 | lithi | yea |
| 20:30.47 | rob0 | I still consider such limits on "unlimited" to be fraudulent, regardless of how common they are. |
| 20:30.55 | cpm | Not interested in selling service, more interested in providing bundled services, and billing for them. Different altogether. |
| 20:31.16 | rob0 | <== got kicked off an "unlimited" dialup ISP for too much use |
| 20:31.20 | cpm | rob0, that's because they are fraudulent |
| 20:31.47 | _Sam-- | in the dialup business....there is a distintion between unlimited..and dedicated |
| 20:31.56 | _Sam-- | i had that problem when i owned an ISP |
| 20:32.15 | _Sam-- | yeah you can have unlimited access...but not dedicated...when you are on the same call for 7 days in a row,. that borders dedicated |
| 20:32.17 | harryvv | so you can sell just the asterisk servers at all the job sites and perhaps make more then hosting them and pray your OC or major carrier goes down. |
| 20:32.18 | scardinal | is there a way to see if the phone actually registered with the proxy(*) ? |
| 20:32.32 | cpm | De facto definition, Unlimited, without limit |
| 20:33.26 | _Sam-- | unlimited use...which means you have to use it |
| 20:33.27 | _Sam-- | not let it idle |
| 20:33.27 | _Sam-- | if you are using it 7 days in a row...that is using it.. |
| 20:33.27 | _Sam-- | if you let it idle for 3.5 out of 7...that is not using it |
| 20:33.28 | upsite | tzafrir_laptop it seems that * is not even trying to acces the sound files |
| 20:33.41 | sevard | docelm0: Did you see? :) |
| 20:33.45 | lithi | _Sam--: I know what your saying, but I think the dialup company would still cut you off even if it was real use |
| 20:33.46 | _Sam-- | in the end, at my isp, we ultimately stopped calling it unlimted. and we did switch to a limited service (this was in 1996 or so) |
| 20:34.05 | _Sam-- | but people werent aware of the distinction between dedicated and unlimited |
| 20:34.18 | _Sam-- | at some people something stops being unlimited and starts being dedicated, almost |
| 20:34.26 | _Sam-- | at some POINT i meant |
| 20:34.40 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) |
| 20:34.53 | harryvv | Well, I wonder if there is any point for hosting a voip service and then have clients phones register to it though the net vs just having the pbx at the customers sites. I know one user here that did this but gave TO MUCH of his service away to the customer. Example would be "here, you use this interface to add more extentions or you can add this feature" I mean how many voip consultant do that? |
| 20:34.54 | sevard | use the sed edit :) |
| 20:35.02 | sevard | at some people something stops |
| 20:35.07 | sevard | s/people/point |
| 20:35.08 | sevard | s/people/point/ |
| 20:35.11 | sevard | grr |
| 20:35.16 | sevard | at some people something stops |
| 20:35.18 | sevard | s/people/point/ |
| 20:35.27 | harryvv | sam, did you see your revenue climb at that point? |
| 20:35.33 | sevard | i don't like having to add the / |
| 20:35.52 | _Sam-- | no, but the costs to provide the service did, because the people who were tying up all the lines had to find a new home, which meant i could service more customers on my existing lines |
| 20:36.04 | _Sam-- | the costs declined |
| 20:36.10 | tzafrir_laptop | upsite, if you replace the Playback() with Milliwatt(), do you hear something? |
| 20:36.16 | _Sam-- | we lost a few customers, but the cost savings more than offset them |
| 20:36.21 | harryvv | sam, so the expenses dropped then. |
| 20:36.27 | tzafrir_laptop | (reload for the dialplan change to take effect |
| 20:36.35 | harryvv | okay, you fired some of your customers :) |
| 20:36.36 | _Sam-- | yeah because instead of 1 guy using 1 line and hogging it...7 people could now use the same line |
| 20:36.38 | _Sam-- | in general |
| 20:36.50 | harryvv | the ones that were a drag on your company |
| 20:36.51 | _Sam-- | the average dial ratio was 7 customers for each 1 modem |
| 20:36.57 | _Sam-- | s/dial/dialup |
| 20:37.03 | harryvv | okay |
| 20:37.06 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.9) |
| 20:37.11 | lithi | harryvv: I cant find a really good example for you (really beacause I am to lazy) but they are there trust me. They usualy classify it as abuse of the service and yes its very much like ISPs. |
| 20:37.33 | scardinal | anyone!? |
| 20:37.59 | lithi | Most DSL/Cable providers who offer 'unlimited' service will give you a phone call if they feel you are abusing (or using) their service too much. |
| 20:38.10 | harryvv | Lithi, I see. What I need is basicly a service level agreement that if something should break or a new feature is needed that thay would agree to it. |
| 20:38.12 | lithi | I got one when I used 350GB in a month |
| 20:38.26 | harryvv | how much lithi? |
| 20:38.32 | MikeJ[Laptop] | heh |
| 20:38.47 | harryvv | I mean that would acomidate alot of sip traffic |
| 20:38.52 | MikeJ[Laptop] | do the math on 1 voip call up 24/7 for a month |
| 20:39.03 | MikeJ[Laptop] | say ulaw for the fun of it.. |
| 20:39.16 | lithi | I mean 350gb is like a dedicated 1 megabit |
| 20:39.19 | MikeJ[Laptop] | it;s a lot |
| 20:39.35 | lithi | but I am sure I used it in spikes up over 3 megabits |
| 20:39.41 | harryvv | lithi, I see, so that 1 megabit every hour of the month. |
| 20:39.43 | _Sam-- | that is alot...64k/sec * 60 * 24 * 31 |
| 20:39.50 | lithi | 1 megabit per second |
| 20:39.53 | harryvv | yes |
| 20:39.59 | lithi | is about 320gb in a month |
| 20:40.08 | lithi | (I think) |
| 20:40.10 | MikeJ[Laptop] | that;s kilobits per sec I think yes? |
| 20:40.15 | Qwell[] | 215gb |
| 20:40.15 | _Sam-- | should be |
| 20:40.17 | harryvv | sam, that duplex right? |
| 20:40.17 | Qwell[] | ? |
| 20:40.19 | MikeJ[Laptop] | so /8 |
| 20:40.22 | cpm | here's a really noob question. Realistically, how many gsm call one stuff over an 'internet' t1 without it sucking ? |
| 20:40.30 | Qwell[] | erm, right |
| 20:40.32 | Qwell[] | 27gb |
| 20:40.32 | _Sam-- | gsm: alot |
| 20:40.33 | MikeJ[Laptop] | ? |
| 20:40.35 | harryvv | you mean call quality sucking |
| 20:40.41 | _Sam-- | over a t1...maybe 50 calls easy |
| 20:40.47 | _Sam-- | er maybe a little less |
| 20:40.50 | cpm | harryvv, yes |
| 20:40.50 | harryvv | before the customers complain or hate the service |
| 20:40.53 | lithi | yea a t1 = 23 calls with ulaw right? |
| 20:40.53 | MikeJ[Laptop] | not on ulaw |
| 20:41.03 | _Sam-- | lithi: 24 |
| 20:41.04 | cpm | Okay, with ulaw |
| 20:41.06 | Qwell[] | lithi: no, there is overhead too |
| 20:41.08 | cpm | 24, that's what I thought |
| 20:41.08 | MikeJ[Laptop] | lithi, 24 |
| 20:41.09 | MikeJ[Laptop] | heh |
| 20:41.15 | cpm | So, at the end of the day, what's the point? |
| 20:41.17 | MikeJ[Laptop] | 23 +1 on pri |
| 20:41.18 | lithi | thats why I say 23 (because of overhead) |
| 20:41.20 | harryvv | 24 with ulaw? |
| 20:41.26 | Qwell[] | lithi: Far less |
| 20:41.30 | Qwell[] | 15-18 |
| 20:41.34 | MikeJ[Laptop] | that's pure t1 |
| 20:41.37 | Qwell[] | data |
| 20:41.38 | MikeJ[Laptop] | not data\voip |
| 20:41.41 | lithi | ah |
| 20:41.54 | _Sam-- | i thought the overhead of the ulaw wwas included in the 64k/sec |
| 20:41.58 | Qwell[] | _Sam--: no |
| 20:42.02 | Qwell[] | ~83k/s |
| 20:42.05 | _Sam-- | wow |
| 20:42.07 | _Sam-- | thanks |
| 20:42.17 | _Sam-- | why does it take so much overhead? |
| 20:42.17 | harryvv | I took a telcom class. I learned it was 56 usable for voice the rest was overhead. |
| 20:42.22 | MikeJ[Laptop] | tcp overhaed is not inlcuded |
| 20:42.27 | _Sam-- | if you took the D channel from a PRI and divided it by 23... |
| 20:42.44 | Qwell[] | _Sam--: tcp, plus sip, plus ethernet |
| 20:42.54 | lithi | Plus you prob need some bandwidth for websurfing and such on that t1 |
| 20:43.00 | gaupe | tcp? |
| 20:43.02 | _Sam-- | if you connect to your provider over UDP (iax) that doesnt mean anything for the overhead? |
| 20:43.03 | lithi | QoS for sure |
| 20:43.17 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 20:43.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:43.32 | Qwell[] | _Sam--: iax has overhead too |
| 20:44.07 | _Sam-- | if you use IAX trunking, does that use a single channel for overhead / signalling like a PRI D channel? |
| 20:44.29 | upsite | tzafrir_laptop if i use a mp3player it's working |
| 20:44.51 | tzafrir_laptop | "it" ==? |
| 20:45.05 | upsite | it's playing a sound |
| 20:45.06 | upsite | ;) |
| 20:45.11 | lithi | When a softphone is offline why does asterisk think its busy? It should be unavailable. CLI reports Unable to create channel of type 'IAX2' (cause 3 - No route to destination) or is that n+101 both for busy and offline? How can I determine the result? |
| 20:45.31 | _Sam-- | dialstatus |
| 20:45.31 | lithi | of the Dial* |
| 20:45.34 | _Sam-- | i think |
| 20:45.36 | lithi | ah |
| 20:45.38 | upsite | i guess its some bug in the app_queue |
| 20:45.41 | tzafrir_laptop | use mp3player how? from within asterisk? |
| 20:45.48 | upsite | yes |
| 20:45.55 | upsite | app_mp3player |
| 20:46.06 | harryvv | So anyway, back to my question. Say I went out and started selling a boat load of ATS or phones and charged only 19.95 per month. Thay used the standard asterisk dial plan and can call anywhere in north america. The rate I am charged is 2 cents per min. With the overhead cost of the rented rack space/bandwith I suspect that most customer will not go over 1,995/.02 which would be 99,700 min per month. I dont think a customer would ever g |
| 20:46.11 | _Sam-- | if you use dialstatus (like the macro stdextension example) then callers get sent to right voicemail based on the dial status |
| 20:46.36 | upsite | i'll add some debug lines in app_queue to see if the playback routine is called |
| 20:46.58 | tzafrir_laptop | upsite, and iff you give full path to the sound file in Playback? |
| 20:47.04 | Qwell[] | harryvv: would ever g...? |
| 20:47.10 | lithi | harryvv: a) get a better rate then 2c a min (at least 1.5 or even 1.1 c a min) do unlimited USA48/Canada (I wouldent recommend unlimited world as theres alot of 'exceptions') |
| 20:47.13 | cpm | harryvv, if you used the same call termination services that xxxxxxx uses, you'd drop a lot of calls, and no one woudl be able to break you. :) |
| 20:47.23 | cpm | like a cellphone :) |
| 20:47.37 | lithi | _Sam--: Thanks |
| 20:47.41 | _Sam-- | if you are paying someone else, and you are charging19.99 a month...i think you will go broke. |
| 20:47.45 | _Sam-- | are at least starve |
| 20:47.49 | harryvv | cpm, my box drop the calls or my wholsaler? |
| 20:48.05 | _Sam-- | s/are/or/ |
| 20:48.12 | upsite | tzafrir_laptop the app_playback is working to |
| 20:48.15 | Derkommissar | what should be the optimal qualify setting on sip.conf ? |
| 20:48.19 | upsite | too |
| 20:48.21 | cpm | harryvv, I was being sardonic. Sorry |
| 20:48.24 | harryvv | sam, then how do some of these retail voip service providers that charge this and alot less stay in bussiness? |
| 20:48.33 | _Sam-- | they get the minutes for a lot less than you can |
| 20:48.34 | synthetiq | derk, crappy networks, 5000 |
| 20:48.39 | tzafrir_laptop | upsite, so what exactly *doesn't* work? |
| 20:48.44 | _Sam-- | they are maybe paying .8c/min |
| 20:48.47 | harryvv | sam, less then a penny per min? |
| 20:49.03 | _Sam-- | or they have PRIs that are flat rate |
| 20:49.08 | upsite | the playback from the voicepromts in the queue |
| 20:49.26 | upsite | like "your are on the first postition" |
| 20:49.27 | upsite | -... |
| 20:49.30 | tuxinator_linux | _Sam--, flat rate PRI exist? |
| 20:49.31 | harryvv | sam, flat rate bandwith limites? ie, unlimited cost ? |
| 20:49.34 | synthetiq | no such thing as a flat rate pri |
| 20:49.39 | cpm | I mean, I'd love to get on voipjet, their backbone looks good, but their eula is just too dodgy for me. |
| 20:49.48 | _Sam-- | tuxinator_linux: they do here...i had a flat rate unlimited PRI with unlimited LD in the us48 |
| 20:49.49 | harryvv | unlimited bandwith usage at one fixed rate? |
| 20:49.56 | lithi | I like using Teliax as an example $24.99/mth USD (residental) for 1500 min USA48/Canada. At 1.5c a min thats $22.50 |
| 20:49.59 | cpm | synthetiq, no such thing? |
| 20:50.05 | synthetiq | well |
| 20:50.07 | _Sam-- | i pad 750/month |
| 20:50.09 | synthetiq | hold on |
| 20:50.10 | tuxinator_linux | _Sam--, were do I sign up? |
| 20:50.12 | synthetiq | i could be wrong |
| 20:50.17 | cpm | I'm pretty sure my day job PRIs are flat rate. |
| 20:50.18 | _Sam-- | tuxinator_linux: www.cavtel.com |
| 20:50.26 | Qwell[] | lithi: That's horrible |
| 20:50.27 | cpm | Of course, we don't come even close to maxing them, ever. |
| 20:50.28 | harryvv | I suspect that with these rates, some voip services have gone belly up. |
| 20:50.43 | tzafrir_laptop | upsite, can you play that specific sound with Playback? |
| 20:50.57 | synthetiq | yea i think you would be paying the upper end for flat rate, you can only use a max of 23 channels |
| 20:51.01 | tzafrir_laptop | (not using full path) |
| 20:51.23 | lithi | Qwell[]: Looks like if you want to make a profit you need less then 1c a min. I mean if people actualy used the 1500 min. Which they dont (but best to be safe then sorry) |
| 20:51.27 | synthetiq | so 60*24*30* typical rate |
| 20:51.51 | _Sam-- | i can scan my service agreement if you really want |
| 20:52.04 | _Sam-- | its 750 / month flat rate 23/b + B US48 LD |
| 20:52.10 | synthetiq | at .02 cents a minite it would be 864 |
| 20:52.10 | _Sam-- | 23B +D |
| 20:52.14 | harryvv | So if eaking out a meger living in selling retail voip service is not a good idea, the only other way of course as somone has said is service/support/consulting of installing asterisk boxes at customers sites and then thay pay the wholsaler voip rates. |
| 20:52.35 | _Sam-- | your number doesnt take into account what the cost of the PRI loop would cost |
| 20:52.52 | _Sam-- | or T1, etc |
| 20:52.53 | lithi | BTW that math doesent even cover server/bandwidth costs. |
| 20:52.54 | cpm | harryvv, do you have the ability to lie and hype like skye and vonage? |
| 20:53.01 | harryvv | hehehe |
| 20:53.09 | *** join/#asterisk detatch (i=detent@dhcp-100.fresno-dc2.brandxnet.com) |
| 20:53.34 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 20:53.36 | harryvv | cpm, sounds like its better just to get a regular job making 12 dollars and up |
| 20:53.41 | cpm | can you fit under a door when it's closed? |
| 20:53.55 | lithi | I mean if your looking at making $2.50 per month per customer (before server costs) your screwed. |
| 20:53.56 | cpm | like most snakes? |
| 20:54.01 | detatch | can someone help me with new voicemail email notification? |
| 20:54.15 | upsite | tzafrir_laptop yes |
| 20:54.26 | znoG | for a production environment, should one use Asterisk 1.2.4 or CVS HEAD? |
| 20:54.37 | upsite | all 'explecit' sound playbacks ar working |
| 20:54.41 | harryvv | I guess at one time, it was a decent way to making a living if your a startup..I guess in this day and age not anymore. |
| 20:54.43 | _Sam-- | for the record, with the calculations regarind the flat rate PRI....i switched to a T1 / remote gateway...and my costs are less. |
| 20:55.11 | upsite | only the playback from within the queue is failing |
| 20:55.16 | Derkommissar | what is this mesage suposed to men ? sched.c:219 sched_settime: Request to schedule in the past?!?! |
| 20:55.28 | detatch | i can send mail from the machine ( running sendmail ) but when i leave a voicemail the notification isnt sent |
| 20:55.32 | lithi | _Sam--: Still depends on you area for the PRI rates but yes I would assume data would be a cheaper in most cases. |
| 20:55.40 | tzafrir_laptop | upsite, one filesystem-related longshot: no free disk space? |
| 20:55.55 | upsite | nope |
| 20:55.56 | harryvv | So mabey I should just sell the boxes do the support and thats it. Get alot more customers this way and chage for driving time to those sites..if needed. |
| 20:56.01 | Katty | hmm. |
| 20:56.06 | upsite | everything is 'green' |
| 20:56.14 | upsite | not run of inodes |
| 20:56.15 | upsite | nothing |
| 20:56.17 | cpm | harryvv, what other services can you offer? If you are a small isp, you can offer this to your customers |
| 20:56.25 | tuxinator_linux | Katty, morning |
| 20:56.27 | tzafrir_laptop | upsite, again: logs, logs, logs. Enable "full" in logger.conf, set verbose 5 and debug 5 and look for hints |
| 20:56.29 | harryvv | I am not a small isp |
| 20:56.40 | _Sam-- | harryvv : there is nothing wrong with wanting to resell the service...but you have to really work the numbers carefully....i did. if you could get 1000 customers....i still think you would not make a lot of money |
| 20:56.47 | harryvv | CPM my experaince in the past has been end user solutions. |
| 20:56.58 | _Sam-- | the overhead of support, hardware, etc...it doesnt scale so well initially |
| 20:57.12 | upsite | tzafrir_laptop im allready diging into it ;) |
| 20:57.17 | cpm | well, you could sell consulting on how your clients can do this themselves |
| 20:57.18 | _Sam-- | again this is just my opinion...not any facts. |
| 20:57.22 | harryvv | Sam, vs other ways of generating better returns in other areas of IT |
| 20:57.24 | lithi | harryvv: I think you may be better off just finding a company that will resell directly (ie they do all the work) but under your brand name. |
| 20:57.26 | Katty | hiya, tuxinator_linux |
| 20:57.28 | synthetiq | support? let the fish flop on dry land! |
| 20:57.44 | cpm | there's lithi's concept too, that's a fair model |
| 20:58.07 | harryvv | Lithi, I would be it. I dont know of any other company that is buying asterisk..its usually cisco. |
| 20:58.07 | znoG | so is CVS HEAD way old compared to 1.2.4? |
| 20:58.10 | _Sam-- | there are no good companies to resell...unless you can buy level3 |
| 20:58.13 | harryvv | buying asterisk services that is. |
| 20:58.17 | _Sam-- | and you need to commit to like 3 million minutes |
| 20:58.21 | synthetiq | yes znog |
| 20:58.32 | _Sam-- | if you just resell other tier 2 and lower companies...i dont know |
| 20:58.56 | lithi | If you find a good tier 2 company, thats really not that bad. |
| 20:59.08 | lithi | One that buys from level3 for example |
| 20:59.09 | brookshire | znoG: if you pull from cvs it should be what is currently in svn |
| 20:59.15 | _Sam-- | yeah but a level 2 will be paying a level 1 already...which cuts your profit margins |
| 20:59.30 | harryvv | Lithi, basicly I dont want to waste my time on some kind of plan that generates so little in revenue its basicly moot point. |
| 20:59.34 | synthetiq | trunk 74?? is in latest cvs |
| 20:59.48 | brookshire | hmm.. |
| 20:59.57 | brookshire | maybe it's broke, lol |
| 20:59.59 | harryvv | I suspect this has been brought up before. |
| 21:00.12 | _Sam-- | i dont remember this specific discussion really |
| 21:00.23 | _Sam-- | notice how all the poeple who do SELL the services arent commmenting :) |
| 21:00.29 | lithi | harryvv: Then at least find a good company (that will give you a commission) and recommend them. |
| 21:00.41 | _Sam-- | if you could find out the avg minutes used per customer per month...you could have an even better idea for your business model |
| 21:01.02 | _Sam-- | in terms of it will work or it wont work |
| 21:01.05 | lithi | harryvv: to your end customers |
| 21:01.09 | harryvv | Sam, basicly rent them a asterisk box and look at there cdr |
| 21:01.17 | harryvv | for the month |
| 21:01.26 | _Sam-- | nah...that wont work for residential users |
| 21:01.31 | harryvv | no for bussiness |
| 21:01.34 | _Sam-- | but you could probably find some industry wide numbers |
| 21:01.40 | cpm | residential customers |
| 21:01.46 | harryvv | hehe |
| 21:01.51 | harryvv | yea, thay are cheap cpm |
| 21:01.55 | Assid | make[1]: *** [pbx_dundi.o] Error 1 |
| 21:01.55 | Assid | make[1]: Leaving directory `/usr/src/asterisk-1.2.4/pbx' |
| 21:02.01 | Assid | i cant seem to install this correctly |
| 21:02.31 | brookshire | assid: what distro? |
| 21:02.43 | Assid | debian |
| 21:02.44 | Assid | wait |
| 21:02.47 | Assid | i think i see why |
| 21:02.49 | *** join/#asterisk meshuga (i=meshuga@c-67-183-24-243.hsd1.wa.comcast.net) |
| 21:02.51 | Assid | zlib1g missing |
| 21:02.52 | cpm | <PROTECTED> |
| 21:03.00 | brookshire | yeah.. that helps |
| 21:03.00 | meshuga | whats a good win32 SIP client that supports transferring? |
| 21:03.05 | _Sam-- | its good ot have a mix of biz and residential |
| 21:03.09 | _Sam-- | because biz uses the service all day |
| 21:03.12 | _Sam-- | and res uses it all night |
| 21:03.16 | harryvv | true |
| 21:03.19 | _Sam-- | that way can you really use your resources |
| 21:04.08 | cpm | Yeah, that's the best way to go. I want folks to use my resources, they are paid for, they should be used. But I'd rather give it away than support residental customers, the cable company is welcome to them. |
| 21:04.19 | harryvv | Just called Ariel. not home has he been here lately? |
| 21:04.50 | harryvv | who knows Ariel really well? |
| 21:05.02 | synthetiq | from the little mermaid? |
| 21:05.06 | Renacor | whats a company that would sell 1800 T lines? |
| 21:05.07 | harryvv | hehe |
| 21:05.14 | Renacor | or lease I should say |
| 21:05.15 | harryvv | 1800 numbers? |
| 21:05.21 | cpm | businesses are similar tight, as they have to be, if they want to viable, but at least they understand cost+overhead+profit, and want good service, and usually will pay for it, (if it's good) |
| 21:05.25 | Assid | i guess i need to install pgsql before i load up * |
| 21:10.10 | badboyz | can anyone reocmmend a voip provider that they are very satisfied with? |
| 21:10.23 | Dandan | BV i (surprise!) Vonage :) |
| 21:10.30 | austinnichols101 | inbound / outbound or both |
| 21:10.40 | austinnichols101 | domestic or international or both |
| 21:10.48 | Dandan | overall |
| 21:11.32 | austinnichols101 | badboyz: using voxee for outbound only |
| 21:11.44 | austinnichols101 | badboyz: using telasip for home in/out |
| 21:12.05 | Dandan | i am looking for someone experienced with VoiceTronix boards! |
| 21:12.52 | harryvv | Can anyone think of any company that ports numbers for the regional BC canada market? |
| 21:13.21 | Dandan | austinnichols101: voxee: http://blogs.zdnet.com/ip-telephony/?p=551 |
| 21:13.31 | badboyz | im using telasip as well, and having issues with not being able to hear callers from time to time |
| 21:13.47 | badboyz | Gene doesnt want to speak with me anymore it appears, hes no longer returning emails |
| 21:13.59 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 21:14.24 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 21:14.42 | badboyz | im looking for in and outbound, domestic only |
| 21:14.57 | badboyz | and i need to be able to have 5 concurrent calls going on ( in or outbound ) @ same time |
| 21:15.07 | *** join/#asterisk exonic (n=exonic@209.172.11.54) |
| 21:15.24 | exonic | Hey there folks, Anyone ever implement line roll over via asterisk? |
| 21:15.41 | *** join/#asterisk HeyEveryBody (n=Aces1Up@ip70-189-157-31.lv.lv.cox.net) |
| 21:15.55 | exonic | I'm planning to devise a method first, but wanted to check w/ the community. |
| 21:16.19 | harryvv | exonic, so if your voip fails, it will roll over to local pstn carrier |
| 21:16.21 | HeyEveryBody | hey all is anyone here familiar with building auto dialers? I would like to talk with them please. |
| 21:16.48 | exonic | harryvv, more so, if line 1 is busy, try line 2, if line 2 is busy try line 3, etc. |
| 21:17.02 | harryvv | exonic for pstn? |
| 21:17.05 | HeyEveryBody | also i have a client that needs a 96-line auto-dialer that can run over voip, is there a such thing as a softphone autodialer? that has many softphones dialing out to voip lines? |
| 21:17.15 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
| 21:17.18 | Flauto | exonic, it should be pretty easy |
| 21:17.19 | exonic | harryvv, PSTN to SIP |
| 21:17.32 | harryvv | exonic I got it. yea its in the dial plan. |
| 21:18.25 | HeyEveryBody | harryvv are you getting my pm's? |
| 21:18.48 | exonic | I would like to do it in a organized manner, I'm also using realtime, perhaps a simple dialplan example? I'm quite fluent with dialplans but not exactly sure of how to check the BUSY status. |
| 21:19.13 | austinnichols101 | Dandan: nice link - thanks |
| 21:20.10 | Luke-Jr | exonic: IIRC, use Dial and make a priority at +100 for If-Busy |
| 21:21.46 | exonic | Luke-Jr, harryvv the problem with that is in a large group of calls, you have to wait for asterisk to attempt a dial to every SIP client only to get a 40X BUSY message, asterisk is not capable of knowing when a SIP peer is busy or not? |
| 21:22.05 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 21:22.48 | Luke-Jr | exonic: not without dialing, I think... How is Asterisk supposed to know? The peer could be in a call with a completely different server |
| 21:23.12 | exonic | ChanIsAvail() comes to mind |
| 21:23.23 | exonic | Luke-Jr, good poin |
| 21:23.42 | *** join/#asterisk dahunter3 (n=dahunter@65.77.168.255) |
| 21:24.07 | exonic | ok, I guess the only thing to attempt is to setup a test environ |
| 21:24.24 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 21:24.48 | badboyz | has anyone ever had the problem, that people calling into an asterisk system, can hear you fine -- but you cannot hear them? whats the typical solution? (this problem is intermittant -- about 50% of hte time) |
| 21:25.13 | *** join/#asterisk FliesLikeABrick (n=Ryan@about/rpi/rawdor) |
| 21:25.19 | exonic | badboyz, I was w/ ya until the intermittant thing. using STUN? |
| 21:25.28 | FliesLikeABrick | jalsot let me know when you're around |
| 21:25.28 | znoG | i wonder if anyone has made an "announce" script. Like one that calls all the registered extensions and tells them "the PBX will be restarted please don't make any calls for the next 2-3 minutes" |
| 21:25.30 | docelm0 | sevard, hay almost done with the weather script |
| 21:25.33 | znoG | it wouldn't be hard to do |
| 21:25.34 | Qwell | hrm |
| 21:25.36 | exonic | badboyz, what's the setep like? Is it simple PSTN => SIP ? |
| 21:25.41 | docelm0 | well its primative.. but does the job for now.. |
| 21:25.47 | docelm0 | something quick and simple |
| 21:25.57 | dahunter3 | I have this weird telecom provider that needs to have callerid sent inband as opposed to out of band, reducing the sampling from 64k to 56k. Can we do this in asterisk? |
| 21:26.19 | badboyz | exonic: caller(pstn) => us (voip / sip ) |
| 21:26.21 | exonic | znoG, Easy to do by placying files into the outgoing call queue or using the manager ORIGINATE command |
| 21:26.21 | Qwell[] | dahunter3: 56k ulaw? |
| 21:26.38 | badboyz | exonic: whats STUN? |
| 21:26.45 | Qwell[] | ~stun |
| 21:26.47 | jbot | from memory, stun is that feeling you get when you realise your SIP call actually got through! |
| 21:26.55 | znoG | exonic: yeah, exactly. not hard |
| 21:26.57 | cpm | heh |
| 21:27.00 | dahunter3 | qwell: I'm not sure what you mean. |
| 21:27.01 | Qwell[] | nice |
| 21:27.10 | Qwell[] | dahunter3: Get a provider that isn't junk |
| 21:27.31 | dahunter3 | qwell: Yeahh, I hear you... but can I do anything currently? |
| 21:27.33 | exonic | <PROTECTED> |
| 21:27.59 | Qwell[] | umm |
| 21:28.09 | Qwell[] | somebody explain something to me |
| 21:28.15 | Qwell[] | for the last 3 days...my ISP has screwed me at exactly 1:25pm |
| 21:28.31 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 21:28.39 | tainted_ | i successfully compiled asterisk but it doesn't load on 'asterisk -vvvvgc'.. no errors either - any ideas? |
| 21:28.40 | cpm | someone firing up bittorrent at the same time every day |
| 21:28.43 | _Sam-- | you didnt know? you are supposed to not be online at that hour...lunch break |
| 21:29.15 | RoyK | hi |
| 21:29.16 | dahunter3 | So, will asterisk get the callerid from an inband transmission as opposed to a d-channel? |
| 21:29.48 | exonic | tainted_, I doubt that. |
| 21:30.01 | tainted_ | exonic what do u mean |
| 21:30.14 | Qwell[] | by screwed, I mean the modem dies |
| 21:30.19 | Qwell[] | unsyncs |
| 21:30.22 | cpm | modem? what's that? |
| 21:30.30 | Dandan | ~jbot STUN i's a NAT traversal protocol |
| 21:30.35 | Dandan | ~jbot STUN is a NAT traversal protocol |
| 21:30.36 | jbot | ...but stun is already something else... |
| 21:30.41 | exonic | tainted_, taht produces a ton of output, w/ no errors? nopaste the full screen |
| 21:30.44 | Dandan | ~jbot |
| 21:30.45 | jbot | i guess jbot is only marginally useful at best, or a silly little bugger |
| 21:30.46 | _Sam-- | probably using too much of your unlimited service! |
| 21:30.48 | Dandan | ~stun |
| 21:30.49 | jbot | stun is probably that feeling you get when you realise your SIP call actually got through! |
| 21:30.59 | cpm | Qwell[], that might be their way of saying they love you |
| 21:31.10 | _Sam-- | maybe the have 24hour connection limit |
| 21:31.11 | exonic | jbot needs to learn what STUN really is :) |
| 21:31.13 | jbot | okay, exonic |
| 21:31.13 | _Sam-- | so they disconnect you after 24 |
| 21:31.14 | Dandan | ~jbot stun is NOT that feeling you get when you realise your SIP call actually got through! |
| 21:31.15 | jbot | ...but stun is already something else... |
| 21:31.17 | _Sam-- | which is the same time, each day |
| 21:31.27 | Dandan | ~jbot stun NOT is that feeling you get when you realise your SIP call actually got through! |
| 21:31.28 | jbot | okay, Dandan |
| 21:31.28 | Qwell[] | jbot stun is also Simple Traversal of UDP over NATs |
| 21:31.30 | jbot | okay, Qwell[] |
| 21:31.30 | tainted_ | http://pastebin.ca/40608 |
| 21:31.35 | Qwell[] | ugh |
| 21:31.36 | znoG | ~stun |
| 21:31.37 | jbot | i guess stun is that feeling you get when you realise your SIP call actually got through!. Simple Traversal of UDP over NATs |
| 21:31.44 | Qwell[] | better |
| 21:31.46 | Dandan | lol! |
| 21:31.50 | Dandan | confusing jbot :) |
| 21:32.00 | exonic | heh, wow, IRC bots |
| 21:32.02 | Qwell[] | jbot forget stun not |
| 21:32.02 | jbot | Qwell[]: i forgot stun not |
| 21:32.06 | cpm | ya'll leave jbot alone. |
| 21:32.18 | Dandan | lol :) |
| 21:32.22 | znoG | ~stun |
| 21:32.23 | jbot | methinks stun is that feeling you get when you realise your SIP call actually got through!. Simple Traversal of UDP over NATs |
| 21:32.24 | Dandan | bbl laterz :) |
| 21:32.27 | Dandan | got a date :) |
| 21:32.29 | znoG | nope, didn't forget it |
| 21:32.36 | badboyz | whats the ideal codec to be using w/ asterisk? (free one) ? |
| 21:32.37 | Dandan | ...with my own wife :) |
| 21:32.37 | Qwell[] | he forgot what I asked him to |
| 21:32.41 | znoG | oh, right |
| 21:32.47 | tainted_ | exonic http://pastebin.ca/40608 |
| 21:32.49 | znoG | ~best |
| 21:32.51 | jbot | best for what? please define what you mean by "best" Gloria Gaynor! Tina Turner! Aretha Franklin! Men without Hats! Women without Hats! |
| 21:32.51 | Dandan | [d] |
| 21:33.13 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 21:33.31 | tainted_ | jbot best is also Flock of Seagulls! |
| 21:33.32 | jbot | tainted_: okay |
| 21:33.46 | lo_tech | ~best |
| 21:33.47 | jbot | best for what? please define what you mean by "best" Gloria Gaynor! Tina Turner! Aretha Franklin! Men without Hats! Women without Hats! Flock of Seagulls! |
| 21:33.49 | exonic | tainted_ wow, sorry I doubted, perhaps an strace is in order? |
| 21:34.11 | znoG | jbot best is also Women without clothes! |
| 21:34.13 | jbot | znoG: okay |
| 21:34.21 | Damin | Alright.. |
| 21:34.28 | Dr-Linux | jbot: i hate you |
| 21:34.29 | jbot | You hate you? |
| 21:34.29 | Dr-Linux | :S |
| 21:34.53 | *** join/#asterisk pointer (i=pointer@aj.catt.com) |
| 21:35.25 | exonic | trouble in #asterisk when people talk to a bot more than other real people |
| 21:35.43 | Luke-Jr | ~best |
| 21:35.44 | jbot | best for what? please define what you mean by "best" Gloria Gaynor! Tina Turner! Aretha Franklin! Men without Hats! Women without Hats! Flock of Seagulls!, or fvwm! Women without clothes! |
| 21:35.45 | lo_tech | ~exonic |
| 21:36.08 | pointer | what's the best option for ISDN support in asterisk 1.2.x? There seem to be 3-4 options |
| 21:36.18 | Qwell[] | pointer: Avoiding it altogether :) |
| 21:36.19 | tainted_ | exonic should i just reinstall? |
| 21:36.20 | lo_tech | libpri for me |
| 21:36.30 | pointer | Qwell[]: that bad? |
| 21:36.37 | Qwell[] | no |
| 21:36.37 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 21:36.44 | exonic | tainted_, is this an upgrade? |
| 21:36.47 | tainted_ | no |
| 21:37.01 | [TK]D-Fender | tainted- : Just do a NOLOAD in modules.conf for chan_modem.so |
| 21:37.10 | pointer | Qwell[]: I've given up on the digium cards, so I was going for a digital circuit |
| 21:37.13 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@d83-179-160-178.cust.tele2.fr) |
| 21:37.13 | dahunter3 | qwell: So is 58k ulaw what I need to solve my problem? |
| 21:37.22 | exonic | pointer, care to enlighten? |
| 21:37.26 | Qwell[] | dahunter3: You need to ask your question again, with full details |
| 21:37.27 | lo_tech | Careful... Qwell is moaning about hid dial-up ISP... might want to consider the source :P |
| 21:37.28 | pointer | Qwell[]: if ISDN doesn't work, we're just going to give our customer a PRI and buy a sangoma |
| 21:37.34 | exonic | pointer, i've been running 4 TE410P digiums for a while now |
| 21:37.34 | lo_tech | s/hid/his |
| 21:37.42 | Qwell[] | boo sangoma |
| 21:37.55 | Qwell[] | lo_tech: Who said I had a dial-up ISP? |
| 21:37.59 | pointer | exonic: I'm not knocking the 4 port cards...I have 2 of those...they seem to work |
| 21:38.12 | exonic | pointer, the TDM cards? |
| 21:38.24 | tainted_ | [TK]D-Fender exonic you know what it was? i didn't 'make samples' so w/ no conf files it just chocked w/o showing any errors |
| 21:38.25 | lo_tech | Qwell: 'modem dies' '1:25PM' |
| 21:38.35 | meshuga | anyone know of any SIP softphone clients for win32 that support call transfer? |
| 21:38.36 | Qwell[] | lo_tech: What, your DSL/Cable doesn't have a mdoem? |
| 21:38.36 | tainted_ | that might be a bug! |
| 21:38.55 | lo_tech | sry, FastEther connection here |
| 21:39.01 | pointer | exonic: the stuff I've tried that didn't work: 3 TDM400Ps, 1 single port T1 card (forget model), and a TDM24xx w/echo can |
| 21:39.08 | Qwell[] | Which connects to what? ;) |
| 21:39.15 | exonic | tainted_, well good to know |
| 21:39.15 | pointer | exonic: by didn't work, I mean "had bad echo" |
| 21:39.20 | exonic | pointer, aye |
| 21:39.25 | lo_tech | Quell: ATM mux :P |
| 21:39.37 | pointer | exonic: I've wasted way to much time and money on the digium hardware |
| 21:40.04 | dahunter3 | Okay, I have what the telecom refers to as a supertrunk, which is basically just a bunch of B channels with no D channel. They are set to do E&M + wink. I am using a digium 1 port T1 card. Asterisk 1.2.2 + patch for sound. Each phoneline (12 total) is sampling at 64k right now ... to get callerid transferred in, according to the telco, I need to lower the sampling to 56k and catch "inband signalling" related to the |
| 21:40.04 | exonic | pointer, the b!tch of digium is supported hardware. I have countless customers who buy these $3k servers, only to find out the board/bios sucks my ass. It' hard to deal w/ 'em |
| 21:40.06 | Qwell[] | Poincare: Don't expect sangomas to "just work" |
| 21:40.18 | _Sam-- | pointer: is your digium hardware for sale? |
| 21:40.19 | Poincare | Qwell[]: what? |
| 21:40.30 | dahunter3 | callerid |
| 21:40.32 | Qwell[] | You think it'll just get rid of the echo magically? |
| 21:40.33 | pointer | _Sam--: no, we'll probably put it on the wall or something ;) |
| 21:41.00 | _Sam-- | a tdm2400 w/EC is an expensive wall sculpture |
| 21:41.01 | dahunter3 | The question is can I do that in asterisk and get the callerid information passed in? |
| 21:41.19 | pointer | _Sam--: yeah, I think we paid just over $1k for that one |
| 21:41.36 | pointer | _Sam--: maybe they'll fix whatever the problem is and it'll be useful for me |
| 21:41.47 | [TK]D-Fender | Qwell[] : Mine got rid of echo technologically :) A104d = 100% echo-free. |
| 21:42.04 | pointer | _Sam--: I'm just not investing any more time into it until I see something exciting in the changelogs or something along those lines |
| 21:42.19 | pointer | [TK]D-Fender: that's the route we're going |
| 21:42.25 | exonic | pointer, I had trouble with echo running on slow systems |
| 21:42.28 | _Sam-- | its odd that it works for so many people but not yourself....what do mean when you say "it didnt work' |
| 21:42.31 | malverian[work] | How do you guys handle recording channels? |
| 21:42.36 | pointer | _Sam--: echo |
| 21:42.39 | _Sam-- | i see |
| 21:42.44 | pointer | _Sam--: we even had digium work on it |
| 21:42.45 | Luke-Jr | malverian[work]: "Monitor"? |
| 21:42.46 | Qwell[] | echo is easy :D |
| 21:42.47 | malverian[work] | Do you use ChanSpy and Monitor? |
| 21:43.00 | malverian[work] | Luke-Jr, Specifically, I want to be able to start recording a specific SIP extension. |
| 21:43.04 | malverian[work] | Luke-Jr, On demand. |
| 21:43.04 | pointer | we're using a vegastream vega50...it works really well |
| 21:43.16 | Luke-Jr | malverian[work]: oh. I just record all calls |
| 21:43.47 | _Sam-- | pointer you have a PRI or POTs? |
| 21:44.00 | _Sam-- | cause you mentioned both types of hardware |
| 21:44.05 | lo_tech | malverian[work]: if you're 1.2+, MixMonitor is better imo... i use it for all my recordings |
| 21:44.08 | _Sam-- | "T1 card and TDM2400" |
| 21:44.09 | pointer | _Sam--: we thought it may be the MB/resource conflicts, but we have tried 4 cards in 6 separate machines |
| 21:44.22 | _Sam-- | what type of connection to the phone company |
| 21:44.58 | pointer | _Sam--: we have several installations.... 2 T1s into a quad, 1 t1 into a single port...2 sites where we used to use TDM400ps....and one where we still do |
| 21:45.10 | pointer | we replaced the TDM400ps with the vega50 at 2 sites |
| 21:46.21 | lo_tech | dahunter3: dont know the exact setting, but it'll most likely be in the zapata.conf if you're using Wildcard's |
| 21:47.18 | dahunter3 | lo_tech: Okay, checking now |
| 21:47.54 | pointer | no ISDN vendor/product suggestions? |
| 21:48.24 | _Sam-- | i am no expert at anything...but it seems EC is a black art |
| 21:48.29 | *** join/#asterisk rene- (n=rene@dsl-201-128-115-222.prod-infinitum.com.mx) |
| 21:48.51 | pointer | _Sam--: vegastream seems to have it mostly nailed...and from what I hear (no pun intended) sangoma does too |
| 21:49.09 | _Sam-- | yeah i dont have any experience..you are a lot more well versed already than i am. |
| 21:49.16 | rene- | hi, i want to compile zaptel in a virtual linux server, im getting a linux/ioctl.h no such file or directory does that means i need kernel headers/sources? |
| 21:49.28 | *** join/#asterisk WasPhantom (n=neil@203.86.192.98) |
| 21:49.31 | _Sam-- | but [TK]D-Fender is has said the sangoma EC is working well for him (maybe he works there?) :) |
| 21:49.31 | rene- | i meant no such file or directory ERROR |
| 21:50.05 | pointer | _Sam--: I've already blown nearly $5k on digium hardware and support, I'm ready to move on...and sangoma seems to be what many people like |
| 21:50.10 | Qwell[] | rene-: Yes, you need kernel sources |
| 21:50.24 | pointer | _Sam--: I spoke with some of their engineers at a convention and was pretty impressed |
| 21:50.34 | _Sam-- | there are some digium folks here, i think |
| 21:50.45 | [TK]D-Fender | _Sam-- : No I just went through 2 TE405P's at work, and a TDM400 at home. |
| 21:51.23 | _Sam-- | someone here works for/at sangoma..just forget who :) |
| 21:51.42 | pointer | _Sam--: read the guarantee on sangoma's website |
| 21:51.46 | [TK]D-Fender | Digium cards work just fine for some, not so fine for others for myriad reasons. If it works for you GREAT. If not it shouldn't hurt to try something different. |
| 21:51.59 | pointer | _Sam--: that's a company that stands behind there product if I've ever seen one |
| 21:52.04 | dahunter3 | The closest thing I see in zapata.conf is: outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT |
| 21:52.07 | dahunter3 | ; inband: Signal Busy/Congestion using in-band tones |
| 21:52.10 | dahunter3 | ; |
| 21:52.12 | dahunter3 | ; priindication = outofband |
| 21:52.27 | pointer | _Sam--: and vegastream is probably going to be dispatching an engineer from out of state to our main site to help us resolve an issue |
| 21:52.32 | dahunter3 | That seems more suited for busy signaling etc.... does it do callerid too? |
| 21:52.33 | cpm | yeah, but it's so easy to like the folks at digium |
| 21:53.09 | pointer | cpm: most of the time, yes |
| 21:53.11 | cpm | they remind me of the smarter younger brother I never had. |
| 21:53.35 | cpm | that would say smart things, and you'd punch them for it. |
| 21:55.16 | pointer | cpm: I just want a hybrid voip/tdm pbx that works... the voip part of asterisk is functional, but the zaptel stuff seems to have problems |
| 21:55.37 | De_Mon | tdm? |
| 21:55.38 | pointer | cpm: we've tried other similar PBXs...(ie. 3com NBX)....they suck worse |
| 21:55.54 | pointer | De_Mon: s/tdm/PSTN/ |
| 21:56.30 | De_Mon | it's beyond me how do you misstyped pstn that badly |
| 21:56.33 | *** join/#asterisk Bakermd (n=bakermd@exchange.i2telecom.com) |
| 21:56.36 | De_Mon | s/ do// |
| 21:56.46 | pointer | heh |
| 21:57.19 | De_Mon | I wonder if I'll ever get use to that |
| 21:57.33 | lo_tech | dahunter: signalling=sf_XXX ? |
| 21:57.37 | Bakermd | All: I have an upgrade question.. my predecessor installed our * server, and I want to upgrade it to the current release. This is a production server, so I cannot afford much downtime at all. The current version is Asterisk CVS-NHEAD-05/18/05-04:13:59 and I am considering upgrading to the latest. Any thoughts / suggestions? |
| 21:57.49 | pointer | [TK]D-Fender: I completely agree with you there...move on and try something else |
| 21:57.55 | Qwell[] | Bakermd: Read the release notes for 1.2.4 |
| 21:58.07 | Qwell[] | Bakermd: Coming from cvs head, you won't have too many issues |
| 21:58.12 | Bakermd | Right, I saw the issues listed there.. |
| 21:58.13 | Bakermd | Okay |
| 21:58.26 | cpm | pointer, I think you are getting close to the heart of the matter. Once upon a time there was rotary and crossbar, and if the lines were up,it worked, then we had 5ess, and everything went |
| 21:58.26 | Bakermd | I have never installed a package from CVS head so was curious |
| 21:58.42 | cpm | to shit and became affordable. Now it's really cheap, and everything sucks |
| 21:58.48 | Qwell[] | Bakermd: just clear out the modules dir, and it'll work fine |
| 21:59.03 | [TK]D-Fender | pointer : I've got an occasional glitch on mine I haven't give any effort to debug or correct where the PRI goes down every few weeks (to which I can't assign blame yet). Aside from that its been a great experience for me. |
| 21:59.04 | pointer | cpm: *nod* we have a dms100....it just works |
| 21:59.04 | Bakermd | Thanks. How should I install? CVS, or source, etc? |
| 21:59.16 | Qwell[] | Bakermd: get the 1.2.4 source...that's your best bet |
| 21:59.30 | *** join/#asterisk KranZ (n=user@imail.bestline.net) |
| 21:59.30 | Bakermd | Okay - Thanks |
| 21:59.57 | pointer | cpm: we're testing a class 5 softswitch now...and from what i've seen, I'm not impressed |
| 22:00.15 | pointer | [TK]D-Fender: that's with the sangoma card? |
| 22:00.48 | [TK]D-Fender | pointer : yup. Then again my time with the others was so rough I couldnt tell if it'd have happened with them too :) |
| 22:00.56 | KranZ | anyone done a custom dir-intro.gsm for Directory()? |
| 22:01.01 | Qwell[] | it'd have? |
| 22:01.06 | Qwell[] | it'd've |
| 22:01.08 | Qwell[] | :P |
| 22:01.11 | pointer | [TK]D-Fender: hrm...what version of zaptel/* are you running? |
| 22:01.15 | [TK]D-Fender | pointer : And I'm not on the most current "stable" release (still using the first stable version since its release) |
| 22:01.26 | pointer | [TK]D-Fender: ah, ok |
| 22:01.31 | KranZ | fender, having pri issues? |
| 22:01.34 | oogle | anyone know why AGI would be returning a blank DIALSTATUS after a dial? |
| 22:01.34 | [TK]D-Fender | pointer : Oh yeah... and 1.0.9 CVS as of OCT somthing-or-other |
| 22:01.58 | [TK]D-Fender | I'm overdue in upgrading but its been hard to schedule the downtime for it (would have to be on a weekend |
| 22:01.59 | KranZ | oogle: you didnt use ${DIALSTATUS}? |
| 22:02.11 | Bakermd | Also, all of our config for extensions, etc. is in a database - will this be an issue? |
| 22:02.14 | oogle | KranZ: I tried both, but i'm using AGI to fetch it |
| 22:02.18 | KranZ | hmm |
| 22:02.23 | Qwell[] | Bakermd: Shouldn't be |
| 22:02.38 | Bakermd | Excellent - thanks |
| 22:02.48 | [hC] | Hey, whats up guys |
| 22:02.52 | Qwell[] | [TK]D-Fender: Just restart when convenient :p |
| 22:02.59 | Qwell[] | [hC]: hey |
| 22:03.04 | [hC] | qwell!@#! |
| 22:03.07 | KranZ | is it possible to have more than one directory intro sound file for different customers? |
| 22:03.11 | *** join/#asterisk FuriousGeorge (n=ads@pool-68-162-29-224.nwrk.east.verizon.net) |
| 22:03.12 | pointer | Qwell[]: heh, and that's when you have to stop taking it seriously |
| 22:03.18 | [hC] | just got back from costa rica again, only this time i got sick out there |
| 22:03.18 | [hC] | doh |
| 22:03.19 | [hC] | :P |
| 22:03.23 | Qwell[] | again? |
| 22:03.27 | Qwell[] | I hate you. |
| 22:03.46 | Qwell[] | [hC]: Let me know when you need an onsite consultant :P |
| 22:03.46 | badboyz | anyone ever get this: Incoming call: Got SIP response 500 "Internal Server Error" ? |
| 22:03.58 | [hC] | Qwell[]: hahah i will for sure :) |
| 22:03.59 | *** part/#asterisk FuriousGeorge (n=ads@pool-68-162-29-224.nwrk.east.verizon.net) |
| 22:04.00 | [TK]D-Fender | Qwell[] : Funny... Doesnt' cut it for a full upgrade :) |
| 22:04.01 | Qwell[] | ;] |
| 22:04.09 | Qwell[] | [TK]D-Fender: Sure it does |
| 22:04.12 | [hC] | here's hoping i can line up some more customers out there and need some help :) |
| 22:04.44 | *** join/#asterisk gbodemantv (n=gbodeman@mail.televerde.com) |
| 22:04.52 | gbodemantv | hello all |
| 22:05.01 | [TK]D-Fender | Qwell[] : Well at the same time there's the sync'd effort with my GUI provider (ScopServ) and the loss of weekend time if itgoes bad (which mean I'm probably running a late Friday upgraed) |
| 22:05.10 | gbodemantv | memory leak hell |
| 22:05.20 | *** join/#asterisk Drew__ (n=foo@zux221-065-169.adsl.green.ch) |
| 22:05.23 | De_Mon | gbodemantv ? |
| 22:05.24 | gbodemantv | updated to 1.2.4 and still getting crashed |
| 22:05.46 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 22:05.49 | Qwell[] | woohoo |
| 22:05.52 | gbodemantv | 3 times yesterday and updated last night |
| 22:05.57 | gbodemantv | once this morning |
| 22:06.01 | Drew__ | New Beta Firmware for the GXP2000 is available (1.0.2.8) @ http://voip-info.org/wiki/view/GXP-2000 |
| 22:06.02 | Qwell[] | 40 minutes of downtime...awesome |
| 22:06.11 | De_Mon | gbodemantv using queues? |
| 22:06.16 | gbodemantv | nope |
| 22:06.17 | KranZ | 1.2.3 been up for 10 days and still have 350mb left in ram |
| 22:06.32 | gbodemantv | by the way we reboot every night |
| 22:06.38 | gbodemantv | to clear |
| 22:06.45 | gbodemantv | and this still happens |
| 22:06.50 | De_Mon | gbodemantv windows? |
| 22:07.10 | gbodemantv | HP DL360 g4p |
| 22:07.14 | gbodemantv | running fedora 3 |
| 22:07.19 | gbodemantv | 2.6 kernel |
| 22:07.27 | wundaboy | what config files are necesary for asterisk to run? |
| 22:07.33 | gbodemantv | dual 186 drives |
| 22:07.43 | Qwell[] | 186 drives? |
| 22:07.57 | gbodemantv | sorry dual 186 GB drives in a RAID 2 |
| 22:07.58 | wundaboy | 18.6? |
| 22:08.00 | *** join/#asterisk [Atlas] (n=Matthew@65.73.185.2) |
| 22:08.03 | Qwell[] | 18,6 indeed |
| 22:08.06 | Qwell[] | . |
| 22:08.31 | wundaboy | im new to asterisk, what config files do i need? asterisk.conf extensions.conf iax.conf sip.conf are those it? |
| 22:08.32 | gbodemantv | 5 GB Ram |
| 22:08.59 | gbodemantv | wundaboy : depends on how you want to connect to outside world |
| 22:09.03 | Qwell[] | gbodemantv: excessive |
| 22:09.07 | gbodemantv | I agree |
| 22:09.10 | KranZ | gbodemantv: took the bait |
| 22:09.36 | wundaboy | gbodemantv: i want to use my sip phone to talk to an IAX voip provider |
| 22:09.47 | Qwell[] | ~docs |
| 22:09.48 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 22:09.52 | Qwell[] | wundaboy: You've got a bit of reading to do |
| 22:09.56 | KranZ | you'll def need those |
| 22:10.16 | gbodemantv | just can't figure it |
| 22:10.16 | KranZ | "make samples" and poke around |
| 22:10.18 | wundaboy | Qwell[]: for sure, ive been reading voip-info all morning |
| 22:10.23 | gbodemantv | have a system on this side of the network |
| 22:10.28 | gbodemantv | same config, no problems |
| 22:10.50 | KranZ | same kernel and drivers? |
| 22:10.51 | De_Mon | gbodemantv sounds like hardware problems then |
| 22:11.01 | KranZ | 64bit vs 32bit? |
| 22:11.03 | gbodemantv | except 2gb of ram and earlier BIOS |
| 22:11.11 | gbodemantv | same |
| 22:11.23 | wundaboy | for the record, no one answered my question |
| 22:11.34 | KranZ | <KranZ> you'll def need those |
| 22:11.57 | gbodemantv | wundaboy : you will need to create a sip peer to Iax provider |
| 22:12.04 | gbodemantv | then set up sip peers for users |
| 22:12.10 | gbodemantv | phones or softphones |
| 22:12.25 | gbodemantv | sorry iax peer to iax provider |
| 22:12.30 | *** join/#asterisk lthnnpwr (i=Gelezini@cpc2-harg1-6-0-cust190.leed.cable.ntl.com) |
| 22:12.36 | De_Mon | i was about to say.. |
| 22:12.38 | *** join/#asterisk saftsack (n=saftsack@p54A7CA87.dip.t-dialin.net) |
| 22:12.47 | wundaboy | im using a polycom ip500 |
| 22:13.42 | gbodemantv | best is to do a make samples |
| 22:13.53 | gbodemantv | then consult the iax provider for settings |
| 22:13.54 | wundaboy | i did that, but the files are so full of examples |
| 22:14.20 | gbodemantv | but the iax.conf for example will tell you how to connect in its examples |
| 22:14.38 | gbodemantv | what iax provider? |
| 22:14.44 | wundaboy | junctionnetworks |
| 22:14.56 | pointer | [TK]D-Fender: thanks for the info on the sangoma! |
| 22:15.05 | jbalcomb | I can not get my Cisco 7940Gs to upgrade to the SIP firmware |
| 22:15.20 | wundaboy | their kinda expensive ($2/month DID $.029/minute), do you know a cheaper one? |
| 22:15.49 | pointer | [TK]D-Fender: I know we bought one for our 7th site, but I haven't played with it yet |
| 22:15.52 | jbalcomb | I'm seeing 'RRQ from 10.0.101.158 filename OS79XX.TXT' and then 'sending NAK (4, Request not null-terminated) to 10.0.101.158' in mt tftp log file |
| 22:15.56 | lthnnpwr | hi all. we've got this problem here. why isn't the connection being established when using the oh323 protocol and calling to the gatekeeper? the gatekeeper IS accepting connections, registering, accepting the numbers, but once you try to connect - the call is instantly being cancelled. the gatekeeper in use is the GnuGK. the xlite client is reporting a 503 error. where could the problem be? asterisk or the gatekeeper? thanks |
| 22:15.58 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 22:16.08 | *** part/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 22:16.18 | jbalcomb | ~jbot go |
| 22:17.39 | jbalcomb | Anyone experienced with converting Cisco IP Phones to SIP firmwares? |
| 22:18.18 | pointer | jbalcomb: somewhat...you have to put an _old_ sip load on them....then you can upgrade to the latest one |
| 22:18.43 | pointer | jbalcomb: please keep in mind that you are supposed to have SIP licenses for each phone that you do this to |
| 22:19.03 | pointer | jbalcomb: that's why we use polycoms |
| 22:19.07 | Qwell[] | meh |
| 22:19.10 | gbodemantv | any ideas where to start on my crash |
| 22:19.12 | Qwell[] | You can go from sccp > sip |
| 22:19.13 | gbodemantv | ?? |
| 22:19.17 | Qwell[] | no problems at all |
| 22:19.44 | pointer | Qwell[]: oh, I thought he had MGCP...that'll teach me to pay more attention |
| 22:20.18 | lthnnpwr | so anyone. no suggestions? |
| 22:20.19 | Qwell[] | well, he could, I suppose |
| 22:20.33 | Qwell[] | That'd be rare though |
| 22:21.33 | saftsack | are here some people with isdn? |
| 22:22.01 | jbalcomb | pointer hrmm.. thats wierd. ok, so must be why it says something about upgrade to 6.3 and then the latest.. hard to follow that document |
| 22:22.12 | jbalcomb | pointer thank you though |
| 22:23.22 | *** join/#asterisk nitestarr (n=knightst@cpe-24-33-15-250.midsouth.res.rr.com) |
| 22:23.47 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:24.02 | *** join/#asterisk lalito (n=erg@201.154.202.128) |
| 22:24.39 | pointer | saftsack: I wish, I'm looking for ISDN card recommendations |
| 22:24.39 | Qwell[] | jbalcomb: What image is on it now? |
| 22:25.06 | *** join/#asterisk nitestarr (n=knightst@cpe-24-33-15-250.midsouth.res.rr.com) |
| 22:25.08 | jbalcomb | Qwell[]: SCCP |
| 22:25.12 | Qwell[] | yes, which one? |
| 22:25.24 | jbalcomb | Qwell[]: lemme check, one sec |
| 22:26.26 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
| 22:26.35 | KranZ | damn, svn down |
| 22:26.44 | saftsack | pointer, germany? |
| 22:26.54 | saftsack | im looking for a better driver than misdn ^^ |
| 22:27.09 | saftsack | and my question is, if bristuff is better than misdn |
| 22:27.29 | Qwell[] | KranZ: Works fine here.. |
| 22:28.19 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 22:28.40 | CunningPike | Anyone up for a SPA-3000 question? |
| 22:29.37 | jbalcomb | Qwell[]: Version: 3.1(MF.G2) |
| 22:29.40 | pointer | saftsack: that's what I'd like to know as well...with 4-5 options, it makes you wonder :-\ |
| 22:30.23 | jbalcomb | Qwell[]: App ID: P0030301MFG2 Bood Load ID: PC0303010200 |
| 22:30.28 | Qwell[] | jbalcomb: should upgrade fine straight to 7.x. I've done so a few times now |
| 22:30.38 | Qwell[] | 7.x sccp, then go to sip |
| 22:30.46 | jbalcomb | Qwell[]: RRQ from 10.0.101.158 filename OS79XX.TXT |
| 22:30.54 | jbalcomb | Qwell[]: sending NAK (4, Request not null-terminated) to 10.0.101.158 |
| 22:31.08 | Qwell[] | NAK usually means the file doesn't exist |
| 22:31.32 | jbalcomb | Qwell[]: vi OS79XX.TXT == P003-07-5-00 |
| 22:31.54 | Qwell[] | for some reason, the tftpd doesn't see it |
| 22:32.16 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 22:32.37 | jbalcomb | Qwell[]: if i rm OS79XX.TXT it says 'file not found' rather than 'Request not null-terminated' |
| 22:32.59 | jbalcomb | Qwell[]: I set chmod 666 on *.* in tftp root |
| 22:33.09 | saftsack | pointer, what do you mean with 4 - 5 options? |
| 22:33.50 | Flyboy-SR22 | anyone know how I can up the volumn coming out of my fsx ports...? Is the the rxgain and txgain settings in the zapata.conf file..? |
| 22:34.06 | *** join/#asterisk FuriousGeorge (n=ads@pool-68-162-29-224.nwrk.east.verizon.net) |
| 22:35.32 | jbalcomb | Qwell[]: is put 'P003-07-5-00' in the OS79XX.TXT all that should be required to get it to pull the image? |
| 22:35.55 | Qwell[] | jbalcomb: sometimes |
| 22:36.41 | wundaboy | how can i tell if i registered with my iax provider correctly? |
| 22:36.52 | wundaboy | *voip provider over iax |
| 22:37.17 | jbalcomb | Qwell[]: what else might I need? |
| 22:37.27 | Qwell[] | jbalcomb: Can you get the file from tftp on your own? |
| 22:37.39 | Qwell[] | tftp to the machine, and get OS79XX.TXT |
| 22:38.29 | KranZ | anyone know if you can have different sound bytes for the Directory() command? |
| 22:38.36 | KranZ | for different voicemail contexts? |
| 22:38.37 | jbalcomb | Qwell[]: trying now.. |
| 22:38.57 | lthnnpwr | hi all. we've got this problem here. why isn't the connection being established when using the oh323 protocol and calling to the gatekeeper? the gatekeeper IS accepting connections, registering, accepting the numbers, but once you try to connect - the call is instantly being cancelled. the gatekeeper in use is the GnuGK. the xlite client is reporting a 503 error. where could the problem be? asterisk or the gatekeeper? thanks |
| 22:39.22 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 22:41.33 | saftsack | are some junghanns experts here? |
| 22:43.48 | pointer | saftsack: chan_capi, mISDN, zap_hfc, chan_modem, and vISDN |
| 22:44.14 | saftsack | pointer, i know them ;) |
| 22:44.23 | saftsack | but i want to find the advantagest one |
| 22:44.28 | pointer | saftsack: same here |
| 22:44.35 | saftsack | and i think that the junghanns driver would be the best |
| 22:44.50 | saftsack | the misdn driver has many bugs if you ask me |
| 22:45.02 | jbalcomb | Qwell[]: my tftp client says file not found. |
| 22:45.11 | jbalcomb | Qwell[]: /usr/sbin/in.tftpd -c -l -vvvvv -u nobody -s /asterisk/tftp is running ok |
| 22:45.16 | Qwell[] | Are you sure the tftpd can see it? |
| 22:45.38 | jbalcomb | Qwell[] the permissions seem fine |
| 22:46.06 | Qwell[] | it has access to /asterisk and /asterisk/tftp ? |
| 22:46.28 | jbalcomb | Qwell[]: the wierd part is that when the file is actually missing it says 'file not found' but if its there it says 'Request not null-terminated' |
| 22:46.39 | saftsack | pointer, do you have any experiences with any isdn driver? |
| 22:46.54 | jbalcomb | Qwell[]: drwxr-xr-x 6 root root 184 2005-10-12 21:12 asterisk |
| 22:47.14 | Qwell[] | and /asterisk/tftp ? |
| 22:47.16 | jbalcomb | Qwell[]: drwxr-xr-x 6 root root 136 2006-02-07 17:44 tftp |
| 22:47.51 | crich1999 | saftsack, please post them at bugs.digium.com if you find some |
| 22:48.03 | jbalcomb | Qwell[]: i just chmod 777 them both and uploaded a file via tftp client |
| 22:48.13 | Qwell[] | jbalcomb: Can you get that file now? |
| 22:48.19 | jbalcomb | Qwell[]: it uploaded and is in the right place |
| 22:48.41 | saftsack | crich1999, ok i have misdn atm |
| 22:49.09 | saftsack | crich1999, i have to "patch" the sourcecode of qozap to let it run with my beronet card, or? |
| 22:49.34 | crich1999 | saftsack, yes you have if you really like to do that |
| 22:49.58 | crich1999 | saftsack, what kind of issues do you encounter with misdn ? |
| 22:50.18 | jbalcomb | Qwell[]: yes, I was able to grab that file. |
| 22:50.28 | saftsack | segfaulting when i pickup a call |
| 22:50.30 | Qwell[] | jbalcomb: But you can't grab the other file? Sounds like an issue |
| 22:50.32 | jbalcomb | Qwell[]: uploading the firmware files now via the tftp client |
| 22:50.53 | tzafrir_laptop | saftsack, does the card have a different PCI ID? |
| 22:51.02 | tzafrir_laptop | it sure looks very similar |
| 22:51.06 | Qwell[] | jbalcomb: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx |
| 22:51.14 | jbalcomb | Qwell[]: maybe the chmod 777 did it |
| 22:51.19 | Qwell[] | jbalcomb: Look at "simplify updates" section |
| 22:51.25 | Qwell[] | it's like one step |
| 22:51.31 | jbalcomb | Qwell[]: ok, im on that page already |
| 22:52.01 | saftsack | tzafrir, its a 4port card |
| 22:52.13 | saftsack | what do you mean with pci id? |
| 22:52.17 | Qwell[] | brb |
| 22:52.21 | jbalcomb | Qwell[]: damn, didn't see that section |
| 22:52.32 | tzafrir_laptop | saftsack, what patch did you apply? |
| 22:54.25 | saftsack | i didnt apply any patch. i have asterisk 1.2.1 so i dont need the stable patch for misdn what runs very stable. but just pickup doesnt work |
| 22:54.31 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 22:54.43 | tzafrir_laptop | I figure nobody has both beronet and junghanns cards. And with the current qozap module not loading automatically by PCI ID, that patch seem safe to add at first glance |
| 22:55.31 | pointer | saftsack: nope |
| 22:55.39 | tzafrir_laptop | (assuming that you could make the beronet card easily work with qozap) |
| 22:55.41 | saftsack | ? |
| 22:55.53 | tzafrir_laptop | saftsack, ignore me. just thinking aloud |
| 22:55.57 | saftsack | ok |
| 22:56.01 | pointer | saftsack: I haven't tried any of the isdn drivers |
| 22:56.31 | *** join/#asterisk Zodiacal (n=hehe@bdsl.66.14.242.199.gte.net) |
| 22:56.37 | saftsack | yes ok |
| 22:56.55 | saftsack | crich1999, are you experienced with isdn? |
| 22:56.59 | Zodiacal | is a software raid ok for a 6 POTS Lines and 12 SIP ext. user system? |
| 22:57.30 | saftsack | crich1999, are you mr. richter? :) |
| 22:57.45 | crich1999 | yep that's me :-) |
| 22:58.08 | jbalcomb | Qwell[]: all good. thank you. |
| 22:58.08 | saftsack | wow :) then you are the perfect man for me :) |
| 22:58.15 | saftsack | i am the guy who talked ca. 4 weeks ago and asking for some bugs ;) |
| 22:58.15 | jbalcomb | FYI: +10 karma point for Qwell[]. woot!! |
| 22:58.21 | crich1999 | well i have a girl friend you know |
| 22:58.27 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 22:58.33 | file | Qwell[]: House tonight |
| 22:58.36 | saftsack | ? |
| 22:59.01 | Qwell[] | jbalcomb: Do me a favor... |
| 22:59.08 | Qwell[] | jbalcomb: Call benjk an idiot. :) |
| 22:59.16 | Qwell[] | He couldn't do that one simple step |
| 22:59.21 | Qwell[] | newb :D |
| 22:59.32 | file | like taking candy... from a baby! |
| 22:59.35 | file | A B C |
| 22:59.43 | file | falling in love with you was easy for me |
| 22:59.46 | Qwell[] | file: I take candy from my baby all the time. ;) |
| 22:59.48 | file | and you can do it too |
| 22:59.53 | file | it's so easy... like taking candy from a baby |
| 22:59.59 | file | baby there's nothing hard about love |
| 23:00.09 | file | basically it's as easy as pie |
| 23:00.37 | Qwell[] | RoyK: How much $ you got? |
| 23:00.41 | Qwell[] | I can hook you up |
| 23:00.47 | wundaboy | so, say i have [junction-networks] |
| 23:00.50 | RoyK | :) |
| 23:00.51 | wundaboy | in my iax.conf file |
| 23:01.01 | wundaboy | and some stuff in that and its registerd with the provider |
| 23:01.11 | wundaboy | what channel would that be? |
| 23:01.16 | wundaboy | iax2/junction-networks ? |
| 23:01.21 | Qwell[] | benjk: You hear that? This guy (who doesn't claim to be a professional) was able to do the one-step upgrade |
| 23:01.26 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 23:01.27 | Qwell[] | </gloat> |
| 23:01.44 | mzo | hahaahahahaha he's STILL on about taht shit? |
| 23:01.49 | Qwell[] | mzo: No :p |
| 23:01.52 | mzo | yay |
| 23:01.58 | Qwell[] | mzo: But somebody else was able to just a minute ago |
| 23:02.02 | mzo | good, someone help me buy a dell replacement fan, dell wants $99 for it. |
| 23:02.05 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167051174.pppoe-dynamic.nb.aliant.net) |
| 23:02.08 | Qwell[] | wtf |
| 23:02.10 | Qwell[] | mzo: measure it |
| 23:02.13 | Qwell[] | file: NOOO!!!! |
| 23:02.15 | mzo | It's a standard part, |
| 23:02.15 | *** part/#asterisk rene- (n=rene@dsl-201-128-115-222.prod-infinitum.com.mx) |
| 23:02.22 | mzo | i think im just gonna gamble and buy one online |
| 23:02.22 | Qwell[] | mzo: 80mm? |
| 23:02.25 | mzo | no, bigger. |
| 23:02.28 | Qwell[] | 90mm? |
| 23:02.32 | mzo | 120ish |
| 23:02.33 | Qwell[] | 120mm? |
| 23:02.44 | Qwell[] | yeah, that's a standard size. $2.99 |
| 23:02.45 | wundaboy | hey, it might be worth $99... |
| 23:02.54 | file | my ISP has decided to send all internet ISP through a single pipe to the other side of the province |
| 23:03.01 | file | it now has a habit of getting congested |
| 23:03.01 | [Atlas] | anyone happen to know where i can grab a cisco 7960 sip flash img without a cisco passwd? |
| 23:03.10 | mzo | it's 120mm x 120mm, but i need it with the techometer sensors. Anyone have a recommendation? |
| 23:03.30 | Qwell[] | file: I've got you beat cold |
| 23:03.40 | Qwell[] | file: My ISP cuts me off at 1:25 every day, for the last 3 days |
| 23:03.45 | file | Qwell[]: beautiful |
| 23:03.49 | wundaboy | so, i am a little confuzed at how channel's are named in asterisk |
| 23:03.59 | Zodiacal | anyone know if software raid1 would be to slow for asterisk? 6 pots lines and 12 sip phones |
| 23:04.24 | Qwell[] | Zodiacal: It'll be fine |
| 23:04.34 | Zodiacal | qwell Thank You! |
| 23:04.48 | [Atlas] | do you have to download the img from the cisco site? how much does a passwd cost? |
| 23:04.55 | *** part/#asterisk pointer (i=pointer@aj.catt.com) |
| 23:04.56 | mzo | support through cisco is worth it |
| 23:04.59 | Qwell[] | [Atlas]: You need a cco support contract |
| 23:05.20 | Qwell[] | it isn't much though. Maybe $20/phone |
| 23:05.41 | [Atlas] | ah ok thats not bad |
| 23:05.55 | Qwell[] | ymmv |
| 23:06.01 | Qwell[] | ~ymmv |
| 23:06.02 | jbot | hmm... ymmv is Your Mileage May Vary |
| 23:06.03 | [Atlas] | i was afraid i would have to get like a 600 dollar smartnet contract =D |
| 23:06.14 | Qwell[] | nah, get the el-cheapo smartnet |
| 23:06.25 | [Atlas] | kk |
| 23:06.47 | [Atlas] | thanks! |
| 23:06.51 | Qwell[] | file: OMG, is it new?! |
| 23:07.34 | file | Qwell[]: OMG LIKE YEAH I THINK |
| 23:07.39 | Qwell[] | omg omg omg |
| 23:07.45 | file | 1337 |
| 23:07.50 | Qwell[] | ubar-krad |
| 23:07.51 | tuxinator_linux | Qwell, [Atlas], they wouldn't sell me the CON-SNT-CP7960 ($13), forced me to get the CON-SNT-PKG1 ($93) |
| 23:08.01 | Qwell[] | tuxinator_linux: Who is they? |
| 23:08.13 | tuxinator_linux | Cisco via CDW |
| 23:08.18 | Qwell[] | go elsewhere |
| 23:08.42 | tuxinator_linux | I'm open to that |
| 23:08.50 | Qwell[] | CDW won't sell it in the US anymore, iirc |
| 23:09.31 | mzo | hahaha, nice |
| 23:09.34 | Zodiacal | atlas 8 bux http://www.sparco.com/cgi-bin/wfind2?spn=A748642 |
| 23:09.45 | Zodiacal | took a few weeks tho |
| 23:09.50 | Zodiacal | but got it |
| 23:10.26 | Qwell[] | tuxinator_linux: ^ |
| 23:12.16 | tuxinator_linux | Zodiacal, are you in the States? |
| 23:12.38 | Zodiacal | yep |
| 23:13.27 | tuxinator_linux | Zodiacal, how long ago? |
| 23:13.35 | tuxinator_linux | did you order yours |
| 23:13.36 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
| 23:13.38 | Zodiacal | got the code yesterday, ordered it like 3 weeks ago tho |
| 23:13.47 | tuxinator_linux | k |
| 23:14.19 | Zodiacal | im acctualy in the next city over from you :P |
| 23:14.20 | dlynes | Has anyone encountered the following error on Asterisk 1.2.3? I can't seem to find any reference to it on Google anywhere |
| 23:14.35 | dlynes | logger.c: Don't know what to do if second ROSE component is of type 0x6 |
| 23:15.01 | tuxinator_linux | Zodiacal, Riverside? |
| 23:15.03 | dlynes | Well, it's not an error...it shows up with a verbose tag |
| 23:15.09 | Qwell[] | eww, Riverside? |
| 23:15.10 | Qwell[] | gross |
| 23:15.14 | Zodiacal | no |
| 23:15.25 | tuxinator_linux | I don't like Riverside or Moreno Valley |
| 23:15.29 | Qwell[] | tuxinator_linux: If Riverside is next to you...you're in...MV? |
| 23:15.37 | tuxinator_linux | yep |
| 23:15.40 | Qwell[] | oh lord |
| 23:15.40 | tuxinator_linux | not by choice |
| 23:15.48 | Zodiacal | tuxinator_linux get FIOS yet? |
| 23:15.52 | Qwell[] | I just moved away from there like 3 months ago :p |
| 23:15.59 | tuxinator_linux | my wife is here on an internership, done in 6 weeks |
| 23:16.00 | Zodiacal | err, obvlusly not.. i should say why not yet? |
| 23:16.15 | tuxinator_linux | moving over to Anaheim Hills or Yorba Linda (where I grew up) |
| 23:16.19 | Qwell[] | Zodiacal: They've got fiber in parts of MV...just haven't lit it up yet |
| 23:16.28 | Skumling | humm. would one via the normal asterisk queues be able to present the caller for a dialing tone right before getting through to an agent? the immediate switch from MOH to the Agent might seem a bit "confusing" to some callers... |
| 23:16.38 | tuxinator_linux | Zodiacal, not sure what FIOS is |
| 23:16.52 | Zodiacal | qwell its on the polls on my street, but not lit up yet either.. other parts of the town are tho |
| 23:16.54 | Qwell[] | hell, it's been there for years too |
| 23:16.58 | Zodiacal | FIOS is fiber to the home |
| 23:17.02 | Qwell[] | tuxinator_linux: What part of MV? Cross-streets? |
| 23:17.06 | Zodiacal | 15Mbps/2Mbps for 45bux |
| 23:17.10 | Zodiacal | www.verizon.com/fios |
| 23:17.21 | _Sam-- | i have it on order for my house |
| 23:17.25 | tuxinator_linux | Moreno Beach Dr & John F Kennedy |
| 23:17.33 | Qwell[] | tuxinator_linux: not so bad then |
| 23:17.34 | Zodiacal | tuxinator_linux they probably have activated where u are |
| 23:18.24 | Qwell[] | Zodiacal: And you're where? |
| 23:18.51 | Zodiacal | chino |
| 23:19.07 | Qwell[] | yuck |
| 23:19.08 | Zodiacal | qwell MV? |
| 23:19.18 | Zodiacal | qwell work not home :P |
| 23:19.19 | Qwell[] | not anymore...thank god |
| 23:20.00 | Zodiacal | can't order fios yet |
| 23:20.12 | Zodiacal | but other people in my town have it.. |
| 23:20.14 | Zodiacal | grr |
| 23:20.35 | tuxinator_linux | no FIOS for me |
| 23:20.43 | tuxinator_linux | but I moving away soon anyways |
| 23:20.45 | mzo | does FIOS still use PPPoE? |
| 23:20.49 | Zodiacal | mzo i think so |
| 23:20.53 | mzo | then it's ass :P |
| 23:21.04 | Zodiacal | oh big deal |
| 23:21.07 | mzo | stupid as hell having to remember an 'internet password' |
| 23:21.08 | Qwell[] | PPPoF? heh |
| 23:21.10 | Zodiacal | its fast |
| 23:21.18 | mzo | and install stupdi client software just to connect |
| 23:21.24 | Zodiacal | mzo they say they are going to change soon |
| 23:21.29 | mzo | haa, i dobut it |
| 23:21.31 | Zodiacal | no client software. they give u a router |
| 23:21.45 | mzo | god, the person who thoght of PPPoE should be shot |
| 23:21.46 | mzo | oh wait |
| 23:21.47 | mzo | he was |
| 23:21.51 | Qwell[] | The only thing special about that router is the 100mbit wan port |
| 23:21.57 | mzo | his name is Haidar Chamas, Director of Verizon High Speed solutions. |
| 23:22.01 | Qwell[] | They say "No, you MUST use our router!" |
| 23:22.06 | mzo | they fired his ass in 2002, after that PPPoE fiasco |
| 23:23.46 | Zodiacal | as well they should but its not going to stop me from gettin my fios |
| 23:24.16 | cpm | pppoe is convenient on the isp side. |
| 23:24.39 | mzo | of course it is |
| 23:24.47 | mzo | which is why they rammed it down their customers' throats, |
| 23:25.06 | mzo | but fuck those assholes (former employer), idiot incompetents :P |
| 23:25.08 | Himeko | no pppoe here which was very cool |
| 23:25.20 | tuxinator_linux | I suppose I can try and return the CON-SNT-PKG1 from CDW. Might be a strugle. What is their reasoning for saying the Cisco won't sell the cheaper one to me? |
| 23:25.39 | Qwell[] | tuxinator_linux: Not supposed to sell it in the US or something |
| 23:26.01 | tuxinator_linux | what differnce does it make? That's all I have been told, also. |
| 23:26.05 | Zodiacal | qwell i even selected a location dropdown on cisco's site when finishing the registration form.. |
| 23:26.07 | Zodiacal | i picked usa |
| 23:26.09 | oogle | crisco won't... it's not in their nature. they're still mad about their stock price |
| 23:26.11 | Qwell[] | dunno |
| 23:26.26 | *** join/#asterisk X10ZION (n=askme@68.63.2.196) |
| 23:26.34 | Zodiacal | qwell i read some where people were having trouble with cdw so someone on a forum suggested i try that sparco or wahtever co. |
| 23:26.54 | X10ZION | anyone know right off where the asterisk call manger log is? |
| 23:29.10 | X10ZION | heh mainly what the name of it is |
| 23:30.09 | *** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu) |
| 23:30.34 | wundaboy | is there an easy-to-learn document that covers the basics of how asterisk works all in one document? |
| 23:30.49 | wundaboy | because im learning pieces on voip-info they just arent all falling together completely yet... |
| 23:30.56 | tzafrir_laptop | wundaboy, at what level? |
| 23:31.07 | wundaboy | tzafrir_laptop: idiot level |
| 23:31.25 | Zodiacal | wundaboy search google for : asterisk handbook |
| 23:31.44 | Qwell[] | ~thebook |
| 23:31.46 | jbot | thebook is, like, Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 23:31.46 | tzafrir_laptop | Not sure. It is more of an install guide |
| 23:32.12 | Zodiacal | wundaboy http://www.digium.com/handbook-draft.pdf |
| 23:33.32 | wundaboy | Zodiacal: i found it |
| 23:35.02 | tuxinator_linux | Zodiacal, shipping on a non-tangable item (sparco)? |
| 23:35.44 | Zodiacal | i think they have a 50 min limit you have to buy |
| 23:35.52 | Zodiacal | no shiping, they email it to you |
| 23:36.02 | Zodiacal | u can buy 50 worth of other crap at their store tho |
| 23:36.52 | Zodiacal | forgot about that catch, sorry |
| 23:36.52 | trelane | anyone using cisco 7905 phones? |
| 23:39.39 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 23:41.10 | tuxinator_linux | Zodiacal, it processed my $9 order |
| 23:41.32 | *** join/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
| 23:44.18 | Zodiacal | tuxinator yeah, i did that too and they called me on it.. then i just went and ordered some other stuff from the site to add up to exactly 50 bux.. |
| 23:44.40 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 23:49.15 | *** join/#asterisk kimosabe (n=kimosabe@dsl-201-128-128-191.prod-infinitum.com.mx) |
| 23:49.27 | kimosabe | is any one runnin asterisk on openwrt |
| 23:50.06 | cpm | brilliant! |
| 23:50.31 | kimosabe | i need a hand im getting drive full |
| 23:50.52 | kimosabe | i only have 1.5 meg 2 install on |
| 23:53.03 | tuxinator_linux | Zodiacal, oh, okay.... My next phone will most likely be a polycom.... too much trouble for cisco's, but I know that going into it |
| 23:53.33 | *** join/#asterisk cj (n=cjcollie@unaffiliated/daman/x-0000001) |
| 23:53.38 | cj | where's the svn live? |
| 23:53.56 | tuxinator_linux | cj, digium? |
| 23:54.08 | Zodiacal | tuxinator i think polycoms need a lisence too |
| 23:54.12 | Zodiacal | dunno for sure tho |
| 23:54.22 | *** join/#asterisk st3v (n=spj@netblock-66-218-41-231.dslextreme.com) |
| 23:54.24 | MstlyHrmls | Zodiacal: no, they don't |
| 23:54.35 | Zodiacal | i stand corrected |
| 23:54.39 | MstlyHrmls | :-) |
| 23:54.47 | st3v | is there a way to have AMP forward to a cell phone instead of going to voicemail? |
| 23:54.49 | tuxinator_linux | if I'm not mistaken, the phone part of cisco's are really poloycoms... |
| 23:54.55 | Zodiacal | mstlyhrmls maybe it was snom? |
| 23:54.56 | cj | tuxinator_linux: got a url? |
| 23:54.58 | Zodiacal | i don't remember |
| 23:55.54 | tuxinator_linux | cj, http://www.asterisk.org/asterisk-converts-to-subversion |
| 23:56.10 | MstlyHrmls | tuxinator_linux: cisco's used Polycom's DSP code. dunno if they still do though |
| 23:56.20 | *** join/#asterisk ketanp (n=ketanp@67.132.43.2) |
| 23:57.02 | tuxinator_linux | cj, also http://www.asterisk.org/download |
| 23:57.38 | cj | ah, I see... it just requires that I choose a subdirectory and not grab the entire repository |
| 23:57.49 | [av]bani | polycom doesnt need a license, you just need a vendor who doesnt suck |
| 23:58.05 | _Sam-- | [av]bani: did you fix your phone? |
| 23:58.05 | cj | thank you all for this software, by the way |
| 23:58.11 | [av]bani | _Sam--: yes thanks \o/ |
| 23:58.14 | _Sam-- | sweet! |
| 23:58.15 | [av]bani | @\o/@ |
| 23:58.24 | [av]bani | @\@\o |
| 23:58.25 | [av]bani | o/@/@ |
| 23:58.35 | TooMe | any idea why the System() would have issues running cp or mv? |
| 23:58.36 | cj | it has the potential to break apart the US telecom monopoly.... again :) |
| 23:58.55 | ketanp | into tiny little pieces this time |
| 23:59.00 | [av]bani | _Sam--: and my cow orker was taunting me when i borked the flash, he was all disappointed when i recovered it :D |
| 23:59.10 | *** join/#asterisk voipD470 (n=A_mail@pool-68-238-244-251.phlapa.fios.verizon.net) |
| 23:59.29 | _Sam-- | poor guy....he had to get back to work :) |
| 23:59.51 | _Sam-- | anything good with that new firmware? |
| 23:59.58 | _Sam-- | they fix the loop? |