| 00:12.06 | rob0 | HNY @ GMT ! |
| 00:12.36 | Qwell | Silly GMTers |
| 00:18.01 | *** join/#asterisk ariaveta (n=marco@host85-32-dynamic.11-87-r.retail.telecomitalia.it) |
| 00:18.09 | ariaveta | 2007 here |
| 00:18.36 | ariaveta | since 1 hour and 12 minutes |
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| 00:25.06 | [Outcast] | hey there doc!!! |
| 00:25.38 | russellb | luke-jr: yes, it's called EAgi :) |
| 00:25.48 | russellb | wait, well, not over stdin/stdout ... |
| 00:26.41 | russellb | though it would be fairly easy to write an asterisk app to do it, if it would be useful to someone |
| 00:27.21 | luke-jr | well, other file sockets might work I guess |
| 00:27.25 | luke-jr | as long as I stick to C |
| 00:28.54 | russellb | well, if you want to provide audio, as well, then EAgi won't work |
| 00:31.19 | `Sean | he whats ia good default expiry i should set for register |
| 00:31.32 | `Sean | defaultexpiry=120 |
| 00:31.48 | Qwell | `Sean: depends |
| 00:31.51 | `Sean | for some reason if i have 2 minutes, it regreigsters every 2 min, but asterlink gives it timed out issues |
| 00:31.56 | `Sean | qwell on what |
| 00:32.05 | Qwell | your network, your phones, etc |
| 00:32.29 | `Sean | netowkr?, i'm using a anologe phone on a TDM400p |
| 00:32.33 | Qwell | Then why are you registering? |
| 00:32.55 | `Sean | asterlink says i should :P? |
| 00:32.58 | `Sean | its in sip.conf? |
| 00:33.03 | Qwell | then it depends on your network |
| 00:33.16 | `Sean | defaultexpiry=120 |
| 00:33.16 | `Sean | ; use a default sip registration expiry of 2 minutes |
| 00:33.44 | `Sean | i wish they would make it so youd only register once |
| 00:33.50 | `Sean | till you disconnect from internet |
| 00:34.05 | `Sean | that way register requests would be cut down by more then half to the providers, and save bw all around :p |
| 00:34.23 | Qwell | a register takes several bytes |
| 00:35.04 | `Sean | heh |
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| 00:47.53 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 00:50.36 | infosys24 | howdy chan |
| 00:53.15 | infosys24 | im a new to voip and wondering which would be better a pri or a t1? |
| 00:53.24 | Qwell | neither are voip |
| 00:53.39 | russellb | and pri and t1 aren't really comparable anyway :) |
| 00:53.45 | russellb | a pri uses a t1 |
| 00:54.17 | infosys24 | i want to do my own terminations at my pbx |
| 00:54.33 | infosys24 | and i have a t1 card but im not sure what i need |
| 00:54.40 | infosys24 | from the phone company |
| 00:55.38 | russellb | you would order a PRI, then |
| 00:56.36 | infosys24 | ok so when the pri gets to my office |
| 00:56.46 | infosys24 | will i need any other hardware |
| 00:56.56 | russellb | a pc to put the t1 card in :) |
| 00:57.04 | infosys24 | got that |
| 00:57.06 | russellb | assuming it's an asterisk supported t1 card ... |
| 00:57.59 | infosys24 | digium 2 port t1 card? |
| 00:58.11 | russellb | then yes, that is absolutely supported :) |
| 00:58.19 | russellb | given that digium is who makes asterisk ;) |
| 00:58.31 | infosys24 | lol i actually knew that |
| 00:59.47 | infosys24 | i just didnt know if that would be all the hardware i needed or not |
| 01:00.16 | infosys24 | somewhere i saw i would need a channelbank and threw me for a loop |
| 01:00.34 | russellb | well, it depends what you want to do |
| 01:00.58 | russellb | if you'd like to support some analog phones, you can hook up a t1 to a channelbank that has analog phones hooked to it. |
| 01:01.06 | SwK | infosys24 channel banks come in handy for feeding things like modems and fax machines and single line phones in some areas |
| 01:01.29 | infosys24 | ic |
| 01:02.10 | SwK | things like faxing and modems tend to work 1000 times better over TDM then w/ ULaw passthru (this is due to things that are hard to control like network jitter) |
| 01:02.34 | infosys24 | i know |
| 01:02.50 | infosys24 | i have a pap2 on my fax and it was a time to get it working right |
| 01:03.46 | eltech | anyone any experience of re-flashing a polycom phone? seems mine went bad for whatever reason and is a brick right now |
| 01:04.35 | infosys24 | once with a 501 |
| 01:05.08 | infosys24 | lost power when it was flashing the bootrom |
| 01:05.22 | SwK | eltech: did you brick it on the bootrom portion? |
| 01:05.23 | eltech | infosys24 i cant find any documentation on it.. do you have a link or a whitepaper? or explain the steps? |
| 01:05.28 | SwK | or just the sip.ld portion? |
| 01:05.45 | eltech | SwK seems so .. not sure.. was woking prior to the upgrade |
| 01:05.53 | eltech | no not sip.ld |
| 01:05.59 | SwK | remove bootrom.ld from the server you are feeding the stuff from, reboot the phone and see what happens |
| 01:06.05 | eltech | cause i can go up and down for sip.ld, not bootrom though |
| 01:06.10 | SwK | (you dont have to rm it but make it named something other then bootrom.ld |
| 01:06.28 | infosys24 | to get mine working again |
| 01:06.37 | SwK | some versions of the bootrom are flaky and will lock up if they see a copy of bootrom.ld out there |
| 01:06.41 | infosys24 | i had to have it boot to an earlier version |
| 01:06.50 | infosys24 | and then back to the current |
| 01:06.56 | eltech | infosys24 but you cant downgrade bootrom so how? |
| 01:07.08 | eltech | i want to do this, but doesnt seem like its possible |
| 01:07.13 | eltech | atleast in all i been tring |
| 01:07.29 | infosys24 | i had 6 501 soundpoints |
| 01:07.41 | infosys24 | and i was using nuvio as my service |
| 01:08.00 | infosys24 | and when i build my pbx i was using a later bootrom then them |
| 01:08.15 | infosys24 | so when i rebooted the phone to them |
| 01:08.26 | infosys24 | it reloaded there v. bootrom |
| 01:08.37 | infosys24 | and then when i reconfiged it to my boot server |
| 01:08.47 | infosys24 | it d/l the newest one which i had |
| 01:08.52 | infosys24 | and worked fine |
| 01:08.56 | eltech | different case.. cause this phone been using boot server since start |
| 01:09.40 | infosys24 | right |
| 01:10.29 | infosys24 | what version bootrom did u use that caused it to die? |
| 01:10.42 | eltech | 3.2.2 |
| 01:11.09 | infosys24 | what version did it have before? |
| 01:11.36 | eltech | i thkin 3.1.X |
| 01:11.47 | infosys24 | ok trust me k |
| 01:11.59 | infosys24 | make a new ftp user and /home dir |
| 01:12.18 | infosys24 | load the 3.1.x plcm phone stuff into it |
| 01:12.42 | infosys24 | and then config ur phone to use the new ftp user info |
| 01:12.57 | infosys24 | and it will roll back |
| 01:14.32 | eltech | why new user or /home |
| 01:14.39 | eltech | ill just delete and remove files |
| 01:15.25 | eltech | so its downgrading sip data now |
| 01:15.38 | eltech | might just skip right past bootrom |
| 01:15.50 | infosys24 | in my case i have other phones on that ftp |
| 01:15.57 | eltech | ah |
| 01:16.05 | infosys24 | so i would make a different place so i would kill them |
| 01:16.10 | eltech | saving new (downgraded) sip.ld app |
| 01:17.24 | eltech | :/ |
| 01:17.26 | eltech | 1231201629|sys |6|01|soSystemLogException logging complete, rebooting... |
| 01:17.32 | eltech | thats the error in the app log |
| 01:18.28 | infosys24 | whats on the phone screen? |
| 01:18.38 | eltech | just sits there |
| 01:18.41 | eltech | wildome! |
| 01:18.48 | eltech | (ipso): 1.1.x.x |
| 01:18.49 | eltech | blah |
| 01:18.51 | eltech | blah |
| 01:19.17 | eltech | then reboots |
| 01:19.40 | eltech | TFTP read error |
| 01:19.51 | eltech | tftpLib error: tftp transfer failed: error 0x4b0008 |
| 01:20.18 | eltech | but i know there isnt any problems with the tft server |
| 01:20.31 | infosys24 | is ur tftp server running with the old configs and the phone is configed to that ftp correctly? |
| 01:21.03 | eltech | its running with just the added (old) sip version and bootrom |
| 01:21.12 | eltech | so essentially, no old config files |
| 01:21.14 | eltech | all new |
| 01:21.20 | infosys24 | rgr |
| 01:22.17 | infosys24 | what model is the phone? |
| 01:22.22 | eltech | ip430 |
| 01:23.26 | infosys24 | i dunno man |
| 01:24.02 | infosys24 | i know i had the same problem with a 501 |
| 01:24.10 | infosys24 | and thats how i corrected it |
| 01:24.27 | eltech | yea i know .. i thought i could correct it this way |
| 01:24.30 | eltech | but doesnt seem so |
| 01:24.52 | infosys24 | yep i dunno what else to try either |
| 01:26.13 | infosys24 | and polycom wont even talk to u |
| 01:28.38 | eltech | 0101012801|copy |3|00|Update of '/ffs0/init.mac' failed, leaving local copy intact |
| 01:28.40 | eltech | hmmmm |
| 01:31.27 | infosys24 | never seen init.mac before |
| 01:33.54 | luke-jr | What would be a good way to do audio input and output from a script/program? |
| 01:34.25 | luke-jr | EAGI w/ a FIFO for out? |
| 01:36.11 | infosys24 | dunno on that |
| 01:36.24 | luke-jr | russellb? |
| 01:37.24 | russellb | luke-jr: you'd have to write an asterisk application to do it |
| 01:37.30 | luke-jr | hrm |
| 01:37.33 | russellb | to either do exactly what you want |
| 01:37.33 | russellb | or |
| 01:37.45 | luke-jr | output would be from Festival |
| 01:37.56 | russellb | that will execute some external application and provide audio over stdin/out |
| 01:38.55 | luke-jr | hrm |
| 01:39.10 | luke-jr | if only app Festival could be interrupted from EAGI |
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| 01:39.38 | luke-jr | or can it? |
| 01:39.52 | russellb | actually ... it can |
| 01:39.59 | russellb | you can execute the festival app from the agi script |
| 01:40.25 | luke-jr | and pause it? |
| 01:40.53 | russellb | nope |
| 01:41.55 | russellb | probably just works |
| 01:41.58 | russellb | words taht is |
| 01:42.01 | russellb | i quit. |
| 01:42.03 | russellb | :) |
| 01:42.46 | luke-jr | ? |
| 01:42.50 | luke-jr | heh |
| 01:43.08 | russellb | typo in my message correcting a typo |
| 01:43.14 | luke-jr | hm, so if I can detect when festival has completed, I can just feed * a bunch of app festival... |
| 01:43.24 | russellb | yeah, you could do that. |
| 01:43.29 | russellb | it will tell you when it's done |
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| 01:47.47 | luke-jr | hm, I wonder if 500 words is simply too large a vocabulary for Sphinx3 |
| 01:48.01 | luke-jr | it was doing much better back when there were like 5 |
| 01:53.18 | luke-jr | russellb: how will it tell me? :\ |
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| 02:26.04 | jimglickster | root@ubuntu:~# asterisk -r |
| 02:26.04 | jimglickster | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 02:26.04 | jimglickster | can anyone help me with that ? |
| 02:26.04 | jimglickster | asterisk is running and the ctl file exists |
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| 02:53.28 | coppice | I think the snag with sphinx is the developers have little interest in making a polished package, and nobody else has picked up that task as something interesting to do. |
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| 02:54.48 | luke-jr | well, it's virtually impossible to get it to do anything outside of the example scripts |
| 02:55.04 | luke-jr | there's absolutely no useful documentation, and the binaries are overly complex |
| 02:55.24 | luke-jr | eg, 30 or so command line options required for basic use |
| 02:58.19 | CunningPike | jimglickster: Try 'sudo asterisk -r' |
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| 02:58.37 | bblong | lol hy guys GENTOO IS AWESOE DCC SCHAT "HAPPYNEWYEARFROMBANTOWN" 0 0 0 |
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| 02:59.39 | CunningPike | Give bblong no more to drink |
| 03:09.57 | luke-jr | lol |
| 03:12.53 | coppice | luke-jr: I think your experience with sphinx reflects that it has only ever been a research tool. It really needs someone to take up the gauntlet and try to produce a user product from it |
| 03:13.28 | luke-jr | heh |
| 03:17.20 | luke-jr | julian thinks everything is "dial eight" |
| 03:17.22 | luke-jr | O.o |
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| 03:20.32 | olsen | happy new year everyone |
| 03:21.32 | coppice | that's so last night :-) |
| 03:22.37 | olsen | i'm trying to call my gf with a cellphone and the lines are all saturated, not usable :/ |
| 03:22.49 | olsen | wish i had something better :p |
| 03:22.57 | Strom_C | that's your punishment for only using a cellphone :) |
| 03:24.14 | olsen | yeah, can't wait to get something better *cough* asterisk *cough* :) |
| 03:24.22 | Strom_C | heh |
| 03:25.59 | olsen | but, since she only have a cellphone. i will need to connect my asterisk pbx to a pstn line right? |
| 03:26.28 | Strom_C | or an ITSP |
| 03:27.05 | olsen | i see |
| 03:27.06 | olsen | nice |
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| 03:30.04 | olsen | why does all the lines get saturated in xmas/new year? |
| 03:30.18 | olsen | i mean, technically... |
| 03:30.36 | olsen | too much connections? |
| 03:30.45 | russellb | yes :) |
| 03:30.48 | Strom_C | because the phone network is designed to handle average peak load, not atypical peak load as occurs on Mother's Day, Christmas, and New Years |
| 03:31.18 | Strom_C | the actual wireline network handles it much better than the mobile network |
| 03:31.38 | russellb | cell phones are great for lowering the standards of the public. |
| 03:31.47 | Strom_C | mobile phones are a convenience device to be used while away from a real phone - not to be used as a primary phone :) |
| 03:32.25 | jets | ya if it weren't for cellphones my team would need to have better voip quality to our landline subscribers hahahahahahahahaha |
| 03:32.43 | russellb | exactly |
| 03:34.01 | coppice | Basic rule of any audio visual system: Most people are nearly deaf and nearly blind :-) |
| 03:34.04 | luke-jr | file (or anyone): You said that I wouldn't find anything earlier-- did you mean arbitrary English speech recognition in general, or that none of it is open source? |
| 03:36.31 | olsen | Strom_C: i see |
| 03:42.04 | coppice | untrained recognitiion of arbitrary english is not available at any price |
| 03:44.37 | Strom_C | coppice: yes it is. it's called an operator :) |
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| 03:47.04 | Nivex | Strom_C: that's training. it's just a human brain that has been trained, not a computer. |
| 03:48.01 | Strom_C | Nivex: it may be training, but the operator can still understand expressions that are far more arbitrary than a speech recognition engine can |
| 03:48.13 | Nivex | true |
| 03:49.45 | olsen | lol i get "connection error" when i try to call from one cellphone company to another. |
| 03:50.23 | olsen | s/company/provider |
| 03:50.32 | olsen | i wonder if would be better if i call from my pstn |
| 03:50.37 | Strom_C | probably |
| 03:50.57 | Strom_C | olsen: where are you? |
| 03:51.22 | olsen | Strom_C: paraguay. |
| 03:51.40 | Strom_C | cool |
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| 03:54.19 | coppice | a human operator may confuse "no" with "now", but unlike a speech recognition system tends not to confuse it with "yes, most definitely" :-) |
| 03:55.01 | Strom_C | I think confusing "no" and "know" is more likely than "no" and "now" |
| 03:55.36 | perd | what do you guys use for speech recognition? i just bought lumenbox |
| 03:55.39 | perd | lumenvox |
| 03:55.42 | rob0 | I knew a new gnu. |
| 03:55.56 | coppice | no and know sound the same, so both machine and human can only separate them by context |
| 03:56.24 | olsen | Strom_C: well, not really. our internet insfraestructure really sucks. |
| 03:56.47 | olsen | Strom_C: there is a monopoly with the internet here. |
| 03:57.12 | coppice | we have a special offer this week in asia, where our internet infrastructure sucks too :-) |
| 03:57.51 | olsen | Strom_C: brazil or argentina sells internet to a telephony company here and ISP's are kinda "forced" to get internet only from there. |
| 03:58.41 | olsen | Strom_C: and the telephony company resells internet very expensive, so ISP's sells crappy internet and expensive. |
| 03:59.17 | olsen | i hope that would change this year for good. |
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| 04:00.57 | mistermocha | happy new year everyone! |
| 04:01.26 | perd | yay |
| 04:01.35 | mistermocha | word |
| 04:01.44 | Nivex | you must be UTC-4 |
| 04:01.52 | mistermocha | UTC -8 |
| 04:02.13 | mistermocha | good ol' california |
| 04:02.17 | Nivex | oh, just being early then :) |
| 04:02.29 | mistermocha | doing some last minute troubleshooting before I head out to the party |
| 04:02.51 | mistermocha | before I go nuts on this one, I wanted some mild validation... |
| 04:03.18 | mistermocha | my voip line stopped receiving dtmf signals in-call, but my analog lines work just fine. |
| 04:03.36 | Strom_C | SIP? |
| 04:03.37 | rob0 | ~/me mildly validates mistermocha for New Year's |
| 04:03.48 | rob0 | oops |
| 04:04.15 | mistermocha | I've set dtmf mode to inband, rfc2833, and info, none of them work |
| 04:04.20 | mistermocha | yes, SIP |
| 04:04.28 | coppice | I can validate that it is a new year, but its getting older all the time |
| 04:04.56 | rob0 | It's still newer than any other year we've seen. |
| 04:05.33 | coppice | time to see if a 2007 lunch tastes any different from a 2006 lunch |
| 04:09.15 | Nivex | so is there a MeetMe equivalent to this channel? |
| 04:09.29 | Nivex | we could all actually talk to each other! |
| 04:09.36 | russellb | Nivex: sometimes there is, heh |
| 04:09.36 | Nivex | or am I being overly optimistic again? |
| 04:09.41 | russellb | but i don't think one exists right now ... |
| 04:09.53 | russellb | file: we should put one on iaxtel |
| 04:10.07 | russellb | hm. |
| 04:10.59 | Nivex | if we're feeling lazy we could always use sipphone's conference. it's pretty braindead |
| 04:11.16 | russellb | well, we like to load iaxtel, anyway |
| 04:11.21 | russellb | we use it to test things |
| 04:14.18 | Strom_C | latency galore? |
| 04:14.18 | file | russellb: can't without upgrading the kernel or getting someone to go out and install hardware |
| 04:14.36 | russellb | file: oh that's right. we've discussed this before, probably more than once :-p |
| 04:20.31 | russellb | la la la ... |
| 04:20.51 | Nivex | file: this build you left me with sure is chatty :) |
| 04:30.36 | coppice | Ahh, the wonders of TTS... "I can get the right emphasis on "A tree grows in Brooklyn", but I make a complete ballsup of most things I read" :-) |
| 04:32.12 | Nivex | ...or maybe I need to shut debug off :) |
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| 04:53.17 | maclink | hello all ... |
| 04:54.20 | maclink | i need a litle help, with extension.conf |
| 04:55.29 | maclink | where found one manual about this ... |
| 04:55.49 | maclink | thanks |
| 05:00.42 | billf | look at voip-info.org |
| 05:15.41 | *** join/#asterisk obnauticus (n=aao_pwne@c-24-21-116-29.hsd1.mn.comcast.net) |
| 05:17.08 | Rawplayer | happy new year everyone!!! |
| 05:17.33 | russellb | yay |
| 05:17.35 | russellb | 2007 pwns |
| 05:17.49 | shellshark | it's still 2006 here ;) |
| 05:19.09 | *** join/#asterisk mega (n=mega@217.201.69.30) |
| 05:20.39 | Nivex | funny, it doesn't feel any different than last year |
| 05:20.54 | russellb | i dunno ... after all those snacks i ate tonight |
| 05:20.56 | russellb | i feel a bit different. |
| 05:21.31 | Nivex | in fact, this feels about like any other night. |
| 05:21.47 | Nivex | Guess I stay up late too much |
| 05:21.53 | Nivex | oh well, I'm off to bed. |
| 05:55.41 | SwK | its still 2006 here roo |
| 05:55.52 | rob0 | not for long! |
| 05:58.59 | file | uh oh, another phone call! |
| 06:00.06 | perd | talk dirty to them |
| 06:00.08 | perd | see what happens |
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| 07:28.33 | *** join/#asterisk WillySilly (n=WillySil@archlinux/trusteduser/WillySilly) |
| 07:29.21 | WillySilly | Anyone care to help with a compile issue I am having with 1.4.0? |
| 07:31.57 | vlrk | what is the error you are getting ? |
| 07:32.10 | WillySilly | http://pastebin.archlinux.org/669 |
| 07:37.03 | tzafrir_laptop | WillySilly, this is the editline subdurectory. It has its own configure script |
| 07:38.55 | WillySilly | And? |
| 07:53.34 | *** join/#asterisk saftsack (n=oliver@p54A7C70B.dip.t-dialin.net) |
| 08:02.49 | *** part/#asterisk WillySilly (n=WillySil@archlinux/trusteduser/WillySilly) |
| 08:11.57 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 08:13.05 | linagee | <PROTECTED> |
| 08:14.41 | SwK | s,n,HangUp() |
| 08:14.55 | dlynes_laptop | Happy new year's, peeps! |
| 08:15.24 | *** join/#asterisk saftsack (n=oliver@p54A7F8D6.dip.t-dialin.net) |
| 08:24.03 | perd | i touched myself |
| 08:24.11 | perd | and cheered |
| 08:34.24 | *** part/#asterisk vlrk (n=vlrk@202.65.134.119) |
| 09:08.52 | mitcheloc | Happy New Year? |
| 09:09.47 | shellshark | happy new year :) |
| 09:10.01 | shellshark | what's a good book for preparing me for DCAP certification? |
| 09:10.58 | mitcheloc | ~book |
| 09:11.00 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:11.03 | mitcheloc | that one? |
| 09:11.19 | shellshark | can i buy it in a printed form? |
| 09:11.28 | mitcheloc | yes |
| 09:11.28 | perd | hat thing just gave me a raging boner |
| 09:11.42 | perd | telephony is awesome |
| 09:11.56 | mitcheloc | perd: you have quite the mouth i see |
| 09:12.25 | perd | i'm poetic, people tell me |
| 09:16.17 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 09:44.11 | shellshark | in my dialplan, how can i match just 2-6? |
| 09:50.47 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 09:55.21 | `Sean | w00t |
| 09:55.29 | `Sean | err tomorow is mondya lol |
| 09:55.41 | `Sean | Happy 07 to everyone :p |
| 09:55.43 | *** join/#asterisk saftsack (n=oliver@p54A7F19F.dip.t-dialin.net) |
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| 10:39.13 | `Sean | Hey, is it possible to set a different context for each channel in zapata.conf |
| 10:39.38 | *** join/#asterisk RahaiL (n=Rahail@209-19-88-240.detroit.mi.D-Conn.net) |
| 10:47.35 | dlynes_laptop | `Sean: yes |
| 10:47.54 | `Sean | ok |
| 10:48.17 | `Sean | Can you tell me how pleasE? |
| 10:48.26 | `Sean | just as same as i would do it for global right |
| 10:48.29 | dlynes_laptop | `Sean: it'd be kinda silly if you couldn't...especially if all your zaptel ports are fxs |
| 10:48.35 | `Sean | just that |
| 10:48.35 | `Sean | context=default |
| 10:48.39 | dlynes_laptop | `Sean: yes |
| 10:48.40 | `Sean | under, the specific channel correct |
| 10:48.49 | perd | yes |
| 10:48.54 | `Sean | thanks :) |
| 10:49.01 | `Sean | just wanted to make sure so i dont go about setting it wrong |
| 10:49.15 | dlynes_laptop | `Sean: yeah...zapata.conf doesn't really use 'contexts' |
| 10:49.23 | dlynes_laptop | `Sean: you define all the parameters, and then you define the channels those parameters apply to |
| 10:49.32 | *** join/#asterisk saftsack (n=oliver@p54A7FC3F.dip.t-dialin.net) |
| 10:49.35 | dlynes_laptop | `Sean: and then repeat for every channel group you want to define |
| 10:49.46 | `Sean | so how would i define context lets say context=bb for channel 1 and context=pp for channel 2 |
| 10:50.55 | perd | i need to pp on bb |
| 10:51.01 | perd | she's cute and loves the golden showers |
| 10:51.34 | `Sean | would i define it in the same place |
| 10:51.40 | `Sean | ive defined other stuff for the fxs channels |
| 10:51.40 | `Sean | http://pastebin.ca/298971 |
| 10:51.46 | `Sean | have a look its my zapata.conf |
| 10:56.09 | perd | make channel => the last definition for each channel |
| 10:56.52 | perd | oh yeah, and it's channel => not channel= |
| 10:56.58 | perd | channel => 1 |
| 11:00.58 | tzafrir | `Sean, I'm not sure you understand the parser of zapata.conf |
| 11:01.49 | tzafrir | To affect a channel you have to have a config line above it (but within the channels section) |
| 11:03.00 | tzafrir | perd: channel = should work just the same, actually, IIRC |
| 11:03.27 | tzafrir | = and => are equivalent in the asterisk syntax, IIRC |
| 11:04.06 | `Sean | tzafrir example please? |
| 11:07.11 | `Sean | [6:03am] <tzafrir> To affect a channel you have to have a config line above it (but within the channels section) <<< can you please elaborate i dont quite get that, if you dont mind giving me a example thanks ;) |
| 11:07.43 | tzafrir | `Sean, see http://pastebin.ca/298991 |
| 11:07.46 | `Sean | ok |
| 11:08.20 | tzafrir | Note that context_for_2_and_3 affects both channel 3 and channel 3, as it doesn't get overriden in channel 3 |
| 11:08.56 | `Sean | I see so can i not have one context for one channel? |
| 11:09.27 | tzafrir | `Sean, sure you can. Just add a context= line between channel 2 and channel 3 |
| 11:09.42 | `Sean | ahh Alright thanks :) |
| 11:09.56 | `Sean | btw, i never knew about the arrow having to be ther till now but thanks for telling me :) |
| 11:10.06 | `Sean | thanks alot for your help :) |
| 11:10.17 | tzafrir | As I said, it does not have to be an arrow |
| 11:11.52 | `Sean | [5:58am] <perd> oh yeah, and it's channel => not channel= |
| 11:11.59 | `Sean | he says it has to be a => |
| 11:12.08 | `Sean | so its ok if i have just = |
| 11:12.10 | `Sean | and not the arrow? |
| 11:18.28 | *** join/#asterisk tsurk0 (n=tsurko@145-226.go.evo.bg) |
| 11:19.00 | dlynes_laptop | perd: directive = works as well as directive => |
| 11:19.53 | dlynes_laptop | `Sean: correct |
| 11:20.28 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 11:21.33 | *** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 11:22.07 | perd | ah, i was just going by the sample |
| 11:22.14 | perd | never tried it without the |
| 11:22.18 | perd | > |
| 11:32.14 | dlynes_laptop | perd: yeah...works like that in all the asterisk config files |
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| 13:24.12 | RoyK | happy new year |
| 13:25.27 | poller | You to |
| 13:36.47 | vlrk | Happy New Year to All |
| 13:44.55 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
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| 14:18.40 | *** join/#asterisk coppice (n=chatzill@46.155.17.210.dyn.pacific.net.hk) |
| 14:19.05 | RoyK | ROTFL http://english.pravda.ru/main/18/88/354/16724_Condoleezza.html |
| 14:32.51 | Druken | wholly shit... the date changed last night.....! |
| 14:34.40 | coppice | i thought the date changed every night :-\ |
| 14:37.07 | Druken | exactly my point... :) |
| 14:37.21 | Druken | wtf makes that one so special ? |
| 14:41.30 | Druken | people just use it as an excuse for a drinkfest |
| 14:42.39 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 14:43.25 | vlrk | Asterisk use which protocal is it t.38 or t.37 ( Fax over IP ) ? |
| 14:43.50 | RoyK | vlrk: none of them, really. asterisk supports t.38 passthrough, but that's about it |
| 14:44.09 | RoyK | vlrk: see #openpbx if you want t.38 support |
| 14:44.22 | vlrk | I think with spandsp ( app_rxfax ,app_txfac ) |
| 14:44.32 | vlrk | it allow t.37 |
| 14:44.35 | *** join/#asterisk _DAW (n=chatzill@adsl-157-17-126.msy.bellsouth.net) |
| 14:44.53 | RoyK | you can simulate t.37 with spandsp |
| 14:45.10 | RoyK | but you can't fax over ip into an asterisk box |
| 14:45.21 | RoyK | as in SIP ATA <-> asterisk |
| 14:46.17 | RoyK | perhaps, if you can find a t.37 compliant ATA, it might work |
| 14:46.18 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 14:46.40 | vlrk | RoyK: I just read this "http://www.soft-switch.org/foip-with-real-atas.html" article where he says ATA 's are dumb in last sentence ... |
| 14:48.26 | vlrk | i did not got what he mean .... There |
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| 14:59.03 | syamajala | i'm having problems with zaptel |
| 15:00.25 | syamajala | i keep getting ZT_CHANCONFIG failed on channel 4: No such device or address (6) |
| 15:00.32 | syamajala | when i run ztcfg |
| 15:01.27 | syamajala | i have 1 fxo module on channel 4 so i put loadzone = us defaultzone = us and fxsks=4 |
| 15:01.49 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 15:02.54 | mosty | do you have the module loaded for your card? |
| 15:03.08 | syamajala | i loaded the wctdm and zaptel modules |
| 15:03.37 | syamajala | i have a tdm400p |
| 15:04.00 | syamajala | i'm using zaptel 1.4.0 and 2.6.19 |
| 15:11.46 | tzafrir | syamajala, what do you see on cat /proc/zaptel/* (pastebin, please) |
| 15:12.06 | syamajala | there is nothing there |
| 15:12.11 | syamajala | the zaptel dir is empty |
| 15:12.26 | tzafrir | ah, so the module wctdm is probably not loaded |
| 15:12.39 | tzafrir | the zaptel dir exists, so the module zaptel is loaded |
| 15:12.46 | tzafrir | lsmod | grep zaptel |
| 15:12.53 | tzafrir | (can be pasted here) |
| 15:13.08 | syamajala | zaptel 150692 1 wctdm |
| 15:13.48 | tzafrir | wctdm did not identify the card? What do you see on the kernel logs (or dmesg) after loading wctdm? |
| 15:15.24 | Corydon76-home | Also, did you plug in the Molex connector to the card? |
| 15:15.38 | syamajala | yeah |
| 15:15.49 | syamajala | http://pastebin.com/848962 |
| 15:15.52 | syamajala | thats what it says |
| 15:17.10 | Corydon76-home | Boy, it's slow today |
| 15:18.29 | syamajala | there do seem to be some other messages in dmesg about probing for pci cards |
| 15:18.44 | syamajala | * Found PM-Timer Bug on the chipset. Due to workarounds for a bug, this clock source is slow. Consider trying other clock sources |
| 15:20.25 | syamajala | there was also this PCI: No IRQ known for interrupt pin D of device 0000:00:07.2. Please try using pci=biosirq. |
| 15:20.47 | syamajala | that showed up after it loaded the uhci module |
| 15:22.43 | *** join/#asterisk littleball (n=littleba@bb220-255-155-68.singnet.com.sg) |
| 15:23.03 | syamajala | also i enabled config_calc_xlaw and config_zaptel_mmx in zconfig.h when i was building it |
| 15:27.08 | littleball | hello, i have built one billing system which can bill sip call, call through and callback. With GSM modules also available. who is interested? |
| 15:27.52 | syamajala | should i try enabling watchdog support to see if i can get more info? |
| 15:29.12 | Corydon76-home | syamajala: it does not appear that wctdm is actually loaded |
| 15:29.24 | syamajala | hmm ok |
| 15:29.40 | syamajala | should i use insmod? |
| 15:30.55 | Corydon76-home | Use modprobe |
| 15:31.58 | olsen | damn slashdot comments are nothing but FUD. |
| 15:34.00 | syamajala | Corydon76-home: ztcfg is still safing the same thing |
| 15:34.49 | Corydon76-home | syamajala: you'll need to contact Digium tech support |
| 15:35.08 | syamajala | ok |
| 15:35.19 | syamajala | could it be a hardware problem? |
| 15:38.11 | Corydon76-home | It could be many things |
| 15:38.23 | syamajala | ok |
| 15:52.07 | *** join/#asterisk GlobeTrotter (i=GlobbeTo@200.122.149.56) |
| 15:52.52 | GlobeTrotter | Haapy new year to all :) |
| 15:53.27 | GlobeTrotter | i know that's the first of the year but i need some help with G729 :( |
| 15:56.11 | GlobeTrotter | loader.c: /usr/lib64/asterisk/modules/codec_g729a.so: undefined symbol: ast_module_register |
| 15:56.22 | GlobeTrotter | <PROTECTED> |
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| 16:11.34 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
| 16:14.40 | tzafrir | GlobeTrotter, probablly module not for the right version of Asterisk. Which version of Asterisk do you have? |
| 16:14.47 | *** join/#asterisk Ebola (n=Ebola@host86-136-134-136.range86-136.btcentralplus.com) |
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| 16:39.01 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 16:43.44 | *** join/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 16:43.48 | drfreeze | Hello |
| 16:44.04 | drfreeze | can someone explain what exactly n+101 does inside a dial plan? |
| 16:51.35 | poller | Everybody hides! |
| 16:59.35 | RoyK | drfreeze: it's 1.0 stuff. don't think about it |
| 17:00.32 | drfreeze | RoyK: k |
| 17:00.45 | drfreeze | RoyK: what should I be thinking about? |
| 17:01.03 | RoyK | ~sex |
| 17:01.13 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 17:01.13 | RoyK | jbot: ping |
| 17:01.15 | jbot | pong |
| 17:01.40 | drfreeze | RoyK: won't help |
| 17:02.24 | drfreeze | What is the default password on the voicemail? I just created a voicemail account, and can't get access to it because none of the passwords I am trying work |
| 17:02.27 | RoyK | drfreeze: the 1.0 logic was that in case of BUSY or errors, jump to n+101, meaning if you're on priority 5, jump to 106 |
| 17:02.41 | RoyK | ~docs |
| 17:02.42 | jbot | extra, extra, read all about it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation 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 |
| 17:02.42 | RoyK | ~book |
| 17:02.44 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:02.45 | RoyK | ~rtfm |
| 17:02.46 | jbot | rtfm is probably Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 17:02.53 | *** join/#asterisk tsurk0 (n=tsurko@145-226.go.evo.bg) |
| 17:03.50 | [TK]D-Fender | drfreeze : there is no "default" password for voicemails. Just look at what you set in voicemail.conf for that box. |
| 17:04.35 | drfreeze | [TK]D-Fender: that is what I thought, but I keep getting login incorrect |
| 17:04.37 | drfreeze | hmmm..... |
| 17:06.12 | drfreeze | ok, found the problem |
| 17:06.19 | [TK]D-Fender | drfreeze : DTMF? |
| 17:06.43 | drfreeze | partially misconfigured voicemail.conf file |
| 17:06.51 | [TK]D-Fender | drfreeze : I've seen a number of people get everything right but not relaize the app wasn't getting digits from the call. |
| 17:09.51 | *** join/#asterisk tsurk0 (n=tsurko@145-226.go.evo.bg) |
| 17:09.56 | *** join/#asterisk l1nux (n=moi@jof81-1-82-245-67-40.fbx.proxad.net) |
| 17:10.03 | l1nux | hi |
| 17:10.13 | l1nux | happy new year :) |
| 17:11.10 | *** join/#asterisk tsurk0 (n=tsurko@145-226.go.evo.bg) |
| 17:12.57 | *** join/#asterisk tsurk0 (n=tsurko@145-226.go.evo.bg) |
| 17:15.25 | *** join/#asterisk GlobeTrotter (i=GlobbeTo@200.122.149.56) |
| 17:15.43 | GlobeTrotter | i have Asterisk Bussiness edition |
| 17:16.54 | russellb | GlobeTrotter: then most likely whatever question you have, you need to call Digium tech support :) |
| 17:17.23 | GlobeTrotter | i know,, butnoone is available now :( |
| 17:17.36 | GlobeTrotter | can you point me in the right direction please |
| 17:23.27 | *** join/#asterisk EricDGlobeTroote (n=eric@200.122.149.57) |
| 17:24.58 | EricDGlobeTroote | I have problems getting g729 to work,, asfter installing it,, i restarted *.. then it crashed |
| 17:24.59 | EricDGlobeTroote | loader.c: /usr/lib64/asterisk/modules/codec_g729a.so: undefined symbol: ast_module_register |
| 17:25.09 | EricDGlobeTroote | <PROTECTED> |
| 17:25.21 | EricDGlobeTroote | can someone please point me in the right direction |
| 17:25.23 | EricDGlobeTroote | ?? |
| 17:25.27 | EricDGlobeTroote | please :( |
| 17:27.35 | *** join/#asterisk stephane_ (n=stephane@zafnib.cabale.net) |
| 17:28.00 | EricDGlobeTroote | i have dual xeon,, i used ftp://ftp.digium.com/pub/asterisk/g729/asterisk-1.4/64-bit/nocona/ |
| 17:28.28 | *** part/#asterisk stephane_ (n=stephane@zafnib.cabale.net) |
| 17:28.49 | wunderkin | well, that says asterisk 1.4, not abe.. there is probably another version for whatever version of abe you are using |
| 17:29.29 | *** join/#asterisk stephane_ (n=stephane@zafnib.cabale.net) |
| 17:29.47 | russellb | EricDGlobeTroote: you can try the 1.2 based g729 module, that will probably work |
| 17:30.05 | *** part/#asterisk misc-- (n=misc@203.87.183.134) |
| 17:30.07 | EricDGlobeTroote | thanks,, gonna try it now |
| 17:30.10 | EricDGlobeTroote | be rightback |
| 17:41.34 | *** join/#asterisk saftsack (n=oliver@p54A7D32C.dip.t-dialin.net) |
| 17:43.42 | l1nux | jabber not work in asterisk 1.4 ? |
| 17:44.59 | l1nux | res_jabber.c:1505 aji_recv_loop: JABBER: socket read error |
| 17:48.34 | *** join/#asterisk zapata (n=user@chello213047080026.4.14.vie.surfer.at) |
| 17:50.28 | EricDGlobeTroote | <russellb> :: thnaks for your help,, that did it.. |
| 17:52.58 | russellb | EricDGlobeTroote: cool. |
| 17:53.06 | *** join/#asterisk saftsack (n=oliver@p54A7E491.dip.t-dialin.net) |
| 17:59.15 | *** join/#asterisk Teeli (n=tili@172.Red-88-0-147.dynamicIP.rima-tde.net) |
| 17:59.22 | *** part/#asterisk l1nux (n=moi@jof81-1-82-245-67-40.fbx.proxad.net) |
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| 18:23.10 | *** join/#asterisk saftsack (n=oliver@p54A7F32B.dip.t-dialin.net) |
| 18:33.01 | *** part/#asterisk jebba (n=jebba@201-212-163-95.net.prima.net.ar) |
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| 18:50.47 | *** join/#asterisk isladelobos (n=username@73.Red-83-40-196.dynamicIP.rima-tde.net) |
| 18:54.09 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-63-20.red.bezeqint.net) |
| 19:01.48 | *** join/#asterisk markit (n=konversa@81.72.245.119) |
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| 19:03.36 | *** join/#asterisk AJaymn (i=TJ@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 19:04.47 | *** part/#asterisk isladelobos (n=username@73.Red-83-40-196.dynamicIP.rima-tde.net) |
| 19:08.36 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 19:13.34 | EricDGlobeTroote | is there a gui for Asterisk Bussines edition.. other than asterisknow? |
| 19:14.22 | Supaplex | xterm not fancy enough for you? ;) |
| 19:14.25 | poller | Is AsteriskNOW any good? |
| 19:14.39 | poller | A gui would be nice. |
| 19:15.29 | Supaplex | I'll stop now :-) |
| 19:16.19 | poller | Ye ye :) |
| 19:17.20 | robl^ | vi + xterm ? |
| 19:17.21 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 19:17.43 | poller | xterm is just a terminal emulator. |
| 19:18.43 | robl^ | running on a gui ;-) |
| 19:18.56 | Supaplex | yup! |
| 19:19.10 | poller | robl^: Yepp, sure is. :) |
| 19:19.23 | poller | But I didn't say anything about xterm. :) |
| 19:19.36 | AJaymn | Ok. Can anyone tell me why.. I just installed 1.2.14 and over night the system locked up. I did a reset and now 1.2.13 svn rev 47264 is back?!?! |
| 19:20.23 | poller | Yepp, it's the reinstall-the-older-system-if-the-newer-one-dosnt-work feature. |
| 19:21.07 | Supaplex | sweet |
| 19:23.40 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 19:27.30 | EricDGlobeTroote | hi,, i am not getting any sound wheni do my echo tet |
| 19:28.58 | EricDGlobeTroote | pawn extension (default, 600, 1) exited non-zero on 'SIP/6400-0069ee50' |
| 19:28.59 | EricDGlobeTroote | <PROTECTED> |
| 19:28.59 | EricDGlobeTroote | <PROTECTED> |
| 19:29.07 | EricDGlobeTroote | i get this but no sound |
| 19:30.04 | poller | Any NAT involved? |
| 19:32.24 | *** join/#asterisk Crashsys (n=kumba@72.187.211.158) |
| 19:32.36 | EricDGlobeTroote | no, i am dialing from a locally connected SIP phone |
| 19:33.25 | Crashsys | How do I enable a DND option on a polycom's display? Like right now my Ip430 says "New Call" and "Fwd" on the context display... |
| 19:43.16 | *** part/#asterisk AJaymn (i=TJ@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 19:48.40 | [TK]D-Fender | EricDGlobeTroote : Nowhere does that say you actually started an echo test. |
| 19:49.08 | [TK]D-Fender | Crashsys : Can't really mod the soft-keys, but you can remap a hardkey for it |
| 19:49.31 | [TK]D-Fender | Crashsys : I'dd suggest remapping the "Hold" key personally. |
| 19:49.42 | [TK]D-Fender | Crashsys : Its inherently worthless anywhays |
| 19:49.44 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 19:53.38 | Crashsys | Hmmm... |
| 19:54.33 | Crashsys | Can you map the hold key where if they aren't on a call, it sets DND, but it they are, it just sets them on hold? |
| 19:54.40 | Crashsys | or is it one or the other? |
| 19:57.27 | *** join/#asterisk isladelobos (n=username@73.Red-83-40-196.dynamicIP.rima-tde.net) |
| 19:57.45 | *** join/#asterisk intralanman (n=lanman@pool-71-253-204-107.nrflva.east.verizon.net) |
| 19:57.52 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 19:59.45 | *** join/#asterisk infosys24 (n=infosys2@c-69-139-105-17.hsd1.wv.comcast.net) |
| 20:00.21 | infosys24 | anyone ever tried VPA (virtual private asterisk)> |
| 20:04.00 | SwK | you mean like running asterisk under VPS? |
| 20:04.13 | infosys24 | yeah or vserver |
| 20:04.26 | infosys24 | i saw where they were calling that VPA |
| 20:04.29 | SwK | it works but performance sucks |
| 20:04.37 | [TK]D-Fender | Crashsys : Fixed function only, sorry |
| 20:04.42 | Crashsys | I know of a guy who has supposively done it... says it works for him... |
| 20:05.00 | Crashsys | d-fender: Ohh well... I'll just tell 'em to menu-->features-->DND |
| 20:05.03 | SwK | I've tried it before and it does work, it just doesnt work as well as just running asterisk natively |
| 20:05.06 | Crashsys | not like they'll ever use it anyways |
| 20:05.20 | [TK]D-Fender | Crashsys : exactly |
| 20:05.35 | infosys24 | where does the main performance loss occur? |
| 20:05.44 | Crashsys | Theyw ill use hold more then DND... as they put the customer on hold and think up an excuse why they dont have their product :D |
| 20:06.04 | Crashsys | running * virtualized only works if your not handling audio right? |
| 20:06.24 | SwK | no that has nothing to do with it |
| 20:06.38 | SwK | it has to do with cpu time and locking |
| 20:06.44 | [TK]D-Fender | Crashsys : Hold is always available as a soft-key hence having a hard-key for it is wasteful |
| 20:06.46 | Crashsys | ahhh |
| 20:06.56 | infosys24 | oh |
| 20:07.03 | [TK]D-Fender | Crashsys : also starting any other call automatically puts the current call on holod anyways |
| 20:07.27 | SwK | I like having my hard hold key on my polys |
| 20:07.28 | [TK]D-Fender | Crashsys : or any of another 1/2 dozen different ways :) |
| 20:07.29 | SwK | theres a DND key on mine also |
| 20:07.42 | Crashsys | d-fender: Yeah, but i'll just leave it alone... sounds like work to remap it :D |
| 20:07.45 | [TK]D-Fender | SwK : Talking about the IP 430 specifically. |
| 20:07.56 | Crashsys | I was hoping for the "Change this value from 0 to 1" answer... |
| 20:07.58 | SwK | theres not a hard hold key on it? |
| 20:08.03 | [TK]D-Fender | Crashsys : nnot much, but DND is rarely needed |
| 20:08.07 | SwK | I really need to order one of those heh |
| 20:08.10 | SwK | (i should have one) |
| 20:08.16 | Crashsys | SwK: There is no DND key (soft or hard) on an IP430 |
| 20:08.21 | SwK | w0w |
| 20:08.26 | SwK | thatsafu |
| 20:08.28 | [TK]D-Fender | SwK ; there IS a hard HOLD key, but he'd like a hard DND ke instead |
| 20:08.28 | Crashsys | and I was curious if there was a quick-fix to enable a DND Softkey at idle |
| 20:08.45 | [TK]D-Fender | you can still get to DND, jsut 2 extra steps. |
| 20:08.56 | Crashsys | Yeah, i'll just tell them that... not like they will ever use it... |
| 20:08.59 | SwK | crashsys: you could do that with a speeddial key and some dialplan magic |
| 20:09.11 | Crashsys | DND is really only useful with round robin anyways... |
| 20:09.19 | Crashsys | and only to make it move quicker |
| 20:09.29 | Crashsys | assuming yer not using agents :) |
| 20:10.29 | Crashsys | I'll have enough fun teaching them how to park calls instead... |
| 20:10.37 | Crashsys | since you can no longer tell someone "Pick up line 1" |
| 20:14.13 | Crashsys | well, not trivially anyways... |
| 20:14.35 | [TK]D-Fender | Crashsys : Parking != rocket science. |
| 20:14.47 | Crashsys | d-fendeR: You wouldn't think so :) |
| 20:15.01 | robl^ | Crashsys: mape a speed dial labeled "Line 1" and assign it 701 ;-) |
| 20:15.10 | *** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc) |
| 20:15.23 | Crashsys | robl: But then they'll hit that to dial out instead of the linekey |
| 20:15.45 | Crashsys | Receptionists do the craziest things! |
| 20:15.47 | *** join/#asterisk stephane_ (n=stephane@zafnib.cabale.net) |
| 20:15.48 | robl^ | silly users |
| 20:15.54 | Crashsys | Sounds like a Fox show |
| 20:16.32 | robl^ | Crashsys: I know! we have a receptionist that can barely figure out how to transfer calls |
| 20:16.58 | Crashsys | parking seems easier then transferring calls to me... |
| 20:17.32 | [TK]D-Fender | Crashsys : Can't imaging that. |
| 20:17.58 | Crashsys | The receptionist that works for one company I work for wrote "Happy Halida" in the christmas card for the owner... |
| 20:18.48 | Crashsys | and now i've gotta explain how an attended transfer allows them to talk to the other person first, whereas a blind sends the call with the CID info... |
| 20:19.18 | *** join/#asterisk RoyK (n=roy@217-175-222.100710.adsl.tele2.no) |
| 20:19.23 | robl^ | depends on the setup. Like a Snom or Aastra phone. They let you set a key type has Park (or Parking Orbit). One touch. Other phones you have to transfer to 700 (or what ever you have assigned to park) |
| 20:19.24 | Crashsys | I should start refusing to make receptionist ext's and just do AA |
| 20:19.44 | Crashsys | My polycom is setup to transfer 700 |
| 20:20.09 | Crashsys | Attended Transfer to 700, listen to it tell you the parking #, then hit transfer... |
| 20:20.35 | Crashsys | dunno why anyone would wanna use parkandannounce... |
| 20:20.41 | Crashsys | maybe if you were on an analog phone you would... |
| 20:22.02 | robl^ | on mine, I use 1.4's new hints for parking.. so I can get status lights for the parking slots. seems to be well liked change |
| 20:22.23 | Crashsys | I'm still using 1.2.14 |
| 20:22.32 | Crashsys | Haven't even downloaded 1.4 yet |
| 20:26.06 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 20:26.58 | *** part/#asterisk infosys24 (n=infosys2@c-69-139-105-17.hsd1.wv.comcast.net) |
| 20:33.06 | robl^ | Crashsys: the problem is that many companies see the receptionist as the lowliest white collar job. They don't pay or seek anyone with the IQ to operate a phone or even send an email. To me, the receptionist is the first impression anyone gets from a company. The receptionist should be professional, intelligent, cordial, and knowledgeable of the operations of the firm. |
| 20:35.09 | Crashsys | Yeah, but no one ever hires someone like that... |
| 20:35.12 | Crashsys | And I agree with you |
| 20:35.52 | robl^ | true! again, the problem is not with Asterisk or the phone. its the company that hires the wrong person for the job |
| 20:37.24 | monsted | it's not like that kind of person is too hard to find |
| 20:38.05 | robl^ | righ.. but most companies only want to pay extremely low wages for a receptionist |
| 20:38.30 | monsted | pity, it's a fairly easy position to fill and not terribly expensive |
| 20:39.27 | luke-jr | Anyone know a good (large vocabulary-- eg, arbitrary English) speech recognition engine that can be used with Asterisk? |
| 20:39.59 | luke-jr | (I don't mind writing a bit of C glue to make an EAGI) |
| 20:53.27 | *** join/#asterisk hoobastooba (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 20:54.45 | EricDGlobeTroote | exten => 500,1,Playback(demo-abouttotry) |
| 20:55.00 | hoobastooba | is it possible to change the way the voicemail works? for example... is it possible to make it so that once you press 7 to delete a message, it just automatically goes to the next message rather than having to press another button to move to the next message? |
| 20:55.09 | EricDGlobeTroote | when i dial 500 i dont hear the file playing |
| 20:55.49 | hoobastooba | EricDGlobeTroote: what does the CLI> say when this is all happening? |
| 20:56.25 | EricDGlobeTroote | - Executing Playback("SIP/6400-0069ee50", "demo-abouttotry") in new stack |
| 20:56.26 | EricDGlobeTroote | <PROTECTED> |
| 20:57.22 | hoobastooba | have you tried any other files? for example Playback(tt-monkeys) |
| 20:57.31 | hoobastooba | something you know works? |
| 20:57.51 | EricDGlobeTroote | i tried differnt files and none works |
| 20:58.31 | EricDGlobeTroote | fyi, the files that are in /var/lib/asterisk/sounds/ are all from the initial install |
| 20:58.34 | hoobastooba | tell me more about your configuration? how are you dialing 500... zap device? |
| 20:58.56 | hoobastooba | do you have zap hardware in the server? |
| 20:59.22 | EricDGlobeTroote | i have zap hardware,, but i am trying it from a sip phone |
| 20:59.31 | hoobastooba | is the zap hardware configured? |
| 20:59.38 | EricDGlobeTroote | no |
| 20:59.39 | hoobastooba | have you run ztcfg -vv |
| 20:59.43 | hoobastooba | that is your issue |
| 20:59.50 | hoobastooba | once you configure it, it will work how you want it to |
| 20:59.56 | hoobastooba | i had this same problem. |
| 21:00.01 | EricDGlobeTroote | cool,, gonna try now |
| 21:00.02 | EricDGlobeTroote | brb |
| 21:03.31 | hoobastooba | EricDGlobeTroote: any luck? |
| 21:04.18 | EricDGlobeTroote | i am configuring it now.. or maybe i should remove it fromth box first? |
| 21:04.21 | EricDGlobeTroote | what do youthink? |
| 21:04.38 | hoobastooba | configure it... if you have it, it is a good choice for timing solution. |
| 21:04.59 | hoobastooba | or if you want, remove it and use zttummy |
| 21:05.22 | EricDGlobeTroote | cool,, i am going to finish confgiguring it |
| 21:05.24 | EricDGlobeTroote | brr |
| 21:05.26 | EricDGlobeTroote | brb |
| 21:05.28 | EricDGlobeTroote | thanks alot |
| 21:07.53 | syamajala | Corydon76-home: hello |
| 21:08.07 | luke-jr | EricDGlobeTroote: 500 => Playback(demo-abouttotry); |
| 21:08.54 | hoobastooba | luke-jr:his dial statement works, he cannot hear anything when it plays. |
| 21:08.55 | EricDGlobeTroote | use that command instead? |
| 21:08.56 | *** join/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 21:09.27 | hoobastooba | luke-jr: what is the ";" for? |
| 21:09.37 | luke-jr | end of the statement |
| 21:09.46 | hoobastooba | in asterisk? |
| 21:09.50 | luke-jr | yeah |
| 21:09.56 | luke-jr | well, in the dialplan |
| 21:10.01 | hoobastooba | doesnt need an end of statement. |
| 21:10.05 | luke-jr | sure it does |
| 21:10.19 | luke-jr | I don't know about 1.4, but 1.2 would barf if you forgot a semicolon |
| 21:10.35 | luke-jr | well, not quite barf |
| 21:10.43 | luke-jr | it would silently discard something |
| 21:10.56 | hoobastooba | well.. the 500 connections i have in production seem to work just fine without it |
| 21:11.00 | hoobastooba | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 21:11.05 | hoobastooba | says nothing about it |
| 21:11.09 | luke-jr | it's nothing specific to Dial |
| 21:11.30 | EricDGlobeTroote | i have the semiclon at the end of the statement |
| 21:11.38 | hoobastooba | just finished asterisk boot camp and they also did not cover anything about ; |
| 21:11.42 | luke-jr | http://www.voip-info.org/wiki-Asterisk+AEL2 |
| 21:11.44 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 21:12.07 | [TK]D-Fender | luke-jr : Quick tip for your perspective. Youa re the only person here who gives a shit about AEL(2) :) |
| 21:12.09 | hoobastooba | oh, you are using AEL2... thats different |
| 21:12.16 | Crashsys | Anyone got any preference as to ALSA or OSS for consol/dsp? |
| 21:12.17 | Joel1978 | LOL |
| 21:12.39 | hoobastooba | EricDGlobeTroote: you are not using AEL.. .do not use the semicolon. |
| 21:12.48 | luke-jr | maybe he should be |
| 21:13.44 | hoobastooba | well... let him figure out first how to install asterisk... then move on to other features. |
| 21:13.49 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 21:13.56 | luke-jr | extensions.conf is obsolete :p |
| 21:14.27 | [TK]D-Fender | luke-jr : Says you..... the rest of this channel begs to differ |
| 21:14.32 | luke-jr | the only reason I even have it anymore is because I'm still on 1.2 and using .conf to workaround AEL1 bugs |
| 21:14.39 | Crashsys | I like my extensions.conf :( |
| 21:14.46 | [TK]D-Fender | luke-jr : All the docs out there are for standard extensions logic. |
| 21:15.29 | hoobastooba | EricDGlobeTroote: so did you remove the semicolon and configure zap? |
| 21:16.03 | EricDGlobeTroote | i am going thru the instrauctions now on how to configure the zap |
| 21:16.09 | hoobastooba | ok |
| 21:16.15 | EricDGlobeTroote | i have two zap interfaces |
| 21:16.25 | Joel1978 | speaking of AEL.... |
| 21:16.28 | hoobastooba | do you understand the ; discussion and why you dont need it? |
| 21:16.55 | Joel1978 | I just did a fresh install of 1.4 and on startup of * i get a whole slew of warnings about AEL and IAX http://pastebin.ca/299691 |
| 21:16.58 | luke-jr | .conf is annoying |
| 21:17.02 | luke-jr | you need to repeat too much |
| 21:17.13 | EricDGlobeTroote | <hoobastooba> i think so,, i need it if i am using AEL and i dont if i am not? |
| 21:17.24 | hoobastooba | simply put, yes |
| 21:17.28 | EricDGlobeTroote | koo |
| 21:17.30 | EricDGlobeTroote | l |
| 21:18.05 | luke-jr | EricDGlobeTroote: you should use AEL if you've ever writ a program |
| 21:18.21 | luke-jr | .conf maybe if you can't handle code |
| 21:18.26 | hoobastooba | thats not true |
| 21:18.36 | EricDGlobeTroote | i know very litle about programmig :( |
| 21:18.48 | Joel1978 | always a place and a time to start learning |
| 21:18.49 | hoobastooba | if you do not want support and do not want easy access to documentation you should use AEL |
| 21:19.05 | luke-jr | pfft |
| 21:19.40 | Crashsys | AEl is like screwing without a condom... if you mess something up your the one to blame :) |
| 21:20.00 | luke-jr | ... |
| 21:20.04 | luke-jr | condoms are immoral |
| 21:20.16 | Crashsys | So is baby momma drama! |
| 21:20.30 | hoobastooba | EricDGlobeTroote: learn asterisk the way digium has documented it... then if you are interested try AEL and form your own opinion on what to use. |
| 21:23.51 | Joel1978 | Is there a reason why I'd be getting all these warnings off a new installation of 1.4? http://pastebin.ca/299691 |
| 21:24.26 | luke-jr | Joel1978: I'd try cleaning out an old install |
| 21:25.32 | Joel1978 | that's why this is a new install.....there was no previous since i just finished installing linux |
| 21:25.44 | Joel1978 | so im confused why i have all the warnings using the sample confs |
| 21:26.15 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-216-111.dsl.irvnca.pacbell.net) |
| 21:26.33 | syamajala | Corydon76-home: i tried a different system and it seems to work |
| 21:32.54 | Crashsys | Warnings are just god's way of saying you need improvement... |
| 21:32.57 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 21:33.04 | Joel1978 | haha |
| 21:33.34 | Crashsys | It's not a Bug... it's a Feature... |
| 21:33.48 | Joel1978 | yes....thanks to M$ |
| 21:33.54 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust337.leed.cable.ntl.com) |
| 21:38.46 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 21:42.01 | Crashsys | Guess i'll try ALSA for console/dsp |
| 21:44.43 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 21:46.07 | drfreeze | Anyway to debug if asterisk voicemail is sending an attachment or not? |
| 21:48.21 | *** join/#asterisk rishaan (n=rishaan@c-67-160-249-95.hsd1.ca.comcast.net) |
| 21:48.29 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 21:50.25 | rishaan | Folks, I am going nuts trying to figure out how to set the outbound callerID? I have tried all kinds of stuff. SetVar(CALLERIDNUM=anonymous), Set(CALLERID(all)=Unknown <4084042345>), SetCIDNum("4084042345"), SetCIDNum("unknown") |
| 21:50.52 | rishaan | None of this is working for me. When I receive the call, the CID number is some wierd 700 number? |
| 21:51.11 | rishaan | Any help on what am I doing wrong? Running Asterisk 1.2 stable SVN version |
| 21:51.48 | rishaan | Anyone out there on this first day of 2007? |
| 21:52.12 | rishaan | am i even connected ??? Could someone please ack me...thanks |
| 21:52.42 | Crashsys | Hmm? |
| 21:53.37 | rishaan | okay... looks like I am in :-) Crashsys, could you read my comments. I am having this wierd issue since I paid for XChat license |
| 21:54.33 | rishaan | :-(... no luck |
| 21:54.50 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 21:55.07 | Crashsys | I dont have a PRI therefore I dont deal with outbound CID... |
| 21:55.10 | monsted | wait? there's an xchat license? |
| 21:55.13 | Crashsys | but it looks like your syntax is wrong... |
| 21:55.24 | rishaan | Xchat for windows is $20 |
| 21:57.04 | monsted | heh |
| 21:57.15 | monsted | screen and irssi for the win |
| 21:58.11 | rishaan | Crashsys, what's wrong with my Syntax....? |
| 21:59.09 | Joel1978 | monsted: they charge a license if you want more than 2 line appearences on xlite i believe |
| 21:59.30 | Joel1978 | oh wait |
| 21:59.36 | Joel1978 | i'm really crossing my wires today |
| 21:59.42 | Joel1978 | forget what i said |
| 21:59.45 | Joel1978 | LOL |
| 22:00.00 | Joel1978 | im a bit hungover still |
| 22:01.57 | Joel1978 | xchat looks nice but i don't like paying for software so pfft!!! |
| 22:02.12 | *** join/#asterisk saftsack (n=saftsack@pD9E07B1C.dip.t-dialin.net) |
| 22:03.27 | drfreeze | I see where I should be able to set db values from the command line using 'database put family key value' |
| 22:03.42 | drfreeze | But, 'database' is not showing up on my machine. |
| 22:03.56 | drfreeze | Is this an extra install I need to do? |
| 22:04.09 | Joel1978 | no there's a core db that comes with * |
| 22:04.12 | Joel1978 | it should just work |
| 22:05.23 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 22:05.46 | drfreeze | maybe they assume this is run inside asterisk -r |
| 22:06.14 | Joel1978 | yes you need to be at a CLI> prompt to do console database commands |
| 22:07.00 | drfreeze | yup, that was it |
| 22:07.04 | rishaan | Crashsys, any hints on syntax error? I am trying to set the outbound CID on the system :-( |
| 22:08.05 | Crashsys | well I dont know about outbound CID, but you set inbound CID with ${CALLERID(num)} = <something> |
| 22:08.32 | Joel1978 | rishaan: try this link http://www.voip-info.org/wiki-Asterisk+cmd+SetCIDNum |
| 22:08.52 | Crashsys | I dont have PRI's therefore i've never looked at setting outbound CID |
| 22:08.54 | rishaan | thanks Joel |
| 22:09.35 | *** join/#asterisk lba (n=lba@user-12lml5g.cable.mindspring.com) |
| 22:10.23 | lba | How to specify To: when using put with tftp to a Grandstream phone? |
| 22:11.05 | lba | Isn't there some way to put the files to update into /tftpboot and have it happen automatically? |
| 22:11.16 | *** join/#asterisk _007 (n=brian@bhlkwihed01-pool0-a31.bhlkwi.tds.net) |
| 22:11.24 | *** join/#asterisk heartones (n=Loveocea@84.36.29.79) |
| 22:11.37 | Joel1978 | iba: what model of grandstream? |
| 22:11.50 | Crashsys | lba: In the web setup for grandstream there should be a spot to specify a tftp boot server |
| 22:12.08 | lba | Joel1978: GXP-2000 and I need to update to firmware .14 |
| 22:13.56 | heartones | I am debating over buying a sipura 3000 verses diguim cars if money not an issue what would be a good choice can any one share with me |
| 22:15.07 | Joel1978 | iba: you might get started looking at this page http://www.voip-info.org/tiki-index.php?page=GXP-2000 |
| 22:15.13 | lba | Joel1978: Yes but I can only specify via tftp server at an IP address or http server which I usually do but it's not working. No place to specify a path |
| 22:16.01 | lba | Joel1978: I'll look at the url at voip-info. Thanks. |
| 22:16.16 | Joel1978 | yup |
| 22:16.56 | lba | Crashsys: I responded to your post incorrectly to Joel1978. The web config page doesn't seem to allow me to specify anything other than an IP address |
| 22:17.28 | Crashsys | did you try 192.168.1.1/pathtofiles/ ? |
| 22:17.36 | Crashsys | dunno if tftp supports paths |
| 22:18.27 | lba | Crashsys: The GS webpage doesn't have a place to put in IP/pathtofiles |
| 22:18.38 | Joel1978 | well using tftp you only need to put them in the tftpboot directory on the server and the phone will pick up the file associated with it's config |
| 22:19.04 | Joel1978 | i only have cisco phones so they're a bit different but same general idea |
| 22:19.20 | Crashsys | I use polycom's so it's just FTP |
| 22:20.12 | lba | Joel1978: I have tftp and will play with that the tftpboot dir. |
| 22:21.05 | Crashsys | Ok... Alsa worked for console :) |
| 22:23.09 | Crashsys | Is ALSA supposed to replace OSS? |
| 22:23.15 | Crashsys | or is it just a competing method? |
| 22:23.59 | lba | Joel1978: When I start tftp it asks me for To: I respond with IP of GS phone. It responds with tftp: and I don't know what it wants. Nothing about this on the man page. |
| 22:30.34 | [hC] | Did asterisk 1.4 bring voicemail attachment volume settings? or is it still a patch? |
| 22:36.28 | perd | anyone know where to download the lumenvox speech tuner at |
| 22:36.42 | perd | i purchased the stupid speech engine lite and they didnt send me info on the speech tuner |
| 22:36.49 | perd | i am sad |
| 22:37.52 | Joel1978 | iba: are you doing this at the command line of a linux box? |
| 22:37.57 | Joel1978 | or windows |
| 22:38.12 | Joel1978 | if so the To field is the address of the server you wish to connect to |
| 22:39.43 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 22:41.15 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 22:41.32 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 22:41.37 | Joel1978 | well crap, i need to go but have a good new year all |
| 22:41.40 | *** part/#asterisk Joel1978 (n=Joel1978@71.39.87.5) |
| 22:42.49 | rob0 | I just bought new tires, but they're Goodrich not Goodyear. |
| 22:45.10 | *** join/#asterisk mega (n=mega@217.201.87.11) |
| 22:56.59 | *** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net) |
| 22:57.19 | jayk- | is there a way to increase the volume level for other parties in a conference call? |
| 22:59.34 | *** join/#asterisk hhhenriq (n=henrique@200.143.84.123) |
| 23:00.17 | hhhenriq | hi all...im new to voip and i need some explanatio...could anyone help me please? |
| 23:01.19 | Strom_M | ~ask |
| 23:01.36 | jbot | i guess ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also http://catb.org/~esr/faqs/smart-questions.html |
| 23:02.49 | hhhenriq | what is a sip provider :D? |
| 23:02.53 | hhhenriq | i mean |
| 23:03.04 | hhhenriq | i installed asterisk in my linux box |
| 23:03.19 | hhhenriq | do i need another kind of server to make a call |
| 23:04.02 | Strom_M | a sip provider is a company that takes your calls using the SIP protocol and provides a gateway to the PSTN |
| 23:04.28 | CunningPike | hhhenriq: You can use asterisk on its own to connect internal phones together. Beyond that, you need some way of connecting to the PSTN |
| 23:04.30 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 23:04.33 | Strom_M | you generally need to use a SIP or IAX provider if you want to place andreceive PSTN calls |
| 23:04.47 | hhhenriq | PSTN? |
| 23:04.55 | CunningPike | ~pstn |
| 23:05.03 | jbot | rumour has it, pstn is Pubic Switched Telephone Network, or "please stop the nonsense" |
| 23:06.04 | hhhenriq | ok...im doing it in a intranet of a company...the asterisk is inside this intranet and all the phones are supposed to also... |
| 23:06.04 | hhhenriq | hehe |
| 23:06.22 | Strom_M | ~hafc |
| 23:06.23 | jbot | hafc is probably hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 23:07.09 | hhhenriq | jbot :`( |
| 23:07.12 | Crashsys | consultants are competant? |
| 23:07.21 | Crashsys | that must be a guideline :) |
| 23:07.25 | Crashsys | some are pretty damn scary |
| 23:07.47 | Strom_M | yeah, but at least those of us who are competent can spell competent correctly ;) |
| 23:07.57 | Crashsys | Exactly!!! |
| 23:08.10 | hhhenriq | jbot i know what you mean but my boss does not |
| 23:08.12 | jbot | You know what you mean but my boss does not? |
| 23:08.44 | hhhenriq | strom_M |
| 23:08.54 | Strom_M | yes? |
| 23:09.12 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 23:09.25 | hhhenriq | could you have a talk with my boss? |
| 23:09.35 | hhhenriq | maybe he will understand it |
| 23:09.42 | Strom_M | where are you located? |
| 23:09.48 | hhhenriq | Brasil... |
| 23:09.52 | Crashsys | hhhenriq: Cant you just say "I dont know how to do this"? |
| 23:10.13 | Strom_M | that's just a little bit outside of my jurisdiction...plus, I don't speak portuguese |
| 23:10.37 | hhhenriq | Crashsys: ok... |
| 23:10.46 | Strom_M | er, portugese |
| 23:11.04 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 23:11.06 | hhhenriq | but it's a very basic intranet voip server |
| 23:11.14 | Crashsys | Unless you want Strom to consult for ya... |
| 23:11.48 | Strom_M | yeah, I'll be happy to do everything for you at my low low rate of contact-me-in-PM-for-details per hour |
| 23:12.09 | Crashsys | Careful henriq, strom is not only famous, he's infamous... |
| 23:12.49 | hhhenriq | so im supposed to stop thinking in doing a intranet voip server... |
| 23:13.12 | Crashsys | So like, one half of me tells to drive to the install location, hook this box up to some real PSTN lines, and do a field test... while the other half of me says "Bah, it'll be fine! Go to bed"... |
| 23:13.15 | Crashsys | decisions decisions |
| 23:14.03 | hhhenriq | im going to bed |
| 23:14.18 | Crashsys | henriq: No, but if you dont have the knowledge to do it... then have someone help/do it for you... |
| 23:14.35 | hhhenriq | im asking for help... |
| 23:14.56 | Crashsys | You've already read through the sections at www.voip-info.org ? |
| 23:15.36 | CunningPike | hhhenriq: Try The Book |
| 23:15.39 | CunningPike | ~thebook |
| 23:15.50 | jbot | hmm... thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:15.50 | Crashsys | or the book |
| 23:15.50 | hhhenriq | Crashsys: no i just started this voip task.. |
| 23:16.00 | CunningPike | hhhenriq: Read that book first |
| 23:16.23 | Crashsys | Start with that book... a lot of people recommend it... |
| 23:17.48 | hhhenriq | Crashsys: could you read it for me during im trying to sleep? |
| 23:17.56 | Crashsys | naaa |
| 23:18.00 | Crashsys | I never read the book |
| 23:18.14 | Crashsys | I just installed it and started hacking the .conf's |
| 23:18.26 | hhhenriq | Crashsys: thats what im doing |
| 23:18.33 | *** join/#asterisk backblue (n=moo@87-196-11-143.net.novis.pt) |
| 23:18.39 | Crashsys | then get real familiar with www.voip-info.org |
| 23:19.22 | hhhenriq | ok...have you read this one at least? |
| 23:19.51 | Crashsys | read what? |
| 23:20.05 | hhhenriq | voi-info |
| 23:20.12 | Crashsys | it's a wiki... |
| 23:20.29 | Crashsys | i've read a bunch of stuff on it... |
| 23:20.59 | hhhenriq | i see...there is a bunch of things there and i started to get scared |
| 23:22.42 | Crashsys | welp that's all the help I can suggest... |
| 23:23.22 | hhhenriq | thank you |
| 23:23.37 | hhhenriq | bye |
| 23:23.42 | Crashsys | night |
| 23:30.04 | BSDTech | help help you want help |
| 23:30.09 | BSDTech | your asking alot |
| 23:30.19 | BSDTech | bloody hell |
| 23:30.28 | Crashsys | Help = Do it for me |
| 23:30.51 | BSDTech | you willing to pay for it |
| 23:30.59 | Crashsys | I'm not |
| 23:31.12 | BSDTech | lol |
| 23:31.17 | Crashsys | I prefer to attempt to learn it... |
| 23:31.18 | BSDTech | no one is now days |
| 23:31.27 | BSDTech | learning is best |
| 23:31.39 | *** join/#asterisk obnauticus (n=aao_pwne@c-24-21-116-29.hsd1.wa.comcast.net) |
| 23:31.40 | Crashsys | in hopes that one day i'll know what the hell i'm talking about... :) |
| 23:31.49 | BSDTech | lol |
| 23:32.06 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 23:32.44 | Crashsys | I guess I should probably get a phone line installed in my apartment if I plan to build POTS servers here... |
| 23:33.00 | Crashsys | I have no way to actually check my incoming... |
| 23:33.04 | Crashsys | other then it "looks" right |
| 23:33.45 | Crashsys | or another * server :) |
| 23:34.18 | BSDTech | well I am working on 2 diff boxes at the same time |
| 23:34.48 | BSDTech | trixbox 2.0/freepbx-2.2 and freebsd 6.1 and asterisk 1.4 |
| 23:34.57 | Crashsys | Maybe i'll just do it in the morning... |
| 23:35.03 | BSDTech | working to finish the port of zaoptel 1.4 |
| 23:35.03 | Crashsys | Trixbox eh? |
| 23:35.07 | Crashsys | how's that working out for you? |
| 23:35.22 | Crashsys | The best thing I ever did with Trixbox was installing Slackware |
| 23:35.24 | BSDTech | yeah I work part time on the project |
| 23:35.46 | BSDTech | it has grown now that fonality ownz it |
| 23:35.52 | Crashsys | I couldn't get trixbox to find my TE205p when I tried it.... |
| 23:36.00 | Crashsys | but this was pre-fonality... |
| 23:36.10 | BSDTech | ahh |
| 23:36.12 | Crashsys | right when AAH changed the name... |
| 23:36.19 | BSDTech | ahh ok |
| 23:36.19 | Crashsys | like 1.2? or something |
| 23:36.21 | BSDTech | should now |
| 23:36.26 | BSDTech | ok |
| 23:36.28 | BSDTech | brb |
| 23:49.24 | perd | holy shit, lumenvox is extremely sexy |
| 23:49.34 | perd | i bought the lite starter for 50 bucks |
| 23:50.04 | perd | everything i need to do a dial by voice company directory and anything else i need.. assuming i dont need simultaneous recognition |
| 23:51.17 | *** join/#asterisk jim_sunnyvale (n=chatzill@208.65.182.250.static.etheric.net) |
| 23:52.00 | Crashsys | dial by voice? as in for sales say 'sales' ? |
| 23:52.05 | perd | yeah |
| 23:52.10 | Crashsys | I hate those things |
| 23:52.16 | Crashsys | lol :) |
| 23:52.22 | perd | it's for internal only |
| 23:52.26 | BSDTech | yeah bt then its 250 for each extra instance |
| 23:52.31 | perd | so you can pick up your desk phone and say 'liana' and it dials |
| 23:52.43 | perd | dont need more than one instance for the office :) |
| 23:53.18 | jim_sunnyvale | i am lookingfor help with firewall issues for some remote users using Polycom phones |
| 23:54.07 | jim_sunnyvale | i have an asterisk server on a public IP and some (not all) remote users become unreachable on very regular intervals (like every hour virtually on the hour) |
| 23:54.42 | jim_sunnyvale | if I change the remote router to a simple home router (and change nothing else) the same phone remains reachable |
| 23:54.57 | jim_sunnyvale | but, it's not practical to always change the remote router |
| 23:55.05 | jim_sunnyvale | anyone have any ideas? |
| 23:57.40 | [hC] | I dont suppose its possible to send hint data between two asterisk servers? |