irclog2html for #asterisk on 20051020

00:00.11spootnickIgbothom_III: change exten => _0.,1,Dial(SIP/${EXTEN}@astrasip-out) to exten => _0.,1,Dial(SIP/astrasip-out/${EXTEN})
00:00.19clyrradgroogs, I found this http://lists.digium.com/pipermail/asterisk-users/2005-March/095420.html but do not see where the actual context is defined and its behaviours
00:00.36Igbothom_IIII had that earlier, didn't work, ppl suggested I change it to how it is now
00:00.41Igbothom_IIII'll change it back, though
00:01.04X-Robspootnick - you are aware that astrasip doesn't work very well?
00:01.13X-Robin fact, they're probably the worst VoIP providers in .au?
00:01.21supaigtrAnyone know if the 411p card has better echo yet?  Or better yet how to disable and use the KB1 software ecan?
00:01.45spootnickX-Rob: no i'm not aware. but still, that's the way to do it. if he's getting no output in * console, then the dialplan is not even recognizing what he wants to do
00:01.57asterisk99anyone using Asterisk on Gentoo ? zaptel won't load after reboot (yes, I followed install instructions)
00:01.58X-Robspootnick - sorry, wrong person
00:02.03Igbothom_IIIspootnick; same result - no output on * console, engaged signal
00:02.04X-Robwas meant to be to Igbothom_III
00:02.18spootnickIgbothom_III: take that line out of the [general] context as well. put it it [astrasip-out] maybe
00:02.31spootnickIgbothom_III: ahm, hold on
00:02.37Igbothom_IIIX-Rob; sure, I'll change to someone else once I get this actually orking.  But paying for a 2nd account when this account doesn't want to play...
00:02.53Igbothom_IIII spoke to Faktortel...
00:02.57spootnickIgbothom_III: see that [astrasip-out] thing you have in extensions.conf?
00:03.02Igbothom_IIIyup
00:03.07spootnickIgbothom_III: it's in the wrong place. put in in sip.conf
00:03.09groogsclyrrad: ... i thought you said you were using AMP? you should have an extensions.conf with from-internal in it
00:03.11Igbothom_IIIok
00:03.33spootnickIgbothom_III: rename it to [astrasip], just to avoid confusion
00:03.34groogsif you're not using AMP then there IS no call waiting stuff defined, you have to do it all yourself
00:03.47Igbothom_IIIk
00:04.18spootnickIgbothom_III: then change the context of [astrasip] to, e.g.: [astrasip-out]
00:04.26spootnicksorry, context=astrasip-out
00:04.28*** join/#asterisk Xbouncethis (n=egg@pcp01534754pcs.huntsv01.al.comcast.net)
00:04.37spootnick(in sip.conf)
00:05.00kuku5groogs: that sucks - what if I park 10 calls - how will i know what is where?
00:05.16spootnickIgbothom_III: done it?
00:05.35groogskuku5: huh?
00:05.42Igbothom_IIIyup
00:05.45kuku5( parked calls )
00:05.58Igbothom_IIIexact same result
00:05.59spootnickIgbothom_III: now go back to extensions.conf, create a context [astrasip-out] and put that dial line i mentioned to you before
00:06.04Igbothom_IIIaha  :)
00:06.33groogsam i missing like 50% of what is happening here or are people just not making sense?
00:06.48spootnickIgbothom_III: reload asterisk, and try dialing a number starting with 0 (as per your Dial line)
00:07.14Igbothom_IIIk
00:07.30Igbothom_IIIexact same result
00:07.47spootnickk, paste your sip.conf and extensions.conf in pastebin again. omit your pws
00:07.51*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
00:07.53Igbothom_IIIyup
00:08.00X-Robgroogs - kuku5 is speaking randomly.
00:08.27groogsok
00:08.54spootnickAriel_: any luck? i'll send you a pack of budweiser if that thing of having a free did work for me  =)
00:11.25kuku5groogs: When you have a regular phone - you  have line 1-6 lets say, and if put it in park - you can see lets say "park3"  and then tell the other person pickup park3, with lets say 7940g, you have no way of seeing the parked calls
00:11.38spootnickIgbothom_III: and?
00:12.07Igbothom_IIIgetting there... conf in a few files - Xorcom Rapid distro splits 'em up into many .conf files  :)
00:12.17X-Robkuku5 - that's right.
00:12.17spootnickah, ok
00:12.30kuku5is there a way to fix that ?
00:12.36X-Robno
00:12.48X-Robthe long answer is "possibly, with openpbx.org"
00:12.55spootnickIgbothom_III: you know, anyway it's a good idea not to use anything like that. write your own dialplan from scratch. it'll take you a couple days, and after doing it, you're good to pull any stunts you wish
00:12.57kuku5like extra hardware
00:13.02kuku5sidecar or something
00:13.32Igbothom_IIIspootnick; I didn't use anything from that - there's no default sip dialplans.  I followed the voip0-info.org site, and this was the result!!
00:14.15Igbothom_IIIhttp://pastebin.ca/26014
00:14.23X-RobIgbothom_III - when you download asterisk, it comes with sample configuration files.
00:14.23spootnickhm, i never looked at it to write my confs. i based myself on the samples from 'make samples'. but ok, not to worry
00:14.32groogskuku5: you mean you can see where it is with a Key system.. and asterisk is NOT a key system ..
00:14.35Igbothom_IIInot in this distro
00:14.44X-Robas I said
00:14.48X-Robwhen _you_ download asterisk...
00:14.52groogskuku5: which is why when you park something, it tells you the extension its parked on
00:14.54Igbothom_IIII've not got a single Linux distro here that wants to work properly without heavy massaging
00:15.02spootnickIgbothom_III: "exten => _0.,1,Dial(SIP/${EXTEN}@astrasip-out) to exten => _0.,1,Dial(SIP/astrasip-out/${EXTEN})"
00:15.08spootnickIgbothom_III: make sure you're using the second one
00:15.16X-RobIgbothom_III - CEntOS+AMP works. Asterisk@Home is built for it.
00:15.30groogsbesides, i don't see what the big deal is. people picking up random parked calls leads for an interesting day at the office
00:15.32Igbothom_IIInot Centos, not Debian sarge, not Debian testing, nothing - they all fail compiling with various issues, hence why I relented and chose a distro (other then *@H) to try and learn on
00:15.37n3u7something tells me that not loading these 20 or thirty app modules to get asterisk to run will cripple my ssetup
00:15.45spootnickIgbothom_III: in sip.conf, change "type=peer" to "type=friend"
00:15.46X-RobIgbothom_III - CentOS works perfectly.
00:15.48X-Robreally.
00:15.52X-RobIt use it for all my asterisk machines.
00:15.59Igbothom_IIII'd like to...
00:16.06groogsdebian works fine here
00:16.10Igbothom_IIIActually, I'd prefer FreeBSD
00:16.11X-Robdebians is good too.
00:16.13spootnickyeah, and so does gentoo, debian, and any other distro who calls itself  a linux distro
00:16.29DaPrivateerOk, still no progress... For some reason after asterisk has been up for a day or so its stops answering 2nd and 3rd slot on my FXO card. The slots all show offhook after receiving their first call, and won't reset. Additionally, I get a polarity warning on all three lines no matter which way I plug the line in (0 -> 1). I am in the US. Anyone have any ideas?
00:16.30*** join/#asterisk wunderkin (n=wunderki@12-219-165-109.client.mchsi.com)
00:16.42Igbothom_IIIwell, want me to try compiling and pastte the legion error messages that each distro showed?  :(
00:16.48X-RobDaPrivateer - upgrade to CVS-HEAD for everything.
00:16.49groogsIgbothom_III: what are your 'issues'? you don't have required libraries to compile? thats not something thats magically fixed
00:16.58groogsunless maybe you install absolutely everything.. :p
00:17.08n3u7lgbothom_III:in the last week I have tried ubuntu and SuSE9.3
00:17.10spootnickIgbothom_III: it's either about old libraries installed by default, or by them not being there at all
00:17.13DaPrivateerX-Rob im running on BSD so thats probably not a good idea
00:17.14DaPrivateerhehe
00:17.18spootnickIgbothom_III: ok, got the last changes i mentioned?
00:17.24n3u7now I'm onto sarge
00:17.25Igbothom_IIIand that is from distros downloaded in the last week!
00:17.36n3u7only yhe apt get will install not the cvs
00:17.43groogsIgbothom_III: usually the first error message or so will tell you what's missing.. sometimes there will be screens of errors after that, but they're just symptoms of an earlier message
00:17.56Ariel_spootnick, sorry wife came home with dinner.
00:18.06spootnickAriel_: lol, sorry man
00:18.07kuku5groogs: where does it tell you which park it parked on /
00:18.08*** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com)
00:18.38*** join/#asterisk Los415 (n=los415@64.201.104.186)
00:18.39groogskuku5: it depends how you have it setup .. when I transfer someone to extension 70, it says "71" back to me .. that means you pick it up with 71
00:18.54kuku5on what phone
00:19.07spootnickAriel_: still got the address of that pastebin? if not, http://pastebin.ca/26012
00:19.14Ariel_looking at it now
00:19.35spootnickgoddamn iax! that's something i never thought i would have any issues setting up
00:19.54groogskuku5: spa-841s, analog connected to ATA's...
00:19.57Ariel_spootnick, what is the error your getting?
00:20.12kuku5does it "say" or does it "show on the lcd"
00:20.12spootnickAriel_: chan_iax2.c:6629 socket_read: Rejected connect attempt from 204.13.233.114, who was trying to reach 's@'
00:20.29groogssay, as in SayNumber(71)
00:20.38kuku5ok
00:20.46spootnickAriel_: my * box is behind nat, but i'm forwarding 4569 udp to it. as i'm doing with 5060 (sip), and sip is working fine
00:21.00kuku5What happens when there are 10 calls all at once
00:21.05groogskuku5: i also have FOP installed, and it shows me all the parked extensions
00:21.08kuku5how does one put them all on hold
00:21.18spootnickIgbothom_III: so?
00:22.28Ariel_spootnick, it's like my setup except I have my inbound type=user context name [878201000060]
00:22.44groogskuku5: well, for one, i don't have that many lines so i dont' get that many calls.. 2, if you have that many calls parked, something is wrong - ie, you should be using queues, or tell their employees to do their jobs.. 3. you can set the extension. i use 70 to park and 71-79 as parking spots.. but you can use 71-89 if you want, or 700 and 701 to 799.. .or 7000 and 70001 to 79999
00:22.59spootnickAriel_: um, you mean "type=user", and context what?
00:23.37Ariel_I have two context one type=peer and and 2nd one called [878201000060] which is where I have type=user
00:23.52kuku5hm
00:23.58Ariel_context=from-pstn for me
00:24.10Ariel_but your heading I had to change it to the number
00:24.15alephcomAnybody here have an experience with POE-Component-Client-Asterisk-Manager?  I think I successfully built a manager driven realtime rating engine. :-)
00:24.35Ariel_POE as power over ethernet?
00:25.19kuku5can I purchase one POE adapter?
00:25.21alephcomportable multitasking and networking framework for Perl    I'm not sure where the abreviation came from.
00:25.31spootnickAriel_: and what's the context name where you have type=peer?
00:25.37alephcomkuku5:  Sure, but I can't sell it.
00:26.06Ariel_I have [goiax]
00:26.22spootnickAriel_: both have context=from-pstn?
00:26.26Igbothom_IIIspootnick; engaged signal, but I get this on the console now... Oct 19 20:18:42 NOTICE[1735]: chan_sip.c:8118 sip_poke_peer: Still have a call...
00:26.36kuku5alephcom: why not
00:26.48spootnickIgbothom_III: ok, now i don't know what that problem means. but now your dialplan knows what you're trying to do
00:26.53Ariel_spootnick, I use in my setup context=from-pstn the context you send it to is up to you.
00:27.11alephcomkuku5: 'cause I don't sell them.  The POE I was referring to is a networking framework for perl.
00:27.28Igbothom_IIIactually, sorry, that was a spurious error.
00:27.35Ariel_you also might have to setup a exten => s,1,NoOp  to see if it's looking for a s extension instead of the number in your case
00:27.39kuku5what about your framework - im in need of a simple billing system
00:27.47Igbothom_IIIThe changes resulted in the exact same result - no output on * console, engaged signal
00:28.05spootnickIgbothom_III: allright, shoot your current conf again in pastebin
00:28.08alephcomkuku5:  I'm the author of ASTPP, I'll msg you privately.
00:28.15Igbothom_IIIok
00:28.34*** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com)
00:28.38kuku5ok
00:29.50clyrradAriel_ I got the * extensions included, and get the feed back the features are enabled or disabled, the the functionality stays the same.  For example *70 to enable call waiting, it sais its enabled, but it still goes to the answering machine.  Any idea why this is?
00:30.01spootnickAriel_: got it. great! it lacked the [87820100xxxxx] in iax.conf
00:30.05supaigtrAnyone know how to turn off the VPM in the 411p and use the software echo can?
00:31.25X-Robspootnick - he obviously doesn't have the Dial* in the context his sip phone is in
00:31.28X-Robcheck that
00:31.37spootnickX-Rob: that's what i'm guessing
00:32.01*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
00:32.50Netgeeksgrrrr, stupid non-functional realtime
00:33.07Igbothom_IIIhttp://pastebin.ca/26015
00:34.00X-Robhah
00:34.02X-Robno he doesn't
00:34.06X-RobIgbothom_III - include => astrasip-out
00:34.10X-Robin your default
00:34.26X-Robduh
00:34.31X-RobIT's a Pan. Duh.
00:34.34X-Rob*rawor*
00:34.50spootnickhold on, it's not only that
00:34.50X-Robthen extensions reload
00:35.00Igbothom_IIIIT is fine, * is my pain right now  :)
00:35.29spootnickIgbothom_III: in sip.conf, set [astrasip] to "type=friend"
00:35.46Igbothom_IIIthat goes in as "include => astrasip-out" in sip.conf?
00:35.51spootnickIgbothom_III: and "context=astrasip-out"
00:36.08X-RobIgbothom_III
00:36.17spootnickIgbothom_III: AND, in extensions.conf, change "exten => _0.,1,Dial(SIP/astrasip-out/${EXTEN})" to "exten => _0.,1,Dial(SIP/astrasip/${EXTEN})"
00:36.19X-Robyour system doesn't know where to load that Dial astrasip-out
00:36.31X-Robyou need to include it in your default context
00:36.35spootnickIgbothom_III: don't bother including anything
00:36.40X-Robcontext's refer to extensions.conf
00:36.41Igbothom_IIIk
00:36.50spootnickX-Rob: no he doesn't my default context doesn't include anything. in fact, it's empty
00:37.06X-Robspootnick - look at his extensions.conf
00:37.09spootnick(i mean, there's a "wrong number dude" there
00:37.20X-Robhe's using a [default] context.
00:37.37X-Robbeing that he hasn't put any other contexts there.
00:37.42X-RobI'd assume that that is all he's got.
00:37.47spootnickX-Rob: i did. but for this to work (he being able to dial using astrasip), there's no need for that
00:38.07spootnickX-Rob: yeah, i understand. but in sip.conf, if the [astrasip] entry points to astrasip-out, it's fine
00:38.09X-Robspootnick - hush. you're wrong 8)
00:38.19X-Robspootnick - that's for _receiving_ calls
00:38.21X-Roblook at 501.
00:38.23Igbothom_IIII'll wait ...  :)
00:38.32X-Robhe doesn't have a 'context=' in there. That's his sip phone
00:38.40X-Robthat means it's going to load the [default] context.
00:38.43spootnickhahaha, man, wish when i was starting there was people fighting over the correct solution for my problem
00:38.53X-Robthe [default] context doesn't include the astrasip-out context.
00:38.54Igbothom_IIIlol
00:38.55spootnickspootnick: allright, do what X-Rob says
00:38.59X-Robtherefore, he can't dial anything starting with 0
00:39.00spootnickops
00:39.05spootnickIgbothom_III: allright, do what X-Rob says
00:39.11*** join/#asterisk pussfeller (n=todd@12.150.129.170)
00:39.17spootnickIgbothom_III: and if it doesn't work, then do as i say
00:39.23Igbothom_IIIok, but what was that again?  :)
00:39.27X-Robig
00:39.36X-Robunderneat [default] in extensions.conf you've got a pile of include=> lines
00:39.42Igbothom_IIIX-Rob; can you please repost as it is mixed up in a lot of other things
00:39.50Igbothom_IIIyup
00:40.00X-Robafter the last of the include=> lines put
00:40.04X-Robinclude => astrasip-out
00:40.35X-Robuh
00:40.36X-Robhang on
00:40.41spootnickX-Rob: if you're right, i'll drop perl in favor of ruby, start using osx, and start telling freebsd is for terrorists
00:40.48X-RobWhatever context you have the Dial(..) in.
00:40.48Igbothom_IIIhanging
00:41.16X-RobIgbothom_III - I'm lookin gat 26013
00:41.21X-Robthat says 'from-astrasip'
00:41.24Igbothom_III26015
00:41.27spootnickX-Rob: http://pastebin.ca/26015
00:41.36X-Robok
00:41.45X-Robyes. include => astrasip-out
00:41.45spootnickaha! that's why!
00:41.47*** join/#asterisk mistral (i=mistral@jstevenson.plus.com)
00:42.03X-Robthen do an extensions reload
00:42.18*** join/#asterisk Insanity5 (n=feaw@ip68-105-215-25.tl.dl.cox.net)
00:42.21DaPrivateerDoes anyone know of any work on the ability to terminate a skype account on an asterisk pbx?
00:42.26X-Robyou also have to fix the dial string which is referencing astrasip-out
00:42.29spootnickIgbothom_III: actually, make that a 'reload
00:42.36X-Robwhich is actually astrasip in your sip.conf
00:42.41spootnickif you changed anything in iax.conf. you'll need it
00:42.44*** join/#asterisk MatsK (n=mk@99.80-202-83.nextgentel.com)
00:42.54Igbothom_IIInot changed iax.cong
00:42.57X-RobAriel_ - I agree. But he's just learning.
00:43.07Ariel_X-Rob, yes your right.
00:43.21Igbothom_III"include => astrasip-out" in the extensions.conf [globals] section?
00:43.25X-RobNo
00:43.27X-Rob[default]
00:43.27Ariel_if you change extensions.conf just do reload extensions
00:43.31Igbothom_IIIok
00:43.34Igbothom_IIIchanging now
00:43.34spootnickdefault actually means nothing. it's just a suggested name for the default context. the point is there should be no "default" context. calls need direction, just like children
00:43.35X-Robyour SIP phone is looking at [default]
00:43.38spootnick(dug)
00:44.01X-Robyour device can ONLY do things referenced from [default]
00:44.09Ariel_no default does mean something anything that does not find a correct context falls there even un authorised calls
00:44.11Igbothom_IIIwoohoo - getting further!
00:44.33spootnickAriel_: isn't that actually what say in the [general] context of sip.conf?
00:44.40X-Robspootnick - so. Perl and evil?
00:44.47Insanity5Am I the only one that has had endless problems with terminatino providers?  Echos, jitter, silence supression, to name a few.
00:44.48spootnickspootnick: yep
00:44.48Igbothom_IIIhttp://pastebin.ca/26016
00:44.58X-Rob<spootnick> X-Rob: if you're right, i'll drop perl in favor of ruby, start using osx, and start telling freebsd is for terrorists
00:45.04spootnickfuck sake, i'm going nuts
00:45.08spootnickX-Rob: yep
00:45.31Igbothom_IIIlol
00:45.34Ariel_spootnick, in sip.conf default context can be changed but others like mgcp, h323, and iax.conf will fall throught to the default
00:45.51pauldywhen you call dbput and dbget where are those values really stored?
00:45.59spootnickAriel_: i always thought you could change it wherever... weird
00:46.22Ariel_spootnick, try it. you will see what happens
00:46.22X-RobIgbothom_III - you've been told about 20 times
00:46.31X-Robchange 'astrasip-out' to 'astrasip' in the dial plan
00:46.32spootnickpauldy: wherever you say it should be in res_mysql.conf
00:46.51X-RobDial command even
00:47.09Ariel_Igbothom_III, you have a great sample extension setup on your system. it's located /usr/src/asterisk/configs/extensions.conf.sample
00:47.12pauldyspootnick, and if you don't have that file?
00:47.23spootnickpauldy: then create one. check voip-info.org for a sample
00:47.23moralewhere do voip providers get their PSTN numbers from or how do they get the routing from VoIP -> PSTN going?
00:47.27Igbothom_IIIAriel_; nope, I don't
00:47.54pauldyspootnick, does * still maintain stateful info on its own?
00:47.58Ariel_Igbothom_III, strange every normal setup does
00:47.58X-RobIgbothom_III - you've been told about 20 times
00:48.02X-Robchange 'astrasip-out' to 'astrasip' in the dial command
00:48.10X-Robthat's 21
00:48.13pauldyor is the db required to make it so that it does remember from one restart to the next
00:48.13Igbothom_IIII read that, X-Rob and changing it now
00:48.14file[laptop]morale: either from another VoIP provider, or they have a PSTN connection
00:48.18X-RobNow
00:48.22X-Robyou also want to drop the 0 from the start
00:48.23Igbothom_IIII didn't ask to have that repeated then
00:48.28spootnickspootnick: ahm, goddamn it, that's right. res_mysql is something different. dbput and dbget work with Berkeley DB
00:48.34X-Robwhich means changing ${EXTEN} to ${EXTEN:1}
00:48.40spootnickman, i'm out of here. i'm freaking
00:48.49spootnickpauldy: ahm, goddamn it, that's right. res_mysql is something different. dbput and dbget work with Berkeley DB
00:49.01moralefile[laptop]: when you say a pstn connection, you mean a trunk from a pstn phone company?
00:49.12Insanity5Anyone here use nufone?
00:49.13pauldyso somehwere there is a nice berkly style hash table that all the data is in
00:49.15file[laptop]morale: PRI, channelized T1, DS3, whatever
00:49.25hcirmorale: PSTN numbers in the US are allocated by nanpa
00:49.25tzangerI use nufone
00:49.32spootnickpauldy: in /var/lib/asterisk/astdb
00:49.39Igbothom_IIIX-Rob; I actually want to keep the "0" for now as astrasip, due to your recommendation, will not be staying around after I use my existing credit up
00:49.45*** join/#asterisk vooduhal (n=christop@67.19.25.178)
00:49.53tzangerapparently NANPA is going to add a digit to area code and digit to extension
00:49.54pauldynice thansk spootnick
00:49.58tzangerin the next 20 years
00:50.16Igbothom_IIIok, changed the dialplan to "astrasip" and still fails
00:50.17Ariel_tzanger, a digit like?
00:50.18pauldynext question can this be changed in 1.0.9 or future version of *
00:50.22moralethanks
00:50.35Igbothom_IIIdiff error message tho - pasting now
00:50.44vooduhalHello all. Probably a simple question.  I'm using rxfax and I'm needing to run an AGI once it is done receiving the fax, but it looks like as soon as the other device is done sending, it disconnects and my AGI never runs.  Any suggestions?
00:51.02Ariel_use the h extension
00:51.04hardwireanybody use snom via poe?
00:51.10vooduhalAriel_: Thank you.
00:51.15Igbothom_IIIhttp://pastebin.ca/26017
00:51.23Ariel_hardwire, I have but the 200 mainly
00:51.50Insanity5Echo cancellation! help!  I cant' bare it any more!
00:52.08Insanity5Nufone to chicago area numbers echos a lot -- for my own voice -- and the other users can't hear it.  How can I fix it?
00:52.13tzangerAriel_: as in NXXNXXXXXX -> NXXXNXXXXXXX
00:52.29tzangerInsanity5: interesting
00:52.30Ariel_tzanger, that sucks
00:52.35tzangeronly to chicago?
00:52.45vooduhalI guess along the same lines, is there a better way to do this.  I hav emy regular dialplan that starts answer,wait  and then a fax extension.  Is there anyway to avoid the wait at the beginning?
00:52.48Insanity5tzanger - at least to 630 area code... I've tried 4 phone numbers
00:52.49spootnickpauldy: changed to what?
00:52.53Insanity5tzanger - I only otherwise dial to idaho
00:53.05Ariel_Igbothom_III, do you have setup in your sip called [astrasip]
00:53.06tzangerInsanity5: can you find any IVRs in that area I can call and test?
00:53.06Insanity5tzanger - which sounds perfect, might I add.
00:53.16Insanity5tzanger - how about a voice mail?  Just talk to yourself
00:53.21Igbothom_IIIhttp://pastebin.ca/26015 is (basically) my config
00:53.53Insanity5tzanger - call my fathers office - 630-887-8640.  Press 0 to shut the gal up and start talking to yourself.
00:53.58tzangerInsanity5: :-)
00:54.04tzangerok where the hell's my phone
00:54.07tzangerdammit
00:54.14tzangeryou leave for a few days and the cat disappears your phone
00:54.41spootnickpauldy: 1.0.9 uses Berkeley DB 1, and so does 1.2.0-beta. in my case it was a shame because i needed to integrate that with a web interface. i "migrated" my dialplan to ODBCget and ODBCput so I could use mysql
00:54.46Ariel_cat ate it
00:54.47pauldyspootnick, mysql/postgres
00:55.01spootnickpauldy: guess i answered your question before i understood it then
00:55.04pauldythats what I'm looking for thanks spootnick
00:55.09Insanity5tzanger - lol
00:55.57hardwireAriel_: aby 360>
00:56.01hardwireany 360?
00:56.07Igbothom_IIIAriel_; tho I ow have "include => astrasip" in default.conf
00:56.13pauldykind of interesting that particular part is not customizeable and that the AMP guys didn't just go ahead and move everything over to it, their dialplan defaults to everyone having call waiting disabled
00:56.34Ariel_what is default.conf??/
00:56.34pauldyand it isn't configurable via the web interface
00:56.44pauldyhave to manualy go to each phone and dial *70
00:56.47Igbothom_IIIit is a file loaded by extensions.conf
00:56.51Ariel_Igbothom_III, you don't include a sip account
00:56.52Insanity5tzanger - Find cat or phone yet?
00:56.54Igbothom_IIIeffectively, it is in [default]
00:57.07spootnickAriel_: he's using some sort of conf generator
00:57.16Igbothom_IIInope, Xorcom Rapid 1.1
00:57.26Igbothom_IIII'm manually editing the conf files - no generator here
00:57.29spootnickIgbothom_III: and does it generate config files?
00:57.29*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
00:57.33Igbothom_IIIno
00:57.40spootnickmy bad then
00:57.49Igbothom_IIIit has its defaults, which allow mw to call internally, the rest I'm adding manually
00:57.59Ariel_Igbothom_III, locate extensions.conf.sample then do some reading there
00:58.05*** join/#asterisk criptos (n=criptos@201.145.227.17)
00:58.37Igbothom_IIIfor example, it built 501.conf and 502.conf which are #indluded in sip.conf, and they each contain their respecting [501] and [502] sections
00:58.48*** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
00:58.50Igbothom_IIIAriel_; I said before that the file does not exist
00:58.56spootnicki could never stand AMP for what it generated. i can't imagine what this other thing does
00:59.15Ariel_Igbothom_III, it does just not in the directory I posted
00:59.26Ariel_rapid puts it in a different location
00:59.31Insanity5spootnick - amp?
00:59.39pauldyspootnick, its at least headed in the right direction I think sure makes managing configs a breaze
00:59.39spootnickInsanity5: asterisk management portal
00:59.44Insanity5ahh
00:59.59pauldydamb internet gremlins took my it
01:00.00Ariel_spootnick, I use amp, normal setups, and others it's just a learning curve
01:00.20spootnickpauldy: it is for sure. there's people using it. so they have to be doing a good job. but the principle is that you can work with asterisk, even in a "lower level" without having to write things on your own
01:00.30Igbothom_IIIthe main issue is that I read voip-info.org and followed the info there, which lead me down the garden path - putting [contexts] in extensions.conf that should be in sip.conf
01:00.49criptoszap/3&zap/4,30,Ttr for a dial commnad is correct rigth? becose, I have this: 1000,1,Dial(zap/3,30,Ttr) 100,2,Dial(zap/3&zap/4,30,ttr)  but the second dial says taht zap/3 is busy and only dial zap/4 :(
01:00.57Igbothom_IIIok - so is there any issue running * (with a wcfxo card) on FreeBSD?
01:01.00pauldytrue but thats how you get it into the hands of people on the other side of the bel curve
01:01.05Igbothom_IIIif I'm gonna download another distro...
01:01.14Ariel_r should never be used.
01:01.20Igbothom_IIIor should I try Centos 4.x again?
01:01.23pauldythen true havok can be achieved
01:01.39Ariel_Igbothom_III, CentOS
01:01.55Igbothom_IIIk, going there again  :)
01:02.24Ariel_rapid uses debian which if you know is a good distro as well
01:02.35Igbothom_III4.2 worth it, or still beta?
01:02.47criptos?
01:02.51Ariel_Igbothom_III, I have not tried it yet. But it's not beta any more
01:02.56Igbothom_IIIk
01:03.07Igbothom_IIIso, should be fine  :)
01:03.15Ariel_besides the people on #centos are very helpful
01:03.37Igbothom_IIIjust need the first 2 ISOs, or all?
01:03.38pauldyif your thinking of using centos for asterisk why not grab the asteriskat home package
01:04.00Igbothom_IIIpauldy; because I'd rather a STANDARD install, thanks to the issues I'm seeing here  :)
01:04.02Ariel_Igbothom_III, I have not tried it yet
01:04.08Ariel_but you don't need xwindows
01:04.18Igbothom_IIIwith 4.1, how many ISOs needed for *?
01:04.25Igbothom_IIIyeah, not wanting that on at all
01:05.00Ariel_Igbothom_III, last one I used 4.1 it used frist 2 disk But I have not tried 4.2 yet
01:05.23*** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net)
01:05.41Igbothom_IIIok, downloading first 3 just in case, and will see what happens then.  :)
01:05.49Ariel_hello shido6 welcome
01:05.52*** join/#asterisk wunderkin (n=wunderki@VDSL-130-13-234-137.PHNX.QWEST.NET)
01:06.03Igbothom_IIIAnything "interesting" needs to be done before installing * on it, and which version of * is the most stable?
01:06.20Ariel_Igbothom_III, I think there is a network setup.  Check with the guys at #centos
01:06.27Igbothom_IIIk
01:06.38shido6?
01:06.40Delta34anybody using cisco 7960 phones?
01:06.42Ariel_Igbothom_III, I use only for production 1.0.9.2 right now
01:06.46shido6I have asterisk@home running as my firewall
01:06.49shido6finally
01:06.53Ariel_nice
01:07.06Ariel_what did you put on it smoothwall or shorewall
01:07.09Igbothom_IIIAriel_; ok, I'd not really like to use Beta in production anyway
01:07.15twisted[asteria]iptables!
01:07.21Igbothom_IIIm0n0wall is better than SmoothWall  :)
01:07.22twisted[asteria]oh wait
01:07.36Ariel_Igbothom_III, yes it is but it's based on fbsd
01:07.41Igbothom_IIIyup
01:07.43Igbothom_III:)
01:07.47Ariel_can't really put it on a asterisk box
01:07.48Igbothom_IIIthat's what I have here
01:07.53Igbothom_IIIon a Soekris net4501
01:07.54Igbothom_IIInice
01:08.00twisted[asteria]no, but you can put asterisk on an fbsd box ;)
01:08.03Igbothom_IIIdoes * work on fbsd yet?
01:08.04twisted[asteria]just without hardware
01:08.10Igbothom_IIIaha  :(
01:08.23kuku5Delta34: i am
01:08.49Ariel_twisted[asteria], how are you doing tonight?
01:09.02twisted[asteria]Ariel_, pretty good... about to leave the office in the next 10 minutes or so
01:09.10Ariel_nice
01:09.20Ariel_putting in a late night (normal)
01:09.28Delta34kuku5: what does your latency times show for sip show peers
01:09.39Delta34do u qualify your phones?
01:09.48twisted[asteria]Ariel_, heh..  i'm usually here pretty long as of lately
01:09.56Ariel_Delta34, what is your problem?
01:10.32Delta34all my cisco phones show up as 70ms for 7.4 code, on 7.5 cisco code they show up as 140ms
01:10.33n3u7well I finally got asterisk running on Sarge with an x100p card
01:10.39n3u7*ghetto
01:10.43Delta34this is all internal lan
01:10.51n3u7can anyone help me set up voicemail?
01:10.52Delta34for xten clients they show up at 8ms
01:10.54kuku5i msg u
01:11.06kuku570-140 ms
01:11.10Ariel_Delta34, well then go back then send an email to cisco about it.
01:11.12twisted[asteria]n3u7, www.voip-info.org/wiki-Asterisk
01:11.19twisted[asteria]n3u7, or, voicemail.conf.sample
01:11.29Igbothom_IIIok, so thanks guys for the helpo.  Downloading CentOS 4.2 now and seeing how I go  :)  Should be a while before I get back in - 512/128 ADSL here  :)
01:11.34shido6i made a script using iptables
01:11.36n3u7thanks twisted
01:11.36shido6no shorewall
01:11.39vooduhalOk, another quick q.  Is there any way to do something like Wait() but that still listens to audio on the channel (ie, fax machine) while it's waiting?
01:11.40shido6none of that crap
01:11.55twisted[asteria]vooduhal, if it's zap, just turn on faxdetect
01:12.00Ariel_shido6, nice do you want to post the script?
01:12.23*** part/#asterisk criptos (n=criptos@201.145.227.17)
01:12.29twisted[asteria]vooduhal, otherwise, google for nvfaxdetect
01:12.35vooduhalI did that, but once it calls my agi it no longer detects it, but if I have it background a wav it will detect it just fine.
01:12.40vooduhalK.
01:12.46Ariel_wait(3)
01:12.47*** join/#asterisk Administrator_ (n=sheva@host-200-94-47-83.block.alestra.net.mx)
01:12.48shido6sure...  now that I know my mouse was dying and not someone hacking into my mactel box
01:12.51shido6:)
01:12.58Ariel_nice
01:13.16Ariel_shido6, thanks for the help last week. (nufone)
01:13.39vooduhalAriel_: I thought wait discarded all audio passed on the channel?
01:13.47Administrator_Desert Zarzamora??
01:14.05Ariel_??????
01:14.22Ariel_vooduhal, I use wait all the time for fax detection
01:15.06vooduhalOk.
01:15.20n3u7wait:is there something I can do to dial into x100p to verify asterisk is working?
01:15.50Ariel_n3u7, plug a phone line to it then dial that number
01:17.24Kattytwisted[asteria]: there's spirit matter drifting about our universe
01:17.49Administrator_I though you were another guy...
01:17.54*** join/#asterisk docelm0 (n=docelm0@pool-70-110-66-127.tampfl.fios.verizon.net)
01:17.57Kattytwisted[asteria]: http://www.cfht.hawaii.edu/News/Lensing/
01:18.12*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
01:18.15*** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net)
01:18.17docelm0whadup?!
01:18.31Ariel_damm wilma is suppose to now come this way... aRgh
01:18.50supaigtrBetter duck.
01:18.55Administrator_see ya
01:19.03X-Robfred will kick your arse if you mess with wilma.
01:19.05docelm0this way?  Ariel where r u?
01:19.25Ariel_miami
01:19.30docelm0TAMPA!
01:19.48docelm0You could have probably guessed that by my hostname..
01:19.55Ariel_docelm0, your also in it's sights
01:20.07docelm0BRING IT ON!
01:20.17Ariel_sick puppy
01:20.32supaigtrMove away from shore ppl.
01:20.43docelm0Dude I made it thru charlie..  And I was stuck in the middle of it in orlando
01:20.52Ariel_supaigtr, haha
01:21.14Ariel_docelm0, I lost a house in Andrew and it's not funny any more
01:21.18*** join/#asterisk Qorky (n=spam@202.173.160.26)
01:21.20*** join/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net)
01:21.23docelm0How old are you?
01:21.27docelm0True.. I wasnt here then
01:21.45Ariel_docelm0, I might have kids your age. I am an old fart
01:21.52docelm0Im 28
01:22.12Ariel_docelm0, my eldest is 22   hahaha.
01:22.21docelm0ok maybe you are..
01:22.27Ariel_docelm0, but my youngest is 2
01:23.14QorkyHow can I create a Caller name when calling an extension? I want when I recieve a particular indial and call a sip extension, for it to say that number X is ringing. and if there is another indial calling that same sip extension, for it to say Y is calling.
01:23.19n3u7:(
01:23.21Qorkyerm. hope that made sense :)
01:23.53docelm0Damn dude..
01:23.54Qorkyat the momment it just says asterisk is calling.
01:24.10Ariel_setCallerID
01:25.07Qorky;exten => s,1,setCallerID('Blah') ?
01:25.53*** join/#asterisk spootnick (n=irc@CPE-144-133-126-245.nsw.bigpond.net.au)
01:26.21Insanity5tzanger - you here?
01:26.26*** join/#asterisk Netgeeks_ (n=Chris@68-185-24-2.static.mdfd.or.charter.com)
01:26.27*** join/#asterisk glomph (n=black@c-24-18-145-249.hsd1.wa.comcast.net)
01:26.32Qorkylike thaqt ?
01:27.23glomphLooking for help connecting w/Vbuzzer..  I can connect with it, it is a working call, but no data passes.   get RTP read error.   This box is NOT behind a firewall.
01:27.46spootnickdid anybody here ever used multiple switch statements (for realtime) in extensions.conf? (same context, like stacking extensions)
01:28.17Qorkysweet. it works :)
01:28.19Ariel_exten => s,3,SetCIDName(Blah)
01:28.23Qorkythants Ariel_
01:28.44Ariel_what is vbuzzzer
01:28.55glomphvbuzzer.com   SIP provider in Toronto
01:29.13Ariel_ahh
01:29.38glomphI see postings that it works with asterisk, but not for me.   (And I have set up many asterisk boxes successfully)
01:29.40Ariel_post your settings for them on pasterbin.ca and give us your error and let us see what we can do
01:30.09Ariel_you can do sip debug when you make a call to see what it's looking for
01:30.47glomphyeah, I think it is doing something funky with the ports.     The data is not getting through.
01:31.35Insanity5Nufone to chicago area numbers echos a lot -- for my own voice -- and the other users can't hear it.  How can I fix it?
01:31.46Ariel_try adding fromdomain= and fromuser=  that sometimes helps with there using SER or BroadWorks setup
01:32.13Ariel_Insanity5, hummmm how about other areas
01:32.25Insanity5Ariel_ - nope
01:32.35Insanity5Ariel_ - Well, I haven't tried anything but chicago and idaho.  idaho is fine.
01:32.36*** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net)
01:32.49Ariel_it could be there servers
01:32.54glomphI have those settings alrready, and I am not having auth. problems, this is data-passing
01:33.00*** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net)
01:33.01Ariel_ask them support@nufone.net
01:33.13Ariel_data-passing??
01:33.24Insanity5Ariel_ - I gave up on that address a while ago.  Is there a way that I can locally compensate.
01:33.25Insanity5?
01:33.29glomphI mean the call sets up all right, but no audio
01:33.48Ariel_Insanity5, sometimes they hang out here and others at #nufone
01:34.02Ariel_glomph, that is another issue
01:34.03*** join/#asterisk P-NuT (n=pnut_@fw.office.unitedip.net.au)
01:34.03Insanity5Ariel_ - Is it possible to correct it myself?
01:34.14Ariel_Insanity5, don't think so
01:34.26Ariel_glomph, are you behind a nat/firewall?
01:34.43glomphNO
01:35.03Ariel_do you have extenip=yourexternIP localnet=192.168.XXX.0/255.255.255.0 in sip.conf
01:35.14NuggetNAT blows goats.
01:35.33Ariel_do you have a firewall on the asterisk box?
01:35.44kuku5Insanity5: if no audio its a firewall issue
01:36.04glomphSIP blows goats.  NAT just makes it more obvious
01:37.17kb1_kanobehowever, I may have to roll back to last May, which has been solid as a rock.
01:37.27X-Robkb1_kanobe - drumkilla did a whole pile of stuff a couple of days ago
01:37.39X-Robroll back to when app_page was submitted
01:37.46X-Rob(eg, about a week?)
01:37.48X-Robit's great there.
01:37.58kb1_kanobeIt's wierd - there's no oops or anything. The machine just stops servicing the nics and the PRI.
01:39.26Ariel_OK it's time to go play with my little girl and get her to bed.  She is still up.  Does she not know she was suppose to be sleeping hours ago....
01:40.10glomphstart talking about linux and asterisk, she will fall asleep IMMEDIATELY
01:40.39*** join/#asterisk dos000 (n=dos000@i216-58-60-251.cybersurf.com)
01:40.57Ariel_glomph, no she will not she actually likes me to talk about anything she is a good listener
01:42.31*** join/#asterisk Tili (i=Tili@218.20.52.122)
01:47.52ManxPower~docs
01:47.54jbotit has been said that docs is 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
01:48.36ManxPowerno audio is usually a NAT or firwewall issue, but can also sometimes be a codec issue or a bindaddr= issue.
01:49.00ManxPowerkb1_kanobe, no IRQ conflict, I assume.
01:49.31*** join/#asterisk Uberbot (n=Uberbot@69.252.219.76)
01:49.54ManxPowerkb1_kanobe, I finally got a tellabs echo canceler working.
01:50.29ManxPower(yes, I was the one that many months ago sent you a message titled "Help me kb1_kenobi, you're my only hope!"
01:50.32X-RobAriel_ - how old is she?
01:51.09Qorkyanyone farmiliar with snom phones? I want to be able to use the extension pad thingo. and be able to monitor if other extensions are on calls etc. for example if an extension is on a call i can see that by there extension on the pad being red.
01:51.17Qorkythis is on a snom 360
01:52.57*** part/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net)
01:53.23ManxPowerQorky, many messages in the mailing lsit archive.
01:53.26ManxPower~mailinglist
01:53.28jbotsomebody said mailinglist was Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
01:53.32*** join/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net)
01:53.48X-RobQorky - you need HINT's
01:53.52spootnickis there a way to call AbsoluteTimeout *during* a call to update it's remaining time?
01:54.01X-RobHINT's tell asterisk which SIP/... relate to which 'number'
01:54.31P-NuTHi all.
01:54.40Delta34can cisco phones do the same, x-rob?
01:54.41P-NuTHow do I make an offsite IAX extension?
01:54.51P-NuTcan someone give me a brief rundown?
01:55.44Qorkyhints eh
01:55.50Qorkyi'll look into it.
01:55.53Qorkyi can do it though?
01:56.18Qorkyit can be done yeah ?
01:56.18glomphexten => 12345,1,Dial(IAX2/user:secret@servername/destination-number)
01:56.27*** join/#asterisk mthem (i=merlintm@64.235.245.133)
01:58.08X-RobQorky - I'm doing in on my 360 now.
01:58.11X-RobUse firmware 4.3
01:58.34Qorkysweet :)
01:58.43Qorkygood phone by the way.
01:59.03Qorkybest i've played with yet. next to my cisco 7960
02:00.03*** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com)
02:00.32tekatiAny idea why I would be getting this when upgrading from RedHat 9 to Fedora Core 1?  ZT_CHANCONFIG failed on channel 3: No such device or address
02:00.50glomphyou probably got a new kernel
02:00.51Netgeeksudev perhaps?
02:01.06glomphand the zaptel module won't load
02:01.15tekatiI did get a new kernel and I recompiled the zaptel stuff.
02:01.21tekatiudev?
02:01.33Netgeeksread the README.udev in zaptel source director
02:01.39tekatiThanks.
02:01.45Netgeeksit will tell you what to do
02:01.56Delta34what version of asterisk do u need to do hint? when i do show applications, hint aint there?
02:03.46*** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net)
02:04.35tekatiDont see udev on my computer.  Anyone else have any ideas for the ZT_CHANCONFIG failed on channel 3: No such device or address after an upgrade from RedHat 9 to Fedora Core 1?
02:04.36wunderkinits not an application
02:06.53*** join/#asterisk bjohnson (n=bjohnson@i216-58-15-5.cybersurf.com)
02:09.10*** part/#asterisk glomph (n=black@c-24-18-145-249.hsd1.wa.comcast.net)
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02:12.37X-Robtekati - fedora core 1 is broken.
02:12.41X-Robyou have to use 3 at least.
02:13.01X-RobDelta34 - it's not an application. Please, RTFW.
02:13.15P-NuTso to make an IAX extension, I make an IAX extension
02:13.21P-NuTin asterisk@home
02:13.33P-NuTand then I put exten => 12345,1,Dial(IAX2/user:secret@servername/destination-number)
02:13.39P-NuTin extensions.conf
02:13.49P-NuTand then I use this extension ofsite?
02:13.54X-RobP-NuT - if you're using asterisk@home, you should be asking on #amportal
02:14.10P-NuTThey are no help though.
02:14.33X-RobNo, especially when you don't ask.
02:14.35P-NuTAll I want is an offsite extension be it SIP or IAX but I can't get any info on how to do it
02:14.36X-Robyou're not even on the channel.
02:14.40brent21was there anything exciting to come out of Astricon 2005? I remember there was a lot of news after Astricon 2004, but i couldn't go this year, and haven't heard anything about it... anyone there?
02:15.01P-NuTI was X-Rob.
02:15.03X-RobP-NuT - ask on #amportal.
02:15.08P-NuTok.. *sigh*
02:15.30X-RobAhh there he is.
02:15.42*** join/#asterisk philm (n=a@r43h15.res.gatech.edu)
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02:17.26kuku5Anyone know cpu usage for   ulaw-*-ulaw ?
02:17.43skyenissue-less, me thinks.
02:17.52kuku5as in?
02:18.00skyenlow. i don't now
02:18.18skyennever even noticed it. not that i run any bigger sites, though.
02:18.33skyenis your system slow?
02:18.35kuku5how many sim. calls?
02:18.46kuku5no - trying to decide on hardware
02:18.47skyen20 at max
02:18.57bjohnsonif both sides use the same codec, should be damn little cpu usage
02:19.00bjohnsonlike a router
02:19.24skyenah, that's usually the case here.
02:20.31bjohnsonI didn't read how many calls kukucachu was expecting
02:20.33Kattysomeone find me an animated movie...specifically an artist's recreation of earth's gravitational field and how light is bent around it
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02:21.27skyenhomework, hey?
02:21.30Igbothom_IIIor, watch more Simpsons  :)
02:21.38bjohnsonI've never seen such a thing
02:21.58skyenI never knew light was bent around planets
02:22.09bjohnsonbig mirrors
02:22.10Kattyskyen: gravity bends light
02:22.34Igbothom_IIIKatty; absolutely
02:22.43Igbothom_IIIand planets bend gravity
02:22.43Kattyif it's powerful enough.
02:22.51Kattyit'll even eat it
02:23.02Kattysort of, anyway
02:23.02bjohnsonyum
02:23.08Igbothom_IIIhence Black Holes
02:23.11bjohnsonnow that's an extreme diet
02:23.14Kattythey're predicting some sort of space time bend around our planet
02:23.22Igbothom_IIIthat emit some radiation, but consume a lot more  :)
02:23.24Kattysort've like a blackhole bending everything around it's event horizon
02:23.41mmlj4Gravity: it's not just a good idea, it's the law.
02:23.52Igbothom_IIInah, gravity sucks
02:23.54Kattymmlj4: well, some things do break the law of gravity
02:23.58Kattymmlj4: dark energy.
02:24.04Kattymmlj4: it sorta holds our universe together
02:24.06Igbothom_IIIdark chocolate, too
02:24.06bjohnsonEVIL ENERGY
02:24.18mmlj4um, right, they just disproved dark matter
02:24.35Kattymmlj4: i've yet to read that.
02:24.39bjohnsonI wish they'd disprove dark turkeu meat
02:24.50bjohnsonhate that stuff
02:24.53Kattymmlj4: post url
02:24.56NetgeeksOff the top of my head I can't think of an animation, however if you want an example....  Think of taking a sheet, stretch out out by the four corners
02:25.10mmlj4don't have it, but it was recent... the last 2 weeks
02:25.14KattyNetgeeks: i'm well aware of what it, in theory, looks like.
02:25.18bjohnsonand wrap it around a planet?
02:25.21NetgeeksThen take a small weight and drop it on the sheep
02:25.31KattyNetgeeks: what i want to see is the extend thing s are being bent around our planet
02:25.36wundaboyi just built asterisk and when i try and start it i get this: Ouch ... error while writing audio data: : Broken pipe
02:25.41Kattys/extend/extent
02:25.44n3u7oh well
02:25.50X-RobNetgeeks - bastard. Why are you throwing rocks at a sheep?
02:25.54Kattymmlj4: i will look for it.
02:25.54n3u7x100p is not cooperating
02:26.03skyenX-Rob: haha, thinkin the same thing :)
02:26.18Netgeeksnow, if you had a ball of zero weight and you rolled it in a non-collision path to the item making the dimple... you'd get an idea of the path
02:26.27Netgeeksthe math isn't too difficult actually
02:26.46Kattymath needs to draw me a pretty picture
02:26.47X-RobNetgeeks - if a ball has zero weight, it has no inertia.How are you going to effect it?
02:26.56X-Robaffect it
02:26.58KattyX-Rob: mirrors
02:26.59mmlj4Katty: http://rss.slashdot.org/Slashdot/slashdot/to?m=738
02:27.06Netgeeksif you assume the mass to be a single point, the math is easy, you could plug numbers in for the mass of the earth...
02:27.07KattyX-Rob: light has no weight, yet it's moving and has mass
02:27.15bjohnsonX-Rob: telepathy
02:27.18X-RobKatty  - uh. Weight == Mas.
02:27.19X-Robs
02:27.27KattyX-Rob: no
02:27.28arp2uhm
02:27.28mmlj4um, weight ne mass
02:27.30KattyX-Rob: mass = energy
02:27.32arp2weight is not mass
02:27.34NetgeeksWeight = Mass * gravity
02:27.39bjohnsonno
02:27.43mmlj4F = M * A
02:27.46bjohnsonWeight = Mass * acceleration
02:27.56arp2acceleration due to gravity
02:28.04Netgeeks*nod* arp2
02:28.10bjohnsonarp2: sometimes
02:28.17X-Robthat's what weight is
02:28.22bjohnsonarp2: could be from other forces too
02:28.34Kattyi'm still pondering this light event horizon thing.
02:28.43X-Robif it has mass, it _does_ have weight. Unless you can find a 'null' part of the universe where there's no gravity from anywhere.
02:28.47mmlj4that'll take forever
02:28.48wundaboyif my box does not have a sound card what kind of sound support do i need to have in my kernel to run asterisk?
02:28.49X-Robwhich, I believe is impossible.
02:28.56Netgeeksyup, impossibel
02:28.57arp2x-rob, thats fine
02:28.58Kattyif the fastest a black hole can pull at is the speed of light, then if a few rays of light travel across a n event horizon....then what happens
02:29.04Kattythere must be some point where the ray is ripped apart
02:29.12Kattyhalf of it goes to the black hole, the other half keeps going
02:29.17arp2but saying mass == weight, is just, incorrect :)
02:29.18Kattyso, that would either mean a photon is rpiped in half
02:29.18bjohnsonpoor little light speck
02:29.26X-Robarp2 - I was simplifing. I do apologise.
02:29.35Kattyin which case both haves are somehow connected with this pilot wave theory
02:29.43Kattyor the photon is stuck in place with equal forces pull on it
02:29.56Kattyor it acts as a wave...the energy basis rather than a partical basis
02:30.20bjohnsonI think it's just magic
02:30.27Kattyin which case the light just dies on itself
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02:30.35NetgeeksThe problem katty is that a photon is both a wave and a particle at the same time
02:30.37bjohnsona big, cosmic Doug Hennesey
02:30.43Kattylight doesn't have infinate energy. but what happens when a photon dies?
02:30.53bjohnsontiny funeral
02:31.02Netgeekswhat do you mean by die?
02:31.06KattyNetgeeks: then what happens to it.
02:31.19KattyNetgeeks: it can't stay still
02:31.20Netgeekswhen it crosses the event horizon?
02:31.24Kattyit going against maxwell's law
02:31.28Igbothom_IIIit doesn't die but transform into a different form of energy/mass
02:31.30X-Robas bjohnson said. There's a tiny funeral.
02:31.42X-Robwith other photon mourners
02:31.54Igbothom_IIIMaxwell Smart's law?  Poor bugger died the other week
02:31.56X-RobBut they can never damn well figure out where the funeral is.
02:32.03bjohnsonis that like .. a photn never dies, it just fades away
02:32.05X-Robor how much the corpse weighs.
02:32.24Igbothom_IIIX-Rob; it doesn't weighmuch, it is very light  :)
02:32.32KattyNetgeeks: at some point a light ray will break in half when it goes across the event horizon. part of it will be pulled into the blackhole
02:32.35bjohnsonexactly, a light particle
02:32.37bjohnsonlike lint
02:32.40X-Robheh
02:33.00KattyNetgeeks: there is some point, whatever it's called, where there are equal forces pulling on that ray of light
02:33.07bjohnsonblackhole = space belly button
02:33.12Netgeeksyou cannot separate the wave/particle attributes of the photon....  it doesn't break apart.
02:33.24bjohnsonI'm starting a new law, the belly button law
02:33.27KattyNetgeeks: it'strying to go it's normal speed, and the blackhole is apply equal negative forces.
02:33.44KattyNetgeeks: so, it cannot remain stationary according to maxwell's law
02:33.45NetgeeksThe problem with black holes is we don't have any to examine, and if we did, we couldn't examine it, so the only models we have of black holes are math models
02:33.53mmlj4armchair quantum physicists, hrmph
02:34.03bjohnsona piece of lint, travelling at the speed of lint, tries to traverse a belly button in space
02:34.07bjohnsonwhat happens
02:34.10X-RobNetgeeks - I believe this is what Hawking's recent stuff was talking about, with the gamma radiation being given off by a black hole when light particles are 'torn apart'
02:34.15KattyNetgeeks: thusly, it either acts as a particle or a wave (or, as you say, both at the same time)
02:34.32NetgeeksAh!  the gamma emmiters!
02:34.32KattyNetgeeks: when you rip apart a photon and send it seven miles apart from each other down fiberobtics, they're still linked
02:34.58KattyNetgeeks: everhything one does, the other does. they blame some sort of 'pilot' wave which connects the two haves. some say they're ocnnected on a different dimension, ...i certainly don't know
02:35.02X-RobKatty - that's called 'Spooky' entanglement.
02:35.08X-Robdeservedly sow.
02:35.11X-Robso.
02:35.20bjohnsonso if one goes on a date?
02:35.24NetgeeksKatty: I'm not familiar with "ripping apart photons"  This probably is something that is new since I actually was a physicisy
02:35.28X-Robbjohnson - the other one does _not_ get laid.
02:35.30KattyNetgeeks: what happens to a wave that's ripped apart?
02:35.32Netgeeksphysicist, even
02:35.48bjohnsonpoor little photon bastard
02:35.52KattyNetgeeks: even if it /was/ stationary light, it would just...stop
02:36.02KattyNetgeeks: it wouldn't have infinate energy to stay..uhh... lit
02:37.06Kattyand if it's a trackable conversion...
02:37.15Kattythat means we can watch it, map it..and find the equal force horizon
02:37.30Kattythat would at least let us /see/ it a little better
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02:37.46mmlj4all these newfangled ideas are really strange to me... photons being ripped apart... who makes this stuff up?
02:38.15Kattyor if the photons are split when pulled opposite directions, then we should be able to track one half and from that figure out what the other half in the blackhole is doing
02:38.20Kattyin theory, of course.
02:38.29mmlj4the little guy just hangs at the event horizon, taking forever to make up his mind, does he fall in, or wriggle out
02:38.32NetgeeksOkay, I don't think I can help, as the ability to 'rip apart a photon' is beyond my comprehension.....
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02:38.44KattyNetgeeks: moment, i'll get the experiment
02:39.10NetgeeksI finished my masters in physics in 88, and at that time, photons were not rippable apartable... hehe
02:39.15mmlj4yes, yes, the experiments, i forgot they were creating black holes in the lab these days
02:39.19*** part/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net)
02:39.19twistedwheeee
02:39.40twistedmv file /dev/bed
02:39.42erickj_azI have an intresting issue.  I have very good ping times to a sip phone (80ms or less), but the sip show peers is showing 250ms or more and most of the time it is unreachable.  Any ideas.  This is just a recent thing.  The only thing I did was configure DUNDi on the private network card, but I think this started before I did that.
02:39.51twisteder no wait
02:39.55twistedthat would mean i made file into a bed
02:40.00fileooh twisted
02:40.03twistedcat file > /dev/bed
02:40.03fileI didn't know you were like that!
02:40.11drumkillaw00t
02:40.14twistedsup dk
02:40.41KattyNetgeeks: http://www.cebaf.gov/news/internet/1997/spooky.html <- by rip in half i mean a pair.
02:41.05Netgeekseinsten's Spooky action at a distance problem?
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02:41.28Kattythe effect of a black hole on one photon /should/ have an affect on its escaped pair
02:41.42twistedturns out it just needed some air.
02:41.49twistedin the tires.
02:41.58KattyNetgeeks: or am i reading into that incorrectly?
02:43.01Netgeekssec, reading the url you sent
02:43.24KattyNetgeeks: The crystal splits the photon in two, producing two new photons that continue on in somewhat different directions, and whose combined energy equals the energy of their parent photon.
02:43.45KattyNetgeeks: The astonishing consequence of this is that the particle's distant twin experiences exactly the same metamorphosis at the same moment, even though there is no physical link or signal between the two twins
02:44.42NetgeeksYep, I'm familiar with this.  It's a quandry and actually has been one of the impetus for the creation of the multi-dimensional and string theories of the universe
02:45.24NetgeeksThe whole problem we are having in this previous discussion is that Einstenian physics breaks as you get into particle physics, where quantum physics takes over
02:45.30X-RobPersonally, I think it's the FSM.
02:45.30Kattyhmm, yet another interesting topic. string theories.
02:45.40NetgeeksYou can't apply a problem in quantum physics to the einstein physics
02:45.45X-RobHe's touching the photons with his holy noodle.
02:46.06KattyNetgeeks: but even using quantum physics...breaking a photon into two equal pairs...
02:46.18Netgeeksso talking about how a photon reacts at an event horizon doesn't work
02:46.20KattyNetgeeks: there should be a way to watch the effects of a black hole on light
02:46.25KattyNetgeeks: why not?
02:46.36KattyNetgeeks: surely it doesn't drag the entire ray in ;)
02:46.48KattyNetgeeks: that concept is absurd.
02:46.55Netgeeksbecause the event horizon is a mathematical function of einstein physics
02:47.07Kattyhmm.
02:47.35Netgeekswhat you would have to say is what happens to a photon when it's mass becomes infinite
02:47.37*** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net)
02:48.05Netgeeksand do all your work in the quantum thoery world
02:48.10*** join/#asterisk CANO-1982 (n=cano1982@201.255.51.6)
02:48.16Netgeeksand that is some very nasty math
02:49.11CANO-1982y have a problem with my tdm400p
02:49.26NetgeeksString thoery (way way way beyond my understanding) apparently predicts and describes how the photon pair (which in string thoery aren't really pairs, they are the same photon whose string passes through our universe at two locations)
02:49.26CANO-1982can y ask here?
02:49.38Netgeekshow the pair reacts the way it does
02:50.10Netgeeksa REALLY bad and basic analogy is this
02:50.37CANO-1982I can´t make fxo module work properly
02:50.39Netgeeksif you lived in a 2d world, think of a piece of paper in which you are stick people drawn on the paper...  it's a 2d world
02:51.05Netgeeksnow fold the paper in half, the people in the 2d world don't know it's folded in half, but since we are in 3d and outside we do
02:51.14Netgeeksnow run a needle through the folded paper
02:51.16CANO-1982have to plu and unplug the tephone cable every incoming call
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02:51.29Netgeeksaccording to the 2d world people they see two dots
02:51.48Netgeekswhen you twist or move the needs to the 2d world people both dots seem to be connected
02:52.10Netgeekstwo completely separate entities to them seem to act in concert at a distance
02:52.11CANO-1982I think dimensions is only a mathematical abtraction, only that
02:53.00NetgeeksThe current evolution og string thoery i think contemplates 11 dimensions....
02:54.35NetgeeksQuantum physics is a very disturbing field, IMHO....
02:55.31NetgeeksYou still there Katty?
02:55.51KattyNetgeeks: i made a phonecall.
02:56.03blitzrageevening all
02:56.28NetgeeksAre you familiar with the age-old experiment that they used to show that a photon is both a wave and a particle at once?  the observation test?
02:56.40blitzragequantum physics r0x
02:56.59blitzrageNetgeeks: yes, I know that test
02:57.11KattyNetgeeks: no, i've not read that.
02:57.14KattyNetgeeks: but i shall.
02:57.16blitzrageNetgeeks: as soon as you observe it, it snaps into place, if not, then you get an interference wave
02:57.20Netgeeksbasically it's this
02:57.26KattyNetgeeks: my phonecall cleared up a few things.
02:57.53Netgeeksif you shoot a single photon through a difracting slit and observe it as a wave, it creates a wave diffracting pattern
02:58.05Kattyyes
02:58.25Netgeeksif you observe it as a particle, it strikes at some random location that wold be part of the diffraction pattern, but only one spot
02:58.34blitzrageNetgeeks / Katty: I *HIGHLY* recommend you get Brian Greene's, "The Fabric of the Cosmos: Space, Time, and the texture of Reality"
02:58.51Netgeeksif you shoot enough of them and observe them as particles, you recreate the diffraction wwave pattern
02:58.58blitzrageNetgeeks: actually, you can't "observe" the waveform
02:58.59erickj_azdoes anyone know how to use the dial cmd to call a sip device in a specific context on a remote server? somthing like SIP/1234@othercontext@sip.domain.com
02:59.04fileBrian Greene is gooooood
02:59.12erickj_azor is that even possible?
02:59.14fileerickj_az: contexts don't exist with SIP
02:59.22blitzrageerickj_az: thats only with IAX2
02:59.29filehttp://www.pbs.org/wgbh/nova/elegant/ -> good to watch
02:59.47erickj_azWhat is the dial command for iax2, then?
02:59.57NetgeeksThe guys who did the photon experiment you read about, used a modification of this experiment, what they did was split a photon run it down the optics and observed it at the end as a particle... each split photon hit on the exact same spot in the pattern
03:00.22Netgeekseach and every time
03:00.29blitzragefile: great show
03:00.47blitzragehttp://www.pbs.org/wgbh/nova/elegant/greene.html
03:01.09KattyNetgeeks: i think i need more base information
03:01.17KattyNetgeeks: i'm having a difficult time with that sinking in
03:01.42erickj_azI want a sip phone on a remore asterisk server to dial into a context on the main server on for extensions on the main server to get tot he sip phone on the remote server
03:01.43*** join/#asterisk CANO-1982 (i=alejandr@201.255.51.6)
03:01.56NetgeeksKatty, go find that particle/wave experiment, it should be in almost any book that claims to delve even into the slightest bit of quantum physics
03:02.06KattyNetgeeks: k
03:02.09Netgeeksyou can even find it in Schroedinger's cat book
03:02.18Kattyyay, schroedinger's cat!
03:02.48erickj_azWill DUNDi do it if I just put the SIP deives oin the same context on both servers?
03:02.51NetgeeksNow, once you read on it, tink what would happen if I had two of these experiments, and I 'split' a photon and send half down one and half down the other....
03:03.00Netgeeksthis is what they did
03:03.07blitzragethat Brian Greene book I told you about is pretty much the only book you need to get started understanding the universe around us
03:03.50Netgeeksand what they 'observed' was that the 'daughter' photons ( or the two photons that were created from the original photon) acted the exact same way at the observation points
03:04.28NetgeeksThus, they concluded that the photons are connected somehow
03:06.13blitzragefile: I leave in 3 days
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03:07.18*** part/#asterisk Xbouncethis (n=egg@pcp01534754pcs.huntsv01.al.comcast.net)
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03:08.34KattyNetgeeks: tunneling also interests me.
03:08.35twistedblitzrage, where are your photos?
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03:09.29NetgeeksI don't know if there is much to say about tunneling...
03:09.55websaecan i use my broadvoice account to use my computer to dialup to the internet?
03:09.57websaeis that possible?
03:10.16NetgeeksIt's just the word given to the wierd fact that quantum entities can 'avoid' physical objects in thier way
03:10.42KattyNetgeeks: and the bigger the barrier, the quicker they seem to travel?
03:10.47Kattythat doesn't really make sense either
03:10.51websaeanyone have any ideas on that?
03:11.03*** join/#asterisk wolfson` (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
03:11.05websaeusing voip account to dialup to an internet connection?
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03:11.34KattyNetgeeks: has anyone tried piggybacking anything onto a photon.uhh, half?
03:12.12Netgeekshehe, I've no clue Katty.  You thinking of teleportation?
03:12.20Kattybank transactions, actually
03:12.27Kattyno ecryption required
03:12.39Kattywhat one photon does, the other should, right?
03:12.46Kattyfaster than the speed of light
03:12.47NetgeeksWell, you could use photons to carry data, right?  fiber optics does this
03:13.10KattyNetgeeks: would the photon break down?
03:13.16Netgeeksso you could in thoery using tunneling get data from one location to another instantly
03:13.33KattyNetgeeks: invent it.
03:13.48KattyNetgeeks: the speed of light is now slow.
03:14.07Netgeeksnot that I'm aware of... from what I understand of tunneling, there is only a random chance that the quantum particle/wave will tunnel
03:14.10Netgeeksnot all do
03:14.15Netgeeksso you would lose data
03:14.27Netgeeksor the data arriving would be garbled
03:15.39Katty:<
03:15.53Kattywe must stablize photon daughters!
03:16.01Kattyby finding something else that will tunnel with it
03:16.08Kattysomething stable
03:16.17Kattyand osmoene bind them
03:16.20Kattysomehow, i mean
03:16.32Netgeeksalas, I threw away my masters in physics to play with computers 15 years ago......
03:16.36*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
03:16.44pauldykatty just use more than one and ask at&t if you can use their crc method and boom
03:16.44Katty:<
03:17.20Kattyi still think there's a way to make space travel much faster usin gnothing but weak magnetic fields.
03:17.45Kattylike how they speed up particles using electromagnetic fields...and switching polarities of 'tubes'
03:17.47*** join/#asterisk saint_ (i=saint@ool-44c4aa93.dyn.optonline.net)
03:17.48saint_hi all
03:18.00pauldyKatty I thinkthe idea that light travels at differnt speeds through differnt mediums means ftl travel is possible to me at least
03:18.10NetgeeksI had a choice, take a job at a nuke plant developing computer software to predict what was going on in the core of the reactor, or go be an assitant at an observatory on the top of a very cold mountain... I chose the nuke plant, and my fate was sealed
03:18.12saint_I have a super question...: Is it possible to use a cell phone for an outgoing trunk ?
03:18.29KattyNetgeeks: hmm.
03:18.34KattyNetgeeks: query.
03:18.47KattyNetgeeks: has there been any advancements to space folding?
03:18.58KattyNetgeeks: like how they think the photons halves are just one, but on different planes
03:19.33Kattyhop the needle for a ride, so to speak
03:19.39Netgeeksum, two thoeries address that, the expanding decision thoery and string thoery
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03:19.52Kattymeh, string theory still confuses me
03:19.58kb1_kanobesaint_: yes, though I beleive you need GSM hardware. I recall it's quite popular in europe.
03:20.10saint_well...
03:20.29Netgeeksthe expanding decision thoery (not the real name, because I forget what it is) says that a separate existance is created for every possible outcome of every event
03:20.34X-Robsaint_ - there's some hardware that lets you dock your mobile phone and it turns it into a FXS port.
03:20.34saint_I install Alcatel phone systems, and one of my customers asked me if it was possible. I did it with Quescom in France, but it is for GSM...
03:20.39X-Robyou can then get an FXO card and use that.
03:20.50saint_In the USA it is all fucked up with the FCC ...
03:21.03Netgeeksstring thoery is understood by less than maybe 50 people in the world.... I'm not one of them by a long shot
03:21.19saint_so I was wondering if there is any kind of hardware that I could plug on an analog port , or on a trunk (like pot lines) to the cell phone to do that ..
03:21.23KattyNetgeeks: every action has an equal and oposite reaction
03:21.33X-Robsaint_ http://www.cellsocket.com/index.htm
03:21.35NetgeeksThats newtonian physics
03:21.39KattyNetgeeks: for ever action in this universe, there is an equal and opposite universe to counteract it
03:21.42saint_i did not find anything, so i was wondering if I could do it with Asterisk over a PC, and then have a Qsig link between the Asterisk and the Alcatel system ?
03:21.52X-Robsaint_ - HELLO?
03:21.55X-Rob*kick kick*
03:22.00X-RobARE YOU PAYING ATTENTION?
03:22.06kb1_kanobesaint_: there is an FXO module for older Motorola phones to allow connecting a portable fax or such - we use them to drive conventional analog phones. Same would work as a trunk interface.
03:22.09X-Robhttp://www.cellsocket.com/index.htm
03:22.13NetgeeksKatty: ahh, hrm, is that another thoery out there?
03:22.33X-Robbugger
03:22.34X-Robit's broken
03:22.40saint_cellsocket is not working ..
03:22.42X-Robtheir website is broken, but they do exist.
03:22.50Netgeeksalthogh that probably falls into the first alternate thoery I mentioned
03:22.51KattyNetgeeks: that would no doubt be a seperate existance
03:22.56Netgeekswish I could remember it's name
03:22.57X-RobThat's what they do. You dock your mobile, and it give syou an FSX
03:23.00KattyNetgeeks: unless you're getting int some sort of anti-particle thingy
03:23.17KattyNetgeeks: for every particle this is an anti-particle pair
03:23.28X-Robsaint_ - google for 'em.
03:23.37Kattyi love how our universe just likes creating things out of nowhere for us!
03:23.39Kattyfun fun.
03:24.01NetgeeksOh, speaking of the anti-particle thingy.. one thoery says you can create a particle if you also create it's anti-particle, because the overall balance of mass energy is maintained constant
03:24.20KattyNetgeeks: i could create fire, and the anti-fire.
03:24.27NetgeeksSo you could create a photon, if you also created an anti-photon
03:24.27KattyNetgeeks: or move and anti-move an object
03:25.08Kattyback to string theory
03:25.16Kattyfrom what i understand there are two types of strings
03:25.17Kattyopen and closed
03:25.36Kattyclosed exist in our dimension because they're trapped...having both ends here.....sort like on a guitar.
03:25.45Kattywhere as closed particles are more of a loop...they have no beginning or end
03:25.49NetgeeksBut you have to remember that 99% of all this is artificial.. it's people trying to take some strange math result and apply reality to the result...  No one has found a way to just 'create' an photon and anti-photon
03:25.53Kattyso they can wander around and not bump into thing
03:26.34KattyNetgeeks: yes but the universe spews out particles and anti-particles at times.
03:26.43KattyNetgeeks: very briefly, anyway
03:27.15KattyNetgeeks: i would take that as something accidently got dumped into our dimension
03:27.21KattyNetgeeks: and to balance it out, something had to give
03:27.38KattyNetgeeks: which then forces it back to where it was...originally
03:28.32Netgeeks*speechless*
03:29.11Kattyi guess that's something like a white hole
03:29.24Kattyi don't know of anything else that would be a theoretical leak
03:29.52Netgeekswell, I have to head out now.  Nice thoerizing with you
03:29.58Kattykthxbi
03:30.36*** join/#asterisk Jzalae (n=sk@216-220-250-204.midmaine.com)
03:34.59*** join/#asterisk eivindtr (n=wingnut-@062016241059.customer.alfanett.no)
03:35.10*** join/#asterisk newmember (i=user@S010600036d1139fb.cg.shawcable.net)
03:36.31websaecurious
03:36.44websaeanyone know about dialing an ISP over your VoIP connection with ATA adapter?
03:36.46websaeis that possible?
03:37.24kb1_kanobeyou mean PC-Modem-ATA-(voip)-ITSP-ISP ?
03:37.57jsmithModem over VoIP won't work :-)
03:38.08jsmith(Well, maybe at 9600 baud, if your latency is *VERY* low)
03:38.27kb1_kanobeagreed. Only if there is no jitter and the bandwidth of the modem is less than the bandwidth of the codec (eg. fax over ulaw)
03:39.20Nuggeteven then, it's a crapshoot.  :)
03:39.54syle2why isn;t this working errrrr
03:40.01Qorkyhas the syntax changed ?
03:40.01Qorkyasterisk*CLI> capi info
03:40.01QorkyContr1: 2 B channels total, 2 B channels free.
03:40.01QorkyContr2: 2 B channels total, 2 B channels free.
03:40.07Qorkyi have 4 channels free
03:40.14syle2if (mysql_real_query(&mysql, sqlcmd2, strlen(sqlcmd2))) { }
03:40.14Qorky<PROTECTED>
03:40.17syle2else {
03:40.21Qorkybut i cant dial anymore :(
03:40.30Qorkyi just updated to the latest cvs-head
03:40.44syle2int nrows; MYSQL_RES *result; result = mysql_store_result(&mysql); nrows = mysql_num_rows(result);
03:40.55syle2pissin me off
03:41.07syle2if(nrows == 0) { always returns true
03:41.32syle2i see the query is getting called
03:41.49syle2just nrows always 0
03:44.18*** join/#asterisk [hC] (n=hardcore@c-24-127-192-210.hsd1.fl.comcast.net)
03:44.45*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
03:49.14*** join/#asterisk Deedubb (n=Deedubb@S010600055d22c57f.vf.shawcable.net)
03:50.45*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
03:50.57*** join/#asterisk bmg505 (n=leon@rndf-146-56-187.telkomadsl.co.za)
03:52.06websaewell im curious if it's possible to have an asterisk server with a DID and unlimited incoming channels---to dial into that DID--and be able to have internet access---VoIP internet or something
03:52.39*** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net)
03:53.30glm2kwebsae: you already have internet, just take out the Vo from the IP
03:54.00Nuggetjust get AOL.
03:54.14dasuberdavidlol
03:54.16glm2khehe
03:55.22Deedubblaugh
03:55.39Deedubb... I was bored, sorry
03:57.11Qorky<PROTECTED>
03:57.21Qorkyany know a bit about it?
03:58.50blitzrageI use AOL. Its awesome.
03:59.09file[laptop]blitzrage: never say that again
03:59.19websaewell say i have a server colocated on 100mb connection----and i have unlimited channels to it---and i dial that did with my modem over a PSTN line---is there some how i could configure internet access-???
03:59.21blitzragefile[laptop]: what... I'm serious. I just switched over a couple of days ago
03:59.55file[laptop]I think my ISP has all my e-mail on a time delay
04:00.01ManxPowerwebsae, no
04:00.04file[laptop]and why am I responding to an e-mail bkw sent me...
04:00.13ManxPowerYou are trying to do IPoV
04:00.22websaeyes
04:00.30blitzrageisn't modem PPP ?
04:00.34blitzragebut either way :)
04:00.50*** join/#asterisk Insanity5 (n=feaw@ip68-105-215-25.tl.dl.cox.net)
04:01.00Insanity5I am getting way too much white noise from a pap2 -- is there an easy fix out there?
04:04.10*** part/#asterisk santiago (n=santiago@208.195.215.158)
04:05.02yxaif i scripted * to call 2 external numbers using 2 fxo, is there a way to bridge the 2 calls?
04:05.58blitzrageblind xfer?
04:06.29jsmithyxa: Yes, you could do it through call files or the manager interface.
04:07.33file[laptop]uh oh it's jsmith
04:08.42file[laptop]eep
04:08.57yxajsmith i understand the call files part but what abt the ami part?
04:09.20jsmithYou should be able to do the same thing over AMI
04:09.45yxajsmith oh i want it totally scripted
04:10.57jsmithOK, gotta get some sleep -- long day tomorrow.
04:11.41*** join/#asterisk glomph (n=black@c-24-18-145-249.hsd1.wa.comcast.net)
04:12.30glomphAsterisk rules, check out the story about the Gumstix-server.  TINY
04:12.41glomphhttp://www.linuxdevices.com/news/NS7497964764.html
04:13.01blitzrageglomph: dude... that's SOOOOOO last week :)
04:13.44glomphOuch!  Lots of venomous eels in here
04:14.17blitzrageglomph: I like Windows as a desktop OS :)
04:14.30glomphI like Windows as a doorstop OS
04:14.49blitzrageglomph: I can only make fun of your posting because I was at AstriCon and saw the gumstix live, and talked to Kristian right after his interview :)
04:15.16glomphWish I could have made the journey
04:15.17dasuberdavidis kristian in here?
04:17.37*** join/#asterisk damned (n=vpol@damned.vpol.org.ru)
04:17.39Qorkybah. i cant get it to work
04:17.44tehdelyass turd dicks
04:17.51QorkyNo channel type registered for 'ISDN1'
04:18.41*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
04:18.44*** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au)
04:19.18spootnickhas anybody ever used multiple switch (realtime) statements in the same dialplan? can't seem to be able to make second one work...
04:19.34blitzragedasuberdavid: nah, he doesn't use IM/IRC
04:19.43blitzragespootnick: I don't think you can
04:19.56blitzragespootnick: but not positive
04:20.03blitzrage(having never used realtime)
04:20.33spootnickquote from voip-info.org. "And YES! You can have multiple switches and multiple family names using this method. "
04:20.43spootnickreferring to having multiple switches
04:21.50spootnicki'm trying to crack this one for a month now. while in the database, i can take a caller "out of" relatime by having a goto that points to my extensions.conf file
04:22.03spootnickbut once there, if i try to
04:22.15spootnickbut once there, if i try to "switch =>" again, it doesn't work
04:39.52*** join/#asterisk Insanity5 (n=feaw@ip68-105-215-25.tl.dl.cox.net)
04:39.53Insanity5Damit --- buying 10 linksys pap2's at staples - $500.  Filling out 10 rebate forms $500 check in the mail.  Unlocking them - 10 minute each.  Finding out that one batch of 5 that your bought at a certain store are all defective with loud annoying white noise... priceless.
04:40.06Insanity5:(
04:40.15X-Robhah
04:40.25X-Robyou got 'em for $0?
04:40.28Insanity5ya
04:40.31X-Robsweet.
04:40.46Insanity5They were free after mail in rebate last week (vonage units), and an unlock hack was on dslreports.com so I went for it.
04:41.30Insanity5They're now buy for $50 get $50 cc gift card in the mail at circuit city.... Christmas shopping or launder the gift card on ebay for 90% value for $45 back.
04:41.54blitzrageInsanity5: wonder if that was the unlock thing that JunK-Y made
04:42.09FuriousGeorgeblitzrage: how was astricon
04:42.10Insanity5blitzrage - what is this?
04:42.22blitzrageFuriousGeorge: pretty damn good!
04:42.27blitzrageFuriousGeorge: and San Fran was even better
04:42.33blitzrageInsanity5: thats all I know :)
04:42.44FuriousGeorgeblitzrage: did you end up using IPCOP for that thing
04:42.56blitzrageFuriousGeorge: nah... m0n0wall is 100 times better :)
04:43.07FuriousGeorgeneverheard of it
04:43.10FuriousGeorgeheard of smoothwall
04:43.11blitzrageFuriousGeorge: really?!
04:43.13Insanity5What's a good 10 minute QOS hack?
04:43.19Insanity5for a linux router?
04:43.20blitzrageFuriousGeorge: crazy... www.m0n0.ch
04:43.28Insanity5so peer to peer doesn't kill downloads
04:43.30blitzrageInsanity5: www.m0n0.ch
04:43.53Insanity5blitzrage - see http://www.dslreports.com/forum/remark,14450684?hilite=pap2+unlock
04:43.55blitzrageInsanity5: www.m0n0.ch <-- yep, go to the traffic shaper wizard, and click the appropriate check boxes
04:44.07Insanity5blitzrage - But I can't use my current linux distro with that, can I?
04:44.09*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
04:44.19blitzrageInsanity5: what do you mean? Its a router/FW (runs off of CDrom)
04:44.32Insanity5blitzrage - Well, I might want to run a web server or other goodies.
04:44.34blitzrageInsanity5: it sits outside of whatever you're using
04:44.39blitzrageInsanity5: so use another server
04:44.40Insanity5on the same box
04:44.41FuriousGeorgei think call parking can be done better
04:44.58Insanity5It's for residential use, for me and 2 roomies.
04:45.21blitzrageInsanity5: then good luck... m0n0wall was the only thing that didn't require a month of time to learn iptables / tcng
04:45.26FuriousGeorgei think you need a pull button, or some way to pull a parked call.  ideally you should be able to control where a call gets parked
04:45.43blitzrageFuriousGeorge: then you want to talk to oej -- he made such a thing
04:45.48FuriousGeorgeor just put a caller on hold, and someone can pull it from you
04:45.57FuriousGeorgeoej?
04:46.09blitzrageOlle E. Johansson (the guy who does the SIP channel)
04:46.27blitzrageFuriousGeorge: oej at edvina dot net
04:47.47FuriousGeorgei think im gonna go make a request
04:48.02*** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com)
04:55.09FuriousGeorgethis would probably be pretty easy to implement.  if i park a call it automatically goes to ${EXTEN}# (or something like that)
04:55.11Qorkyhmm since i loaded now head and capi. i get this.
04:55.11QorkyContr2: 2 B channels total, 2 B channels free.
04:55.11Qorky<PROTECTED>
04:55.11Qorky<PROTECTED>
04:55.11Qorky<PROTECTED>
04:55.12Qorky<PROTECTED>
04:55.14Qorky<PROTECTED>
04:55.17Qorkywhen just tryig to make a call.
04:55.30Qorky== ISDN1: CAPI Hangingup
04:55.30Qorky<PROTECTED>
04:55.38*** join/#asterisk jeffik (n=Jeff@node-423a160a.mdw.onnet.us.uu.net)
04:55.39Qorkylooks like its making several calls ?
04:55.53FuriousGeorge~pb
04:55.54jbotit has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
04:59.43*** part/#asterisk glomph (n=black@c-24-18-145-249.hsd1.wa.comcast.net)
05:00.12DeedubbKitKat is good
05:02.42Insanity5Is it possible to have asterisk  ring multiple pots lines, and once one answers, stop ringing the others?  IE:  Ring my cell and home phone?
05:04.57Insanity5I love how my isps dns server has a reverse entry for 172.16.0.50 :(
05:05.06Insanity5Always messes with my home computer's hostname, hehe.
05:05.33KattyDeedubb: mew?
05:05.35FuriousGeorgeInsanity5: thats the default behavior
05:06.15FuriousGeorgedial(zap/cell&zap/home)
05:07.00*** part/#asterisk Uberbot (n=Uberbot@69.252.219.76)
05:09.29Insanity5FuriousGeorge - I know.  but isps shouldn't poison their dns servers with that crap :)
05:09.59FuriousGeorgei was talking about your first question.  with the ringing pots lines
05:10.57*** join/#asterisk Smi|k (n=smilk@adsl-66-159-200-157.dslextreme.com)
05:11.44Smi|kDoes anyone here have experience with asterisk / crm / shopping cart / google api? I really want to make an all-in-one turnkey solution
05:13.05Insanity5FuriousGeorge - Ahh, ok.
05:13.22Insanity5FuriousGeorge - Now, what if the damn cell phone voice mail picks up?  and how does it know th difference between a ring and a pick up?
05:14.30FuriousGeorgeInsanity5: by default, fxo kewlstart signaling (i guess) detects the pickup on pots.  if the voice mail picks up then it answered
05:14.54Insanity5FuriousGeorge - I wondered how it detected the difference between a ring and say, a pbx generated ring.
05:15.21Insanity5Is there a special signal sent over the line once its picked up?
05:16.24FuriousGeorgeInsanity5: actually i dont know how it works.  i think the kewlstart signalling (vs. loopstart) just detects a hangup
05:16.47kuku5Is the 7940 a good secretary phone ?
05:17.04kuku5I'm looking at the two buttons and im worried it wont be...
05:18.49Smi|kor any idea where I'd look for that matter?
05:19.43Insanity5Hmmm, my spa-2002 is misbehaving.  Random dropped packets, ping times spiking to 2000 ms on the local subnet, no dialtone.
05:19.46Insanity5wtf
05:20.47*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
05:21.08wunderkinhopefully this will be the last time the modem gets put to sleep or unplugged
05:21.21Qorkyusing the new chan_capi (0.6)
05:21.22Insanity5?
05:21.36Qorkyhow can i define multiple incomingmsn's ?
05:22.02Qorkylike?      incomingmsn= XXXXXXX, YYYYYYY    ?
05:22.04*** join/#asterisk JonR800 (i=jon@p1mp.org)
05:22.11*** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus)
05:29.43*** join/#asterisk NewSole (n=dave@d38-53-48.commercial1.cgocable.net)
05:32.15FuriousGeorgeQorky: im not sure what ur trying to say
05:32.28FuriousGeorgeincomingmsn?
05:33.01Qorkydo you know you way around the capi.conf ?
05:33.02FuriousGeorgei guess its an isdn thing, which i dont know about
05:33.11FuriousGeorgeno sorry bro
05:33.17Qorkyno worries :)
05:33.19Qorkycheers anyway
05:36.45Insanity5Will this record natively in ulaw (so no ulaw>wav>ulaw transcode) -- I'm trying to record a greeting:
05:36.45Insanity5<PROTECTED>
05:39.09Corydon76-homeI think the ulaw suffix is .ul
05:39.38Insanity5[Description]
05:39.38Insanity5<PROTECTED>
05:39.55Corydon76-homeHave you tried it yet?
05:39.58Insanity5So how do I get it to record in ulaw?  do filename.ul?  ulaw is what is used everywhere else in asterisk for ulaw.
05:40.54blitzrageAstriCon 2005 pictures: http://leifmadsen.com/gallery/astricon_2005
05:41.23Corydon76-homePretty pretty blue eyes
05:41.39blitzragelol
05:41.52blitzragesurprisingly, I didn't get hit on that much at AstriCon :D
05:42.05Corydon76-homesurprisingly
05:42.06blitzragemust have been that beard I had for the first few days... my plan worked :D
05:42.07*** join/#asterisk orlok (n=jwr@202-44-174-4.nexnet.net.au)
05:42.24orlokAnybody know a place to get cisco sip images apart from cisco?
05:42.25orlok:)
05:42.33Insanity5orlok - google?
05:42.53Insanity5orlok - Or buy a service contract on one for 'overseas' off cdw.com -- it's like $10 or so.
05:43.47orlokInsanity5: yeah, i found lots of people asking the same thing
05:43.53orlokcisco shit me to tears
05:44.08Dr_Raythere is an ebay cd
05:44.10orlokthey ship routers that cannot function as advertised, unless you upgrade the IOS
05:44.32Dr_Raywe just paid $80 for a smartnet contract for 1 phone
05:44.34*** join/#asterisk gambolputty (n=gambolpu@72.240.242.4)
05:44.36*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:45.09Insanity5orlok - you're supposed to pay extra for that functionality too :).
05:45.09Insanity5hehe.
05:45.18orlokwhat, working hardware?
05:45.23Insanity5Dr_Ray - does your cco account get access to router ios images too?  Just courious.
05:45.29orlokthis is a VPN router with a crypto card
05:45.35orlokand they reboot every 15 minutes when you se the VPN
05:45.36Dr_Rayno, I believe I only get 7960 images
05:45.43wunderkinhehe, how come i'm getting disconnected when i press * even though i'm not using the h option anywhere :(  and theres nothing mapped in features
05:45.52Dr_Rayso I doubt I could get 7970 or 7940 images
05:45.54Insanity5We have some on some Cisco pixes, and I can get everything.
05:46.12Dr_RayI've never looked actually
05:46.53*** join/#asterisk MGSsancho (n=user@ppp-67-126-240-180.dsl.irvnca.pacbell.net)
05:47.04Dr_RayI've thought of mirroring the cisco images with a chinese/russian web hosting company
05:47.19Insanity5lol.  I believe it's already been done.
05:47.21Insanity5heh
05:47.38Insanity5Are the cisco ip phones really that great?
05:47.59blitzrageInsanity5: yes
05:48.12Dr_Rayonce you get sip 7.5 on it, it's like wonderfull
05:48.21Dr_Rayit was worth the $80
05:48.26Insanity5I mean, isn't it just like, it's another phone?
05:48.29blitzrageI have a 7960, a Polycom IP500 and a pair of Uniden UIP200's
05:48.37blitzrageInsanity5: you don't own one do you? :)
05:48.40Insanity5Nope
05:48.46Insanity5Do you use it for business use?
05:48.48blitzrageInsanity5: you'll understand when you get one :)
05:48.53Dr_Raydon't get me wrong, I like the budgetone 101..
05:48.55blitzrageInsanity5: yes... I use it for conference calls and such
05:49.03blitzrageI hate grandstream stuff :)
05:49.12Dr_Raybut for my front desk staff, the cisco was a no brainer
05:49.15Insanity5Home/soho or coporate use?
05:49.31orlokInsanity5: we have about 2 dozen here, business use.
05:49.37*** join/#asterisk sgorilla (n=tlp@cpe-24-160-119-179.houston.res.rr.com)
05:49.44wunderkinah i did a show features and * is hangup by default doh
05:50.07blitzrageInsanity5: home run business (my bedroom)
05:50.15e-HernickI command bits to move and they do.
05:50.22blitzrageInsanity5: but I do a lot of conference calls, and the Cisco phone just works great for it
05:50.45Insanity5blitzrage - Ahh -- didn't think it would be too hard to do the ccalls asterisk side with the flahs button and an ata
05:51.04Insanity5$18 software upgrade solution: http://cgi.ebay.com/Cisco-7960-7940-SIP-7-5-Firmware-Upgrade-CD_W0QQitemZ5819481213QQcategoryZ51204QQrdZ1QQcmdZViewItem
05:51.07Dr_RayI want to say we spent $400 on our phone, that's for phone, power adapter, and smartnet contract.. plus teh 15 hours of so figuring out how to get the UAL to run
05:51.20sgorillathats hella expensive
05:51.28sgorillahas anyone used the aastra phones
05:51.29Dr_Raywell, it's been hella rock solid
05:51.44sgorillawhat is the most rock solid phone?
05:51.44Insanity5Don't they have some color ones now too?
05:51.46*** join/#asterisk DougRoyer (i=doug@70-67.69-92-cpe.cableone.net)
05:51.46blitzrageI got my 7960 for free by running the Asterisk Documentation Project :)
05:51.52sgorillacool
05:52.07blitzragesgorilla: I think the 7960 is rock solid, along with the Polycom series of phones
05:52.14Dr_Raywhen I sold my boss on asterisk to replace our mitel sx-50, the only part of the page he did not glaze his eyes over was the cisco
05:52.38sgorillaDr_Ray: dont you need some other piece of hardware for the phones?
05:52.43sgorillalike a callmanager
05:52.44Dr_Rayhe'd seen cisco, and he knew that cisco made networking/voip
05:52.57Dr_Raysgorilla - no you can put a sip image on it
05:52.59blitzragecall mangler? nah
05:53.11e-HernickI have analysed the structural integrity of the 7960 phone, and let me tell you that your feeble "rocks" are unworthy of the name.
05:53.21sgorillasgorilla: you can use asterisk and cisco phones without having to buy anything else?
05:53.25Dr_Raynot trying to start a flamewar, but I personally don't like the polycoms
05:53.28Insanity5hmmm - check this out: http://cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0900aecd8030e546.html
05:53.39sgorillathe polycoms i heard take a long time to boot up
05:53.42Dr_Raythe 7970 does not have a sip image
05:53.46sgorillai dont want to wait 3 minutes for a phone to boot up
05:53.57Dr_Rayit takes the cisco 30-40 seconds to reboot
05:53.58e-HernickDr_Ray, what alternative conferencing speakerphones do you prefer ?
05:54.00sgorillai want somethign that is simple and have good speakers
05:54.04Dr_Ray7960
05:54.22Dr_Raymy dislike of polycom is purely based on looks
05:54.25Dr_Raynot actual use
05:54.30sgorillaoh
05:54.31*** join/#asterisk MGSsancho (n=user@ppp-67-126-240-180.dsl.irvnca.pacbell.net)
05:54.45*** join/#asterisk brookshire (n=mbrooks@esbrooks3.traveller.com)
05:55.13Insanity5Free after mail in rebate vonage unlocked adapters work for me I guess :)
05:55.16e-HernickI mean, surely the 7960 cannot compare to the Polycom SoundStation ?
05:55.35e-HernickAh, I see that you are a superficial person.
05:55.37wunderkinif i do a show features, what shows under current is what it should use right? Disconnect Call           *       #2       Attended Transfer                 *      if i do a * it still disconnects argh
05:55.41sgorillaInsanity5: can you hook up directly to vonage using sip
05:55.46sgorillaor do you use an fxo card to the router?
05:55.57e-HernickHe's using unlocked adapters.
05:56.08sgorillawhats a good simple phone for a business?
05:56.12sgorillathat is cheap
05:56.14e-HernickHe's using illegal hacking software to disable their coupling to Vonage.
05:56.17Insanity5sgorilla - fxo, but I unlocked them.
05:56.26Insanity5I disagree with illegal hacking software.
05:56.34sgorillasgorilla: how did you unlock it?
05:56.35e-Hernicksgorilla, what about a Sipura and a good analog phone ?
05:56.37Insanity5I bought the hardware, I'm entitled to it.  Doctrine of first use.
05:56.49sgorillaInsanity5: but you could of violated the DMCA
05:56.57Insanity5sgorilla - that is about it.
05:57.04e-HernickI like the Plantronics CT12.
05:57.06Insanity5sgorilla - But i'm using the software supplied with the phone.
05:57.13e-HernickBut it doesn't like WiFi.
05:57.40sgorillaInsanity5: its a phone that came with vonage?
05:58.01e-HernickThough for around 200$ you can get a nice corded phone, a Plantronics CT12 and a SPA-2002
05:58.01Insanity5sgorilla - No, a phone I bought at staples that said vonage ready on the box.
05:58.09Insanity5sgorilla - With no contract or service plan required.
05:58.36Insanity5sgorilla - see http://www.dslreports.com/forum/remark,14450684?hilite=pap2+unlock
05:58.38sgorillaoh
05:58.43sgorillacool, so they lost money on the phone
05:58.50sgorillathe probllem is there is no QoS
05:59.05Insanity5sgorilla - Best handled at the router, but it does suck.
05:59.11Insanity5The adapter marks the packets though.
05:59.47sgorillaInsanity5: what type of QoS do you do on the router?
05:59.49e-HernickWe all know that the DMCA is the fair and just law of the land, and that the infortainement lobby group has only our best interests at heart. Indeed, their innovative legislative approach helps the citizen stay on the digital path that has been laid by the glorious american corporate software companies.
06:00.40Insanity5sgorilla - you ahve two choices... by port, or by the ToS flag on the IP packets hwich the spa-2000 can flag.
06:01.06sgorillai am looking to do it by port
06:01.10sgorillaflagging the ToS bit
06:01.14e-HernickWho here would dare defy America by illegally perverting a Vonage-ready adapter for use with open source software ?
06:01.18sgorillathen using iptables to mangle it
06:01.28sgorillawell i guess its already mangled
06:01.36sgorillajust need to figure out how to use it with tc
06:01.42e-HernickWhy would you mangle a packed that's already been marked as high-QoS ?
06:01.49sgorillait seems that *bsd QoS is better
06:01.57e-Hernickhtb3 rules
06:02.01sgorillayeah i retracted my statement
06:02.04sgorillawhats htb3?
06:02.07Insanity5You just have to set the router to give it higher priority.
06:02.22e-HernickOkay, sgorilla, have you ever configured Linux QoS ?
06:02.28sgorillayes
06:02.32e-Hernicksgorilla, you complain about tc and you don't even know about htb3 ?
06:02.35sgorillabut very inexperienced
06:02.51sgorillai dont complain, just from reading over different mailling lists
06:02.56sgorillapf vs tc
06:02.57e-HernickWell then, I'll tell you what: Linux QoS is a pain to learn, but it's very powerful.
06:03.09e-Hernickpf certainly sounds rather easier to manage.
06:03.11sgorillae-Hernick: i have ntop set up so I can see traffic flows
06:03.19sgorillayeah it is easier, easier syntax
06:03.31e-HernickI'll give you that, anything has easier syntax than tc
06:03.46sgorillai am looking for a system to do bgp routing
06:03.52sgorillaand failover firewall/router
06:04.01sgorillawithout going the cisco router because of cost
06:04.04e-Hernickzebra ?
06:04.12sgorillayeah i was looking at that
06:04.16e-HernickWait, I think it has a new name.
06:04.18sgorillaactually its not zebra anymore
06:04.26sgorillaits quagga(sic)  or something like that
06:04.39e-HernickRight
06:04.47sgorillaopenbsd has this really nice redundant routing protocol
06:04.58e-HernickWell, BSD might be a good choice.
06:05.00sgorillawhere you can put a crossover cable on two routers, and do a ha setup
06:05.09sgorillai am much more familiar with linux though
06:05.19e-HernickYeah, same here.
06:05.22Insanity5bgp with anything but cisco... buy a damn $200 cisco off ebay if you have the money to buy bgp capaable links.
06:05.27e-HernickI've had an OpenBSD box for a while
06:05.31e-HernickI didn't like it so much.
06:05.40Insanity52600 3620 are good starts.
06:05.59e-HernickA used cisco. That's a thought.
06:06.03sgorillai am probably going to get some EOL cisco box to start playing around with
06:06.07sgorillayeah definetly
06:06.18Insanity5but you probably dont' need bgp :)
06:06.20e-HernickTell you what - you need redundancy ? Get both a cisco and a Linux box.
06:06.22Insanity5hehe
06:06.28sgorillai need to do vlan
06:06.34sgorillawhy not bgp?
06:06.35Insanity5vlan is layer 2
06:06.41e-HernickWhy BGP ?
06:06.42sgorillahave two multihomed fiber connections
06:06.45Insanity5cisco routers do routing ... layer 3
06:06.51e-HernickWhat kind of connections ?
06:06.55sgorillacisco routers do layer 2 also
06:06.58sgorilla100 mb
06:07.02Insanity5sgorilla - And... you can't afford a bgp capable router but are buying big fiber pipes?
06:07.06e-HernickNice
06:07.13sgorillahaha i am not buying the pipes =)
06:07.14Insanity5usually they don't come in 100 megabit unless they're fiber ethernet handoffs.
06:07.27sgorillathey are fiber ethernet handoffs
06:07.29Insanity5Most of the time the only thing that comes in a multiple of 100 is ethernet.
06:07.32e-HernickYeah, you get two 100mbit pipes
06:07.34sgorilla100 mb port
06:07.47e-HernickDo you even have the right to run BGP on these pipes ?
06:07.50Insanity5And very few people hand off the fiber at 100 megabit anymore unless distance is a concern.
06:07.57e-HernickI mean, are you gonna get registration with your upstream routers ?
06:08.08sgorillayeah
06:08.10Insanity5What isp?
06:08.12Insanity5cogentco?
06:08.16sgorillatime warner telecom
06:08.17Insanity5metro area network?
06:08.22Insanity5Is it in a data center?
06:08.30e-HernickIs it in a university ?
06:08.33sgorillanewly built small datacenter
06:08.35Insanity5Are you sure it's fiber?
06:08.39sgorillayes
06:08.40Insanity5Most likely fast ethernet handoff.
06:08.45sgorillawell starting out 20mb
06:08.50Insanity5Well, that's pretty stupid.
06:08.54e-HernickYeah, most likely they have fiber coming in the builiding and you get ethernet.
06:08.57Insanity5Not many people use fiber for data centers.
06:09.00sgorillawith pri
06:09.06sgorillawhat do they use?
06:09.23e-HernickSo, you have two 100mbps links coming in your small datacenter, from two different providers ?
06:09.25Insanity5there used to be a time when it was the only way to get gigabit speeds, but even now gigabit ethernet handoffs are reaidly available.
06:09.45Insanity5e-Hernick - I kind of doubt they're terminated with fiber into his cage.
06:09.50e-HernickSo, are you trying to get equipment to route the whole datacenter ?
06:09.51sgorillaright now there is just 1 20 mb
06:09.53Insanity5Someone else is probably doing that for him and handing them ethernet.
06:10.01sgorillai am thinking about the future
06:10.07sgorillait needs to be physicall diverse
06:10.09e-Hernicksgorilla, are you getting a mere datacenter cage or are you renting a unprepared small room
06:10.23sgorillarenting unprepared small room
06:10.24e-HernickI mean, here in quebec city, most datacenters are actually in small rooms in the basement of the big hotels
06:10.51sgorillawhat is good price for 20 mbit ethernet handoff per month
06:10.55e-HernickThose hotels are full of datacenters and most of the fiber of the city comes there, and most of the big antennas are too
06:11.05sgorillaok telecom hotels
06:11.06Insanity5sgorilla - I seirosuly doubt you are getting fiber handoff from an ISP.
06:11.08e-Hernicksgorilla, depends where you are, depends what kind of QoS, if you get oversold..
06:11.14e-Hernicksgorilla, what provider
06:11.17sgorillaInsanity5: why?
06:11.18Insanity5sgorilla - You are getting ethernet.  You probably won't have fiber unless it was an ATM link.
06:11.20sgorillatime warner telecom
06:11.20e-HernickI mean, cogentco is cheap and dirty
06:11.25Insanity5sgorilla - there is no good reason.
06:11.25e-Hernicktime warner
06:11.29e-Hernickare they reselling from somebody
06:11.30sgorillafiber runs on the telephone pole nearby
06:11.50e-Hernicksgorilla, and you're getting two fiber runs or just one ?
06:11.58sgorillaone starting out
06:12.04e-Hernickso you have just one link
06:12.07Insanity5sgorilla - fiber ethernet if very unlikely except in some metro area networks.  almost all datacenters hand off etherent now adays, and the only other time you would use fiber would be for a long distance link, AKA ATM/oc3
06:12.09sgorillarun another to make it physically diverse
06:12.10e-Hernickhow are you gonna get redundancy
06:12.32e-HernickOkay, so let's recap the situation.
06:12.35sgorillanot going to be redundancy at first
06:12.48sgorillait has fiber on the pole next to it
06:12.57sgorillagoing to a cisco 5650
06:13.01sgorillawell not sure the number
06:13.06e-HernickYou've rented a small room that you want to convert into a datacenter. You get a 20mbps fiber drop with Time warner tier-2 bandwith. You want to have a redundant pair of routers to route the data to your small datacenter.
06:13.12Insanity5I think he's a little confused :)
06:13.14sgorillaits like an old cisco that handles fiber, with a ethernet handoff
06:13.18Insanity5And he's buying non-existant routers.
06:13.20e-Hernicksgorilla, is that accurate ?
06:13.35e-HernickHe's probably getting the 5650 from time warner
06:13.37sgorillatier-1 bandwidth i believe
06:13.39sgorillawhere im at
06:13.46sgorillayeah thats coming from time warner
06:13.47Insanity5time warner is cogentco is crap.
06:13.48e-Hernicksgorilla, okay, is the rest accurate
06:13.58*** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com)
06:13.59Insanity5but, it's cheap b/w
06:14.00e-Hernicksgorilla, do you care much about traffic shaping ?
06:14.01sgorillaoh timewarner uses cogent?
06:14.06sgorillayes i do
06:14.08Insanity5nope, but it's only one step above them.
06:14.09kuku5hm
06:14.13e-Hernicksgorilla, older ciscos are not the best choice
06:14.15sgorillawhat is the best ?
06:14.18Insanity5He shouldn't need to shape traffic in a data center.
06:14.32kuku5well he has only 20mb
06:14.32Insanity5Somethings wrong with your network allocations if this is the case.
06:14.37*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
06:14.39sgorillawhy wouldnt you shape traffic?
06:14.51e-Hernicksgorilla, tell you what. Build two routers for your datacenter. One uses gentoo and one uses freebsd. You put them in a failover setup.
06:15.10e-HernickThat forces you to configure two routers identically. Then, you choose whichever router works best as primary, and the other one as failover.
06:15.21e-HernickYou get to use both pf and tc. That's a win/win situation.
06:15.28sgorillayou wouldnt use two identical for failover?
06:15.29sgorillahaha
06:15.36sgorillawhat is cheap b/w?
06:15.38e-HernickNo, you want to different machines.
06:15.40Insanity5bandwidth
06:15.44e-Hernickcheap bandwith is like, cogentco
06:15.53Insanity5e-Hernick - time warner isn't much better.
06:15.58sgorilla2000 - 20mbit a month
06:16.07e-Hernick2000 is reasonable.
06:16.16Insanity5that's not dirt cheap, but locatino location location is what matters.
06:16.24Insanity5where is it?
06:16.29sgorillaincludes full e1 -  and 100 dids
06:16.38sgorillawell 2 mbit off the 20 for voice
06:16.45sgorillanot paying for the fiber install
06:16.50e-HernickIf I were in your place, I'd get a few 100$/2TB dedicated servers across the country and use them to boost your own bandwith
06:17.03sgorillayeah
06:17.13sgorillai am thinking i should colo
06:17.17e-HernickSo you get a full e1 ?
06:17.18e-HernickNice.
06:17.21sgorillaand just get ports to connect to other datacenters
06:17.24kuku5e-Hernick: have  you done that ?
06:17.37Insanity5sgorilla - I'm afraid you're confused.  If your term of "data center" is the back of "your office", your using the wrong terminology.  It's also clear that you have a long rocky road ahead of you :)
06:17.38sgorillathen you can use other providers
06:17.47sgorillaInsanity5: true =)
06:17.54e-Hernickkuku5, yes, I've done that. My on-location servers route some data through my hub servers which have better connectivity
06:18.09sgorillaInsanity5: what do you need for a datacenter?
06:18.17kuku5you have a direct link between them ? ( say no )
06:18.20sgorillait will have hvac with humidity control
06:18.26sgorillabut not n+1 hvac
06:18.28e-Hernickkuku5, no, I go through the internet
06:18.32e-HernickWell, it's like the boat and ship distinction.
06:18.35Insanity5sgorilla - What are you doing?  data centers are generally points where people can purchase bandwidth?
06:18.35sgorillajust be able to put in mobile cooling
06:18.48Insanity5sgorilla - As in, you order a cage and some b/w.  usually run by telco's or major isps.
06:18.49sgorillawell its just a server farm
06:18.49e-HernickYou're describing a small computer room. A boat, not a ship. Not a datacenter.
06:18.52kuku5that $100 idea is not so bad
06:19.00kuku5didnt think of that
06:19.01sgorilla=)
06:19.12e-HernickI mean, most businesses with a few employees have more power than that in their computer room and way more e1s than you.
06:19.13kuku5what do you route ?
06:19.18e-HernickCalling it a datacenter is pretentious.
06:19.29sgorillaok what should i call it?
06:19.34Insanity5sgorilla - server room?
06:19.35kuku5a room
06:19.38sgorillaserver room
06:19.39sgorillaok
06:19.41sgorillaheh
06:19.43wunderkinthe dungeon
06:19.48Insanity5sgorilla - Now let's get back on track.
06:19.53e-HernickI move Web traffic, IAX2, SIP, Jabber, rsync, bittorrent and more.
06:20.07e-HernickOh, lots of mail too.
06:20.10sgorillaok
06:20.14e-HernickI use front-end servers with excellent connectivity
06:20.26Insanity5sgorilla - So your ordering internet access, 20 megabit.  If it's fiber, at 2k/month, it's probably a fractional ds3 over ATM (not ethernet).  you'll need cisco hardware to terminate that most likely, and it isn't cheap.  You'll pay $1k for the router, and another $2-3k for the damn cards.
06:20.40e-HernickThey're providing the router.
06:20.42sgorillanot paying for the router termination
06:20.47sgorillaor the fiber installation
06:20.58e-HernickHe gets an ethernet port in a 5650
06:21.00sgorillaor actually the internet connection itslef
06:21.04Insanity5sgorilla - Often termination equipment is included.  Then your isp is going to give you an ethernet cable off that router.
06:21.10sgorillayeah
06:21.11sgorillayes
06:21.18Insanity5sgorilla - Your best bet is to use the router they provided for you.
06:21.29sgorillabut i dont think i have access to configure it
06:21.30Insanity5sgorilla - I don't know what your doing that's special that needs further routing.
06:21.36sgorillawell nothing at first
06:21.41sgorillajust need some type of firewall
06:21.43e-HernickYeah, your ISP is not going to send a couple of grunt cable layers with dirty fingers who leave you 3 feet of dirty unterminated raw fiber
06:21.45Insanity5sgorilla - Well that's betwen you and them.
06:21.46sgorillawell redundant firewalls
06:21.48e-HernickThough that would be fun
06:22.02Insanity5sgorilla - Well two cisco pix w/ failover cables.
06:22.09Insanity5sgorilla - For any kind of large enterprise setup.
06:22.11e-Hernick"and remember, don't look into the fiber laser; it's live !"
06:22.21sgorillahaha
06:22.30Igbothom_III"Please don't look into the Laser with remaining eye"
06:22.32kuku5i think you guys are overkilling this redundancy thing
06:22.39Insanity5sgorilla - You can do port forwarding from there... but you still don't have failover (and never will) on your top end router, but that's ok.  Some things you have to accept.
06:22.43sgorillawhat is the difference between pix and linux
06:22.49sgorillai want to be able to run snort with barnyard
06:22.56brookshirepix sucks
06:22.57brookshire:)
06:22.57Insanity5snort doesn't need to be on the router.
06:23.01drumkillabrookshire: your mom
06:23.02sgorillatrue
06:23.03Insanity5brookshire - but at least it's reliable.
06:23.06e-Hernicksgorilla, why are you still looking for a solution ? I've given you the perfect solution already.
06:23.17sgorillathe pix?
06:23.22brookshireInsanity5: and very hackable (last month)
06:23.27Insanity5sgorilla - you are dealing with an enterprise level task here.  You need to spend the money to buy something that is reliable and will do the job.  Linux ip tables code quite frankly, isn't up to par for large scale routing.
06:23.33e-Hernicksgorilla, you need to build two reliable, identical computers with no mechanical storage, just flash. You configure one with FreeBSD and the other one with Linux.
06:23.39kuku5image_version : "P0S3-07-3-00"
06:23.39kuku5FirmLoadID : "PC030301"
06:23.39kuku5DSPLoadID : "PS03AT38"
06:23.47e-HernickNow, whichever works best will become your primary, and the other will be the failover.
06:23.49kuku5Does this mean it has old software?
06:23.55kuku5(cisco 7940)
06:24.03brookshireInsanity5: we have a linux firewall.. it pushes our 10mbit fiber and 100mbit colo
06:24.05sgorillae-Hernick: : that sounds like a good idea
06:24.15brookshireno problems yet and it's been there for a couple of months
06:24.28drumkillathat's because linux is hot
06:24.42Insanity5brookshire - It can be done, but I really don't advise it once you move one step further into mission critical tasks.  It's just not worth betting pc hardware on.
06:24.51brookshiredrumkilla: and it's you're mom ;)
06:24.57Insanity5brookshire - optimized code, risc processors, and processor switched routing just handle the nasty stuff so much better.
06:25.05brookshire"you're" must misspell it correctly ;)
06:25.27sgorillai heard cisco pix can get unstable under DoS
06:25.29e-Hernicksgorilla, it is a good idea. You have two routers that can do the same thing, and if one fails, the other takes over. And even if one of them you can't get to work just perfectly, the other one will.
06:25.35Insanity5sgorilla - I imagine any firewall can.
06:25.36brookshireinsain: it runs digium.com ;)
06:26.07Insanity5brookshire - but seriously, if you reboot that for 5 minutes at 3 am, just a minor inconvience.
06:26.14Insanity5Might loose an order or something.
06:26.22e-Hernicksgorilla, once you get a running setup, if you decide that either FreeBSD or Linux works much better than the other, you have two identical machines; just put them both with identical software.
06:26.30sgorillaok
06:26.42brookshireInsanity5: nah
06:26.45sgorillais there anytype of fastboot option with linux or bsd?
06:26.46Insanity5I just don't advise it... you... and your eventual replacement may not like it's  it's not industry standard for routing, and god, is it a pain in the ass to great a policy-map for QoS.
06:26.54sgorillalike be able to load up memory right away
06:27.00sgorillalike a laptop does with a snapshot
06:27.16orloke-Hernick: only if that one is working imperfectl enough that the other realses it
06:27.22*** join/#asterisk tengulre (n=tengulre@222.90.66.4)
06:28.04Insanity5I've used linux boxes for doing certain things... like routing T-1's, but the moment you touch BGP, some high levle QoS stuff (police/service maps), source based routing, or other advaned traffic mangling, I avoid linux.
06:28.17brookshireos anyways... i think the datacenter is more unreliable than the setup we have
06:28.21sgorillai heard once you go over 100mb its not good
06:28.37sgorillato use linux or vanilla bsd
06:28.45wunderkinbrookshire: which one?
06:28.54Insanity5sgorilla - You'll be applying kernel hacks at minimum... assuming that 100mb is composed of tons of individual flows.
06:29.20brookshireapi-digital... they only have like 2 generators and 10 connections ;)
06:29.25Insanity5sgorilla - But for the price of the grade of hardware I'd recommend running a production linux router on, you can pick up a cisco that can do the job.  you don't need any iterfaces other than ethernet.  Troll ebay.
06:29.46brookshirebut cisco is overpriced
06:29.51brookshireand we can do it cheaper :)
06:29.56Insanity5brookshire - Depends, I don't buy new cisco
06:30.11brookshireand we can do nifty stuff with it too.. because it's open
06:30.13Insanity5brookshire - Ok, $230 on ebay, cisco 3620, add $50 for ethernet nic, and it's routing 15 megabit nicely.
06:30.29e-Hernicksgorilla, you have no need for BGP
06:30.29Insanity5brookshire - I can't do that... reliabily... with linux... for that price.
06:30.32brookshireInsanity5: we're firewalling 100mbit
06:30.38brookshire+10 mbit fiber
06:30.45drumkillawith a beefy server :)
06:30.53Insanity5brookshire - Well the fiber is likely not termianted into the linux box, right?
06:30.56drumkilla</stupid router war>
06:31.00sgorillahaha
06:31.04e-Hernicksgorilla, all you need is a simple router with failover.
06:31.09sgorillayup
06:31.10brookshireInsanity5: it terminates to cat5 :)
06:31.11e-Hernicksgorilla, answer this: why do you need BGP ?
06:31.14Insanity5brookshire - Somethings in the middle, handing it off to ethernet.
06:31.18sgorillai dont need bgp
06:31.21brookshirebut we do the routing
06:31.25Insanity5e-Hernick - He probably doesnt' even have an AS number.
06:31.26sgorillajust something i was thinkg about for the future possibly
06:31.30orloki hate cisco
06:31.31drumkillablah blah blah blah ...
06:31.32orlokwith a passion
06:31.36e-Hernicksgorilla, the future is the future. When you get there, make your choice.
06:31.46sgorillatrue
06:31.48brookshireInsanity5: it's not a cisco.. nor a router
06:32.00brookshirejust an ethernet converter
06:32.01e-Hernicksgorilla, the redundant router solution that I propose will cost you very little. How many network interfaces do you need in each router ?
06:32.11Insanity5brookshire - You can do that for $3k in cisco equipment.  I'd rather the flexibility and more mature ip stack for routing.  I'm sorry, but I don't want to keep screwing with tcp memory buffers and otherwise to keep the linux box roting 100 megabit
06:32.21Insanity5brookshire - And i'm quite suprised you need 100 for digium.
06:32.50e-HernickInsanity5, he's going to be routing 20mbit, and he's going to have both a FreeBSD and a Linux router. If either proves to be unsuitable, both will be converted to the same OS.
06:32.51Insanity5brookshire - And if you're already getting it in ethernet, you may onlyn eed a firewall, better/cheaper cisco choices for that.
06:32.59sgorillae-Hernick: probably 2 ports
06:33.23sgorillais it possible for switches to failover ?
06:33.24e-Hernicksgorilla, very well then. Do you know what kind of computer you want to build ?
06:33.25X-Robpix -is- not cheaper than a 3620.
06:33.36Insanity5e-Hernick - I just can't pay $2-3k in monthly internet costs, and not justify a damn $400 router in the budget, sorry.
06:33.39e-Hernicksgorilla, not with cheaper switches. You're going to have to trust the switches.
06:33.40sgorillae-Hernick: probably a linux box 512 megs ram, celeron
06:33.43Insanity5X-Rob - It isn't, but I was referring to 100 mb
06:33.44brookshirepix is a piece of crap.. i use to use one daily
06:34.02sgorillawhat router do you suggest?
06:34.10sgorillaold cisco router from ebay?
06:34.20sgorillai need sometype of switch that can do vlan atleast
06:34.21brookshiredigium pci card :)
06:34.29brookshireat least for t1
06:34.29brookshirehehe
06:34.52Insanity5sgorilla - you can get a 24 port maanged vlan capapble cisco switch for $500 on ebay.
06:35.48sgorillawhat model would that be?
06:35.50X-Rob29xx switches are 'ok'
06:36.00Insanity52940xl (make sure it's the 8 meg flash one) will work.
06:36.04X-Robdon't get a 19xx, they don't do 802.1q (eg, 'standard' vlans)
06:36.11Insanity5the 4 meg flash doesn't do vlan
06:36.18Insanity5that's the cheapeast option
06:36.31Insanity53500 series are a better choice for something more modern, but you sound like you're cheap today.
06:36.39sgorillayes
06:36.48kuku52500 is doable
06:36.59Insanity5?
06:37.00e-Hernicksgorilla, I highly suggest that you go with high-quality hardware. I suggest a S754 board with a dual ethernet Intel NICs and one onboard ethernet port. No CD, HD or FD - you run the machines off a 512mb flash drive. Get a very good power supply for the machine, and UPS for both. Link them together through the serial port for failover.
06:37.26e-Hernicksgorilla, use the machines to do traffic shaping, routing, but also VPN and transparent proxying
06:37.37sgorillaok
06:37.42Insanity5trasnparent proxiyng... iick
06:37.45sgorillarun snort on a different box?
06:37.46drumkillaso how about that Asterisk
06:37.49drumkillait's pretty cool, huh
06:37.57sgorillayeah running asterisk also
06:38.06brookshiregentoo + digium pci card = one heck of a router :)
06:38.08X-Robdrumkilla - asterisk eh? I've heard something about that.
06:38.08sgorillanot sure how the t1 is going to be handed off
06:38.10e-HernickYou have two routers, one primary, one secondary
06:38.14drumkillait's pretty awesome
06:38.20drumkillait lets you make phones ring
06:38.22Insanity5e-Hernick - the TCO for most companies just won't pan out for custom programming and distrobution creation.
06:38.24e-HernickYour data logging and monitoring will run on the secondary
06:38.31X-Robphones eh? Wow. Fancy that.
06:38.35sgorillaok
06:38.38X-RobThere should be an IRC channel for talking about asterisk, you know.
06:38.43sgorillahaha
06:38.48Igbothom_IIIX-Rob; what for?  :)
06:38.52drumkillaX-Rob: I agree, let's make one
06:38.54brookshiredrumkilla: i wish chatrooms had a mute button ;)
06:39.00Insanity5it's /ignore :)
06:39.00sgorilla##asterisk
06:39.02drumkillahow about #asterisk-forrealthistime
06:39.07e-HernickInsanity5, you know, I agree. But I like open source, and this is open source.
06:39.14X-RobI was thinkging of #asterisk-no,really.
06:39.20brookshireInsanity5: but that's like 8 buttons.. i just need one
06:39.24e-HernickOkay, so let's talk about asterisk
06:39.30e-HernickWhat about we set up a conference call for the channel ?
06:39.31brookshirea big red one :)
06:39.32sgorillahow would the routers physically be hooked up?
06:39.33*** join/#asterisk glomph (n=black@c-24-18-145-249.hsd1.wa.comcast.net)
06:39.35wunderkinX-Rob: you don't want to join me in #3,000?
06:39.40drumkillalet's play name a feature, and it gets in!
06:39.43Insanity5e-Hernick - for 20 megabits.... I can't even buy one machine for the cost of a cisco router that will do it right the first time with an excellent suppotr staff if a crisis arises.
06:39.49drumkillasomeone name a feature and I'll write it real quick
06:39.49X-Robdrumkilla - porn-on-demand!
06:39.52*** part/#asterisk glomph (n=black@c-24-18-145-249.hsd1.wa.comcast.net)
06:39.52brookshirehaha... /disco inferno
06:39.58brookshirebest irc command ever!
06:39.58e-Hernicksgorilla, you put a switch between the routers and the 5650
06:40.05drumkillaalright, to continue this discussion, join #2,000
06:40.08Insanity5e-Hernick - If you want to put the linux box behind the router for VPN, proxy, etc, I can see that.
06:40.09e-Hernicksgorilla, get an old cisco switch, those rule
06:40.18sgorillaok
06:40.20X-RobHeh
06:40.21e-HernickInsanity5, Linux routers are way more flexible.
06:40.31brookshiree-Hernick: agreed :)
06:40.36sgorillabut that switch will be different from the server switch correcT?
06:40.39drumkillakids just aren't curious anymore these days ...
06:40.47e-Hernicksgorilla, yes
06:40.54sgorillalike a small switch
06:41.00sgorillawhat would you use in that case?
06:41.07Insanity5e-Hernick - Yes, but they shouldn't be your core routing equipment.  Use it to mangle and tag the packets, and let the cisco apply a little to no cpu usage policy-map on the ToS bit...
06:41.21sgorillaliike 4 ports
06:41.25sgorillabut reliable
06:41.39e-Hernicksgorilla, (the internet) --> (5650) --> (small, old cisco 100mb switch) --> (primary router and secondary router) --> (server switches)
06:41.40sgorillalinksys switch
06:41.44sgorillabhaha that is cisco
06:41.50e-Hernicklinksys is low-end.
06:41.53sgorillaok cool
06:42.16e-HernickOne thing, if you decide to go with a cheaper switch, you can make it more reliable in two ways: install passive cooling, and get a better power supply
06:42.19sgorillahow about running snort
06:42.29e-Hernicksnort can be run on either your primary and secondary routers
06:42.33sgorillaok cool
06:42.37e-Hernickyour secondary doesn't have to be entirely idle either
06:42.44Insanity5snort can run anywhere on the network.
06:43.00sgorillathat sounds like a decent setup
06:43.07e-HernickYes, but he might have more than a single server switch
06:43.10wunderkindrumkilla: speaking of features...
06:43.16e-HernickAs a matter of fact, I would recommend at least two server switches.
06:43.25sgorillawhy is that? for failover?
06:43.26Insanity5e-Hernick - Now... if he's at this stage, let's see how many weeks this will take to set up.
06:43.33e-HernickPut 4 network interfaces in each of the routers
06:43.33Insanity5:)
06:43.47e-Hernickhave two server switches
06:43.47wunderkinwhen i do a atxfer and try to disconnect the call while its dialing, i get put into  never-never land, this is what i see http://pastebin.ca/26029
06:43.51sgorillaok i see
06:43.57e-Hernickyou can have redundancy on the server level too
06:44.11e-Hernickor you can simply group off your servers
06:44.16Insanity5For some reason, I was always someone who advocates the best solution to a problem... and I have no problem recommending AD to an enterprise for directory services, while staying by apache to large scale vhost/ssl deployments.
06:44.21sgorillai guess each computer can have redundancy hooking up to two switches
06:44.23Insanity5or asterisk for phone
06:44.29sgorillaatleast have the core servers run on two switches
06:44.31sgorillafor failover
06:44.40e-HernickYou don't necessarily need to configure failover at first
06:44.45brookshireAD is lamer than cisco
06:44.51sgorillaActive Directory?
06:45.03Insanity5brookshire - But it works excellent for certain corporate situations, on a TCO basis.
06:45.04brookshireldap!
06:45.07X-RobAdal Dilaton?
06:45.10e-HernickBut just get enough equipement to do it when you have a problem. If one of your switches were to fail, you have spare equipment which is connected and live
06:45.13brookshirejust use ldap
06:45.29Insanity5brookshire - And hire 4 more people to manage the 500 w2k desktops?  nope.
06:45.31brookshireyou know how many applications support ldap?
06:45.33sgorillaok that sounds good
06:45.41e-Hernicksgorilla, make sure to have a very good monitoring system that contacts you immediately if any problem sis detected
06:45.41brookshirehow many *more*
06:45.49sgorillae-Hernick: like nagios?
06:45.50brookshiread is ldap you fool
06:45.52e-Hernicksgorilla, a part of this system must be installed outside your datacenter
06:45.53brookshireand free
06:46.02Insanity5brookshire - But any ldap doesn't integrate into a win32 environemnt in the same manner.
06:46.04e-Hernicksgorilla, like nagios or zabbix yes
06:46.06sgorillaits a "server room"
06:46.11brookshireor if you want to pay.. but novell's directory
06:46.11sgorilla=)
06:46.12e-Hernickyes, it's a small server room
06:46.16sgorillavery small
06:46.17brookshireit's far superior ;)
06:46.20e-Hernicka datacenter is something bigger
06:46.30Insanity5brookshire - novell's stuff is good and comes with the necessary tools (included) to keep TCO down.
06:46.34sgorillanever heard of zabbix
06:46.41*** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net)
06:46.57e-Hernicksgorilla, redundancy buys you reliability. If you can lose any hardware due to failure and still keep your system running, that is better than having "invincible" hardware that cannot go down
06:47.02Insanity5brookshire - I like Novells stuff, and it is often a tossup.  Typically, novell works better then there are more non-win32 hosts, or many older win32 platforms (pre-w2k).
06:47.10*** part/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
06:47.19Insanity5brookshire - Novell is quite cheaper, but sometimes a little bit more to maintain.
06:47.27drumkillawunderkin: yes?
06:47.34Insanity5brookshire - but at least it includes a lot in the licenseing, like their mail server.
06:47.34shido6Novell?
06:47.39sgorillazabbix screenshots look cool
06:47.44shido6what Novell product is worth buying?
06:47.45Insanity5shido6 - scroll up a but :)
06:47.49Insanity5bit
06:47.50e-HernickAs for my two server switches, if I were on a buget, I'd go with one gigabit switch and one fast ethernet switch
06:48.04sgorillai guess you want to collect information on all the computers
06:48.05brookshireshido6: their ldap stuff is unmatched
06:48.08X-RobSTP (Spanning Tree Protocol) is your friend, people.
06:48.12X-RobLearn to love it.
06:48.13sgorillaand store them in a database, also do remote syslog monitoring
06:48.14Insanity5e-Hernick - and why does he need gigabit?
06:48.14shido6ok I know nothing about ldap
06:48.24e-Hernicksgorilla, yes, you want to check all servers and all services, as well as all connectivity
06:48.31brookshireldap is like a glorifed nis
06:48.32e-Hernicksgorilla, if at all possible you want gigabit between the servers, because of backups and data transfers between servers.
06:48.34Insanity5e-Hernick - Even with over 600 servers here running multi-gigabit sql queries, I don't need gigabit anywhere but the core.
06:48.44sgorillaalso have a dialup to access a terminal
06:48.49sgorillaand be able to ssh on the network
06:48.50wunderkindrumkilla: ? did you see my question after that :D
06:48.52kuku5Oct 20 01:15:14 WARNING[10101]: chan_zap.c:770 zt_open: Unable to open '/dev/zap/pseudo': No such device
06:48.52kuku5Oct 20 01:15:14 ERROR[10101]: chan_zap.c:6731 chandup: Unable to dup channel: No such device
06:48.52kuku5Oct 20 01:15:14 WARNING[10101]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device
06:48.52kuku5Oct 20 01:15:14 WARNING[10101]: app_meetme.c:230 build_conf: Unable to open pseudo device
06:48.53sgorillaif the fiber goes down
06:48.55kuku5What does this mean ?
06:48.57Insanity5PASTEBIN.COM!
06:49.00e-HernickIf you are building a small network, gigabit is not much more expensive.
06:49.00sgorillaalso be able to access serial port on cisco
06:49.02kuku5ergh - sorry
06:49.07e-HernickFor large networks, gigabit is much more expensive per port
06:49.14Insanity5e-Hernick - it is expensive for Cisco (quality) stuff.
06:49.15shido6ztdummy in Makefile kuku5  if u have no digium gear
06:49.22brookshirekuku5: install ztdummy?
06:49.28kuku5i thought i did
06:49.35shido6recompile zaptel
06:49.43e-HernickInsanity5, that's why you don't get cisco stuff. You run two parallel server networks, one fast ethernet and one gigabit, when on a budget. Then, you can have failover if either switch fails.
06:49.44Insanity5kuku5 - modprobe both, remove them, readd
06:49.52e-HernickI like the netgear gigabit switches.
06:49.58kuku5modprobe ?
06:50.12Insanity5e-Hernick - Yeah, non-name brand switch gigabit vs 100 megabit name brand at your core backbone.  Not a good idea.
06:50.14e-HernickThey're fast, have vlan, management, jumbo frames and cost around 25$ per port of gigabit
06:50.32Insanity5e-Hernick - and so much for running netdisco, or millions of other utilities that make a large netowrk amangeable throguh cisco switches.
06:50.36sgorillai heard SMC are decent
06:50.46kuku5dell are ok
06:50.53Insanity5dell are rebranded JUNK
06:50.55Insanity5Don't bother
06:50.58e-HernickInsanity5, I fully agree. If you're running a larger network, you've got to get bigger, more expensive switches. And then you can't afford gigabit.
06:51.19e-HernickBut I'm very good at running small networks, and nowadays I deploy gigabit, and it's worth it.
06:51.22Insanity5We have 50 switches, 10 dells, 40 ciscos.  Guess what?  1 cisco failed, and 3 dells in the last 2 years.
06:51.22e-HernickNot much more expensive.
06:51.33shido6dood
06:51.36shido6dont do it
06:51.45sgorillaInsanity5: using what router?
06:51.46e-HernickYou are running a large network, your considerations are very different.
06:51.47Insanity5e-Hernick - But dual router and crappy switch?
06:51.55Insanity5sgorilla - doesn't matter, they're switches.
06:52.03sgorillai mean what switch
06:52.06shido6buy the giga-hardware with mad reviews and can handle a a lot of packet per second
06:52.13shido6packets
06:52.14brookshirenot all switches are equal
06:52.17Insanity5sgorilla - dell 48 ports, and one dell 24 port gigabit
06:52.20shido6hell no they are not
06:52.24brookshirearp floop a cheap router and find out :)
06:52.34brookshireINSTANT HUB!
06:52.37sgorillahaha
06:52.54Insanity5We finally just ebayed our dell switches.
06:52.54sgorilladell should not be vulnerable to that
06:52.55syle2where can i view the struct table for cdr?
06:52.57sgorillathat is old school stuff
06:53.13Insanity5sgorilla - dell is rebranded garbage.  and they're gigabit swtiches have trouble syncing to some nic cards, big time.
06:53.24Insanity5sgorilla - Combine that with no way to get error counters on the interface, and you have problems.
06:53.43wunderkinsyle2, README.cdr is that what you mean?
06:54.10sgorillaInsanity5: netflow looks good on cisco
06:54.11Insanity5If you must have a cheap managable switch, intel 510T's are OOOOOLD buy reliable.
06:54.15Insanity5and big.
06:54.17sgorillayou can integrate that with ntop
06:54.21syle2no the actual c struct
06:54.30wunderkinok
06:54.38syle2hmm think i found it
06:54.42Insanity5sgorilla - netflow is awesome.  cpu and i/o heavy though.  In many cases, nbar does the trick, and in msot cases, I questino why you actually need it.
06:54.42syle2--/usr/include/asterisk/cdr.h
06:54.46drumkillagrep "struct ast_cdr" include/asterisk/*
06:54.47drumkilla:-p
06:54.55shido6I stil own and ude my old DS104 netgear hub on my local lan for xbox xfers
06:55.02shido6its not plastic
06:55.13syle2just making sure src and dst are chars hehe
06:55.17sgorillawhat is a good progam for monitoring load, temperature, cpu usage on windows boxses?
06:55.28sgorillashido6: you mod xboxes?
06:55.35shido6:)
06:55.37Insanity5sgorilla - Umm, snmp + your favorite front end?
06:55.59Insanity5sgorilla - manually populate them in mrtg/rrdtool, or take the lazy way out and use cacti and/or big brother.
06:55.59shido6blame Mark, I saw his xbox with asterisk on it and felt the need to screw M$
06:56.12wunderkindrumkilla: after i press * to hangup while the other call is ringing, i just get the reorder tone, the original call is still active but no moh
06:56.20kuku5MODULES=zaptel tor2 torisa wcusb wcfxo wcfxs \
06:56.20kuku5<PROTECTED>
06:56.29kuku5I had this compiled already
06:56.31brookshireshido6: the pbxbox :)
06:56.36shido6yeah
06:56.59kuku5Any ideas?
06:57.01sgorillahaha xbox with asterisk
06:57.11Insanity5hehe
06:57.12sgorillathat is nuts, i have modded about 30 xboxes
06:57.20sgorillaxbmc is really nice
06:57.20Insanity5sgorilla - Want to fix mine? :)
06:57.22kuku5Insanity5: what should i do with the /dev/za/pseudo
06:57.30sgorillawhat is wrong with it?
06:57.42sgorillai have fixed out xboxes from just changing out compenents
06:57.43Insanity5kuku5 - I wasn't patying attention, but I fixed my ztdummy problems by removing and adding the proper module.
06:57.45brookshirekuku5: modprobe ztdummy ?
06:58.04Insanity5kuku5 - It in turn will add the depentent modules.
06:58.07sgorillai havent done any type of stuff with logic analyzers
06:58.10syle2hmmmm
06:58.16kuku5[root@mypbx zaptel]#  modprobe ztdummy
06:58.16kuku5[root@mypbx zaptel]#
06:58.17Insanity5sgorilla - I have to hit the power button about 40 times to get it to power up.
06:58.17syle2char src[AST_MAX_EXTENSION];
06:58.29Insanity5sgorilla - oncei t's on it's fine.  It has a homebrew mod applied to it from many years ago.
06:58.30brookshirekuku5: mow try your conf
06:58.31syle2anyone know what AST_MAX_EXTENSION is set to or where i can find it
06:59.09kuku5WO!
06:59.10kuku5WOW!
06:59.13kuku5it worked ... WHY ?
06:59.21brookshireconfs need to be in sync
06:59.36brookshireztdummy uses the cpu to help with timing
06:59.53kuku5ok - so now my paging works :)
06:59.55Insanity5Somebody needs to make a stupid "timing device" that plugs into serial/usb/pci
06:59.58Insanity5for dirt CHEAP
06:59.59Insanity5:0
07:00.04kuku5I have a big problem with this 7940
07:00.07Insanity5It can't be that hard of a problem.
07:00.12brookshireInsanity5: it's called a x100p
07:00.13Insanity5kuku5 - send it to me.
07:00.13X-RobInsanity5 - you're nts.
07:00.13kuku5I ahve no clue how this will work for a secretary
07:00.15X-Robnuts
07:00.20kuku5I mean - 2 extensions
07:00.21X-RobX100p
07:00.23X-RobIt's like $2.
07:00.24X-Robget over it.
07:00.37Insanity5X-Rob - I could use an ebay clone, don't know how well it will work though.
07:00.39X-RobHowever, it's _more_ accurate to use RTC
07:00.48kuku5Which phones you guys use for secretaries?
07:00.54Insanity5X-Rob - I thought ztdummy was the worst option.
07:00.56X-Robyou don't care about the audio quality, you just want timing.
07:01.02brookshirekuku5: we love polycom :)
07:01.13brookshirethey have a neat one with addable extensions :)
07:01.18brookshireblock
07:01.19X-RobInsanity5 - RTC is the _best_ option on 2.6 kernels
07:01.20brookshirethingys
07:01.31kuku5can you transfer with one push ?
07:01.35brookshiremmmhmm
07:01.41kuku5or do you ahve to hit transfer, and then the line
07:01.43Insanity52.6.8-24.5-default #1
07:01.47Insanity5hmmm ok so I guess I'm good.
07:01.55brookshirethey can also be programed with xml
07:02.05brookshiresay like.. placed in a certain spot on your website
07:02.10kuku5yeh
07:02.16kuku5but thats not needed
07:02.23Insanity5Are there cheap cisco phones (old ones) that like, work only with mgcp?  Or have tehy all been firmware flashed over?
07:02.25brookshireit's good for redoing extensions though
07:02.25kuku5I need a girl to answer 5 calls withing 15 sec
07:02.36brookshireon like 400 phones
07:02.37brookshireat one time
07:02.38brookshire:D
07:02.48Insanity5kuku5 - You need an automated interactive girl attendant lol.
07:02.53kuku5what do you mean redoing ?
07:03.08kuku5Insanity5: no... if you have a good system you can do it
07:03.18kuku5but you need to be able to TRANSFEr a call with 1 button
07:03.41Insanity5kuku5 - Difficult without being rude to people (3 seconds to ask them who they want?  I can't say hello that quick).
07:03.43e-HernickThat's correct. You don't need a girl to simply answer the phone for you.
07:03.49e-HernickYou need a girl to record a very nice IVR
07:03.59kuku5no
07:04.07kuku5you need to answer the phone - period
07:04.13Insanity5Who wants to make me an intro prompt?
07:04.27brookshireallison can
07:04.31Insanity5I need some girl to make me a press 1 for blah thing.
07:04.33brookshirewhy do you buy one?
07:04.36drumkilla~thevoice
07:04.42drumkilla~allison
07:04.43jbot[allison] The IVR Voice, http://theivrvoice.com/ and http://thevoice.digium.com/
07:04.44*** join/#asterisk memic (n=memic@chicago089.server4free.de)
07:04.44Insanity5I need 7 words, that's it :)
07:04.50memicwow
07:05.00memichow to limit ring time to 10 sec? Dial(SIP/memic) .. ?
07:05.11brookshireInsanity5: it's like $12
07:05.15brookshirefrom digium
07:05.16memicDial(SIP/memic,10) seems not to work
07:05.24Koshatulmemi: isn't it Dial(SIP/memic,10) ?
07:05.26brookshirei'm sure you can afford it with all that expensive cisco gear you own
07:05.31kuku5these sidecars are expensive !
07:05.34Insanity5brookshire - It's for my personal use :)
07:05.48Koshatulmemi: try Dial(SIP/memic,10,) ?
07:05.54drumkillayou can use the Record dialplan applicaiton to make your own prompts
07:05.57drumkillaif you don't care about quality
07:06.15Insanity5drumkilla - That's what I'm going to do.  It can't sound... that bad :)
07:06.18memick
07:06.22drumkillait works ...
07:06.37Insanity5It's just... Press 1 for Me, Press 2 for joe (roomate #1) press 3 for bob (roomate #2).
07:06.43Insanity5like 7 words, literally.
07:06.52KoshatulInsanity5: you can record that yourself easily
07:07.01e-HernickGetting a good IVR recording can take some work.
07:07.07memicKoshatul problem is phone is ringing but my second phone is not
07:07.11drumkillaall of those words are there except for the names ...
07:07.17drumkillaand there are even some names, but not sure if those are included
07:07.20KoshatulInsanity5: actually, i got an old girlfriend to do the IVR for my company, she has a great voice :)
07:07.20e-HernickIf you're not used to making recordings, you might have trouble getting the right quality.
07:07.22drumkillacheckout asterisk-sounds from cvs
07:07.25drumkillaand check out what is in there
07:07.28e-HernickYou need to use a good microphone among other things
07:07.33memic(SIP/phone1,7,&SIP/phone2,7,)
07:07.41e-HernickA phone won't do for quality recordings.
07:07.42memicbut only phone1 is rining
07:07.52Insanity5drumkilla - I'm sure the prerecorded words are out there, except for the names.
07:07.56drumkillamemic: wrong syntax
07:07.57Koshatulmemic: nah, it's Dial(SIP/phone1&SIP/phone2,7)
07:08.07memicKoshatul thx
07:08.08memicwill try
07:08.25Koshatuli'm starting to hate engin, i can't seen to get DID working
07:08.32drumkillag'night
07:08.36e-HernickInsanity5, try writing a better script too. If you record it yourself or convince a female friend of your to do it for you, make your IVR nicer
07:08.36Koshatulnight
07:08.46brookshireb'killa ;)
07:08.54brookshirekillad
07:08.54drumkillabrookshire: !!!!!!!
07:09.02KoshatulInsanity5: you can use audiology and cut and paste celebrity voice as well
07:09.03Insanity5e-Hernick What else would you do?
07:09.15Koshatulmy friend did that for his answering machine, he cut and pasted star trek voices
07:09.17Insanity5e-Hernick - For a basic 3 roomate setup?
07:11.35memicdrumkilla & Koshatul thx everything is working now..
07:11.38memic:)
07:11.42e-Hernick"Hi! You've reached the insane trio of crazyness! If you wanna speak to Bille-Bob, press one. For Joe, press two. For Bob, press three. If you're not calling any one of us in particular or your touchtone is broken, stay on the phone and somebody may well pick up the phone. We laugh at you, puny caller."
07:12.04Insanity5What is festival?
07:12.08Insanity5e-Hernick - lol
07:12.23memici have strange messages in asterisk log
07:12.24memicOct 20 10:53:03 WARNING[12460]: Unable to forward frame
07:12.32Insanity5e-Hernick - That would be good, and scare more telemarkers away.  You think simple having the IVR will get you off most lists?
07:12.42memicafter zap driver reload problem was gone..
07:12.57Insanity5Oct 20 02:12:52 WARNING[17966]: app_festival.c:350 festival_exec: festival_client: connect to server failed
07:13.00wunderkincan someone check http://pastebin.ca/26029 for me? trying to do an atxfer and when i try to disconnect while the call is ringing out it plays a reorder tone, it wont play the beep : (
07:14.14Insanity5wunderkin - As long as it isn't optioned out, it should play with Recorder.
07:14.23e-HernickInsanity5, my personal IVR has a menu option for telemarketers, which demands that if they want to establish unsolicited business contact with me, they must fax me a proposal containing a list of items and an explanation as to why I would want to do business with them. They are also told to never call me again.
07:14.34*** join/#asterisk psk (n=psk@golia.caltanet.it)
07:14.46e-HernickMy IVR also filters based on callerid
07:15.01brookshirepsk!
07:15.02e-HernickMost of the calls I get, I know who they're from. I greet the person by name and offer them only the appropriate menu options.
07:15.11brookshireoh.. different psk
07:15.21brookshiremy bad :)
07:15.21Insanity5e-Hernick - lol.  I figured the way most telemarketing setups work (the moment carrier is dedicated, an agent comes on the lines... usually a few seconds late -- have you ever said hello to them a few times?)  that an IVR period would throw this off.
07:15.29wunderkinInsanity5: as long as what isn't? recorder? :)
07:15.49Insanity5wunderkin - show application recorder.  It's in there about supressing the tone.  As long as that's not set, you'll be ok.
07:16.01e-HernickYeah, the predictive dialers. They get my IVR, so they have to follow the procedure I lay out for them.
07:16.22Insanity5e-Hernick - Well I figured they'd just hang up immediately and give up on the ivr.
07:16.23Insanity5hehe.
07:16.27e-HernickPrior business contacts get sent to a voicemail.
07:16.37brookshirethere is a nifty trap app out there somewhere
07:16.39e-HernickBut they rarely get to me.
07:16.50e-HernickThe trap app is a joke.
07:16.54brookshiretelemarketers come in and get caught in a loop
07:17.01brookshiree-Hernick: i know
07:17.06e-HernickThe way my app is set up, they get told exactly how to make a business proposal to me.
07:17.11brookshireyou can have whitelists though :)
07:17.27Insanity5Seems like too much hassle... over a "telemarketer... go away" greeting message.
07:17.35wunderkinInsanity5: im not sure what an application would have to do with options, and there is no application called recorder.. im doing an attended transfer
07:17.39e-HernickBut it's fun to make an IVR.
07:17.45e-HernickIt's your voice.
07:18.16sgorillaok im back
07:18.54*** part/#asterisk Deedubb (n=Deedubb@S010600055d22c57f.vf.shawcable.net)
07:18.55Insanity5hehe
07:19.35Insanity5wunderskin - I thought you were asking why it didn't beep when you recorded.  I meant show application record, sorry.
07:19.58wunderkingotta be something simple, ive had this working before but totally different setup.. im not doing recording and yeah thats what i thought you meant
07:20.18sgorillae-Hernick: is it one of those telemarketer torture ivr setups?
07:20.43Insanity5Do they even listen of just hang up?
07:21.01e-HernickWell, my logs show that some of them have listened.
07:21.11e-HernickThough, I'm gonna make a new version in the next few days.
07:21.16e-HernickI'm going to make it bilingual this time, too.
07:21.20e-HernickRight now it's only in french-canadian.
07:21.29Insanity5e-Hernick - How do you know that they were telemarkers?
07:21.29Insanity5hehe
07:21.52wunderkine-Hernick: oh yeah, one of those "for spanish press 9" things huh? haha i love those, how ignorant
07:22.24e-HernickInsanity5, I don't, but it took me a few minutes to record and program in AEL.
07:22.37Insanity5I had a phone number once of a humours IVR -- it was some company of "monkeys"....
07:22.48e-HernickYeah. It's "For english, press nine"
07:22.52Insanity5had options to accomodate all the most annoying things about IVRS and telephones.
07:22.56e-HernickDefault is french-canadian.
07:23.01*** join/#asterisk djin (n=djin@213-132-172-4.multikabel.nl)
07:23.03Insanity5hehe.
07:23.42Insanity5Yawn... I hate compiling crap on pentium 3's.
07:24.19e-HernickI'm getting great sound quality out of it. Using uncompressed wav files for the sound files and they have audio processing applied to them, they're not straight out of the microphone.
07:25.30wunderkin<PROTECTED>
07:25.45*** join/#asterisk Miggidy (i=user@dsl-202-72-180-171.wa.westnet.com.au)
07:26.21djinI need the run * in realtime , what would be the most stable release to go for (CVS of 1.2b1)?
07:26.26Insanity5e-Hernick - better than recording straight to ulaw with no transcoding?
07:26.28wunderkinah zap49 is me.. hmm
07:26.38e-HernickInsanity5, significantly
07:26.41sgorillathey need a ivr with simon says with dial tone
07:26.49Insanity5e-Hernick - how so?  Doesn't make sense.
07:27.22e-HernickSure it does.
07:27.29e-HernickI'm recording at 16-bit 48khz
07:27.38e-HernickI'm processing the audio at that resolution
07:27.41e-Hernickand then I downsample
07:27.57*** join/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net)
07:28.07movermorning
07:28.12mover:-/
07:28.46Insanity5how do you process to increase quality?
07:28.50X-Roboooh
07:28.52Insanity5I mean you can't make something of nothing.
07:29.03X-Robschlockmercenary has become part of blank label comics
07:29.04moverwho has got t38-bits get working and if so, how? :-)
07:29.09X-Roband *hah* he's got star billing!
07:29.11X-Robwoo howard!
07:29.39e-HernickInsanity5, http://www.l3i.ca/accueil.wav
07:30.00e-HernickThis is an example of a file in my IVR
07:30.46e-HernickI tell you, without some audio processing you won't get that kind of quality.
07:31.26sgorillawhy do you need audio processing?
07:31.43e-HernickYou don't need it, but it increases the subjective quality of the sound.
07:31.49e-HernickWhich is good.
07:32.03Insanity5how about a phone # since I don't have speakers hookedu p? :0
07:32.33*** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
07:32.39sgorillai wonder how that sounds over a phone network
07:32.45sgorillacompared to listening to the actual wav
07:33.05wunderkinfigured it out
07:33.47e-Hernickwell, call 8457382479
07:34.02*** part/#asterisk newmember (i=user@S010600036d1139fb.cg.shawcable.net)
07:34.23X-Rob8675309
07:34.52Insanity5Yikes, foreign language :0
07:35.01e-Hernickbut it sounds good
07:35.17Insanity5It does, but it's also hard to tell if you don't know the language.
07:35.22e-Hernickbetter than if I had recorded with a cheap mic and done no processing
07:35.26Insanity5hehe
07:35.27e-Hernickwell, the next version will be bilingual
07:35.32kuku5Did anyone use the gxp 2000 phones?
07:36.15*** join/#asterisk Aze` (n=aze@85-18-136-114.ip.fastwebnet.it)
07:36.51kuku5I cant get ulaw to work on them
07:36.54kuku5the best is gsm
07:36.57kuku5which sucks
07:37.41orlokDoes anybody know how to unlock a cisco ip phone?
07:37.57kuku5yah
07:38.15orlokahh
07:38.17kuku5via keypad?
07:38.17orlok**# apparently
07:38.22Aze`Anyone has last florz's patch ?
07:39.03orlokkuku5: apparently yeah
07:42.18kuku5which version
07:43.22X-Robkuku5 - they're good phone
07:43.44X-Robs
07:44.11*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
07:45.54*** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl)
07:46.08*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
07:54.25*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
07:55.33*** join/#asterisk oej (n=Olle@apollo.webway.se)
07:55.41puzzledmorning oej
07:55.50oejMorning
07:56.35djinI need the run * in realtime , what would be the most stable release to go for (CVS of 1.2b1)?
07:57.11djinoh, did I mention it's on a production machine? ;)
07:57.17sgorillahaha
07:57.41sgorillawhat do you mean by realtime?
07:57.58djinUuuh, asterisk realtime?
07:58.01sgorillalike QNX?
07:58.26djinNope, running SIP and IAX from a DB instead of config files.
07:58.41djinor at least thus users
07:58.43oejbeta 1 is rather old, so go for CVS head
07:58.48sgorillaok
07:58.54sgorillago with cvs head
07:58.56sgorillatest it in a lab
07:59.05djinI will :)
07:59.07sgorillathen rollback if neccessary if you catch some bugs
07:59.09sgorillaor fix  the bugs =)
07:59.32djindepends on the bugs
08:00.01djinoej, will there be an Astricon Europe next year?
08:00.26djinOr is that something you don't want to think about at this time .  .
08:00.27syle2extconfig.conf , do i have that just because i;m in cvs or installed asterisk-addons? or is this not in stable?
08:01.01syle2just wondering if cvs stable has extconfig.conf
08:01.03oejSome kind of Astricon Europe will be happening
08:01.14oejTrying to find a proper format for it and a proper location
08:01.36djinCool, looking forward to visting again.
08:01.49djinMight I suggest Amsterdam ;)
08:02.26djinGives a new meaning to the Golden Pub, I guess.
08:02.40oejWell, a bit too many distractions... Was thinking Karesuando. No one will find anything to do up there, apart from participating in the conference
08:02.58djintrue :)
08:06.20*** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk)
08:07.09*** join/#asterisk kippi (n=chrisfro@host86-133-85-206.range86-133.btcentralplus.com)
08:07.20shido6am I still connected here
08:08.33kippiis there away i can find out what part of my SIP registation is failing?
08:10.04Insanity5watch asteirsk log
08:10.06Insanity5easiest way
08:10.14Insanity5in console, set verbose to like 3
08:10.23Insanity5asterisk -cvvvp starts it this way.
08:13.21Koshatulhas anyone else here setup DID with engin ?
08:17.06*** join/#asterisk RoyK (n=roy@80.239.107.70)
08:17.29*** join/#asterisk Striker`Work (n=striker@lunar-linux/developer/pdpc.bronze.striker)
08:18.32kippiI am just getting  Oct 20 09:17:28 NOTICE[3312]: chan_sip.c:9001 handle_request_register: Registration from '<sip:6696@10.69.69.20>' failed for '10.69.69.25'
08:21.33RoyKkippi: that's usually just a bad password or so.......
08:24.34kippiif I look in /etc/asterisk/extensions.conf I should have a SIP setting for extension 6696?
08:28.18kippiok, thats where the problem is, my amp isn't writing to the same place as where asterisk is loading my config from
08:33.55*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
08:36.18*** join/#asterisk diamon (n=diamon@c-67-191-83-209.hsd1.fl.comcast.net)
08:39.58Koshatulwhat could be causing a "SIP/2.0 404 Not Found" when i have incoming calls ?
08:42.42Aze`Anyone has last florz's patch pls (mirror is down)?
08:43.39kippianyone know how to find out where amp is writing its config files?
08:44.14*** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc)
08:45.35*** join/#asterisk Inv_arp (i=junya@adsl-144-134-75.mia.bellsouth.net)
08:45.51Aze`kippi> /etc/asterisk/sip_additional.conf
08:46.09Aze`example
08:46.38loudKoshatul, user cannot be located by the server.
08:47.38Aze`kippi> /etc/asterisk/extensions_additional.conf
08:47.51Aze`kippi> ls -l /etc/asterisk7 ?
08:50.15kippiI am getting this error now
08:50.17kippiAsterisk Dynamic Loader Starting:
08:50.17kippi<PROTECTED>
08:50.17kippi<PROTECTED>
08:51.04kippifixed
08:51.06syle2is an area code always 3 digits?
08:51.10syle2even international/
08:52.00syle2crap i think some are like 7
08:52.26Dr_Raywoo... I expected this adit 600 to be bigger
08:52.32*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
08:53.28syle2what a pain in the ass
08:53.45syle2how am i suppose to select an area code when they aren;t same number of digits
08:53.51syle2from database
08:54.59Koshatulloud: thanks, i found the problem, my dialplan was "exten => s,1,Dial(${ENGINEXT},60,Tt" instead of "exten => 0731111111,1,Dial(${ENGINEXT},60,Tt)"
08:55.13Koshatuli didn't realise it wouldn't just use the s, exten
08:56.24*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
08:56.49syle2whats the internation format?
08:56.53syle2international
08:57.46iDunno00<countrycode><numberwithoutleading0>
08:57.50iDunno(usually)
08:58.10syle2isn;t it 001 then country code
08:58.19syle2err
08:58.20syle2011
08:58.25kippihow easy is CDR to get working?
08:58.55syle2i'm coding cdr software lol
08:59.01iDunno001 would be america, wouldn't it?
08:59.18syle2no it would be calls from canada/NA to overseas i think
08:59.33iDunno'oh'
08:59.36*** join/#asterisk cjk (n=cjk@80.92.64.103)
08:59.47syle2same if you were in europe
08:59.52iDunnofrom the UK I'm fairly sure it's 00<countrycode><numberwithoutleading0>
08:59.52syle2011 back to canada/US
09:00.17taec_From the UK + Ireland, it is.
09:00.17cjkhi,did anyone get the Playback application working in features.conf (DYNAMIC_FEATURES)
09:00.29taec_001 dials the states from the UK.
09:00.44kippiif I ring a number that isn't on the switch how can I get a message to the phone saying no extension found? or somthing like that?
09:00.51iDunnoI'm glad I'm not going mad :)
09:01.03taec_:)
09:02.13syle2well how about the phone number itself
09:02.25syle2is it always 7?
09:02.30syle2maybe i can parse backwards
09:02.36syle2for teh country code
09:03.52*** join/#asterisk shanky (n=shanky@238.Red-80-33-29.staticIP.rima-tde.net)
09:03.58shankygood afternoon
09:04.15diamonOr morning, as the TZ may be!
09:05.04shankyjust a newbie question, I'd like to use retrieve_extensions_from_mysql.pl (and also the sip one) but I don't know where have I to configure asterisk to invoke them
09:08.49Gibster11:00 -!- ShintaLT [sdfsdfsdf@sg-749.telkomadsl.co.za] has joined #wdsl
09:10.38kippiis there a command to list all registered sip connections
09:15.03loudkippi, as in users or carriers.
09:18.55kippiusers
09:19.10loudsip show peers
09:22.30*** join/#asterisk Newbie___ (i=me@60.48.53.104)
09:23.16Newbie___hi all, i am using Playtone() to play dial tone to user. but user can't enter any DTMF. any suggestion ?
09:24.57*** join/#asterisk rob112 (n=robert@host217-35-76-74.in-addr.btopenworld.com)
09:24.58shido6yeah
09:25.06shido6Background
09:25.49Newbie___shido6: i am playing a dial tone exten => s,3,Playtones(350+440)
09:26.35shido6and you expect something back?
09:26.38kippiI have my two sip phones connected, but if I try and ring either phone from the other phone it dosn't ring at all
09:27.05Newbie___expect user to enter DTMF after the tone is played
09:27.08shido6when you receive dtmf what do you want to do with those tones?
09:27.34Newbie___stop the tones and read the dtmf
09:27.39shido6and do what
09:28.01Newbie___and then dial the desired number
09:29.34shido6what number
09:29.46shido6is there a pattern
09:29.47shido6?
09:29.53shido6that all the numbers might match
09:29.54shido6?
09:31.00*** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com)
09:31.44cjkdamn i get no functions working of features.conf attended transfers etc... any idea why. when i reload asterisk it show on the cli that the feature XY has been mapped to the keys.....
09:32.10shido6Newbie___:
09:32.12shido6http://pastebin.ca/26037
09:33.52rob112Hi can anyone help with dsp call progress?
09:36.35kippiit says the two lines have connected but the other phone never rang. Anyone got any ideas?
09:37.05shido6kippi: just a sip.conf / extensions.conf snafu
09:37.15shido6users and peers
09:37.31shido6use a user to make a call and a peer to dial a phone
09:39.50*** join/#asterisk pashah (n=pashah@ns.itconnection.ru)
09:39.55pashahhello
09:40.11pashahwill digium's te110p work in pcix slot?
09:40.11RoyKolleh
09:40.21RoyKguess so
09:40.28RoyKi'm using te410p in one
09:40.44pashahRoyK: thanks
09:40.46pashahcheers!
09:43.55kippishindo6: so where do I need to look to make the changes?
09:45.34RoyKkippi: read the docs
09:45.39RoyK~docs
09:45.40jbothmm... docs is 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
09:46.07RoyKkippi: http://pastebin.ca/26038
09:46.28kippiok
09:58.18syle2could i get some feature requests plz
09:58.27syle2for asterisk billing system
10:02.44*** join/#asterisk Tjief (n=Tjief@r0x0r.dk)
10:05.00iDunnoright - what do people use as linux soft clients for asterisk?
10:05.04*** join/#asterisk Aze` (n=aze@85-18-136-114.ip.fastwebnet.it)
10:05.21X-Roblinphone apparentl
10:05.22X-Roby
10:05.44MatsKiDunno: Try X-Lite for linux
10:06.02Aze`Anyone has last florz's patch for zaphfc ?
10:07.00rob112Hi does anyone know the best way to detect tones over IAX2 ALAW ?
10:07.55iDunnolinphone appears to suck ;)
10:08.03iDunnoMatsK: hmm. is there source?
10:09.10*** join/#asterisk squirrelv5 (n=squirrel@203.131.188.170)
10:09.30squirrelv5hi there to all
10:11.22MatsKiDunno: I don't know
10:12.40iDunnodoesn't look like it.
10:12.40*** join/#asterisk zotz (n=zotz@24.231.36.100)
10:16.55*** join/#asterisk ful|work (n=fulgas@213.58.130.46)
10:16.59ful|workhey
10:25.17*** join/#asterisk w0w0 (n=w0w0@114.Red-83-41-6.dynamicIP.rima-tde.net)
10:28.59*** join/#asterisk MozartsGhost (n=mg@machinery-of-the-night.za.net)
10:29.09MozartsGhost~dialtone
10:29.16MozartsGhostbleh :/
10:30.12*** join/#asterisk shanky (n=shanky@238.Red-80-33-29.staticIP.rima-tde.net)
10:30.15shankyhi again
10:30.24shankyjust a newbie question, I'd like to use retrieve_extensions_from_mysql.pl (and also the sip one) but I don't know where have I to configure asterisk to invoke them
10:35.32MozartsGhostanybody know how to detect ringtone on a pstn line, off a Digium TDM400P card ?
10:35.51MozartsGhostwant to detect dialtone before it tries to dial.
10:37.08MozartsGhostcan't seem to find any available docs/info on the subject.
10:37.24MozartsGhostguess thats a no brainer then hey.
10:37.25MozartsGhostoh well.
10:38.33MozartsGhostek gan vandag my account kry hoopelik.
10:38.44MozartsGhostewps, wrong window.
10:38.53X-RobMozartsGhost - can't be done. Sorry.
10:39.34MozartsGhostk, ta.
10:41.41*** join/#asterisk tuxinator_linuxM (n=spabin@24-53-55-28.ontrca.adelphia.net)
10:47.33*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
10:48.29trymIt seems the expiry for outbound registration is too high.. because asterisk stops getting SIP requests when I call my sip number. If I call just after a registration it always works. Any ideas?
10:49.48X-Robyour nat is broken?
10:49.54RoyKtrym: well. decreasing the register timeout might be a good idea :)
10:50.12RoyKX-Rob: nat ususally needs far lower register timeout anyway
10:50.26X-RobRoyK - I hate nat. lots.
10:50.50RoyKX-Rob: well. we have a few thousand customers, most of them behind nat.....
10:50.52trymroyk: Sounds like a good idea? ;) Do you know what the paramter is called? And asterisk also does not monitor it.. any way to make it do that?
10:50.55RoyKyou learn to deal with it
10:51.17RoyKtrym: from sip.conf
10:51.17RoyK;maxexpiry=3600                 ; Max length of incoming registration we allow
10:51.17RoyK;defaultexpiry=120              ; Default length of incoming/outoing registration
10:51.28RoyKplay with those
10:51.34trymthanks
10:51.47RoyKtrym: hvor er du kobla opp?
10:51.51trymip24
10:51.54RoyKah
10:51.54trym:/
10:52.16RoyKtrym: kom til asterisk-no
10:52.21trymja husker deg.. du jobbet i briiz før
10:52.23RoyKeller #ip24
10:52.26trymså ble de "slått sammen" right ?
10:52.31RoyKvi spiste ip24.....
10:52.35trymhaha
10:52.38trymnice
10:55.33*** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-43-69.44-151.net24.it)
10:55.44kippiI have added the bits to sip.conf and extension.conf, but now i am geting 404 error. Would anyone be able to point me in the write direction
10:59.42RoyKkippi: pastebin the output and the config
10:59.48RoyK~pastebin
10:59.49jbotpastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca/
11:04.30*** join/#asterisk gordonjcp (n=gordonjc@cpc3-broo2-5-0-cust232.renf.cable.ntl.com)
11:04.42kippihttp://pastebin.ca/26043
11:04.59gordonjcphey folks
11:05.15e-Hernickhey folk-man
11:05.33gordonjcpdoes anyone in here have any experience of the Astralis X101P card?
11:06.32e-Hernickisn't that a cheap 20$ digium ripoff card ?
11:07.39gordonjcpseems to be
11:08.07gordonjcpe-Hernick: worth bothering with, or not?
11:08.53e-Hernickwell, it's never going to give very good results, but it's a fun toy for the price
11:09.15gordonjcpwhy would it be any different to an X100P?
11:10.02e-Hernickcheap amps. cheap caps. cheap everything. no support.
11:10.24e-HernickAnd the X100P isn't a very good card to start with.
11:10.32cjkhi, i tried a lot of features of features.conf, hangup transfer but i get non of them working. not in 1.2 beta and not in a older head version. here is my features.conf   http://pastebin.ca/26045
11:10.56gordonjcpe-Hernick: tbh, I doubt there's much of a difference between the components.
11:11.22e-Hernickwell then, it should work just fine
11:11.28e-Hernickwhat's wrong with yours ?
11:11.52gordonjcpe-Hernick: mmm.  For 3 quid I can hardly go wrong, can I?
11:12.24e-Hernickwell, it can sound so bad as to be useless
11:13.08kippihas anyone been able to look at http://pastebin.ca/26043 and can see why i am getting 404's
11:13.21*** join/#asterisk bintut (i=krcnmz@gr-155-5.eglobalreach.net)
11:13.47X-Robkippi - you have two '1's
11:13.59kippiah
11:14.08bintutwhere can i read the basics of asterisk? its concepts and definitions?  the bandwidth requirement?  difference between sip and aix?
11:14.16X-Rob~docs
11:14.18jboti guess docs is 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
11:15.10X-Robkippi - I also get the feeling that you're using asterisk@home
11:15.22kippinope
11:15.45kippiasterisk@home just works
11:15.47*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
11:16.20bintutjbot: ok. thanks..
11:16.37bintutwhat's the usual codecs being used in asterisk?
11:16.56X-Robbintut - ulaw/alaw, gsm, ilbc, g729 etc.
11:17.06X-Robbintut - you also want to look at http://voip-info.org
11:17.15*** join/#asterisk t0ke (n=toke@120.Red-83-57-33.dynamicIP.rima-tde.net)
11:17.16X-Robkippi - you're not using A@H?
11:17.23kippinopw
11:17.28kippiI have AMP installed
11:17.33t0kehello
11:17.35X-RobOK.
11:17.42X-RobIf you're using AMP, you don't want to mess with extensions.conf
11:17.49X-Robyou use exensions_custom.conf
11:17.50t0keanyone know if there is any limitation about E1 cards to use with Asterisk free?
11:18.17X-Robkippi - and amp questions should be on #amportal
11:18.23X-RobBug me over there.
11:19.10bintutX-Rob: i'm checking the site.. thanks.. :)
11:23.38t0keanyone know if there is any limitation about E1 cards to use with Asterisk free?
11:24.38power1Im using asterisk @ home 1.5 whenever i start up the system "kudzu" allways says its removing the digium card then andding a configuration for it, also when i shutdown the system. i allways get an error when it triues to stop the zaptel driver saying "device or resource busy" any ideas?
11:26.50bintuthow will i know how much bandwidth will i need for my voip setup?
11:26.58X-Rob~bandwidth
11:27.00jbotrumour has it, bandwidth is This is a measure, in some amount of bits per second, of theamount of data that can be sent over a particular cable, interface, orbus.
11:27.00bintutthat is, voip+pbx?
11:27.18X-Robbintut - all this, and more, is available on voip-info.org
11:27.19X-Robreally.
11:27.23X-Robthere's a pile of information there
11:27.24iDunnohmm. iaxcomm kinda works.
11:27.32X-Robtry googleing for 'site:voip-info.org bandwidth'
11:27.53bintutX-Rob: yeah. but i don't understand it really the bandwidth calculator?
11:28.11*** part/#asterisk darkskiez (n=darkskie@194.247.78.146)
11:28.23bintutX-Rob: in your opinion, for a 10 local lines, how much bandwidth will you allocate?
11:28.56X-Robbintut - 1mbit.
11:28.58bintutlet's say, how much bandwidth will you shape for each line?
11:29.09X-RobBecause I'd be using ulaw.
11:29.15X-Robalaw even
11:29.32bintutX-Rob: you mean, 100kbps per line?
11:29.37X-Robyep
11:29.45bintutoh! that's big enough
11:29.58bintutwhat are the other options besides "alaw"
11:30.00bintut?
11:30.20X-Robshow codecs
11:30.36X-Robbintut - it would be easier if you grabbed a spare PC, downloade asterisk@home 2.0beta[something] and have a play
11:31.56bintutX-Rob: yeah, i'll do it later.. i want to get some info first.. i hope you have a little patience on me.. :)
11:33.39*** join/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net)
11:37.05power1Im using asterisk @ home 1.5 whenever i start up the system "kudzu" allways says its removing the digium card then andding a configuration for it, also when i shutdown the system. i allways get an error when it triues to stop the zaptel driver saying "device or resource busy" any ideas?
11:37.45MatsK<power1> check /var/log/messages for info
11:37.51tzafrir_laptophmm, app_segfault from bristuff just proved itself very useful for the case of asterisk is run withwith -p
11:38.33t0keanyone have more one T1/E1 with Asterisk?
11:38.39tzafrir_laptopThough a segfault is known to be a, well, feature, of some other parts of asterisk in some extreme circumstances
11:38.47bintutthanks guys.. gtg now..
11:39.00gordonjcpls
11:39.58power1MatsK, looks normal except for this "Oct 20 19:31:29 asterisk1 init: Id "s0" respawning too fast: disabled for 5 minutes"
11:41.00power1I am having a weird problem with asterisk @ home calls come in perfectly on digium fxo and ring extensions but when the person hangs up, asterisk has the line open indefinately..
11:41.13power1you think this is a hardware problem.
11:41.17tzafrir_laptoppower1, why the &*^*^%&%^ would asterisk be started directly from /etc/inittab ?
11:41.40MatsK<power1> What card is it, is it a cloned X100P ?
11:41.49power1tzafrir_laptop, this is a clean install. and thats what in the logs...you tell me?
11:42.09power1MatsK, Digium TDM400P FXO/FXS Card
11:42.18tzafrir_laptop"asterisk1" is the hostname, then
11:42.24MatsKok
11:42.28tzafrir_laptopSo the message is unrelated to asterisk, I gather
11:42.33power1tzafrir_laptop, yip thats right
11:42.42X-Robtzafrir_laptop - that's saying there's a mgetty on ttys0 that's trying to talk to the modem and failing
11:42.43tzafrir_laptopgrep ^s0 /etc/inittab
11:42.57X-Robignore that
11:43.21power1tzafrir_laptop, s0:12345:respawn:/sbin/agetty -i -h -L 9600 ttyS0 vt100
11:44.05tzafrir_laptopsed -i -e 's/^s0:/#&/' /etc/inittab && telinit -q # assuming there aren't typos and mistakes here
11:44.40tzafrir_laptopThis should help you to ignore that...
11:44.41power1I dont know why its doing this.....it was working properly and started doing this not hanging up thing. Ive now done a fresh install and its still doing it. I have checked irq and the digium card is not sharing.
11:45.45MikeJ[Laptop]morning
11:46.56power1could this be a problem " Registered tone zone 0 (United States / North America)" I am in Africa?
11:51.46MatsK<power1> You are talking about two problems! Hangup problem and kudzu detection.
11:51.50t0keAnyone have Asterisk with more one E1/T1 interfaces or know if there is any limitation about number of E1/T1 supported?
11:52.15MatsK<power1> Have you updated kuzu ?
11:52.59Ahrimanest0ke: i'd say number of cards supported is mostly a performance issue and pci bus slot issue
11:53.08power1MatsK, yes every time kudzu comes up It tells me to remove a configuration then it finds the digium card again and i tell it  to configure it.
11:53.45t0keAhrimanes: someone comment me that I needed to buy Asterisk Business Edition if I wanted more 120 channels
11:53.59syle2where can i get a list of every area code in the world
11:54.20power1MatsK, when i restart the computer should the zaptel module be able to unload cleanly or is it normal to get an error saying "device or resource busy?
11:54.23MatsKsyle2: e164.org
11:54.34t0keAhrimanes: talked me about limitation between commercial and free license
11:54.51t0keAhrimanes: I dont know if it is true
11:54.57syle2looking at the page, where?
11:54.59Ahrimanest0ke: i doubt there's such a limit.. but then again, most servers probably are better of with around 120 channels perfromance wise
11:55.04Ahrimanest0ke: i dont think it's there
11:55.17t0keok..tnks
11:55.54MatsK<power1> No, but try update the system (yum update)
11:55.55t0keAhrimanes: maybe then it will be more depurate for supporting more 120 channels
11:56.07Ahrimanest0ke: depurate?
11:56.17MozartsGhostwho
11:56.20MozartsGhostewps.
11:56.27t0keAhrimanes: optimized
11:56.57e-HernickWell, if you're going to be running that many channels on *, it's a good thing to have technical support and vendor-approved and tested builds.
11:57.00Ahrimanest0ke: that might well be, asterisk business edition is based on cvs code newer than 1.0.9 and optimized by digium
11:57.45e-HernickAnd we're talking below 10$ per channel
11:57.53t0keAhrimanes: I understand
11:58.04t0keAhrimanes: tnks by your feedback
11:58.16Ahrimanese-Hernick: many things are good ideas, but bottomline ends up with what decision management makes
11:58.20Ahrimanest0ke: np :)
11:58.34e-HernickSure, but the decision is based upon the recommendations they receive.
11:58.46t0keAhrimanes: do you know what about perfomance between Digium and Sangomas cards?
11:58.55power1MatsK, thanks...im busy updating yum.........could this be a problem in logs when zaptel driver loads " Registered tone zone 0 (United States / North America)" I am in Africa?
11:59.14Ahrimanest0ke: no, havent tested, but i'd say newer cards from digium with hardware echo cancellation would perform the best
11:59.32*** join/#asterisk Bonzai009 (n=pirch@wbs-146-191-120.telkomadsl.co.za)
11:59.49t0keAhrimanes: yeah..and surely will have best support for Asterisk
12:00.08supa_thygarpower1 where in Africa ?
12:00.46kippihas anyone got madplay to work with asterisk for music on hold?
12:00.58Ahrimanest0ke: i should think so yes :)
12:01.07power1supa_thygar, Johannesburg South Africa
12:01.16supa_thygarPretoria SA here
12:01.18Ahrimanest0ke: allthough sangoma do seem to take pride in asterisk support
12:01.37power1supa_thygar, kewl <grin>
12:02.05power1supa_thygar, hows ur asterisk skills?
12:02.12t0keAhrimanes: Sangoma is going out one new card next monday with echo cancellation and it is 126 ms..really nice
12:02.27supa_thygarwell not good but we have a system up and running
12:02.37e-HernickThe digium card with EC is only 16 ms when running all channels, right ?
12:02.38t0keAhrimanes: I received an email from Sangoma yesteday explained it.
12:02.46Ahrimanest0ke: oh nice, how many ports?
12:02.52t0kee-Hernick: yes
12:02.56t0keAhrimanes: quad
12:03.04supa_thygarpower1 : i have a problem with asterisk dipping but thats about it
12:03.05Ahrimanest0ke: ok, any pricing info?
12:03.12t0keAhrimanes: and will go out next year 8 ports
12:03.26t0keAhrimanes: one sec
12:03.30Ahrimanest0ke: :)
12:04.02t0keAhrimanes: look this: "If you are seeking hardware based echo cancellation, the A104d- T1/E1/J1
12:04.03supa_thygarpower1: you can use ZA in the zone
12:04.34power1supa_thygar, did u have a problem with cellular phones calling in and asterisk not dropping the fx0 channel when the person hangs up?
12:04.36t0keAhrimanes: for resellers or isp...$1890
12:04.39*** join/#asterisk damned (n=vpol@prior.lanck.net)
12:04.53e-Hernickouch, a104d is 2700$ list
12:04.57supa_thygarpower1 nope
12:04.57power1MatsK, its found about 15 packages it want to upgrade...should i let it?
12:05.11t0kee-Hernick: yes..in list it is price
12:05.16Ahrimanest0ke: oh.. that's not too bad
12:05.17RoyKe-Hernick: what is that d for?
12:05.17power1MatsK, sorry for the juvenile questions..ive never used centos before ..i only know gentoo.
12:05.24e-Hernickthe d is for echo cancellation
12:05.31Ahrimanesno, for dsp
12:05.36power1supa_thygar, <grin> I allways get the weird and wonderfuls!
12:05.37supa_thygarpower1 : i know what is wrong
12:05.48power1supa_thygar, im all ears
12:05.55t0keAhrimanes: http://www.sangoma.com/products/p_aft-104-specs.htm
12:05.56Bonzai009hi all
12:06.04Ahrimanest0ke: yes, found it :)
12:06.07t0ke;)
12:06.12supa_thygarpower1 in one of the config files there is a setting that you can reverse polarity put it on yes
12:06.15Ahrimanest0ke: know the price for te411p digium card?
12:06.37power1supa_thygar, what does that do?
12:06.58t0keAhrimanes: I have price for it here in Spain..in EUR...go to convert it to dollars if you want
12:07.00supa_thygarit makes the fx0 drop the line if a cellphone hangs up
12:07.03power1supa_thygar, and where do i put ZA in the zone?
12:07.10Bonzai009is there a way to get the incomming calls to my agents faster. whats happening is that when   i have only one agent of 10 loged on it poles all the pthers before it gets to the  logged one one
12:07.28Ahrimanest0ke: just euro is fine :)
12:07.32supa_thygarpower1 : in zaptel.conf
12:07.38power1supa_thygar, thanks....so u must have had that problem ...if you know how to sort it out.
12:07.39t0keAhrimanes: 2100 Eur
12:07.56supa_thygarpower1 yea just could not remember
12:08.11Bonzai009power1  i had to put it in my indications.conf as well
12:08.23Ahrimanest0ke: wow.. ~$700 price diff
12:08.24supa_thygarpower1 : i have a problem with clarity and dipping at the moment
12:08.33Ahrimanest0ke: maybe one should try out a sangoma card.. :)
12:08.50Bonzai009has any one had issues with asterisk on xeon servers?..
12:08.59t0keAhrimanes: there is many difference about to have echo cancellation on hardware or no. I was thinking about to use PIV with one E1 port and then to do echo cancellation via CPU
12:09.18Bonzai009particularely were the server  just hangs or the pri just freezes up
12:09.18t0keAhrimanes: then to have 4 cpus with one E1 on each one
12:09.23power1supa_thygar, is dipping kinda like volume fluctuation?
12:09.33*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:09.47Bonzai009power1  kinda like telkom trying to provide a 1st world service hehehe
12:10.08supa_thygarpower1 yea
12:10.34supa_thygarpower1: it's like somtimes the calls are 100% and then all the sudden it turns into crap
12:10.39power1Bonzai009, LOL, good old hellkom......did you see telkom dropped the lawsuit against hellkom.co.za
12:10.55Ahrimanest0ke: hm try it out on a 2 cpu system with 2 cards.. the impact of having hardware dsp should be rather big.. digium should launch ds/3 card soon, which in part is possible due to dsp
12:11.00Bonzai009power1  yeah :> makes me happy now we need a sno to come in :|
12:11.10power1supa_thygar, weird....are u running asterisk @ home..or normal version on top of a distro?
12:11.27Bonzai009any one had issues on xeon servers with asterisk 1.0.9 ?..
12:11.31supa_thygarpower1 running it at my office
12:11.40Bonzai009my pri keeps freezing on xeon
12:11.47Bonzai009all ports lock up until i reboot
12:11.49power1Bonzai009, yeah...ASAP....did u read the icasa finding on telkom ADSL practices?
12:11.53Bonzai009hellkom tested the line
12:12.12supa_thygarpower1: using 2 TDM400p
12:12.29Bonzai009power1  yeah what a rip off R2k a month  for a 30gb adsl cap
12:12.49supa_thygarpower1: running normal version
12:12.58Bonzai009power u have a link to that icasa adsl practice.
12:13.54Bonzai009power1  u on adsl ?..
12:14.01power1Bonzai009, yeah....there are ways around that.... ive been paying @260 per month for a 30 gig account for the last couple of months..also for some reason telkom did not complete my adsl installation on their system...so im not getting billed for line rental either <smile>
12:14.22supa_thygarpower1 lol :)
12:14.32power1supa_thygar, what distro u running it on ( gentoo bsd etc)?
12:14.40supa_thygarpower1: @least some ne is winning
12:14.43Bonzai009power1  they dont that to me as well self install and they never sent me a bill hehehe
12:14.49supa_thygarnormal dist on gentoo
12:14.54Bonzai009power1  can i private u quickly?..
12:16.09power1Bonzai009, sure go 4 it
12:16.46*** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net)
12:17.37power1supa_thygar,  do i do that reverse polarity in the file zapata.conf?
12:17.59supa_thygarpower1: cant remember which file
12:18.01*** join/#asterisk razu (n=razu@tln-kontor.norby.ee)
12:18.18supa_thygarpower1: let me check and see if i can find it
12:18.53power1supa_thygar, thanks man!
12:21.56kippihas anyone got any ideas why when i do a echo test I am geting nothing back
12:22.22_m_you're not standing between mountains.
12:22.30_m_no mountain -- no echo
12:23.09kippion voicemail to I don't hear the voice asking for the password
12:23.42_m_Do you have any zaptel based timing source?
12:24.51kippii have a TE110P card installed but there is no line in it at the mo
12:25.32_m_That shouldn't matter.
12:25.32synthetiq<PROTECTED>
12:25.34Bonzai009any one had issues on xeon servers with isdn pri card where all ports just freeze up untill you reboot asterisk
12:25.40*** join/#asterisk Corndawg_ (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net)
12:31.02Bonzai009any one had issues on xeon servers with isdn pri card where all ports just freeze up untill you reboot asterisk
12:31.20Ahrimanesreboot asterisk or the server?
12:31.38*** join/#asterisk coppice (n=chatzill@219.199.17.210.dyn.pacific.net.hk)
12:31.43Bonzai009Ahrimanes  done that  give it 10 to 15 minutes and it freezes
12:31.47Bonzai009its a genuine intel board
12:32.24AhrimanesBonzai009: i meant.. you said until you reboot asterisk.. do you mean reboot the server or restart asterisk?
12:32.46Bonzai009Ahrimanes  reboot the whole server even riped power outa ups
12:33.14AhrimanesBonzai009: ah ok.. hm havent had a problem like that
12:33.28Bonzai009its plain random can be any time but normaly with in 15 minute max
12:33.32Bonzai009hmmm
12:33.48Bonzai009maybe 1.2 will fix the problem any idea when tis going to be stable?..
12:34.09Ahrimanesnot really, rsn i hope :)
12:34.58Bonzai009Ahrimanes  dont want to use  head on a production system thats like asking for sleepless nights
12:35.12*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:35.23AhrimanesBonzai009: well many in here say it's better than stable.. but i do tend to stick to stable as well :)
12:35.26e-HernickHey, I'm looking for reference PSQM/PESQ/PAMS audio test files
12:37.40*** join/#asterisk docE (n=docE@66.237.242.41.ptr.us.xo.net)
12:38.26supa_thygarAny one in here have a problem or had or know how to fix where asterisk is inconsistent on some calls every thing is fine then on others you can hear the person fine but they struggle to hear you and it sounds like you are speaking from far away etc
12:38.30*** join/#asterisk Inkubot (n=inkubot@200.75.4.7)
12:38.37Inkubotgood morning
12:39.01Inkuboti've got a cisco ata 186 and when i make a call it takes about 15 seconds, any ideas why is that ?
12:39.15Bonzai009supa_thygar  check the codecs on your phones and in your iax.conf and sip.conf
12:39.20supa_thygaro by the way i use 2 x TDM400P Digium Cards one with 4xfxo and other 3xfxo and 1x fxs
12:39.36*** join/#asterisk fenlander (n=fenlande@82.152.81.57)
12:40.39supa_thygarBonzai009 is the iax.conf not just if you connect to a nother asterisk server ?
12:42.02supa_thygarBonzai009 what codecs are you using ?
12:43.44Bonzai009supa_thygar  i have voip phones on iax connecting to me and they all over the country i find the iax works better over the internet imho
12:44.18supa_thygarBonzai009 i'm not using it via the internet
12:44.34Bonzai009supa_thygar  well just incase u were i sugested the iax lol
12:44.46power1supa_thygar, whats your phones using " ulaw" ?
12:45.01supa_thygarpower1: yep ulaw it is
12:45.09power1Bonzai009, hmmm i think my phones only supports sip...what a bummer.
12:45.39iDunnoIAX is better for over the internet as it's, like, 1 port and everything travels over it...
12:45.42iDunnoSIP is a nasty mess.
12:45.50kippifor moh do i need to add anything extra for sip?
12:46.03Inkubotno kippi
12:46.16iDunno(but if sip is required, you set up a local asterisk with the sip setup for the phones and an IAX trunk to the main server)
12:46.22power1iDunno, Bonzai009 so would u reccomend i credit my iphones and get a phone that can do IAX?
12:46.25*** join/#asterisk lehel (n=lehel@82.79.20.17)
12:46.28lehelhello there
12:46.45iDunnopower1: we're using SIP phones with a local asterisk server
12:47.07iDunnowhich seems to work for us, so far.
12:47.25supa_thygarAny one in here have a problem or had or know how to fix where asterisk is inconsistent on some calls every thing is fine then on others you can hear the person fine but they struggle to hear you and it sounds like you are speaking from far away etc
12:47.49kippiany ideas why I can't hear anything? is there a * number I can dial you see if it is working or not?
12:48.26Inkubotkippi musiconhold=default in zapata.conf
12:48.51Inkubotand some mp3's in /var/lib/asterisk/mohmp3
12:48.57mmlj4kippi: yes, if you have the demo installed, dial 500
12:49.21*** join/#asterisk w0w0 (n=w0w0@36.Red-83-50-231.dynamicIP.rima-tde.net)
12:49.36Bonzai009iDunno  lol i know sip gave me gray hairs so i upgraded my phones out there to iax
12:49.39Inkubotdamn cisco phone, take so long to register
12:50.17Bonzai009I've contacted a VOIP phone supplier in chin and they busy developing a voip pay phone that takes smart cards and coins.. lemme know if any oen is interested
12:51.17*** join/#asterisk supaigtr (n=yurplsl@152.53.16.10)
12:51.22supaigtrHello.
12:53.58kippimusiconhold=default is in zapata
12:56.38*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
12:59.42Kattymew.
13:00.47KattyRoyK: :<
13:01.56Inkuboti resolve the problem.. extension+# and dials
13:03.14diamonSo, anyone willing to suggest a decent VoIP service provider in the US with decent rates and the ability to dial into/get calls from POTS lines?  I'm trying to test mixed-mode, calling from line to line...
13:03.28*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
13:04.04gordonjcpdiamon: stanavoice?
13:04.07Ariel_Morning everyone
13:04.07Kattydiamon: lots.
13:04.15Kattyhihi, Ariel_ *hug*
13:04.23Ariel_hello Katty
13:04.54iDunnomorning Katty :)
13:07.16diamongordonjcp: Cool, I'll look them up...
13:07.27mutilatormm good breakfast today, reheated breadsticks
13:07.34diamonKatty: I'm just fishing for names people suggest.  Usually hurts less that way.
13:08.29diamonIf nothing else, sometimes I get names to avoid, which is almost as good.  :)
13:08.40Kattyk
13:08.46Kattyhihi, iDunno
13:10.16RoyKiPod, and iChat, and iDunno?
13:12.52*** join/#asterisk Pikoro (n=webmaste@db.sunny-net.ne.jp)
13:13.30Pikoroi got a wierd problem here.. ;D
13:13.53Pikoroi can make outbound sip calls, but cannot receive incoming sip calls
13:14.07Pikoroand internal calls don't work
13:14.21Pikoroi've been working on this for like 2 weeks now :D
13:15.26supa_thygarAny one in here have a problem or had or know how to fix where asterisk is inconsistent on some calls every thing is fine then on others you can hear the person fine but they struggle to hear you and it sounds like you are speaking from far away etc
13:15.37Ariel_Pikoro, are you using AMP or some other predone system?
13:15.44*** join/#asterisk CANO-1982 (n=cano1982@201.255.50.206)
13:15.45PikoroAriel_, no
13:15.49Pikorotried it though
13:16.23iDunnoRoyK: I am in no way affiliated with Apple ;)
13:16.24Pikoroi am positive that all I'm missing is some kind of internal route.. like it can't find the context or something
13:16.39Ariel_Pikoro, When you call extension to extension what does the CLI say in sip debug on
13:16.44Pikoroall i get is a busy signal
13:16.44iDunnoRoyK: I was iDunno loooong before they started prefixing things with i, dammit ;)
13:17.25Pikorowell, in my client i get a 486: Busy Here
13:17.51Pikoroextension is reporting busy and has no voicemail
13:18.23Pikorowhat sucks is that at any one time, I have had all 3 parts working
13:18.34Pikorohad internal and from pstn working but no outgoing
13:18.39*** join/#asterisk hadi57 (n=al_moghr@83.136.8.206)
13:18.48Pikoroso i tried to fix outgoing calls and now no incoming calls work
13:18.49RoyKiDunno: sue them :)
13:19.15Pikoroi'm about ready to just pay someone to log in and fix it for me :\
13:19.17Ariel_Pikoro, post your dial section in pastebin.ca for us to take a look
13:19.39Pikorook, the conf files i have now were created with amp but i no longer have amp installed
13:19.46nextimeanyone with the latest spandsp 0.0.2pre21 on a debian sid with libtiff 3.7.4 and a TE400P quad pri digium card?
13:19.49Pikorodial section from extensions.conf?
13:19.58coppiceiDunno: they used eXXXX, then iXXXX. They'll soon move on to oXXXX and leave you alone
13:20.00lehelPikoro, from CLI
13:20.27iDunnoheh - true, true ;)
13:21.47Pikorodamnit... i can't copy and paste from this stupid dos window.. scrolling it gets everything all screwed up
13:21.52Pikoroi have debug logs running.. want those?
13:22.49*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
13:23.51*** join/#asterisk viLeR (i=1000@66.128.47.232)
13:24.44lehelpaste it
13:25.02Inkubotasterisk doesn't recognize the incoming ID
13:25.09tzangercoppice: heh
13:25.11Inkubotany ideas ?
13:25.20tzangerthe iDunno... haha
13:25.29tzangerfollowed by the iDontGivaShit
13:25.58*** join/#asterisk ast_freak (n=jesse@hades-out.universalsystems.net)
13:25.58Pikorohttp://pastebin.ca/26050
13:26.43*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
13:26.52RoyKcoppice: iirc the eMac came out _after_ the iMac, so perhaps aMac comes next.....
13:26.55RoyKAA Mac
13:28.26PikorobMac
13:29.01Ariel_Pikoro, your not sending the correct info to dialparties
13:29.38Pikorook, where should I go to fix that?
13:29.42Pikoroextensions.conf no?
13:29.47*** join/#asterisk nexis (n=nexis@12-207-56-108.client.mchsi.com)
13:30.19Pikoroanyplace I can grab some known good conf files and just change the registration string, etc...?
13:30.40Pikorook, that was just an internal call
13:31.03Pikoroshall i paste an internal call now/
13:31.15Pikoroerr.. incoming
13:31.35nexisPikoro, dont paste in the channel, it can get spammy, use a pastebin
13:31.45*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc058.dialup.mindspring.com)
13:31.46Pikoronexis, that's what i did :D
13:33.29Pikorohttp://pastebin.ca/26051  <- a call coming in from the PSTN
13:33.35Pikoroor not coming in rather.. heh
13:33.45Ariel_Pikoro, since your using amp presetup files you have to continue with amp. But you can also change your own files read the sample that asterisk has /urs/src/asterisk/configs/extensions.conf.sample
13:34.19tzafrir_laptopAMP's trunks setup is not very helpful. Still requires much docs reading
13:34.21*** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m)
13:34.30Pikorook
13:34.36tzafrir_laptopIn fact, I found it even more confusing than hand-configuring asterisk
13:34.41Pikoroi wanted to do this without amp :D
13:34.48tzafrir_laptopAt least for a simple setup
13:34.49Pikoroi went back to the default conf files
13:34.54Ariel_Pikoro, then remove them all start with the samples
13:34.55Pikorothis should be a simple setup
13:35.23PikoroI did that one time and couldn't get anything to work...
13:35.30Pikoro:(
13:35.35Pikoroi'll give it another shot
13:35.43Pikorotime to go home now, it:s almost 11pm
13:35.54Ariel_Pikoro, print them out then follow the logic that they have.
13:36.01Pikorok
13:36.11Pikorobelieve it or not, it IS all starting to make sence :D
13:36.20Pikoroat least I'll know what I'm doing when i'm done
13:36.21Pikoroheh
13:36.21Ariel_you need to do some reading as to rules and use macro which will make things easy to setup.
13:36.57nexisis it possable to set flags with a call file like with the dial string, such as H or L
13:37.21Ariel_Pikoro, now you have also doc's and the wiki. When I started the standard reply here was read the code to see what it does
13:38.55Ariel_Pikoro, also for some strange reason extensions like 666 don't work in asterisk. Just for your info it does strange things.
13:39.36*** join/#asterisk santiago (n=santiago@208.195.215.158)
13:40.06*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
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13:43.07*** join/#asterisk CANO-1982 (i=alejandr@201.255.50.206)
13:44.19Ariel_Pikoro, there is a great book on line now: http://www.asteriskdocs.org/modules/news/
13:44.39*** join/#asterisk iCEBrkr (n=icebrkr@242858hfc41.tampabay.res.rr.com)
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13:46.51Pikoroi got it
13:46.53Pikorobeen reading it
13:47.05Pikoroproblem is there is nothing about sip in there really
13:47.08Pikorojust iax
13:47.17CANO-1982I have a problem with my TDM400p board
13:47.18Pikoroit very breifly covers sip
13:47.18*** join/#asterisk Los415 (n=los415@ssf-office.corp.race.com)
13:47.20Inkubotmi asterisk in some calls doesn't hungup and the lines is busy for a time.
13:47.48Inkubotany idea ?
13:47.52CANO-1982I have to unplug and plug again each time y have an incoming call on my FXO module
13:47.59CANO-1982any idea?
13:48.38syle2feature requests on billing system?
13:48.52CANO-1982Ive tried aswerpolarity.., callooprogres, busydetect, hanguponpolarity.., loopstart ang kewlstart
13:49.13snittInkubot: search for disconnect provision
13:49.30Los415syle being able to re rate calls
13:49.37*** join/#asterisk astoria (n=haydenth@66.235.201.217)
13:50.06syle2what do you mean?
13:50.17Inkubotthnks snitt
13:50.18CANO-1982Inkubot, have you tried unpluging the phone cable?
13:50.20Inkubotgoogle now
13:50.39Inkubotyeps.. but thats it is an ugly solution
13:50.40Pikorowell, talk to ya'll tomorrow
13:50.53Los415like if a call gets rated at a certain rate
13:50.54*** part/#asterisk Striker`Work (n=striker@lunar-linux/developer/pdpc.bronze.striker)
13:50.56Los415and customer calls
13:51.21CANO-1982well, its a start
13:51.38syle2well what i was thinking was this....
13:51.38Los415need to re rate calls to
13:51.47CANO-1982I cant try teh groundstart signaling
13:51.51CANO-1982maybe that works
13:52.19syle2editable screen that brings up all the worlds area codes, and you can put any rate you want in there, and in customer section a field for a discount, that would just subtract from rate in worldarea codes
13:52.32syle2if you wanted
13:52.58Los415hrmmm
13:53.05Los415billing is a nightmare
13:53.15Los415i even hate looking at the mile long code
13:53.25Los415it's so complex
13:53.28syle2you;d hate to see c code i have so far doing this
13:53.36Los415we wrote some but it's some complex stuff
13:53.53CANO-1982Inkubot, have you tried the things that I told before?
13:54.06syle2is that what you mean by re-rate though?
13:54.11Inkubotwhen the lines is busy
13:54.14Inkuboti restart asterisk
13:54.19Inkubotand just work fine..
13:54.29Inkubotwhen i call from a external number
13:54.38Inkubotthe ivr answer.. but if i end the call..
13:54.40Kattywhere do you find porabolas?
13:54.43Los415well like we have had customers call in say this call was rated higher
13:54.46Kattymagnetic fields?
13:54.47Inkubotthe ivr keeps going and then end in the vm
13:54.54Los415so we had to re rate the calls to a differnt rate
13:55.02Los415which magically changes there invoice up ect.
13:55.02Inkuboti really don't know why is that
13:55.11syle2owww
13:55.14*** join/#asterisk _santiago_ (n=santiago@208.195.215.158)
13:55.25syle2for that month only?
13:55.39Los415it could be anything
13:55.41Los415hehe
13:56.04*** join/#asterisk gambolputty (n=gambolpu@72.240.242.4)
13:56.05tzangerKatty: some women?
13:56.31tzangerdepends though, it ranges from x=y^2 to x=0.01y^2
13:56.47CANO-1982well, in my case Ive got that fixed playing with the aswerpolarity.. and  hanguponpolarity parameters
13:57.09*** join/#asterisk jimmy_deanPB (n=jhodapp@72.244.232.226)
13:57.24tzangerActually when I was a kid I bent sheet metal into a parabolic shape and put a microphone at its focus... I could hear the dog panting 200 feet away
13:58.48CANO-1982have you tried that?Inkubot?
13:58.57iDunnotzanger: are you *sure* it was the dog?
13:59.14tzangeriDunno: ha
14:00.56*** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
14:01.13*** join/#asterisk felipex (n=dsfdsf@85-18-136-75.fastres.net)
14:01.32QbYanyone here using Broadvoice.com ?? and have an 800# with them??  is yours working?
14:06.03*** join/#asterisk mkrufky (n=mk@68.160.103.77)
14:06.33*** join/#asterisk Starcode (n=jan@dslb-082-083-086-069.pools.arcor-ip.net)
14:06.48*** join/#asterisk spiekey (n=spiekey@p549D19B2.dip0.t-ipconnect.de)
14:06.55spiekeyhowdy #asterisk!
14:08.23supaigtrsup
14:08.50ful|workanyone got this type of error -> http://pastebin.ca/26053 ?
14:09.37supaigtrThats a warning.
14:09.45StarcodeI have a problem with a iax <-> iax connection between 2 servers.
14:10.10ful|worksure
14:10.18ful|worknevetheless smth must be wrong
14:10.52StarcodeIn iax.conf is [server_a] type=friend; username=abc; secret=pass; host=10.2.2.2; context=iax
14:11.40InkubotCANO-1982: i don't try that
14:11.41StarcodeIf go a Dial(IAX2/server_a/${EXTEN:2},30,r} I can see:
14:12.09Starcode"Executing Dial("Zap/1-1", "IAX2/server_a/1130|30|r") in new stack
14:12.17StarcodeCalled server_a/1130
14:12.21Kattytzanger: mew?
14:12.26Kattytzanger: what are we talking about again?
14:12.37StarcodeTx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass: NEW
14:12.39*** join/#asterisk santiago (n=santiago@208.195.215.158)
14:12.57StarcodeTimestamp: 00012ms  SCall: 16384  DCall: 00000 [10.2.2.2:0]
14:13.11Kattytzanger: parabolic orbits?
14:13.25StarcodeBut no ip packet is really sent to 10.2.2.2... i testet it with tcpdump
14:13.35sivanaanyone know what a supervised transfer event looks like for the manager api?
14:13.42StarcodeIn the other direction, everything works.
14:14.12StarcodeI don't know, why asterisk calls 10.2.2.2:0 with port 0... on the other side it calls ip:5036
14:14.17*** join/#asterisk toddf (n=toddf@adsl-65-70-118-15.dsl.okcyok.swbell.net)
14:14.29tzangerKatty: you said parabolas
14:14.33synthetiqfuck
14:14.36tzangerI said some women have impressive parabolas
14:14.41astoriaparabola=dangerous amazon snake
14:14.42synthetiqim getting now foudn errors on ym sip phones
14:14.47synthetiqand calls cant go out
14:14.50synthetiqbut they can come in
14:14.56synthetiqwhat could be the problem?
14:15.01synthetiqi change dnothing in the dial plan
14:15.10Kattytzanger: i found orbits (some of them) to be a porabola
14:15.22tzangerI know nothing about astronomy
14:15.28Kattytzanger: except it's a porabola, so it's not really an 'orbit'
14:15.33Kattyjust more like a swing-about
14:15.48tzangerwell yeah if it escapes the gravity well it's not an orbit anymore
14:15.58tzangerthere are circular and elliptical orbits
14:16.05tzangerbut orbits by definition are closed shapes
14:16.26Kattythere's also hyperbola based orbits
14:16.35Kattyi'm reading all about such things right now (=
14:16.42tzangerI thought hyperbolas weren't closed
14:16.49Kattythey're not
14:16.55Kattythey're still considered an orbit though
14:17.00InkubotCANO-1982: msg
14:17.03tzangerreally... how the hell does it go around ?
14:17.29Starcodethis is the asterisk-channel, isn't it?
14:17.30Starcode;-)
14:17.58*** join/#asterisk lehel (n=lehel@82.79.20.17)
14:18.08tzangerStarcode: nope
14:18.08Kattytzanger: it doesn't
14:18.16tzangerKatty: hmm interesting
14:18.24spiekeyi bought a "octoBRI PCI ISDN" from junghanns.net.  Do i only have to plug my telco cable into my FXO port of the card?
14:18.41Kattytzanger: if you take the earth's orbit 18.5 m/s
14:18.53Kattywell the velocity around the sun
14:19.04Kattyand kicked it up to 26 miles a second
14:19.11tzangeryou're gettin me all hot, talkin astrophysics
14:19.22Kattyit'd go shooting off into space in a parobolic orbit
14:19.26sivanaanyone know how to do an attended transfer via Manager API? huh huh
14:19.35*** join/#asterisk jpm_SD (n=jpm@207-40-115-38.sugardog.com)
14:19.49iDunnoonly till it found something else with enough gravity, though ;)
14:20.10Kattytzanger: it's not a trajectory, because those end, somewhere.
14:20.28Kattya porabola, in theory...never ends.
14:20.46gordonjcpno, that's an ellipse, isn't it?
14:21.04Kattygordonjcp: the earth is on an elliptical orbit right now, yes.
14:21.19gordonjcpI thought a parabola wasn't closed
14:21.38StarcodeDoes someone have a working IAX <-> IAX setup?
14:21.40tzangera parabola isn't
14:21.44iDunnoKatty: true, but of course, with a planet flinging about, you'd expect it to find something else with some gravity to pull it in to another form of orbit ;)
14:21.46tzangerhence why I don't think it's an orbit
14:21.50supaigtrStarcode: Define working?
14:21.54gordonjcpah, but it seems you *can* have a parabolic orbit
14:21.56tzangeryou have  circular, elliptical or irregular orbits
14:22.17KattyiDunno: you're getting off the topic.
14:22.33iDunnoKatty: I thought that was the idea ;)
14:22.48Kattytzanger: some comments have a parabolic orbit
14:22.58*** join/#asterisk gambolputty (n=gambolpu@72.240.242.4)
14:22.59Kattytzanger: they swing about the sun and keep on going
14:23.01tzangerKatty: interesting, my comments usually stay put :-)
14:23.05StarcodeWhen I dial from asterisk-server A to server B I get an outgoing call in the debug console to ip of server B, port 0 and ip packet is sent
14:23.09Kattyteehee!
14:23.31StarcodeTx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass: NEW
14:23.31gordonjcphttp://en.wikipedia.org/wiki/Orbit
14:23.36StarcodeTimestamp: 00012ms  SCall: 16384  DCall: 00000 [10.2.2.2:0]
14:23.48gordonjcpahaa! if you launch something at escape velocity, then it is a parabolic orbit
14:23.55Kattytzanger: http://www-history.mcs.st-andrews.ac.uk/history/Java/Parabola.html
14:23.57gordonjcpthat is, it *doesn't* loop round on itself
14:24.16*** join/#asterisk lcars (n=lcars@gentoo/developer/lcars)
14:24.19*** join/#asterisk supa_thygar (i=thygar@tpr-165-255-171.telkomadsl.co.za)
14:24.48*** join/#asterisk rob314 (n=rob314@207.58.194.2)
14:24.49lcarshi folks..what's the proper way for debugging a 'Requested context didn't get merged' error message ?
14:25.00supaigtrStarcode: I just have iax.conf setup on both sides and have one side behind nat registering to the other side.  Besides dropped calls it seems to work. Do you have trunking=yes?
14:25.40StarcodeNo, because my server B is SIP/IAX only and trunk need Zap-driver for some reasons
14:25.52Kattyif you have two stars that go past each other, and their gravitational fields interact, then they usually end up having a hyperbolic trajectory
14:26.04supaigtrYea.  PRI -> * -> IAX - IAX * -> SIP?
14:26.13StarcodeWARNING[229391]: chan_iax2.c:5460 build_user: Unable to support trunking on user 'server_b' without zaptel timing
14:26.32StarcodeYes
14:26.52supaigtrI have the same thing working more or less.
14:27.02StarcodeBut when calling on pri *A does not send ip packets to *B
14:27.14spiekeyPSTN stands for what?
14:27.18supaigtrpastbin your iax and extension where the dialing happens.
14:27.24Kattypurple soy tuna nuggets
14:27.41spiekeyKatty: i am kind of serious ;)
14:27.41supaigtrPissing Standing Top Neutral
14:27.53Kattyspiekey: so? doesn't mean i have to be
14:28.02Kattyspiekey: you could always google you know
14:28.19supaigtrPublic Switched Telephone Network
14:28.23tzanger~pstn
14:28.25jbotrumour has it, pstn is Pubic Switched Telephone Network, or "please stop the nonsense"
14:28.25spiekeythanks
14:28.52synthetiqguys i have a big problem....i have a production amchien with 500 users who can recieve inbound but not make out abnd via lcoal t1 or ds3 on voip..any iddeas?  i see no errors pop up on the console
14:29.12supaigtrds3 voip? GW or *?
14:29.20synthetiq*
14:29.33supaigtrds3 with * how?
14:29.43spiekeyPSTN is the line/cable/port to my telco, right?
14:29.49synthetiqi pump my voip out bound thru a ds3
14:30.04KattyiDunno: also, planets rarely 'fling' about.
14:30.06*** part/#asterisk lcars (n=lcars@gentoo/developer/lcars)
14:30.09*** join/#asterisk Bonzai009 (n=pirch@wbs-146-191-120.telkomadsl.co.za)
14:30.12supaigtrYea what * hardware does that at this moment?
14:30.26supaigtrThey were flung once.
14:30.37iDunnoKatty: well, true, but asteroids kinda do, and they're only a little bit smaller ;)
14:30.39Kattynot..exactly.
14:30.42Starcode[general]
14:30.42Starcodebandwidth=low
14:30.42Starcodedisallow=lpc10
14:30.42Starcodejitterbuffer=no
14:30.42Starcodetos=lowdelay
14:30.42tzangersynthetiq: if you're playing around with DS3 or even T1 you've got enough brains to string enough words together to make a coherent sentence, and enough left over to string the right words together to give us enough info to try and help.
14:30.43Starcode[server_b]
14:30.45Starcodetype=friend
14:30.46Kattysupaigtr: they didn't fling, the grew outward.
14:30.46[TK]D-FenderSangoma has a DS3 card.
14:30.47Starcodeusername=server_b
14:30.49Starcodesecret=pass
14:30.58Starcodehost=10.116.168.250
14:30.58tzangerStarcode: don't do that
14:30.59Starcodecontext=main-routing-table
14:31.01Starcodepermit=0.0.0.0/0.0.0.0
14:31.01KattyiDunno: ye, comments flinging about teh sun
14:31.02tzangerStarcode: use pastebin
14:31.05supaigtrOuch!!!
14:31.06tzanger~pastebin
14:31.07jbotmethinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
14:31.08synthetiqim under pressure thats hwy im having toruble typing
14:31.32KattyiDunno: but if you can graph a parabola using the usual x^2-x+1 thingy
14:31.37tzangersynthetiq: a lack of planning on your part does not constitute and emergency on ours.  We'd love to help but please slow down and give us the information we need to try to help.
14:31.44KattyiDunno: then you can map out if they're going to smack into anything on the way
14:32.14KattyiDunno: and, in fact, it'd be a rather easy graph
14:32.16iDunnoKatty: you need to calculate the gravitational fields around things that they're passing to get it accurate, though ;)
14:32.39iDunnoKatty: though, probably don't need to be *that* accurate ;)
14:32.45Kattythat's what GRACE is for ;)
14:32.50*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
14:32.51supaigtrsynthetiq: Explain your setup?
14:33.14Kattyi'm waiting for GRACE to give us a good idea of earth's gravitational field
14:33.24Kattygravitation being my favorite topic.
14:33.38synthetiqit jsut randomly stopped letting us from makign out bound calls
14:33.39Kattycurious as to how much space-time bend we might be dragging.
14:33.42synthetiqinterneal calls are fine
14:34.02supaigtrsynthetiq: What hardware do you have doing TDM?
14:34.21synthetiqte411p
14:34.40supaigtrOk so what does zttool tell you?
14:34.48synthetiqi rebooted the machines
14:34.54*** join/#asterisk fiXXXerMet (n=Kyle@ip67-154-236-201.z236-154-67.customer.algx.net)
14:35.02synthetiqits not the zap cards, because i cant make calls via IP outbound
14:35.16fiXXXerMetHi everyone.  I'm in the process of setting up an A@H box, and I need a carrier.  What's a nice cheap good one to use for testing?
14:35.37supaigtrOk so pastbin your dialplan.
14:35.42tzangersynthetiq: never reboot a machine unless you know what's going on
14:35.53iDunnoKatty: I think anti-gravity is more fun...
14:36.04iDunnoKatty: involves less crashing to earth with a bump, for a start :)
14:36.13supaigtrEspcially a 411p it may never boot back. :)
14:36.31KattyiDunno: not to me :)
14:36.36tzangersynthetiq: so you have a DS3 coming into an M13 to break it out into individual T1s, which you have going in to TE411 cards.
14:36.46vader-wrkanyone know where to get some super deals on cisco ip phones? other than ebay?
14:36.49tzangersynthetiq: and incoming calls work just fine, which means your switch setup and everything is set up
14:37.35tzangersynthetiq: but outgoing calls fail.  What's the PRI failure cause?  Is your DS3 provisioned for two-way calls?  Are you setting the pridialplan correctly?  you should be on the phone with your switch tech, not on IRC with us
14:38.09tzangernot to be rude, but it kind of sounds like you got a brand new setup and no requisite experience running anything like it
14:38.56*** join/#asterisk tuxinator_linuxM (n=spabin@24-53-55-28.ontrca.adelphia.net)
14:39.12*** join/#asterisk RussC (n=face@216.157.205.211)
14:39.45StarcodeIn the dailplan the only interesting thing for iax <-> iax is in [call-groups]: exten => _391130,4,Dial(IAX2/server_b/${EXTEN:2},30,r)
14:39.52RussCDoes any one have experince with polycom SoundPoint IP 301's?
14:40.03supaigtrRussC: sure
14:40.08astoriaRussC: I have experience, with 300s,500s, and 600s
14:40.21supaigtrsynthetiq: Is your dialplan tring to send out over TDM?
14:41.03RussCWell I am unable to get the 301's to access my server correctly I have them configured with the server IP but all I get when I dial is a busy tone
14:41.04synthetiqicomign calls are on separate ptri
14:41.17shido6ZzzzZZZ
14:41.19vader-wrkdo any of you use softphones at your places?
14:41.24synthetiqoutgoign calls go either thru t1 or IP
14:41.24astoriaRussC: Does teh Asterisk Debug CLI show anything?
14:41.34astoriaRussC: Is it registering?
14:41.35synthetiqvoice IP or data DS3 (logn dist)
14:41.37RussCno it does not
14:41.58astoriaI suggest your run ethereal or something and see what is happening to the packets.
14:42.05RussCAstoria: that is what has lead me to belive that the phone and server are not talking
14:42.11ian_kfiXXXerMet - use freeworlddialup if you just want to test
14:42.24astoriaRussC: well, obviously.. what does :sip show peers show?
14:42.26RussCAstoria: I will try that
14:42.40RussCAstoria: shows them as unknown
14:43.07astoriaRussC: what is in your sip.conf?
14:43.22supaigtrsynthetiq: Your not answering the question. Can you verify that * is trying to send the calls?  Do you see ISDN errors?
14:43.24astoriaPaste your sip.conf and your sip show peers to pastebin and ill look at it.
14:44.05RussCok thanks
14:44.30RussCAstoria: what is a pastebin?
14:44.44clyrradhttp://pastebin.ca
14:44.48vader-wrkdo any of you use softphones at your places?
14:44.49astoria~pastebin
14:44.50jboti guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
14:45.05RussCok thanks all
14:45.13*** join/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au)
14:45.19fiXXXerMetCan anyone recommend a VoIP provider for testing?  Something cheap?
14:45.28ian_kfiXXXerMet freeworlddialup - free
14:45.28astoriafiXXXerMet: asterlink or nufone..
14:45.39supaigtrvoipjet
14:45.43fiXXXerMetI have Verizon Voicewing and a PAP2 device, but I doubt they support this.
14:45.43ian_kfiXXXerMet free for toll-free numbers anyway
14:45.45fiXXXerMetThank you both
14:47.33RussCAstoria: http://pastebin.ca/26059
14:48.45iCEBrkrSo, does fax detection work with VoIP calls? :P
14:48.50iCEBrkrNot that I care to send a fax
14:48.58astoriaRussC: yeah, it looks like their not communicating correctly.. Your config looks OK.. I'd fire up ethereal and see where your packets are going.
14:49.05iCEBrkrI'm just wondering if Asterisk will land in the 'fax' extension after dial()
14:49.28RussCAstoria: thank you is there anything I should look for inpaticular?
14:49.42astoriaRussC: No, but I'd look for packets coming from the IP of your phone..
14:49.56astoriaRussC: just see where they are going..
14:50.11RussCAstoria: ok thanks
14:50.36*** join/#asterisk _Sam-- (n=sam@mail.kneedraggers.com)
14:50.41*** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net)
14:50.41RussCAstoria: would you be kind enough to take a look at my extensions.conf?
14:50.58_Sam--can someone recommend a good IAX provider to terminate my asterisk to?
14:51.10supaigtrvoipjet,nufone
14:51.16astoriaRussC: sure, but that's probably not your problem here.
14:51.21_Sam--im using teliax, just wondering about other options...thanks
14:51.48RussCAstoria: I would just like to see it is correctly setup :-)
14:51.58astoriaRussC: sure, just pastebin it.
14:52.18RussCok
14:52.32_Sam--vader-wrk:  check out www.teliax.com they are the best place ive found so far, ive been using them for about a month no problems
14:52.59vader-wrksam whats your voip setup like internally?
14:53.03vader-wrkie. phones, etc.
14:53.06RussCAstoria: http://pastebin.ca/26061 thanks again
14:53.13vader-wrkim trying to learn as much as i can for my own setup here
14:53.20_Sam--small office setup...12 SIP Clients to asterisk, data t1 to teliax
14:53.23vader-wrki wanna build an asterisk box with some softphones
14:53.47_Sam--it should be a pretty straight forward project these days
14:54.01astoriaRussC: Your contexts are wrong. In your sip.conf, you had two contexts, 03 and 04.. So in your extensions.conf you need to have a [03] and a [04] with the extensions for those below it..
14:54.09vader-wrkmy system is going to have to intergrate with a good chunk of old analog phones
14:54.22vader-wrkanyone have experience with that?
14:54.22_Sam--you will probably want those iaxy thingies
14:54.40astoriaRussC: In your extension lines, rather than having 03 or 04, use the incoming number that your carrier is sending you, like your DID.
14:54.42vader-wrkbecuase we have alot of places that don't have network cable run to them
14:54.45vader-wrkonly telephone lines
14:54.54RussCAstoria: I had it that way and it did not work I forgot to change it back thanks. Other than that?
14:55.09astoriaRussC: Yeah, it's OK, other than that...
14:55.14vader-wrksam what phones are you using?
14:55.21astoriaRussC: But again, it has nothing to do with your phones registering..
14:55.22_Sam--you will be in kind of a quandry, vader...since VOIP implies running over a network
14:55.34vader-wrkya but you can use channel banks for the analog phones
14:55.55astoriaI won an iaxy at cluecon and the guy never sent it to me! what a jerk!
14:55.56vader-wrkatleast thats what im gathering
14:56.16vader-wrkthe ADIT 600 looks like what i would use
14:56.19vader-wrkwith an E1 connection
14:56.23astoriai heart that adit 600
14:56.29iCEBrkrIs there some reason why I can't call my home voicepulse number with my work voicepulse number?? Two seperate accounts
14:56.29vader-wrkyou have an ADIT 600?
14:56.34iCEBrkrWeird
14:56.49astoriaYes. I have an adit 600.
14:56.58astoriaWell, I don't own it.. I lease it from XO.
14:57.08vader-wrkdo you use it to power analog phones?
14:57.38astoriaNo, I use it to split my data and voice traffic on a single T1
14:57.42*** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl)
14:57.45vader-wrkoh
14:58.05_Sam--it does say the adit supports 6 pots
14:58.07vader-wrki was looking to pick up like two 24 port models with FXS cards
14:58.51astoriaadit is pricey, but plays very well with asterisk!
14:58.57_Sam--how many regular analog phones are you looking to connect?
14:59.07vader-wrknot sure the exact number
14:59.10*** join/#asterisk n0where (n=kc@s199048.ppp.asahi-net.or.jp)
14:59.18astoriaYou might want to look at Rhino callbanks too..
14:59.19vader-wrkbut it's going to be in the ballpark of probably 40-50 phones
14:59.30vader-wrki found some adit 600s on ebay cheap
14:59.54vader-wrknot in my exact configuration but i can probably make due
14:59.57*** join/#asterisk pattieja (n=pattieja@adsl-69-153-174-41.dsl.stlsmo.swbell.net)
15:00.15astoriaLooking on ebay now - not that cheap...
15:01.05_Sam--how do those rhino channel banks interface with asterisk?
15:01.20vader-wrkyesterday they had some ADIT 600s with cards and stuff for under 500
15:01.22astoriaVia T1
15:01.28astoriaDamn...
15:01.32vader-wrkadit 600 can connect through ethernet
15:01.36vader-wrkinstead of t1
15:01.40vader-wrkso you don't need a card
15:01.41_Sam--so you would plug the rhino channel bank into like, a digium type card?
15:01.45vader-wrkjust a network connection
15:01.49vader-wrkya
15:02.00astoriavader-wrk: SIP?
15:02.08vader-wrki dunno
15:02.28astoriaHmmm. I'm curious to know which protocol it uses..
15:03.02vader-wrkya not sure
15:03.13vader-wrkim just sooo leary about doing this system
15:03.24astoriaWhy?
15:03.25vader-wrkbecause it's soo many phones,extensions, etc. and quipment
15:03.29vader-wrkequipment
15:03.46vader-wrkand trying to intermix analog and digital
15:03.58*** join/#asterisk gambolputty (n=gambolpu@72.240.242.4)
15:04.03astoriaWell, if you're at ~48 extensions, you can pick up two call banks, interface with asterisk via t1, and keep it analog...
15:04.12*** join/#asterisk glomph (n=black@c-24-18-145-249.hsd1.wa.comcast.net)
15:04.22jake1932looks like MGCP
15:04.26_Sam--why would that be any better than just keeping his existing setup?
15:04.45_Sam--i guess you would get some of the benefits of asterisk...voip termination, some features etc
15:04.51astoriaI dont know what his existing setup is, but you'd get * ..
15:04.57vader-wrkwell this whole voip idea came along because in about 8-9 months we are moving some of our offices temporarly from one part of our building to another
15:04.57jake1932(with a CMG card)
15:05.00astoriaSure, you can do alot more..
15:05.03vader-wrkfor construction reasons
15:05.16astoriaYeah, then when you move, drop only cat5, that's what we did.
15:05.22[TK]D-FenderChannel banks are an iffy thing I find.  Seems better to use ATA's (more hardphone functionality)
15:05.48jake1932<PROTECTED>
15:06.00[TK]D-FenderPlus you have to factor in the T1 card.
15:06.18[TK]D-FenderSipura's are great, and AudioCodes is #1.
15:06.28astoriaAll those ATAs would make my head spin..
15:06.30jake1932my Sipura sucks
15:06.38astoriaI heart Sipura.
15:06.39vader-wrkya well where we wanted to move the people we have cat5 but no telephone
15:06.55vader-wrkso we figured that we would integrate this we the current pbx system or replace the pbx system with this
15:07.02vader-wrkor a combination of both
15:07.03[TK]D-FenderDepends if you use a high density gateway like the AudioCodes rackmount stuff.  1 box.
15:07.18vader-wrkuse this and then phase it from temp solution to full solution
15:07.22glomphKeeping a separate wiring plant just for phones is archaic
15:07.31astoriaYeah, good point [TK]
15:07.33[TK]D-FenderOr if you want even better functionality try Citel's digital phone_. SIP gateway.
15:07.54jake1932from what I've been hearning - channel bank + T1 is the way to go
15:08.19vader-wrkya alot of places in the building have computers now but not all
15:08.25vader-wrkso that becomes a problem
15:08.29vader-wrkif we roll out IP phones
15:08.36vader-wrkbecause some places won't have cat5
15:08.36astoriaAre there any rotary ATAs out there?
15:08.40*** part/#asterisk glomph (n=black@c-24-18-145-249.hsd1.wa.comcast.net)
15:08.55_Sam--through up some short term wireless
15:09.03[TK]D-Fenderjake1932 : its does work, but what is your total scenario looking like?  I find *'s support of straight analog phones quite lacking compared to ATA's and I can't say I blame them for their focus
15:09.05astoriajake1932: actually, I like TKs idea better... I don't have any experience with SIP gateways though.
15:09.21RussCAstoria: could you look at http://pastebin.ca/26062 please
15:09.33vader-wrkwe have a wireless system
15:09.41[TK]D-FenderNorstar phones on a Citellink gateway would eb a cool alternative so as to avoid transformers for your analog phones, etc.
15:09.43vader-wrkbut those wireless phones would be booookuuu bucks
15:09.57_Sam--no, wireless softclients for the desktops
15:10.03RussCAstoria: I am now getting this error, I setup the phones to register with the server. Oops
15:10.11vader-wrkwe were thinking that too
15:10.12jake1932[TK]D-Fender: mine is real simple - SPA3000 FXO = big echo - no matter what I do
15:10.16vader-wrksoftphones
15:10.21vader-wrkbut i haven't found a good client
15:10.26[TK]D-Fenderjake1932 : Really?  I'm about to get one for home.....
15:10.26vader-wrkand i haven't found good reviews
15:10.38_Sam--i like the nero sipps client, it works ok for us, ram hog but good features
15:10.38jake1932[TK]D-Fender: maybe it'll work for you
15:11.02[TK]D-Fenderjake1932 : I can only hope.  what phones are you using apart from the FXS on it?
15:11.35jake1932tried 3 - cisco 7960, analog + DTA310, SPA841
15:11.43[TK]D-Fenderjake1932 : and is its all phones?
15:11.44_Sam--vader-wrk:  sipps:   http://ww2.nero.com/sippstar/eng/SIPPS_What_Is_SIPPS.html
15:11.46jake1932yep
15:11.58jake1932now i'm totally VOIP and no echo
15:12.11jake1932(but network issues occasionally cause blips)
15:12.16*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
15:12.18[TK]D-Fenderjake1932 : wow.... Sipura & Cisco pretty much tell you that something serious needs to be fixed.  Done a F/W upgrade on it?
15:12.24jake1932yep
15:12.29jake1932was at the newest
15:12.36jake1932the phones are fine
15:13.07astoriaRussC: you need to fix your exten=>03 in your [03] context... 03 doesn't exist..
15:13.13funxionanyone know the best way to configure an e1 from an digium card to a cisco
15:13.19astoriaRussC: wait..
15:13.38[TK]D-Fenderjake1932 : :/  Ugh.  I'd still like to ditch analog altogether, but we're still a little while away from Dry-line DSL
15:14.08jake1932yep - I can't do dry DSL - either - I just fwd my honme number for now
15:14.18astoriaRussC: Your dial cmd is jacked up...
15:14.19jake1932still cheaper
15:14.41RussCAstoria: ok
15:14.43jake1932(especially for intl)
15:14.53*** join/#asterisk JASON-0 (n=jason@jason.unitz.ca)
15:15.32*** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv)
15:15.36*** join/#asterisk NewSole2 (n=dave@d38-53-48.commercial1.cgocable.net)
15:15.41JASON-0Hello, I had installed Asterisk@Home which has the AMP portal. There is a Maintenance section on that. But when I installed Asterisk and AMP in Redhat, there is no Maintenance?  Do you where where to get that?
15:15.45*** join/#asterisk zobia (n=laura_sh@218.6.242.212)
15:15.52zobiaHello everyone
15:15.58zobiai have strange issue
15:16.34zobiawhen i enter the number from my softphone, the asterisk always got it repeated twice. if i input 1 , it will receive 11. does any one know why this happened?
15:16.53*** join/#asterisk R3DB0x (i=nobody@66.142.28.36)
15:17.53jpm_SDJASON-0, The Maintance tab is Asterisk @ Home specific..   it's not part of AMP
15:18.20jpm_SDand as far as I know there is no easy way to add it...
15:18.41RussCAstoria: what might I do to fix it?
15:19.23astoriaRussC: Hmmmm... Can you just copy that single command from your extensions.conf in here?
15:19.26astoriahold on
15:19.46fiXXXerMetHas anyone got the Linksys PAP2 working with A@H.  If you could give me a link to a walkthrough or help or whatever, that would be great.
15:19.52RussCAstoria: exten=>03,1,dial(SIP/03,20)
15:20.29astoriaRussC: what are you trying to do with that?? When someone calls in, ring 03?
15:20.55astoriaAlso, it should be dial(SIP/thecontextyouhaveinsip.conf)
15:21.13JASON-0jpm_SD: Thanks
15:21.14zobiaHello any one knows about the press repeat problem?
15:21.24RussCAstoria: ok so what is the ,20
15:21.31astoriaso, pete and his brother repeat were on a boat
15:21.40jake1932oh no
15:21.47astoriaWell, the time to ring the extension.
15:22.04RussCAstoria: alright
15:22.50astoriabless you child
15:22.50jake1932bless you
15:22.50Kattyastoria: i'm not a child.
15:22.50*** join/#asterisk bintut (n=bintut@202.128.40.243)
15:22.50Kattyastoria: but thanks anyway.
15:22.56jake1932gazoontight
15:23.09RussCAstoria: so it needs to match the line context= in the sip.conf?
15:23.09jpm_SDOnline alergies... scary thought.
15:23.12jake1932<PROTECTED>
15:23.32Kattyjpm_SD: dimensional based allergies.
15:23.39jpm_SDTimestamps make my skin itchy
15:23.59astoriaRussC: No. I needs to match the context (the stuff in the brakcets)
15:24.11jake1932it's especially scary considering virus control on the net has not been great
15:24.32RussCohh I see
15:24.42*** join/#asterisk skrusty (i=muad@xdev.net)
15:24.46NewSole2I just replaced all my Digium junk with Sangoma hardware.. selling the digium cheap... got 10 TE410P Cards.... Any takers Best offer
15:25.08skrustyanyone know why cdr_odbc would say it's updated the table, but doesn't?! using odbc->tds->mssql
15:25.10astoriaNewSole2: how much want for one?
15:25.43NewSole2I said best offer
15:25.51astoria50 bucks
15:25.55jpm_SDGot one the Sangoma Anolog beta cards...Mmm Mmm hardware echo can. yummy
15:25.59*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
15:26.02jpm_SDAnalog too.. :)
15:26.07jake1932<PROTECTED>
15:26.14jpm_SDjake1932, yup
15:26.17jake1932<PROTECTED>
15:26.26jpm_SDjake1932, Negitive
15:26.42Kattywhat goes well with mushroom soup?
15:26.51jake1932salt?
15:26.58astoriaA roast beef sandwich.
15:27.04jpm_SDKatty, Toast?
15:27.24Kattyastoria: anything else?
15:27.36fileNewSole2: how about a signed autograph of myself?
15:27.36jake1932maybe i should unload my digium stuff too before this sag stuff becomes mainstream
15:27.40zobiaHello who know why the dtmf tones are double always?
15:27.42sivanaKatty: rice
15:27.46astoriaKatty: no
15:27.46NewSole2current bid is a Night with the Wife....
15:27.46RussCAstoria: ok the phones can call its self but not each other lol
15:27.47Kattysivana: oooh, good idea.
15:28.02Kattysivana: what sort of protein goes well with rice and mushroom soup?
15:28.21astoriaWell, in each of the phone's contexts (in extensions) you need to put an entry if you dial the other phone..
15:28.26sivanaKatty: hrm.. you need enough to counter the rice and soup :)
15:28.33tzangerrice and mushrooms?  I'd throw some chicken in there but I know that's not what you're thinking
15:28.40sivanahehe
15:28.43jpm_SDjake1932, Not having the 1000 Interrupt polls per second is a nice bonus too.
15:28.48Kattysivana: well, i'm plotting an enchilada recipe
15:28.55Kattysivana: where the 'enchilada sauce' is mushroom soup
15:29.00sivanaKatty: maybe some man-made chicken meat?
15:29.01jake1932jpm_SD: when are they expected?
15:29.16Kattysivana: possibly.
15:29.18*** join/#asterisk h3x0r (n=h3xor@64.192.116.16)
15:29.25sivanahrm.. not sure..
15:29.30Kattysivana: that and a can of corn should compliment nicely.....i'd think
15:29.32jpm_SDjake1932, Doug said 3 weeks when I spoke to him at Astricon.
15:29.37Kattyor maybe rice, corn, and refried beans
15:29.41sivanaya
15:29.48sivanaand pepper
15:29.50sivana:)
15:30.17jake1932best offer
15:30.26astoria2o bucks
15:30.41RussCAstoria: sorry I dont understand. I have 03 and 04 the only two in my setup, in sip.conf and extensions.conf
15:31.14astoriaYou need to have an exten=>OTHEREXTENSIONNUMBER with a dial cmd in eachother's extensions..
15:31.19jake1932hey you offered the other guy 50
15:31.37astoriaWell, supply clearly exceeds demand now.
15:31.39RussCAstoria: ok, I see
15:31.56jake1932true
15:31.59astoriaRussC: i mean "eachother's contexts"
15:32.18bjohnson21 bucks
15:32.41jake1932increments of $2.50 at least
15:32.44astoriabjohnson, stop bidding. we don't need to bid.  there are digium cards a plenty!
15:33.14bjohnsonI don't want the card, just my cut for driving up the price
15:33.34astoriabjohnson: do you sell insurance ? :)
15:34.45jake1932actually - I'm thinking of listing for $125 (w/ 1 FXO)
15:35.03_Sam--if you make changes to queues.conf how do you reload?  is it reloaded with extensions reload?
15:35.16jake1932maybe someone else will have better luck
15:35.42RussCAstoria: outstanding thank you so much for your help
15:36.20astoriaRussC: no problem
15:36.38*** join/#asterisk jsaunders (i=js@S01060060971c5817.va.shawcable.net)
15:37.20jsaundersIs it possible to enable a message waiting indicator with asterisk, ie...  when I lift the handset it will make a noise indicating I have messages?
15:37.48filejsaunders: SIP phone, or what
15:37.58*** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
15:37.59jsaundersyes, sip ata]
15:38.05fileit's up to the ATA to do it
15:38.15fileAsterisk merely sends a "there's messages waiting" packet to it
15:38.18jsaundersreheally
15:38.19filethe ATA decides what to do with it.
15:38.21*** part/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net)
15:38.22jsaundersThanks for the tip.  :)
15:38.35fileSIP devices do a lot of stuff themselves.
15:38.40jsaundersDoes asterisk send this packet automagically?  Or is it a setting?
15:38.44[TK]D-FenderHey I'm trying to find a decent IAX softphone for Windows and Firefly's latest build won't even let me ASNWER an incoming call.  Forget about such fun and obvious options as Hold, XFer, etc....  Any suggestions?  I need IAX for NAT reasons (We are on a fixed IP, but behind a SonicWALL router)
15:38.50astoriaMy server plays "Beleieve it or not" on the speakers when a voicemail is left.
15:38.57filejsaunders: you have to tell it the mailbox in sip.conf, it's not psychic
15:39.05mishehubah.
15:39.05jsaundersfile:  Heheh
15:39.10jsaundersfile:  Tnx mang.
15:39.11*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
15:39.13astoriamishehu, how you doin?
15:39.27filejsaunders: you now owe me one DJ Sammy MP3
15:39.35fiXXXerMetHas anyone got the Linksys PAP2 working with A@H?  I simply don't know how to configure the device.  If you could give me a link to a walkthrough or help or whatever, that would be great.
15:39.36fileor your soul
15:39.36iDunnofile: dammit! why isn't is psychic!
15:39.38mishehuastoria: getting killed here.  3 classes at college, and trying to run a business!  heh
15:40.09astoriamishehu: yikes! I hear ya.
15:40.29fileiDunno: I wrote a psychic module for SER when I was sitting at a conference
15:40.39astoriaHey, anyone here one of the organizers of cluecon, back in July.
15:40.44fileastoria: hi.
15:40.58astoriafile: cylogistics never sent me my iaxy I won!
15:41.14fileastoria: weren't you supposed to talk to them? if not I have that guy's business card here
15:41.29astoriafile: yeah, I have it and i've emailed him a bunch of times with no reply
15:41.39filetried phone?
15:41.41supaigtrAnyone have trouble ticket software recommendations?
15:41.45astoriafile: so i am going to start emailing him naked pictures
15:41.52astoriafile: no, all i have is his email addy
15:41.54zobiaplease help with my dtmf tones double problem
15:42.05BladeRunner05How can I use mohmp3 ? I have to write code in extensions.conf ? or what ?
15:42.11fileastoria: see privmsg
15:42.28astoriathanks file
15:42.58mishehuBladeRunner05: you have to pay $$ to the RIAA, you pirate.
15:43.08mishehusince we all know that only pirates use mp3 files.
15:43.08Ahrimaneslol
15:43.31mishehuand l33t pirates use aac or ogg.
15:44.47*** join/#asterisk _tekati_ (n=captain@cpe-66-75-215-63.bak.res.rr.com)
15:44.51BladeRunner05its not a commercial music
15:45.06BladeRunner05How can suggest me ?
15:48.35bintutwhat's the recommended hardware specs for an asterisk box with 50 local soft phones with the following basic features: pbx, ivr, voice mail, call logging/recording, conference bridging, call snooping, call forward, predictive dialer and music on transfer?
15:49.57mmlj4bintut: there's a page on the wiki about asterisk dimensioning, that might help
15:50.17bintutmmlj4: do you know where it is in particular?
15:50.50_m_The mic on my snom190's headset doesn't work when I dial out. It works when I receive a call. The handset works in either direction.
15:50.53mmlj4yes: on the wiki
15:50.54_m_Any ideas?
15:51.03astoriabintut: a 486 at least.
15:51.34*** part/#asterisk n0where (n=kc@s199048.ppp.asahi-net.or.jp)
15:52.06bintutastoria: yeah, i know at least 486.. if you will be deploying the above specs, what will you choose?
15:52.47astoriabintut: i'm not sure. EDO ram is so expensive these days.
15:52.53*** join/#asterisk gaspiz (i=gaspi@86.34.6.164)
15:53.13gaspizhi there, can you hepl a rookie?
15:53.29*** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
15:53.32astoriabintut: i'm honestly not sure, look on the wiki
15:53.40*** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
15:53.42*** join/#asterisk mutilator (n=animenod@65.111.201.79)
15:53.44gaspiznot so rookie
15:54.06*** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
15:54.14cjkhi, i tried a lot of features of features.conf, hangup transfer but i get non of them working. not in 1.2 beta and not in a older head version. here is my features.conf   http://pastebin.ca/26045
15:54.41bintutastoria: where's the wiki? sorry, i'm a newbie.. :((
15:54.49astoriabintut: www.voip-info.org
15:54.59gaspizI have a problem when I set up sip users
15:55.09gaspizwhen I set there names numbers it's ok
15:55.20*** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3685577.sympatico.ca)
15:55.32gaspizwhen I put names i can't place or receive calls
15:55.45gaspizso: name=1001 it's ok
15:55.52gaspizname= gasparz
15:55.57gaspizproblem...
15:56.22*** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus)
15:56.49gaspizideas?
15:57.07bintutastoria: ok. thanks.. :)
15:57.08astoriawhat file is this in??
15:57.31gaspizI use realtime for sip users
15:57.42funxiondid I read wrong but I see hat in the new 1.2 cvs head there is support for t38 pass through?
15:57.45jsaundersfile:  I gots rizapp if yer lookin' for it, no DJ Sammy though.  Jew want somethin'?
15:57.57jsaundersfile: I'll send ya a few of my faves, np.
15:58.16filejsaunders: nah I'm in a DJ Sammy mood
15:58.34jsaundersfile: Hmm, never heard of him.  I'll have to snag some.  Any faves?
15:58.49fileWhy... Rise Again... The Boys of Summer
15:59.01*** join/#asterisk _Thor (i=CS@user-vc8fl7n.biz.mindspring.com)
15:59.10iDunno(but that covers a lot of the time, really)
15:59.31jsaundersfile: denamrk?  What genre?  Oh, techo/rave kinda?
15:59.37filetrance-sorta
15:59.45jsaundersfile: Sweet!  You like Royksopp?
15:59.47_Thorhello, I recently installed oh323, what does this msg means?: Oct 20 11:59:18 WARNING[580]: chan_oh323.c:4014 oh323_gk_check: Gatekeeper discovery failed.
15:59.47_Thor<PROTECTED>
16:00.24gaspizanyone using letters for sip usernames?
16:00.26*** join/#asterisk dstruct (i=dstruct@unaffiliated/dstruct)
16:00.27funxionhas anyone tried T38 passthrough in new CVS head
16:00.28dstructyo
16:00.36bintutguys, what type of phone do you suggest for a 50 ends/local phones: an analog, ip phone or softphone?
16:00.40dstructanyone here have any suggestions on running a pure "meetme" server, IAX2 based..  It seems like I need to include all sorts of other config files to please asterisk...
16:00.44filejsaunders: can't say I do
16:01.01jsaundersfile: Hmm, because you've never listened to them?  Or because you simply aren't partial to their music?
16:01.09astoriabintut: if you are deploying a 50 node voip network, you might want to get more assistence than people in an irc roomm...
16:01.18*** join/#asterisk t0ke (n=toke@120.Red-83-57-33.dynamicIP.rima-tde.net)
16:01.26t0kehello
16:01.42dstructastoria: heheh
16:02.06filejsaunders: never listened
16:02.08astoriaYou'll probably be dropping at least 10 grand on equipment...
16:02.13joelsolankiHello all, i had compiled asterisk-addons after that it is logging cdrs in mysql database. but it has stopped logging the cdrs in /var/log/asterisk/cdr-csv/Master.csv   i want to stop mysql cdr logging and enable the raw /var/log/asterisk/cdr-csv/Master.csv   any way out ?
16:02.19skrustyanyone any good with cdr_odbc? :)
16:02.45jsaundersfile: Reheally...  dude!  Do yourself a favor and grab 'em.  You will enjoy, promise.
16:02.56jsaundersfile: btw, dj sammy's pretty tight.  Thanks for the heads up.
16:03.47fileaye
16:04.53funxionwhat genre is htat
16:05.28CANO-1982I have a problem with my TDM400p board
16:05.32CANO-1982I have to unplug and plug again each time y have an incoming call on my FXO module
16:05.39CANO-1982any idea?
16:05.42CANO-1982Ive tried aswerpolarity.., callooprogres, busydetect, hanguponpolarity.., loopstart ang kewlstart
16:06.08supa_thygarCANO-1982 private me i'll tell you
16:07.37jsaundersfunxion: rave/trance
16:07.59funxionlike house?
16:08.35funxionwhat genre within rave/trance
16:08.44gaspiz<PROTECTED>
16:08.54jsaundersfunxion: Couldn't tell yeah...  that's something I've never learned, the distinction between the various 'dialects' of rave.  You'd have to ask my bro on that one, whom ain't here.
16:09.03funxiontru
16:09.23funxionI like the slower IDM
16:09.58jsaundersfunxion: This is definately tight, you'd dig it.  For a sample without downloading checkout www.djsammy.de
16:14.05*** join/#asterisk gaspiz (i=gaspi@86.34.6.164)
16:16.06*** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
16:16.06*** mode/#asterisk [+o twisted[asteria]] by ChanServ
16:16.58mutilatoranyone have java 1.4.2 or older installed that can compile an applet for me?
16:20.10bintutanyone here can i ask about hardware configurations not being suggesting to hire consultants because i'm here to learn not to look for consultants?
16:21.23skrustybintut: what's up?
16:21.28*** part/#asterisk jsaunders (i=js@S01060060971c5817.va.shawcable.net)
16:21.49cybertank```````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````
16:21.50cybertank```````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````
16:21.55cybertank```````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````
16:21.59SarahEmmuhh...
16:22.00cybertank````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````
16:22.01SarahEmmcybertank?
16:22.02skrustyheh :/
16:22.18cybertankkeyboard problem :)
16:22.19SarahEmmbintut: sure
16:22.19skrustylets hope he just fell asleep while working :)
16:22.22skrustyhehe :)
16:22.23SarahEmmcybertank: lol
16:22.25RoyK~lart cybertank
16:22.49*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
16:24.27*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
16:25.15*** join/#asterisk jac]Z[oby (n=me@62.218.44.178)
16:25.18jac]Z[obyhi
16:25.42jac]Z[obyanyone knows where to set the MSN to which the bristuffed zaptel thingy listens so?
16:25.45jac]Z[obyto
16:26.07*** join/#asterisk tainted_ (n=somewher@mail.k2usa.com)
16:27.20bintutskrusty: we're planning to try asterisk.. basically from my country and we'll subscribe to voip gateway in the us. basically, only 1 leased line connection to the internet.. now, we have 50 analog phones inside.. if we'll do switch to asterisk, what will be my hardware concerns? first, for the server? second, additional hardware i need? third, what is more scalable to you in the long run?
16:29.02SarahEmmbintut: err... that's a little more than i expected heh. hardware-wise you need one T1/E1 port for the outside PRI, two T1/E1 ports for the inside, and two E1 channel banks for the phones to plug into. the channel banks will plug into the two T1/E1 ports... (one quad-port card will do it)
16:29.07skrustyserver hardware requirements can be quite different depending on the call types etc that are passing through it. I would have a read of voip-info relating to server requirements, quite a few people have posted about that
16:29.22SarahEmmbintut: as for capacity, i have no idea.. there's a wiki article on it tho
16:29.27skrustySarahEmm: was a longer question than i expected :)
16:29.48SarahEmmbintut: http://www.voip-info.org/wiki/view/Asterisk+dimensioning
16:29.52SarahEmmskrusty: me too :)
16:30.08bintutSarahEmm: honestly, i don't know those terms you've mentioned like t1/e1, pri, channels, etc.. sorry..
16:30.20skrustybintut: then you really need to learn them
16:30.22astoriaand you're deploying a 50 node network?
16:30.39SarahEmmbintut: then you need to learn about telco.
16:30.43skrustyyou need to have a good look through the wiki, start at the beginning! :)
16:30.51SarahEmmbintut: before deploying a big system, you need to learn a Lot about this :)
16:31.07astoriaIt's really easy to mess up if you don't have everything planned out right the first time.
16:31.19skrustyand it can cost a lot more if you get it wrong :D
16:31.20bintutskrusty: i'm reading the success stories but some terms i really don't yet understand. i'm planning to have 50 phones at the same time
16:31.35skrustyright, then that's your starting point
16:31.35SarahEmmbintut: well, you should learn those terms then :)
16:31.37bintutSarahEmm: i'm already on the asterisk dimension site
16:31.58skrustydecide on codec's etc and read up on what hardware you'll need to interconnect with telco's in your area (if required)
16:32.12bintutSarahEmm: i know computer networking terms (some) but this asterisk/voip thingie is different.. :(
16:32.29skrustybintut: can i suggestion whatis.com!
16:32.37jac]Z[obyanyone knows about bristuff and msn settings
16:32.39jac]Z[oby??
16:32.41SarahEmmbintut: it uses telco terms, because it's a telco system. you need to learn about telco if you're deploying a telco system.
16:32.41skrustyit should bring you upto speed quickly if you know a term you want to look for
16:32.53skrustye.g. if you don't know what E1 means, look it up on there
16:32.59skrustyor another technical resource site
16:33.14*** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net)
16:33.38*** join/#asterisk twista (i=twista@p50845908.dip.t-dialin.net)
16:33.57skrusty``follow the wiki`` :)
16:34.11twistahey there ;)
16:34.15bintutok..
16:34.16skrustyhi
16:34.34skrustybintut: it'll make it far simpler for us to help you, if you know that extra little bit!
16:35.09skrustyand on that note... hometime!
16:36.03*** part/#asterisk gaspiz (i=gaspi@86.34.6.164)
16:36.04twistaprobably "simple" question: is it possible to force asterisk to "proxy" the rtp packets (like when it processes them to change the codec etc.) - because in my case there is just no direct connection possible between the two sip phones (but no nat)
16:36.05*** join/#asterisk mm_pt (n=mm_pt@a83-132-248-28.cpe.netcabo.pt)
16:36.38mm_pthi, any one could help me with xlite register in asterisk?
16:36.48iDunno*without* using Dial(SIP/phoneblah&SIP/phoneblip&SIP/phoneblop)
16:37.01skrustyan agi? :)
16:37.16iDunnoskrusty: that was almost the conclusion that I'd come to :/
16:37.24skrusty:]
16:37.32skrustyl8r people!
16:37.34iDunnoAGI can return to the dialplan, right? ;)
16:37.51FuriousGeorgeiDunno: there are call groups for calling out w/ in the same technology
16:37.55*** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com)
16:38.03twistadoesnt anybody have any hints regarding my problem? :S
16:38.26iDunnoFuriousGeorge: it's an incoming call, and I want it to dial the available phones, really.
16:38.38SarahEmmtwista: what's the problem?
16:38.38mm_ptany one could help me on registering  x lite in asterisk
16:38.45twista<twista> probably "simple" question: is it possible to force asterisk to "proxy" the rtp packets (like when it processes them to change the codec etc.) - because in my case there is just no direct connection possible between the two sip phones (but no nat)
16:38.53iDunnoit's not a big problem, because at the moment I can just specify them all in a long line seperated by & ;)
16:39.00bintutSarahEmm, skrusty and astoria: for the asterisk server specs, do i need a dual opteron with 4gb ram and sata hdd for the following features: pbx, ivr, voice mail, call logging/recording, conference bridging, call snooping, call forward, predictive dialer and music on transfer?
16:39.25astoriabintut: You don't NEED one for any of those features.
16:39.26FuriousGeorgeiDunno: im pretty sure it can be done with one of the three types of groups (dial(sip/1) (where 1 is the group number).  i think what you really need is a queue
16:39.40twistabecause in case of matching codecs the phones are trying to establish a direct rtp exchange (like it should be)..but that doesnt work for me
16:39.54FuriousGeorgeive never implemented a queue, but it cant be hard
16:40.02iDunnoFuriousGeorge: tried the Dial(SIP/1) earlier, that doesn't work ;)
16:40.04bintutSarahEmm, skrusty and astoria: if you will to deploy those features, will you use a dual opteron with 4gb ram and sata hdd for the asterisk server having 50 phones?
16:40.07astoriabintut: You may want to do more research than asking us, and actually look at the codecs and transcoding issues you may have.. It really depends..
16:40.15*** join/#asterisk SwK[Work] (n=SwK@border0hsv.asterisksgi.com)
16:40.15FuriousGeorgeiDunno: you have to set the groups up first
16:40.28mm_ptSarahEmm do you recommend me some one that could help me on with xlite register? i have asus router and asterisk installed with Fedora core 4
16:40.35iDunnoFuriousGeorge: pickup and call groups where specified in the sip.conf
16:40.41bintutastoria: yeah, i will.. i just want to get some suggestion and later on, i should know..
16:40.46SarahEmmmm_pt: i now have 5 people asking me questions
16:40.50astoriamm_pt: look on the wiki
16:40.57SarahEmmmm_pt: i'll try to help in a few minutes, but just ask your question here and someone might answer
16:40.59iDunnoanyways - home time :)
16:41.11FuriousGeorgeyou dont want a pickup group, you want a call group.  put your sip friends in a group, restart asterisk, and call that group when an incoming call comes in
16:41.14SarahEmmtwisted[asteria]: why can't they talk directly?
16:41.22mm_ptThanks for answering me! Sarah Emm !
16:41.28mm_pti could wait
16:41.31FuriousGeorgetwista: yeah, how come they cant talk
16:41.33SarahEmmerr
16:41.42SarahEmmi didn't say wait, i just said ask your question here and someone might be able to help
16:41.48mm_ptok
16:41.58twistawell thats the scenario my company wants me to work on ;)
16:42.01jac]Z[obyhow does the bristuffed active ACER ISDN card know on which msn to listen to?
16:42.04astoriamm_pt: http://www.voip-info.org/wiki/view/Asterisk+phone+xten+xlite
16:42.08twisted[asteria]SarahEmm, huh?
16:42.14mm_ptthanks astoria
16:42.16SarahEmmtwisted[asteria]: gah wrong person sorry
16:42.16twistalike a situation where the asterisk server has two interface but is not "allowed" to route between them
16:42.27FuriousGeorgetwista: the scenario has to coincide with reality.  why, hypotheticallh cant they talk
16:42.35twisted[asteria]SarahEmm, heh, k
16:42.39SarahEmmtwista: so the situation doesn't actually exist?
16:42.49SarahEmmtwista: so it's not allowed to route between them, but it can proxy between them?
16:42.55SarahEmmsounds like you want a SIP proxy, not *
16:43.07twistakind of, yes
16:43.15twistabut it also handles the connection to the psdtn
16:43.17twistapstn
16:43.27drumkillatwista and twisted, how cute
16:43.29FuriousGeorgetwista: if its a nat issue you can set externip and localnet on the * server
16:43.47FuriousGeorgeand two clients that couldnt talk directly can now be friends with *
16:44.01FuriousGeorgeor "friends in* *"
16:44.12SarahEmmtwista: so there's no NAT, but they're on seperate subnets then? and they're not allowed to talk directly to eachother?
16:44.36FuriousGeorgein that case u'd use a vpn right?
16:44.51twistayes...imagine one interface with 192.168.0.0/24 and one with 172.16.0.0/16 (just as an example)
16:44.53astoriabintut: I wouldn't tell that to your boss...
16:45.02FuriousGeorgeor use an IAX client and tunnel the 4569 port (i think it is)
16:45.10twistawell a vpn would be the easy way that'll work in any case
16:45.53SarahEmmsemihere for a few, on a call
16:46.04twistabut actually my question is just if its possible to make asterisk "handle" all connections (not just the ones where the codec needs to be changed/translated)
16:46.12bintutastoria: :)
16:46.21FuriousGeorgeif the 182.X is a subnet in 192.X wouldnt you just needd to put * in the 192. subnet and both would see it, no?
16:46.48twistahmm?
16:47.16SarahEmmtwista: ahh.... i'm not sure offhand on that...
16:47.34twistaFuriousGeorge its like two phones in two different nets connect to the same asterisk server (which has two interfaces) but they can't establish a direct connection between them
16:47.46*** join/#asterisk Juxt (n=Juxt@sfl-dsl-64-135-113-4-cust.host.net)
16:48.01twistaso right now the phones ring (because asterisk takes care of the sip-part) but of course the rtp packets get lost
16:48.07FuriousGeorgetwista: two interfaces=two ethX?
16:48.08Juxthello
16:48.15Juxtcan someone explain how asterisk -p option works?
16:48.25Juxtthe pseudo-realtime thing
16:48.31Juxtdoes it actually improve performance?
16:48.37*** join/#asterisk power1 (n=marktren@rndf-146-4-251.telkomadsl.co.za)
16:48.45twistaFuriousGeorge yes...right now its eth0 and eth0:0 for testing purposes
16:49.07twistaoh and i switched to the cvs head today (from 1.0.9)
16:49.59FuriousGeorgehmm, i have no idea how that would work from an * config perspective, but maybe you need SER
16:50.08FuriousGeorgeask the guys in #SER
16:50.19power1can some 1 help me with this issue, I have asterix @ home 1.5 running with a digium tdm400p , everything works except when there is a call received via an fxo and the person hangs up, asterisk keeps the line open indefinately.....any ideas?
16:50.51FuriousGeorgepower1: does a@h use kewlstart signalling for fxo?
16:51.18twistaFuriousGeorge hmm i read a lot of stuff about proxys, ser etc. today but thought i probably could avoid having to setup additional software
16:51.44FuriousGeorgetwista: i can't say for certain that you cant, but ive jest never considered it
16:51.47power1FuriousGeorge, ummm I dont know, how do i check this?
16:52.04FuriousGeorgepower1: nano -w /etc/asterisc/zapata.conf
16:52.12FuriousGeorgethen paste that on pastebin.ca
16:52.14FuriousGeorge~pb
16:52.16jboti guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
16:52.19*** part/#asterisk Juxt (n=Juxt@sfl-dsl-64-135-113-4-cust.host.net)
16:52.46FuriousGeorgetwista: also will need /etc/zaptel.conf
16:53.19FuriousGeorgei mean power1:
16:53.27twistadoes anyone have experience with that iptables conntrack/nat sip module (i just saw that today during my research)...because i guess i've to deal with the nat-problem in the next days too
16:54.04FuriousGeorgewhat sort of nat problem?
16:54.17mm_ptany one who had experience problems regestering xlite with asterisk? call not aproved...
16:54.52FuriousGeorgemm_pt: what if it has registered and your dialplan isnt setup
16:54.53znoGif i want to connect to a h323 gateway, i just need h323 gateway functionality on Asterisk, not gatekeeper.. right?
16:55.36mm_ptin winxp computer it doesn register, i've configured sip.conf and extensions.conf
16:55.41FuriousGeorgeznoG: i think you need a gatekeepser, im not certain.  i think i remember reading somewhere that "Asterisk cannot function ax the gatekwwper,atm"
16:55.46twistaFuriousGeorge sip phones + asterisk behind a nat (or probably asterisk will get a public ip, too - dont know yet) + calling remote sip clients (probably with public internet ips)
16:55.50mm_pti was just trying echo *45
16:56.14power1FuriousGeorge, http://pastebin.ca/26068
16:56.16*** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com)
16:56.16twistabut i'll figure that out by myself...or at least i'll try before i'll ask about that stuff again here
16:56.25FuriousGeorgeas long as your asterisk server has the right ports and the right settings in sip.conf, local and remote clients behind their own nat will see *
16:56.39twistayep, thats what i expected
16:56.40FuriousGeorgeright ports forearded*
16:57.06mm_ptis there any channel for freshman with asterisk?:)
16:57.15mm_pti don't want to bother hete
16:57.16mm_pthere
16:58.45astoriamm_pt: read stuff on the wiki.. thats the best way..
16:58.46twistaoh but FuriousGeorge, any experience (or at least anything about) the nat modules for netfilter? ( http://www.iptel.org/sipalg/ )
16:58.49FuriousGeorgemm_pt: its called www.voip-info.org
16:58.49mm_ptor other way, because i just want to see it working in some way, may i install sip phone in the same machine of asterisk server?
16:59.19FuriousGeorgetwista: i use IP Cop and its been all i need so far
16:59.28power1FuriousGeorge, thanks, did u get my pastebin?
16:59.49twistaoehm does it come with those modules? (or just the regular stuff?)
16:59.58FuriousGeorgepower1: u gotta post the link for me here
17:00.32power1FuriousGeorge, i did look above
17:00.41FuriousGeorgemy bad :)
17:00.50power1FuriousGeorge, <grin>
17:01.07FuriousGeorgepower1: there must be more
17:01.10FuriousGeorgescroll down
17:02.06power1FuriousGeorge, nope thats it 48 lines
17:02.38power1FuriousGeorge, u sure u r not looking for /etc/zaptel.conf
17:03.33FuriousGeorgepower1: oh i see
17:03.59FuriousGeorgepower1: i guess i gotta see zapata-auto.conf and zapata_additional.conf which are being included
17:04.08FuriousGeorge# isnt a comment
17:04.20*** join/#asterisk damned (n=vpol@damned.vpol.org.ru)
17:04.38power1FuriousGeorge, ok lemme paste em.
17:04.55[TK]D-Fenderrepeat of my earlier question :
17:04.58[TK]D-FenderHey I'm trying to find a decent IAX softphone for Windows and Firefly's latest build won't even let me ASNWER an incoming call.  Forget about such fun and obvious options as Hold, XFer, etc....  Any suggestions?  I need IAX for NAT reasons (We are on a fixed IP, but behind a SonicWALL router)
17:05.27SarahEmmyou can NAT SIP, it's just a pain...
17:05.32FuriousGeorge[TK]D-Fender: i dunno, dot use'em myself.  have you looked at iaxtel
17:05.55FuriousGeorgefile[laptop]: are you around?
17:06.00twistaoh and btw, i put a little effort in developing a little php-driven frontend for the asterisk config (or at least adding/deleteing/updating iax & sip clients, adding/deleting/updating the dialplan, listing current calls (manager api), viewing call history in mysql db etc.)
17:06.09[TK]D-FenderFuriousGeorge : I need a softphone, not termination
17:06.23FuriousGeorge[TK]D-Fender: i meant to say iaxphone, i think its called
17:06.37power1FuriousGeorge, heres the 1 http://pastebin.ca/26074
17:06.38[TK]D-FenderSarahEmm : I'm worried about double-NAT scenarios.....
17:06.39twistaand all by using & parsing the existing config, without templates or profiles which overwritte your existing configuration
17:06.43brimstoneiaxcomm [TK]D-Fender
17:07.13FuriousGeorgebrimstone: thanks
17:07.22brimstoneFuriousGeorge: np
17:07.25*** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc)
17:07.39power1FuriousGeorge, zapata_additional.conf is empty...
17:07.40*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
17:08.01FuriousGeorgepower1: what about zapata-auto.conf, and /etc/zaptel.conf
17:08.34*** join/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com)
17:09.40[TK]D-FenderI tried IAXCOMM, and it was kinda flakey
17:10.11brimstone[TK]D-Fender: you might be better off with setting up an Asterisk box inside both networks and a IAX trunk between them
17:10.13jarrodweeeeee
17:10.21brimstonethen you won't have NAT issues and can use SIP phones locally
17:11.06[TK]D-Fenderbrimstone : An idea if I wasn't going to use it for salespeople's laptops.....
17:11.38brimstoneif people are going out in the field, may i recommend an IAXy ?
17:12.26power1FuriousGeorge, the second paste is of zapata-auto.conf
17:12.51FuriousGeorgepower1: what number?
17:13.02*** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net)
17:14.02power1FuriousGeorge, here is zaptel.conf http://pastebin.ca/26076
17:14.39power1FuriousGeorge, zapata-auto.conf is number 26074
17:14.43FuriousGeorgepower1: hmm, all that work for nothing, you are using kewlstart signalling, which is supposed to detect a remote hangup
17:14.53FuriousGeorgeis this happening incoming outgoing or both
17:15.16power1FuriousGeorge, what field in what of the conf files denotes kewlstart signaling..
17:15.39FuriousGeorgefignalling fx*ks
17:15.48FuriousGeorgesignalling=fx*ks
17:16.18power1FuriousGeorge, you think it could be a faulty module on the digium board..... all was working perfectly before and then it started having this problem...I have even done a fresh install and it does the same thing.....I just dont understand it.
17:16.44FuriousGeorgepower1: i had a problem once where it wouldnt pickup
17:16.50FuriousGeorgewhat version of the driver are you using?
17:16.54FuriousGeorge1.9.2 is latest
17:17.19*** join/#asterisk santiago (n=santiago@208.195.215.158)
17:17.22FuriousGeorgesomeone else is gonna have to tell you were to look for that b/c tbh, i dont know.  1.9.2 is in my gentoo portage
17:17.23brimstone1.0.9.2
17:17.27FuriousGeorgeso thats how i check
17:17.34FuriousGeorgebrimstone: do'h thanks
17:17.51brimstoneFuriousGeorge: no problem, got to earn this hostmask some how
17:18.02FuriousGeorgetwista: listen to brimstone :)
17:18.21*** join/#asterisk loick (n=loick@APuteaux-151-1-34-29.w82-120.abo.wanadoo.fr)
17:18.36power1FuriousGeorge,  im using the standard a@h iso that runs on top of centos..i dunno?
17:20.02moralew
17:20.31FuriousGeorgedoes it answer the line and call out?
17:21.28FuriousGeorgetwista: i mean:  does it pickup the line when you call out
17:21.38*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:21.57copantlcan asterisk be interconnected to a pstn via C7?
17:23.27FuriousGeorge!C&
17:23.32FuriousGeorge~C7
17:23.50power1FuriousGeorge, under trunk sequence if i have 2 fxo modules and 1 fxs module - should there be more to select than just " ZAP/g0" ???
17:24.02*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-202.rockynet.com)
17:24.06FuriousGeorgeg is the group
17:24.15FuriousGeorgeso it picks a free one
17:24.48*** join/#asterisk txbobw (n=non@c-67-174-69-147.hsd1.tx.comcast.net)
17:25.30[TK]D-Fenderbrimstone : Iaxy is a nifty idea, but they need WiFi access, etc.
17:25.47brimstoneah
17:25.47[TK]D-Fendermc
17:26.22brimstonecould just use a crossover cable and ICS on the winderz laptopo
17:26.30brimstonebut that starts to get crazy
17:27.47*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
17:27.52*** join/#asterisk cripito (n=ncripito@67.96.197.99)
17:28.01cripitohi
17:28.37*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
17:31.57cripitothere is any way to totally disable the cdr?
17:32.15cripitoCDR logging: enabled
17:32.15cripitoCDR mode: simple
17:32.26skyengot secrets, hey?
17:32.32cripitono :)
17:32.43skyensorry, I dunno
17:32.50cripitoi have 5 asterisk servers and i wanna disable the cdr in the ones that have digiums
17:35.01*** join/#asterisk [Lamer] (i=Lamer@61.47.112.209)
17:35.22[Lamer]Hi, I just reinstalled * and found the pbx.c:1293 pbx_extension_helper: No applicati
17:35.23[Lamer]on 'Math' for extension (from-pstn, t, 1)
17:35.51[Lamer]what package does the 'Math' come with?
17:38.29cripitoi found the way...
17:41.25[Lamer]cripito: what is it?
17:48.53astoriai love technology
17:49.02astoriai've been chatting with babes all day
17:49.20lehelnice astoria;)
17:50.00funxionu mean old men pretending to be babes
17:50.07snittlol
17:50.41*** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3685577.sympatico.ca)
17:52.33iCEBrkrCrap, my cover is blown
17:52.58funxionso has anyone tried t38 pass through on new cvs head
17:53.59supaigtrt.38 is in head?
17:54.09funxionaccording to wiki
17:54.15*** join/#asterisk zeedo (n=zeedo@80.68.92.188)
17:54.31funxionCVS as of Oct. 2005 now has very limited and experimental T.38 pass-through support for SIP (bug/patch 5090).
17:54.55supaigtrHmm.
17:55.12funxionso i take it no one has tried it
17:55.15supaigtrfunxion: Have you tested it?
17:55.21supaigtrI could try it.
17:55.33funxionI plan to upgrade later today
17:55.46supaigtrI'll try it real quick.
17:55.58astoria<PROTECTED>
17:56.03supaigtrfunxion: what device are you sending to?
17:56.03astoriaoh shit
17:56.09astoriai did not mean to paste that!
17:56.14funxioncisco 2620xm
17:56.18astoriawell, please don't steal my license code.
17:56.21funxionlol
17:56.26*** join/#asterisk darkskiez (n=darkskie@host-84-9-237-137.bulldogdsl.com)
17:56.39astoriawell, that's going to get archived..
17:56.40Kattyastoria: i'll get right on that ;)
17:56.44supaigtrfunxion: 5300 and MaxTNT
17:56.48astoriaand i'm going to get a nasty letter from pdf lib
17:56.58mmlj4use free software, no license issues :-)
17:57.10Kattyastoria: you could contact the archivers.
17:57.23astoriawho are the archivers?
17:57.33Kattythe Powers That Be
17:57.52jontow:o
17:58.03astoriajesus. today is not my day..
17:58.12jontowjbot is the archiver, iirc
17:58.19astoriai might as well just post my social security number here too..
17:58.26astoriawho runs jbot?
17:58.26jontowand if you /whois jbot
17:58.31supaigtrThat would be great lemme get a pin.
17:58.35jontowyou'll see that its TimRiker's bot.. so.. if you /whois TimRiker
17:58.39jontow;)
17:58.43astoriathanks jontow
17:58.46jontownp
17:58.56jontowdon't know if anyone else archives it
17:58.56twistaFuriousGeorge? sry...i was afk... well of course it does not pick up the line
17:58.59jontowbut worth a shot.
17:59.39supaigtr:) disable_vpm + KB1 = nice 411p.....
17:59.47twistathe phones ring but they send their rtp packets straight into a wall
18:01.04jontowi will admit though, from a quick google, looks like its more than just jbot archiving
18:01.12astoriaOh well...
18:02.19SarahEmmvoip handsets really aren't designed to be used with acoustic couplers.
18:02.23astoriaoh well, it's an old license code anyway..
18:03.12supaigtrSarahEmm:  I haven't seen one of those in years...
18:03.16SarahEmmheh :)
18:03.45*** join/#asterisk damned (n=vpol@prior.lanck.net)
18:04.05supaigtrSarahEmm: What u doing with it?  I think we have a couple hear they have on a polycom...werking.
18:04.12*** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
18:04.31SarahEmmsupaigtr: you're using polycoms with acoustic couplers? for what
18:04.32SarahEmm?
18:05.00supaigtrThey were testing the hearing impaired devices. 300 baud modem like thingys.
18:05.01Ahrimanesacoustic coupler = aid for hearing impaired?
18:05.08*** join/#asterisk Exomorph (i=Exomorph@216.251.134.2)
18:05.15jontowhttp://www.digitallyunique.com/koupler.html?src=247
18:05.22jontowthey still make them
18:05.26*** join/#asterisk dstruct (i=dstruct@unaffiliated/dstruct)
18:05.30yxadoes * generally run better with 2.4 or 2.6 kernels?
18:05.48*** join/#asterisk RickTick (n=rpulido_@71.196.17.112)
18:05.53cripito2.6 works better for me
18:05.59SarahEmmsupaigtr: ahh. that's what i'm using it for too. the polycom handset on my TTY's cups for testing :)
18:06.05supaigtr2.6 and 2.4 here works fine either way.
18:06.17SarahEmmAhrimanes: no, acoustic coupler is cups you put the handset of a phone on and it 'listens' and 'talks' on the phone for data use
18:06.28Exomorph<PROTECTED>
18:06.29AhrimanesSarahEmm: ah ok
18:06.37jontowsarah :)))
18:06.38yxasupaigtr is 2.6 faster than 2.4? or rather makes better use of cpu?
18:06.39SarahEmmExomorph: it's on the digium store
18:06.41SarahEmmhihi jontow
18:06.53jontowawesome project, that is
18:06.54supaigtryxa: ?? never tested?
18:06.56SarahEmmExomorph: store.digium.com
18:07.01SarahEmmjontow: me?
18:07.09jontowyes, the tty code
18:07.14SarahEmmahh :)
18:07.15ExomorphSarahEmm: Ahhhh... I was just on the main site, and didn't see it.  Thanks.
18:07.18supaigtrSarahEmm. * has TTY code in it.  Support?
18:07.32RickTickAnyone care to comment on using Quintum 4 port FXS/FXO gateway with asterisk?
18:08.28SarahEmmsupaigtr: it has TTY support, yes. rudimentery, but it's there. onyl accessable from AGI or apps, so you have to write your own apps to do what you want
18:08.40SarahEmmi'm working on a basic set of apps to do stuff that you can do with sounds, but with text
18:08.52SarahEmmso you could have an IVR-like menu system for TTY users
18:09.02supaigtrAhh. Thats cool.  I wonder if it would meet requirements eventually of softswitch market?
18:09.25supaigtrYea.  We are trying to support 3 rd party and 411 type stuff for TTY users.
18:09.33kuku5What do you guys think of hp switches ?
18:09.48jontowkuku5; i have good luck with the procurve 2524's
18:09.52supaigtrkuku5: they are cute.
18:10.00kuku5I used them
18:10.02SarahEmmsupaigtr: ahh nifty :o)
18:10.09kuku5but I dont know if they are good ;) they work fine for me
18:10.14kuku5Maybe someone had problems
18:10.19SarahEmmsupaigtr: my first goal is to be able to make/receive TTY calls from a web interface, and let it take messages when i'm not around
18:10.36supaigtrkuku5: Have had some ports go bad but other than that ok.
18:10.40astoriatty is for the hearing impaired right?
18:10.48astoriaIs that the thing where an operator calls you and repeats things?
18:11.09*** join/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au)
18:11.20supaigtrSarahEmm: Realtime interface?  the stuff we use know is a vb.net interface into a 250,000 based BLVI system.
18:11.20kuku5What parts?
18:11.34SarahEmmastoria: it's a text telephone. for hard of hearing, d/Deaf, speech impaired users.
18:11.54SarahEmmsupaigtr: realtime interface? whatcha mean?
18:12.06supaigtrTTY is a very cool application in terms of the PSTN and VOIP.
18:12.12SarahEmmastoria: that's relay, where an operator relays in between voice and TTY users
18:12.25astoriaoh yeah.. relay is the fun one..
18:12.38SarahEmmfun?
18:12.40Kattyyou know what's really fun?
18:12.44Kattystring theory.
18:12.45Kattynow /that/ is fun
18:13.08*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
18:13.19astoriayeah, kind of, but not as much fun as getting calls at 3am from a relay operator when one of your buddies is drunk and having fun..
18:13.30SarahEmm*sighs*
18:13.34supaigtrWe are implmenting a third party calling solution so a normal operator with a computer and sip phone can do a thrid party call out or in.
18:13.55Kattyheh 'solution'
18:13.55*** join/#asterisk Xen^ (i=linux@202.5.131.110)
18:13.58Xen^hello all
18:14.08Kattyi'm always amused at the lack of information that 'solution' seems to create
18:14.09astoriai always feel bad for the relay operators, they have a pretty crummy job..
18:14.16Xen^do i need to install zaptel in order to use MOH ?
18:14.46supaigtrsolution = $250,000 software and hardware combo that basically sucks
18:15.11astoriaXen^: no
18:15.14supaigtrsolution = AKA nortel
18:15.22SarahEmmastoria: mew?
18:15.23Xen^astoria : but its not working :(
18:15.34Xen^getting this error local_ast_moh_start: No class: default
18:15.35astoriaSarahEmm: mew?
18:15.36jontowpersonally.. i'd love to be a relay operator
18:15.46astoriayou would??
18:15.51jontowthe fact that you can't really get in trouble unless you DON'T say what they want you to
18:15.54jontowhehe
18:15.57astorialol, that's true!
18:17.01supaigtrSarahEmm: Has anyone turned tty into IM on SIP?
18:17.01Xen^can some one please help :(
18:17.29astoriaIs there any output?
18:17.30SarahEmmsupaigtr: that's the next step for me i think. but it'll be more difficult, TTY is character oriented, IM is block oriented.
18:17.41Xen^astoria : yes
18:17.51astoriaXen^: what does it say?
18:17.51Xen^astoria : res_musiconhold.c:864 local_ast_moh_start: No class: default
18:18.03*** join/#asterisk sgorilla (n=tlp@cpe-24-160-119-179.houston.res.rr.com)
18:18.40supaigtrnortel has a char limit type system.  Its really strange.  Uses operator messaging which is NORTEL only.  Really hard to deal with.
18:19.02Xen^astoria : this is what i get when i press hold res_musiconhold.c:864 local_ast_moh_start: No class: default
18:19.06supaigtrTTY device send char limit message pops us sends char limit again message pops up.
18:19.07SarahEmmsupaigtr: hrm? *confused*
18:19.16SarahEmmi don't understand..
18:19.30JASON-0Every extention I dial seems to be busy. When I look at the logs I see...  Oct 20 14:18:57 WARNING[24241] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_NE, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
18:19.31astoriawhat do you have in your musiconhold.conf??
18:19.34JASON-0Every extention I dial seems to be busy. When I look at the logs I see...  Oct 20 14:18:57 WARNING[24241] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_NE, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
18:19.39JASON-0oops
18:19.40Xen^astoria : wait
18:20.10astoriaYou need to have an entry like this: default => quietmp3:/var/lib/asterisk/mohmp3
18:20.32Xen^astoria : yeah but musiconhold.conf is different :)
18:20.43Xen^astoria : check this http://pastebin.ca/26078
18:21.10astoriaXen^: are you running cvs-head?
18:21.15Xen^yupz
18:22.13astoriaXen^: well, then i'm not sure how much I can help you.. But your error: No class: default probably indicates that you have no class named default, and you probably should.
18:22.51Xen^but if you see my config it have it
18:22.52Xen^:(
18:22.57Xen^check this http://pastebin.ca/26078
18:23.00astoriawell, that's a context.
18:23.10Xen^umm
18:23.31Xen^default => quietmp3:/var/lib/asterisk/mohmp3 ?
18:23.58astoriaWell, try making a context named [classes]
18:24.03astoriaand putting that in it...
18:24.06astoriasee if it does anything.
18:24.11Xen^ok
18:24.12astoriaAre you using the default .conf?
18:24.17Xen^yupz
18:26.32supaigtrSarahEmm: The way the app works is they dial a DID.  It forwards to the BVLI line which then sends tty messageing to the NOrtel operators console.  They read that off in blocks or sentences to the other party and vice versa.   The nortel units we us have 30 char limit so press next next .  So the nortel in effect gives you only 30 char block
18:27.48SarahEmmblvi?
18:28.03SarahEmmahh okay re: blocks
18:29.07supaigtrblvi is just an operator line.
18:29.14supaigtrYou can transfer etc.
18:29.21*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
18:30.06*** join/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net)
18:30.15*** part/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net)
18:30.16*** join/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net)
18:30.20MuppetMasterHello
18:30.27MuppetMasterInteresting presentation posted from Astricon here:  http://www.jivesoftware.org/community/servlet/JiveServlet/download/47-16175-105681-2242/astricon-presentation-cleaned-2.ppt
18:30.50rob112hey all, anyone up on dsp call progress?
18:31.06SarahEmmgaaah
18:31.16SarahEmmi get three lines of code on an app written and find more bugs int he core * code
18:31.21SarahEmmsupaigtr: ahh okay
18:31.22harryvvI have a sixtel account to make long distance accounts. I ordered a DID a week ago and thay gave me a phone number and no other info. Is it my assumption that this is tied to my existing account with no other config info needed? this is my first did. BTW, the did thay gave me was tied to a existing cell account.
18:31.24astoriaXen^: try installing zaptel.. i'm not positivie, but perhaps it requires zaptel timing.
18:31.38SarahEmmharryvv: that's likely what they did, yeah
18:31.53Xen^<astoria> : well thats fix now but now i get no sound
18:31.53harryvvokay
18:32.11Xen^i just edit /etc/asteisk/zapata.conf
18:32.28Xen^-- Started music on hold, class 'default', on SIP/1-2c71
18:33.13harryvvbtw, I am looking for a sip wholsaler with a quick turn around time on DIDs. Also, in the future would like them to be involved in a
18:33.38harryvvnumber switch from local carrier to sip provider. Aka, get rid of the local clec
18:35.04SarahEmmsupaigtr: you don't happen to know of any ring-signalling devices that support SIP or IAX do you?
18:36.54supaigtrSarahEmm: ring-signalling ?
18:37.31cripitoi love this... 92 zap calls not a single drop today
18:37.37supaigtrU mean like a garage ringer?
18:39.11SarahEmmsupaigtr: any kind of device that's just a ringer :)
18:39.15harryvvcripto, ive never had a drop on local calls
18:39.21SarahEmmideally with visual ringing built in too, but that could be modified in
18:39.39supaigtrWe cheat and use a sipura or a iaxy with a viking or other PSTN type device.
18:39.55supaigtrI don't know of any that have direct support for a VOIP call.
18:39.55*** join/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4159391.sympatico.ca)
18:40.01ian_kSarahEmm - take a shoebox... toss a sip.iax phone inside.. when it rings, just pretend it's a ringing shoebox.
18:40.10SarahEmmheh, yeah....
18:40.19JASON-0Anyone see anything wrong with this..
18:40.20JASON-0exten => s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} != ${RGPREFIX}]?4:3)
18:40.22SarahEmmsupaigtr: that's what everyone says...
18:40.28SarahEmmi think i'll design/build a real device.
18:40.31cripitoharryvv is 92 concurrent call from a dialer (predictive) connecting agents. b/f than today.. lots of drops
18:40.47ian_kJASON-O yeah.. looks like something only asterisk @home could generate.. :)
18:40.55supaigtrSarahEmm: the viking boxes are big enough to take an IAXY board and integrate into the box.
18:41.10cripitoan entire wct405 :)
18:41.32*** part/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net)
18:41.43JASON-0ian_k: yes, I'm using Asterisk@Home
18:41.51JASON-0my logs complain about that line..
18:42.02ian_kJason-o - what is the complaint?
18:42.15JASON-0Oct 20 14:39:06 WARNING[27518] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
18:42.28JASON-0with an arrow under the !
18:43.00ian_kis RGPREFIX defined?
18:43.25SarahEmmsupaigtr: they're just audio tho right?
18:43.52harryvvI have a agency that wants 6 remote in country offices tied together to make sip calls between offices. A total of 30 phones will be involved. What kind of processor should be nessesary? Thay wanted to go on the cheap for a intel pbx box but I did not recomend a desktop and only a server for reliability. Was thinking of dell.
18:44.20*** join/#asterisk ^X-works (n=drttrtr@81-208-62-98.ip.fastwebnet.it)
18:44.32GeneGHi all. Installing a SIP Asterisk box for the first time, getting unintelligible voice-prompts when I dial in. Outbound and incoming voice quality is fine, just the prompt playback seems broken. CPU utilization is low, and I'm playing the demo gsm prompts that came with Asterisk. Any ideas ? (oh using ztdummy, not actualy hardware)
18:44.36supaigtrSarahEmm: They have strobe, relay,audio interface and audio only devices
18:44.52SarahEmmsupaigtr: ahh. :)
18:45.00JASON-0ian_k: There is no reference to RGPREFIX prior to this line in that file
18:45.40JASON-0ian_k: I dont really know if thats causing a problem.. but every extention I dial gives me a busy signal
18:45.45ian_kdefine at as something random and try it again
18:46.50ian_kharryvv - what codec will you be using? will this be a soft pbx, or will you use digium zap hardware?
18:47.18ian_kJASOn-O it probably has a lot to do with it, depending on what priority 3 and 4 is for that exten
18:48.14harryvvian, well the idea would be to replace the phones so will need local access of telus. it would need some cards for local access.
18:48.32harryvvno T-1 that I am aware of is invoved but I need to look further into it.
18:48.51harryvvthere is very low call volume involved.
18:49.01harryvvat any one time.
18:50.20ian_kwill the asterisk box be transcoding channels?
18:50.35ian_k(connecting different codecs)
18:51.04harryvvall ulaw
18:51.07JASON-0ian_k: I dont know how to define..
18:51.35*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:51.52ian_kharryvv you could run that on a low-end box easily
18:52.33harryvvsure
18:52.40harryvvbut i want dependability
18:52.55harryvvthis server will be located in the next province.
18:52.57SarahEmmwhat cards are you using harryvv? that may affect your decision too
18:52.59ian_kget a dependable low-end box. :)
18:53.24harryvvSarahEmm well the idea is thay want to save on long distance calls.
18:53.32harryvvBut still need local access.
18:53.39SarahEmmerr... that doesn't answer my question :)
18:53.39*** join/#asterisk loick (n=loick@APuteaux-151-1-50-203.w82-124.abo.wanadoo.fr)
18:54.08*** part/#asterisk oej (n=Olle@apollo.webway.se)
18:54.19harryvvsara, no T-1 involved to my knowladge. So say a X100p or equal. mabey two per box.
18:54.23SarahEmmuhh
18:54.27SarahEmmthey only need to make two calls at once?
18:54.27*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
18:54.28ian_kJASON-O: RGPREFIX=XYZ
18:54.32SarahEmmerr
18:54.33SarahEmmone call
18:54.37SarahEmmplus leave one line for incoming
18:54.44SarahEmm(and you don't want to use an x100p really)
18:55.05SarahEmmif you really only do need one call out and one call in at a time, get a BRI
18:55.08SarahEmmbut that sounds low
18:55.18harryvvits a 20 phone system and at what i was told 4 calls at any one time. So mabey 6-8 incomming calls.
18:55.28harryvverr
18:55.35harryvvincomming/outgoing calls I mean
18:56.49ian_kharryvv a 500mhz system could run a pure sip/iax network with ulaw
18:57.35astoriaI run a webserver/voicemail system on a 500mhz and it runs fine under lots of webtraffic and moderate voip traffic..
18:57.41astoriait's even on the public internet.
18:58.18harryvvian i know
18:58.57*** part/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4159391.sympatico.ca)
18:58.59harryvvso as long as a used dell sc420 had parts that are avaiable then it would work.
19:02.19*** join/#asterisk oej (n=Olle@apollo.webway.se)
19:03.52*** join/#asterisk DarthClue (i=user76@wsip-68-99-73-32.tu.ok.cox.net)
19:04.08*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
19:04.39SarahEmmharryvv: okay, so you need a lot more than 2 x100ps then :)
19:04.59SarahEmmif you have 8 lines, and they're all analog, you'd want 2 TDM400s.. that's not a great solution tho, going digital would be a lot better
19:05.39Inv_arpSarahEmm: what technology would be best for 8 lines?
19:06.14harryvvyes I know
19:06.26harryvvand this is only one site
19:06.29*** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
19:06.30harryvvthe main site
19:06.32*** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
19:06.35harryvvwhere the pbx is located
19:06.45harryvvnow
19:07.28harryvvI need to look behind this site to see if its T-1 or not.
19:09.22greekmananyone know if you can force irqs on the te410p?
19:10.57SarahEmmInv_arp: probably a PRI with 8 channels active.
19:11.12SarahEmmyou could also go with 4 BRIs... or get termination via IAX
19:11.14SarahEmmor SIP or whatever
19:12.40astoriaI would go with a partial T1 if it was mission-critical, otherwise i'd just use SIP termination..
19:13.08astoriagreekman: it depends on the bios...
19:18.11*** join/#asterisk pointer (i=pointer@aj.catt.com)
19:18.23*** part/#asterisk pointer (i=pointer@aj.catt.com)
19:18.30harryvvWhat cards can except more then one pstn connection ?
19:19.04astoriaThe Quad TDM cards, with four FXO modules, can.
19:19.11astoriaSo can channel banks.
19:20.18SarahEmmas can the dual/quad T1/E1 cards
19:21.38*** join/#asterisk rocket (n=rocket@gentoo/developer/rocket)
19:22.08vader-wrkany of you guys use softphones?
19:23.16vader-wrkhehe
19:23.21vader-wrkguess no one likes softphones
19:23.23vader-wrk:)
19:23.34harryvvwhat model
19:23.52vader-wrkhi harryvv
19:24.35harryvvtdm40b
19:24.41harryvvhi vadar
19:24.54vader-wrkim starting to get a tighter plan together for this system
19:25.21vader-wrkbut the road block i keep running into is the people that are involved with this wanna just do a test of the 15 people moving instead of replacing the entire system
19:25.35vader-wrkwhich i think will be a major headache and cause the asterisk system to look bad
19:25.49inspiredis NoCDR a good idea before a timeout (t)?
19:25.58vader-wrkthey wanna seamlessly intergrate asterisk with the old pbx
19:26.05vader-wrkand i don't know if thats going to go smoothly
19:26.20jarrodhey how do i play back a variable to a customer so they hear the digits
19:27.09jarrodlike Playvariable(${var}) and var=1111111111
19:27.09inspireda customer has problems where some of his calls are flagged as ANSWERED and billsec = 10 and my CDR is NO ANSWER and billsec = 0. I see the calls on my server got to timeout, so I'm wondering if I can use NoCDR or something, or will that only affect my CDR, not his?
19:28.35*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)
19:28.48harryvvif a company has multiple phone numbers comming into there site does that nessesarly mean thay dont have a T-1 and only pstn connections?
19:29.11harryvvWant to know this so I wont have to dig though the telco closet and do a physical check.
19:29.18SarahEmmharryvv: no. could be either.
19:29.20SarahEmmyou need to dig ;)
19:29.22harryvvThis would be for a small site
19:29.24SarahEmmor look at their bills.
19:29.25harryvvokay
19:29.27syzygyBSDany ideas why I would be getting a "post-install wcte11xp failed" when I try to do a `modprobe wcte11xp`?  it worked yesterday
19:29.37SarahEmmwhat's in dmesg?
19:29.41harryvvWhat would there bills say?
19:29.46syzygyBSDSarahEmm: bootup messages
19:29.58harryvvI dont think it would show the type of connections would it?
19:31.16SarahEmmharryvv: it might
19:31.18SarahEmmit should..
19:31.35SarahEmmsyzygyBSD: uhh.. yeah, i know. i meant related to the post-install failure :)
19:32.03syzygyBSDlol.. oh...
19:32.31syzygyBSDnothing about that module in it
19:33.05syzygyBSDfirst time i have seen IRQ redirection tables in dmesg though
19:33.26*** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
19:35.36syzygyBSDlooks like the card was scanned and started right in dmesg, though I commented out the modprobes from startup so I could run them in the console because of the problems I am having
19:36.08syzygyBSDwanted to see what it said the first time, if i remove all the modules then modprobe them again everythign works
19:36.37mcf3782I have a SIP ATA device I'm playing with. It has an IP address on my LAN, I've told it what the IP address of my Asterisk server is, given it an extension, and configured that extension in extensions.conf. I can call the extension from the asterisk console, and the phone plugged into the ATA rings and I have two-way audio. So far, so good.
19:37.16mcf3782But if I pick up the phone plugged into the ATA and try to dial an extension (like 8500 for voicemail) on the Asterisk server, it doesn't work.
19:37.30SarahEmmmcf3782: your dialplan isn't set up right, likely.
19:37.33SarahEmmcan you pastebin your extensions.conf?
19:37.34mcf3782Any ideas where I should start looking for config issues?
19:37.51*** join/#asterisk fifer (n=sirfifer@207.202.227.161)
19:38.33fiferAny Aastra 480i users here (This is not a call for help, rather the opisite)
19:38.41syzygyBSDmcf3782: what does the console say?
19:38.44SarahEmmmcf3782: extensions.conf... can you pastebin it?
19:39.27_ThorOct 20 11:59:18 WARNING[580]: chan_oh323.c:4014 oh323_gk_check: Gatekeeper discovery failed.
19:39.27_Thor<PROTECTED>
19:39.31mcf3782SarahEmm, I will go 'sanitize' a copy of my extensions.conf file and do that and let you know when it's there.
19:39.36mutilatoryays
19:39.42SarahEmmokie
19:39.47mutilatorgot a few te405's in today
19:39.48Delta34any cisco 7960 users here, was wondering what there sip show peer status latency times looked like, all my cisco phones on internal lan showing up at 70+ms
19:40.05_ThorHello, I recently installed oh323, why am I getting this message? : Oct 20 11:59:18 WARNING[580]: chan_oh323.c:4014 oh323_gk_check: Gatekeeper discovery failed.
19:40.05_Thor<PROTECTED>
19:40.17mcf3782syzygyBSD: The console doesn't say anything for an inbound attempt.. even running in vvvvvv mode.
19:40.31syzygyBSDonly 6 v's?
19:40.33syzygyBSDlol
19:40.37mcf3782:)
19:40.51vader-wrkany of you guys use softphones?
19:40.59syzygyBSDvader-wrk: I have one
19:41.07vader-wrkwhat client do you use?
19:41.11syzygyBSDx-lite
19:41.19vader-wrkhow well does it work?
19:41.23syzygyBSDdon't really use it though
19:41.40mcf3782I finally got the x-lite softphone to work with my asterisk server. What an adventure that was.
19:41.46syzygyBSDworks well enough, couple issues a friend said he had with it, but mainly phonebook issues
19:41.59vader-wrkdoes the phonebook link to asterisk?
19:42.13syzygyBSDlol, ya, took a while till i found a good configuration FAQ
19:42.21syzygyBSDvader-wrk: I dont' think so
19:42.25vader-wrkdoes it sound ok?
19:42.54mcf3782the voice quality in my limited use/tests so far has been acceptable.
19:45.03syzygyBSDsound quality is good as far as I have used it
19:45.38*** join/#asterisk davidnicol (n=chatzill@rrcs-67-53-67-115.west.biz.rr.com)
19:46.02davidnicolwho can point me to some /dev/phone ioctl documentation?
19:47.00*** join/#asterisk pussfeller (n=todd@12.150.129.170)
19:47.14twistare
19:47.27mcf3782/m SarahEmm http://pastebin.ca/26090
19:47.37iDunnoguys: what IP handsets do you recommend?
19:47.43mcf3782ok. that didn't work like I thought it would.
19:47.49mcf3782I hate learning a new client
19:48.07twistaoh another question...what kind of hardware do i need to connect asterisk to an analog port of a local pbx?
19:48.15mcf3782SarahEmm - you have a pastebin link. :)
19:48.19SarahEmmiDunno: i'm a grrl, but i use a polycom 501... i may not be the best person to ask, but it suits my needs :)
19:48.34twistaone of those ports where you usually connect e.g. a fax machine to
19:48.34iDunnofeatures that I'd like: programmable buttons, tiltable, clear screens, and easy to use ;)
19:48.35SarahEmmtwista: a TDM400 with one FXO port per port on the PBX
19:48.38syzygyBSDstupid modules.conf
19:48.53SarahEmmiDunno: okay, the polycoms don't have tiltable screens, but i've not seen a need to tilt them...
19:48.54syzygyBSDI will run the commands that need to run, don't do it for me
19:48.59mcf3782My extensions.conf is.. a work in progress.. I'm still at the early stages of building this system. :)
19:49.07twistathx SarahEmm
19:49.22twistai'll look it up in a second
19:49.27SarahEmmmcf3782: what context does your sip.conf drop the call into?
19:50.25twistaSarahEmm: is just the TDM400 supported or are there alternatives on the market?
19:50.26davidnicoldoes this channel have an answerbot? (hi, I'm new here)
19:50.46loudyes, type ~docs
19:50.56davidnicol~docs
19:50.57jbot[docs] 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
19:51.08iDunnoSarahEmm: hmmm - there seems to be a few issues with the polycom 501, though - like slow boot and horrible configuration interface...
19:51.22iDunnootherwise, it fits most other parts ;)
19:51.53mcf3782SarahEmm: [kphone]
19:52.11*** join/#asterisk nobell (n=jdegraff@70.103.228.158)
19:52.25*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
19:53.05SarahEmmiDunno: horrible config interface? huh?
19:53.12SarahEmmand yeah, they boot slow. but how often do you reboot?
19:53.32SarahEmmtwista: it's the only one i know of. there's actually the single-port x100p too, but it's not what you want.
19:53.57SarahEmmmcf3782: uhh, the *context* is kphone?
19:53.58twistaerm no...one channel is not that much at all
19:54.02SarahEmmyou don't have a kphone context.
19:54.30mcf3782Hmm
19:54.34davidnicolhttp://www.linuxjournal.com/node/4468/print appears to be what one gets.
19:54.59mcf3782*studies config files some more*
19:56.38SarahEmmmcf3782: you need a kphone context then, with whatever extensions you want the phone to be able to dia
19:56.38SarahEmml
19:57.10iDunnoSarahEmm: well, when reconfiguring things, lots ;)
19:57.22mcf3782What if I want that phone to be able to dial anything that's configured on the PBX? :)
19:57.29iDunnoI like the look of the GrandStream GXP-2000 :)
19:58.12iDunnowe've currently got the bottom end of the GrandStream range, the BudgeTone 101, which is not bad, but hasn't got the features that $boss wants.
19:58.29iDunnoI've just pointed him at the GXP-2000, so I might have a new phone to play with soon \o/
19:58.32SarahEmm*blinks*
19:58.37SarahEmmthe budgetone 101 is not bad?
19:58.51SarahEmmmcf3782: then you set up a context with all the extensions.
19:59.16mcf3782Sounds like I have a lot of typing to do. :)
19:59.17iDunnoit's not bad. it's got an easy to use config front end, and it can grab setup from tftp or http...
19:59.35iDunno(and I wrote an evil python script that would talk to the web interface and reboot it and things ;)
19:59.58*** join/#asterisk paryl (n=paryl@209.236.78.59)
20:00.13iDunnothe buttons are crap, the screen is crap, and it's not got any programmable buttons... but other than that, it's a nice cheap phone that works as a phone.
20:00.52SarahEmmmcf3782: well, you can use patterns :)
20:01.09SarahEmmiDunno: so 'it's not bad' means 'it works' ;)
20:01.19twistaiDunno: we just ordered an elmeg 290 phone
20:01.22twistawhich is also not bad
20:01.24*** join/#asterisk hans (n=fugalh@falcon.fugal.net)
20:01.24syzygyBSDdirty te110p takes longer to modprobe then any of the other cards, so it has to be last, or a delay build in
20:01.27SarahEmmi dunno, personally i thought the 101 was pretty bad. plus mine is now broken...
20:01.32parylwhenever i have calls go to voicemail, the greeting is never played, it just plays the standard greeting.  any idea why that would happen?
20:01.34SarahEmmi've been really happy with the olycoms
20:01.46twistabut as far as i know its the same as the snome 2xx-something
20:01.50twista*snom
20:02.01harryvvidunno, what phone?
20:02.02hansfor those not on -dev, http://hans.fugal.net/src/radp (create dialplans in ruby)
20:02.14twistaand it wasnt too expensive
20:02.21mcf3782SarahEmm: Can you give me an example so I have a place to start?
20:03.01SarahEmmmcf3782: i.e. _9XXXNXXXXXX,1,Dial(...)
20:03.15SarahEmmmcf3782: that'll match 10 digit local numbers prefixed with a 9
20:03.49mcf3782Ahh! Ok!  *light goes off* I think I'm starting to get it now. :)
20:04.27iDunnoSarahEmm: well, yes, it's not bad means that it actually talks to asterisk, and has reasonable sound :)
20:04.41parylin one of my extensions i have "exten => 6001,2,Voicemail(u6001)"... shouldn't that play the unavailable greeting?
20:04.53davidnicolspecifically, has anyone written, that you know of, software that pretends to be a /dev/phone but on the back of it is, for instance, asterisk?
20:05.19davidnicolThat is, can Asterisk pretend to be a /dev/phone rather than using a /dev/phone as a hardwware?
20:06.06davidnicoldoes openh323 have an IRC channel? I guess I'll ask their mailing list.
20:06.39gordonjcpcan anyone recommend a SIP phone where the message indication light works?
20:06.57gordonjcpI want to compare what goes on with it to what's happening with my Avaya 4602
20:07.08iCEBrkr<PROTECTED>
20:07.08iCEBrkr<PROTECTED>
20:07.10iCEBrkrerrr.
20:07.12iCEBrkrWTF
20:07.26iCEBrkrI can call my 216 number, but not my 727 number?
20:07.51iDunnotwista: ohh, that doesn't look bad :)
20:13.34twistawell it works for me ;)
20:13.52twistabut i actually had some problems with the grandstram budgettone 101 for a while
20:14.05twistawell actually they still exist
20:14.33twistai don't remember how it exactly was but i think the elmeg phone was able to call the grandstream without problems
20:14.53twistabut the other way round you ended up with something like a 'star trek spacehsip background noise'
20:14.59twistasounded quite funny
20:15.38twistaand you couldn't here the guy on the other end of the line (wasnt able to figure out the problem yet)
20:15.43twistahere = hear
20:16.02twistaanyways...i'm off for the bed..hope that helps iDunno ;)
20:16.10twistan8 everyone
20:17.29vader-wrkdo any of oyu use the web gui's to edit asterisk?
20:18.13*** join/#asterisk nmsclera (n=no-spam@70-56-136-106.albq.qwest.net)
20:18.16*** part/#asterisk hans (n=fugalh@falcon.fugal.net)
20:18.37*** join/#asterisk oej (n=Olle@apollo.webway.se)
20:18.40nmscleraIn a small-office scenario, is it wise to use Zap PRI Channels and something akin to AstFax as a fax machine replacement?
20:18.48nmscleraand if not, what's the most elegant solution aside from a separate line and fax machine for outbound faxing from asterisk?
20:20.38tzangerI have very very few issues with Tx/RxFax
20:21.23Ahrimanestzanger: does Tx/RxFax require you to have pstn in the receiving asterisk or should it work over iax trunks?
20:22.19tzangerAhrimanes: fax over IP is inherently tricky
20:22.22tzangerunless you use t.38
20:22.51tzangerwell t.37/t.38
20:22.59*** join/#asterisk Weezey (i=Weezey@206.210.109.229)
20:23.00Ahrimanestzanger: ok.. any semi-usable solutions with t.38 currently available?
20:23.15tzangerbut if you're IP link is rock-solid (hint: it's not) it works just fine
20:23.31SarahEmmbbiab.
20:23.31tzangernot within asterisk
20:24.06*** join/#asterisk sneak (n=sneak@64.220.234.21)
20:24.13*** join/#asterisk fulgas (n=fulgas@a81-84-116-219.cpe.netcabo.pt)
20:24.24e-HernickThere has couple thousand $ in bounty for t.38 support but nobody is taking it
20:24.37e-HernickIf you want to do fax handling with *, don't count on t.38
20:24.51Ahrimanestzanger: well i'm 4 hops from the other *
20:24.51e-HernickYou could run * to do store-and-forward though. It fully receives the faxes and then resends them
20:24.58nmscleratzanger: for my own edumacation, what's your config setup?  What's, say, the "process" for an end-user to get a tif to the destination machine
20:25.10WeezeyIs it possible to set up a 7900 phone without the need for DHCP and TFTP?  It's nice and all, but I want to travel with it.
20:25.24Ahrimanestzanger: but does it require anything at the pstn terminating asterisk?
20:25.32nmscleratzanger: or have I just not read the wiki thouroughly enough?
20:25.40*** join/#asterisk Mw3 (i=mw3@national.t-error.hu)
20:26.14tzangernmsclera: I believe my solution's already on voip-info
20:26.27tzangerAhrimanes: no, I take a PRI and the fax DID I send to rxfax
20:26.39tzangerrxfax receives, converts to PDF and scp's to my server
20:27.14tzangerI am 1 IAX2 hop from my PRI on a dedicated VOIP link
20:27.32tzangeractually I don't go to rxfax now, we have two real fax machines we send and receive from
20:27.58Ahrimanesok, i'll play with it next week then, should be able to get something working
20:28.05*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
20:28.10parylwhen running asterisk with full debugging, i see "Unable to create lock file: No such file or directory" when voicemailmain is called... what 'file or directory' is it referring to?
20:28.11*** join/#asterisk kippi (n=kippi@cpc4-hatf3-6-0-cust33.lutn.cable.ntl.com)
20:28.23kippihey
20:28.24e-Hernickparyl, the lock file.
20:28.34tzangere-Hernick: hahah
20:28.49tzangerparyl: I'd check app_voicemail.c and see what it's trying to do
20:28.52tzangeroffhand I don't know
20:29.16e-Hernickparyl, you say it works when you're not debugging ?
20:29.25kippiis there way to see if ztdummy is loaded?
20:29.44parylyeah, voicemail works... i just happened to notice it when debugging something else
20:29.46*** join/#asterisk darkskiez (n=darkskie@host-84-9-237-137.bulldogdsl.com)
20:30.00Ahrimaneskippi: lsmod ?
20:30.06*** join/#asterisk pbd (n=plancomm@12.144.118.37)
20:30.07e-Hernickparyl, so everything works fine but you get a warning message ?
20:30.22pbdGreetings, all.
20:30.32e-HernickGreetings, pbd.
20:30.36parylright... though i'm not sure if you'd call that a warning or an error
20:30.56*** join/#asterisk zotz (n=zotz@24.231.36.100)
20:31.07e-Hernickparyl, well, you say that voicemail works fine. does anything fail to work ?
20:31.36parylnot so far, afaik
20:31.37kippiif I have a TE100P card should i need to load ztdummy?
20:31.47parylbut it's obviously a concern
20:31.48Ariel_I would not worry about all the warnings and notices There not an error.
20:31.55Ahrimaneskippi: dummy is if you dont have hardware
20:32.04Ariel_kippi, no
20:32.18kippiif I don't have a line going into it, could this effect it?
20:32.29e-Hernickthe only warning you want to look out for is "Unable to create lock file: Preparing to restore directory tree / to clean state"
20:32.42e-HernickIf you get it, you must exit asterisk within 10 seconds
20:32.43pbdkippi: No, the timing is done on PCI bus interrupts only.  Connection is not necessary.
20:33.17kippiok, one last question, is there away to make sure the card is installed probley?
20:33.35e-HernickWell, if the card works with asterisk, then you know it's installed properly.
20:34.17kippihow can I tell if the card is working with asterisk?
20:34.36e-HernickCan you get a line running into the card ?
20:35.04*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
20:35.05kippinot for another few days, and my voicemail isn't working, and i think it could be because of the card
20:35.08parylsomething else i'm noticing... if i hang up while the voicemail greeting is played, it records a blank message.  it's like it's not detecting the hangup event (tdm400p card) in time
20:35.12syzygyBSDhas anyone here tried to port a cell phone number to a CLEC?  I am not sure how the number would get routed to the new carrier
20:35.33e-HernickHow would a disconnected card break your voicemail ?
20:35.49e-HernickIt don't make no sense.
20:36.21sgorillaok it makes sense
20:36.31e-HernickHOW?
20:36.39e-Hernickspeak, gorilla man !
20:36.40kippibecause of the timings, if the card isn't working then I need to get it working so the timing is working
20:36.53sgorilladon't make no (double negative) is positive
20:37.11e-HernickI've been watching firefly again.
20:37.13sgorillayou debug that thread?
20:37.22e-HernickIt's affected my language.
20:37.36e-Hernickkippi, what you need is to do is plug your card into either a live line or an ATA
20:37.52sgorillakippi: you are saying the zaptel card is broken
20:37.58sgorillabut its using the kernel module for timing
20:37.59e-Hernickkippi, lest you do that, there can be no hope for proper debugging.
20:38.02sgorillasense its broken
20:38.05sgorillait messes stuff up
20:38.10sgorillais this correct?
20:38.12pbdKippi:  You should be able to read the dmesg log for any problems with the Zaptel card.
20:38.19cripitommmmm zaptel and voicemail related?
20:38.22pbdOr do a cat /proc/interrupts, and see it listed there.
20:38.22e-HernickHe's not saying anything, I'm not sure he knows what's going on
20:38.23sgorillayou should put gdb on it
20:38.31sgorillathat is true
20:38.37*** join/#asterisk clive- (n=pirch@ndn-165-130-127.telkomadsl.co.za)
20:38.48e-Hernickkippi, you could just disable the zaptel card and see if it fixes your VM
20:38.57sgorillaset a breakpoint where it starts recording the messages
20:38.57e-Hernickkippi, take it out of the box if you want to be sure
20:39.00cripitoor just bring ztdummy
20:39.06sgorillae-Hernick: thats what i would do
20:39.11sgorillapossibly easiest fix
20:39.37kippiif I install ztdummy will that effect it latter when I put a line into the card?
20:39.50sgorillahaha
20:39.55sgorillayes it probably will
20:39.55e-HernickIf you got a zaptel card and you don't got no line for it, you gotta ask yourself what you got it for. Plug it out, man, till you got a line. That's the only way.
20:39.59sgorillaif you are having problems
20:40.05sgorillajust physically take out the card
20:40.12sgorillaand reinstall asterisk
20:40.15sgorillago from there
20:40.20sgorillaonce you get voicemail working
20:40.24sgorillainstall the card, see if it works
20:40.32sgorillathen compile the module, load it, see if it works
20:40.38sgorillastart from least complex to more complex
20:40.40kippiok
20:40.47e-HernickYou can also recompile your kernel while offering the sacrifice of a blue-eyed kitten to the God of Digital Telephony.
20:41.44kippiok cool
20:41.48kippithanks for that
20:42.07e-HernickHey, do we have an IAX2 conference channel
20:42.10e-Hernicklike #openpbx does
20:43.23*** join/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net)
20:43.34BentleyI recently saw a post that suggested HEAD had a native mp3 player for MOH (ie: no mpg123/madplay required).  I can't find any evidence of this .. is it true?
20:44.11e-HernickIt depends on what your definition of "truth" is.
20:44.42Bentleywell, opposite of false
20:44.56e-Hernick"
20:44.56e-HernickYou may also use the format_mp3 module available within the asterisk-addons package. Simply download asterisk-addons and do a make; make install of /usr/src/asterisk-addons/format_mp3."
20:45.03e-Hernickhttp://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf
20:45.33Bentleyahh, thx
20:46.11jarrodwhats a good commercial softswitch to use
20:46.24e-HernickAsterisk Business Edition
20:46.30jarrodbesides asterisk
20:46.43e-HernickOpenPBX
20:46.49FuriousGeorgeincludes will catch dialpatterns in the included context so long as you put it before the catchall patter (1NXXNXXXXXX)
20:46.51jarrodnot free
20:47.12jarrodfor a provider platform
20:47.21FuriousGeorgeright?
20:47.35e-HernickMy new fork of *, which I call "Expensive Commercial PBX Championship Edition"
20:47.51e-HernickWhat about SER ?
20:48.23sgorilla"The Truly Open Source PBX" TM
20:48.29kippido many ISP/VoIP companys use asterisk?
20:48.31FuriousGeorgeiow if i "include=> local" then i have "exten => _1NXXNXXXXXX,1,Dial(Zap/2/$EXTEN)" patterns in the local context will get caught before the next line
20:48.42e-Hernickkippi, one hundred billion
20:48.43jarrodkippi: not the very successful ones
20:48.54kippihow comes?
20:49.05sgorillacool offshot of asterisk
20:49.12sgorillai never liked the asterisk license
20:49.15kippiis there a better system to use to build voip server?
20:49.15GXTisometimes companies just need something from scratch
20:49.17e-HernickThe very successful ones use a combination of * and SER
20:49.28GXTibut in most cases asterisk is more than sufficient
20:49.40kippianyone got a link for SER?
20:49.42sgorillai looked into contributing some patches to asterisk
20:49.51*** part/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net)
20:49.52sgorillabut you had to jump through all these hoops to do it
20:49.57e-HernickYes, * is very good, and combined with SER it's even better. But for smaller-scale setups, there is little need for SER.
20:50.13kippiah
20:50.15GXTisgorilla: could be worse
20:50.21cripitoand u have to be real real big to put sip in from an asterisk server
20:50.23GXTisgorilla: apple just throws out patches and redoes them from scratch
20:50.59NetSkiere-Hernick: Where is the tradeoff point, where adding SER makes things work better?
20:51.17jarrodasterisk does not perform well on the ip centrex provider platform
20:51.37e-HernickWell, the tradeoff point is when r > 0.42, where r is the SER-requisiteness factor of your * installation.
20:51.37jarrodwhere things like FOP is the only presence utility available
20:51.52jarrodand scalability/redundancy is key
20:52.01cripito:) that depends ;) expecially if u have u own tool that can replace FOP
20:52.11jarrodyes cripito :)
20:52.33*** join/#asterisk clint_ (n=clint@snap.helixsystems.com)
20:52.58cripitoi had my own tool for manage the asterisk.. and an asterisk system managing 320 sip phones... without big issues in that part
20:53.13cripitoso i am thinking that the rule of ser not always apply
20:53.25cripitoat least u need more than a super huge system to need the ser
20:53.35rocketI am new to asterisk so I hope this isnt a stupid question .. but is there a way to test an incomming call to the asterisk server from the asterisk server itself .. but not call an internal extension .. ie I want to test my autoattendant scripts
20:54.45jarrodjust create an extension that dumps to that context
20:54.45e-Hernickrocket, what about you install a softphone ?
20:55.00dstructanyone here doing a pure meetme server?
20:55.31rockete-Hernick: I did .. and I can dial ext 200 for example .. maybe I am just confusing myself .. :/
20:55.47rockete-Hernick: but how do i get the autoattendant to answer?
20:56.04e-HernickHave you gotten the autoattendant to answer any phone at all ?
20:56.12e-HernickOr are you still trying to do that ?
20:56.29*** join/#asterisk jeremywhiting (n=jeremy@71-37-101-103.slkc.qwest.net)
20:56.39rocketnone at all .. still trying to do that .. basically I am really trying to learn it and understand it ..
20:56.47e-HernickAh, I see.
20:56.57cripitoibye the way FOP is not a bad tool anyway
20:57.01e-HernickWell, if you're starting with *, I might recommend that you learn AEL instead of extensions.conf
20:57.05e-HernickWith * 1.2.0-beta1
20:57.22cripitoand i known asteria have his own too
20:57.24e-HernickI figure you're not going to go in production very soon, and the AEL is like 100 times nicer than extensions.conf
20:58.33FuriousGeorgehttp://pastebin.ca/26107  >  if a user in the long_distance context dials 19735551212 will it be caught by the include or will it go through the long_distance context first
20:58.40rockete-Hernick: ok .. I guess I was just messing with asterisk at home .. but it sounds like I need to go more low level
20:58.50e-Hernickrocket, what do you mean more low level ?
20:59.18e-Hernickrocket, asterisk is a very programmable system, and that's the strength of *.
20:59.45rocketwell I am just using Asterisk @ home at the moment as I figured everything was working with it .. so I will have to get the new sources and figure it out more manually
21:00.17harryvvhow many here have integrated a legacy pbx with asterisk pbx?
21:00.17e-Hernickrocket, the core logic of your PBX or IVR system is the extensions file. The old-style extensions.conf works with * 1.0, but it has been replaced by the much better extensions.ael in 1.2
21:00.17rocketif that makes sense ..
21:00.26e-Hernickrocket, yeah, it does. Well, extensions.conf will do, and I suppose it's a good starting point for learning AEL.
21:00.28*** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
21:00.32e-Hernickrocket, but you've got to read about extensions.conf and how to program for it
21:00.39kippiis there away you can send txt/email messages to your phone on asterisk?
21:00.41e-Hernickharryvv, I've integrated a norstar with * using sipuras
21:01.13rockete-Hernick: ok .. thanks .. I know I am missing something there so I will read more .. thanks for the pointer
21:01.51harryvve-hernick, is there any web sites on how to integrate lagacy pbxs with asterisk?
21:03.49FuriousGeorgeok pop quiz
21:03.55FuriousGeorgeeverybody ready?
21:03.58FuriousGeorgehttp://pastebin.ca/26108
21:04.05FuriousGeorgejust shout out when you know the answer
21:04.29*** join/#asterisk fanguin (n=user@p548F3EF0.dip.t-dialin.net)
21:05.28FuriousGeorgeanyone?  i thought that would be pretty basic.  i can't test because the system is in use right now
21:05.43fanguinthere is an "application: " line in callfiles to run an application. is it also possible to execute two apps?
21:08.00FuriousGeorgeblitzrage: wake up im asking you a question :)
21:09.36Ariel_FuriousGeorge, no
21:09.53*** join/#asterisk stickyhorsey (n=blah@rrcs-24-73-191-194.se.biz.rr.com)
21:09.59*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
21:10.07stickyhorseysup folks
21:10.11Ariel_It will match the ld settings first before the local one
21:10.24harryvvAre there any cartoonish diagrams how asterisk would integrate with a old legacy pbx and also asterisk pbx diagrams how it works with other machines? something simple for a customer to understand.
21:11.06cripitothere is something in the wiki... but not customer level
21:11.22FuriousGeorgeAriel_: thats kinda anoying because the long distance context has to be a catchall so i would have to prefix long distance calls, which i really dont wanna do
21:12.44cripitowhy customer love that much the parked calls :((
21:13.05*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
21:13.07*** join/#asterisk oliverqg (n=oliverqg@dsl081-096-215.den1.dsl.speakeasy.net)
21:13.18Ariel_wow strange got knocked off
21:13.30FuriousGeorgei think parked calls is designed wrong.  calls should automatically be parked when you put them on hold, and every extension should have its dedicated parking spot
21:13.37*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
21:13.46cripitothe old legacy pbx do that
21:13.52FuriousGeorgeso if you wanna "pull" a call from me just transfer to ${MYEXTEN}# or something
21:14.06sgorillakippi: yes
21:14.07stickyhorseyanyone here have time to help me figure out why i cant get app_backticks installed?
21:14.28cripitokippy lsmod | egrep ztdummy ;)
21:14.35e-Hernickthere are a great many ways to integrate with a legacy pbx
21:14.44e-Hernickthere is no "one size fits all"
21:15.16sgorillae-Hernick: how about digital legacy pbx?
21:15.26sgorilladon't you need to reverse engineer it?
21:15.38sgorillato get it to work with asterisk
21:15.38e-Hernicksgorilla, what difference does that make? You're going to interface through analog anyway.
21:15.46e-HernickI'm integrating with a digital norstar
21:16.00e-HernickThrough the ATA and the line ports
21:16.28cripitoyeap..
21:16.36oliverqghi there....  I have a SIP phone connecte to an asterisk phone.  My dial plan forwards the call to an FastAgi server which at the end dials to an IAX softphone...  in the IAX softphone I can't hear anything from SIP phone..  it's ok the other way around...  any clue?
21:17.00sgorillae-Hernick: i though to do some of the funciontlity of the pbx
21:17.19sgorillae-Hernick: like transfer lines, etc etc, like using the legacy phones functionality
21:17.28e-Hernicksgorilla, what kind of PBX
21:17.36oliverqgsorry I meant asterisk box..
21:18.27sgorillae-Hernick: nortel
21:18.37sgorillatoshiba
21:19.07sgorillahow would you reverse engineer those?
21:19.18sgorillawith an A-D card?
21:20.38cripitoi think that he connected analogic... i did that with mine any way too
21:21.01cripitoi just make a crossover T1 btw then
21:21.45sgorillawhats a crossover T1?
21:22.09cripitothe only thing that don't work soo well is the voicemail. b/c the as.IKKJSW of quest put the voice in a *XXX number
21:22.24harryvvIm asuming that a T-1 is the same cable that is a dce/dte cable hooked to a cisco router?
21:23.14FuriousGeorgemy idea is to have three types of callers:  internal, local, and long distance.  several area codes and exchanges are local, the user has no way of knowing for certain whats local and what isnt, so a prefix is out of the question.  i cant think of any way of doing this if "include=>"s dont pre-empt matching dialpatterns below them
21:23.41FuriousGeorgethere has to be a way, but i cant picture it
21:23.51e-HernickIt depends on the size of your target system
21:24.08*** join/#asterisk p0lar69 (n=kvirc@155.101.179.19)
21:24.23e-HernickAnd on a great many other factors.
21:25.00cripitohttp://www.cisco.com/en/US/products/hw/routers/ps233/products_tech_note09186a00800a3f09.shtml#topic2
21:25.03e-Hernicksgorilla, what kind of nortel are you looking at ?
21:25.15e-HernickThey make systems of pretty much every shape and size
21:25.30e-Hernicktheir 250000 extension systems are nice
21:25.42e-Hernickrequire a lot of wiring
21:25.44cripito:D that's too way big for me :P
21:26.03e-Hernickasterisk is pretty useless at these scales too
21:26.18cripitoalmost everything is uselles at these scales too
21:26.23harryvvscales?
21:26.26harryvvwhat scales
21:26.33FuriousGeorgefish scales
21:26.34*** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com)
21:26.39e-Hernicka PBX with 1/4 million phones
21:26.39cripito250K extensions
21:26.45harryvvgood grief
21:26.56harryvvthat would take banks and banks of pbxes
21:27.06e-Hernickwell
21:27.10e-Hernickthink pentagon
21:27.14harryvvyea
21:27.25cripitoi am trying to figure out how many asterisk servers u need for that :D
21:27.27FuriousGeorgeone hundred dual opterons?
21:27.27e-Hernickor, think your average big-city CO
21:27.29harryvvI suspect thay have several pbx rooms.
21:27.31FuriousGeorgedual core?
21:27.45e-HernickFuriousGeorge, nah, these systems are probably running antiquated 233mhz cpus
21:28.03cripitoyup
21:28.05FuriousGeorgee-Hernick: i was thinking in terms of Asterisk hardware requirements
21:28.17cripitobut they are totally segmented or break in modules
21:28.26sgorillae-Hernick: any nortel, i guess, not any particular model
21:28.50sgorillawhat is the most stable pbx?
21:29.09sgorillathat has hardware failover
21:29.25sgorillanot sure how you would have failover when you have a pri card that is connected
21:29.41sgorillacould you physically make a switch that goes between two asterisk boxes
21:29.51sgorillaand if one dies
21:29.59sgorillait would physically switch the connection for the pri
21:30.15FuriousGeorgelets say i have an exten => _.X,1,goto(checkforlocal,1,${EXTEN}) and nothing in that context matches, does it come back to the original context at priority 102?
21:31.20FuriousGeorge???
21:31.40cripito<--- is back to his parked calls ;)
21:31.57davidnicolpacket-switching does failover much more gracefully than circuit-switching
21:32.42e-Hernickwith circuit switching, failover usually means you have a hot spare standing by with instructions on how to connect it
21:32.54e-Hernickwhen a problem is detected, somebody onsite has the training to connect the hot spare
21:33.13*** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
21:33.21e-Hernickit's not going to automagically failover
21:33.44QbYthere was a project a while back that did queue reporting..  anyone know the name?  (shows calls, hold times, etc..)
21:33.50FuriousGeorgehow can i catch a goto(context,1,${EXTEN}) that doesnt have a matching extension
21:33.51cripitonot really
21:34.00cripitothink that u have 2 pri
21:34.07cripitothe telco will do the failover for u
21:34.14FuriousGeorgedoes it come back on priority n+100?
21:34.23e-Hernicksure, if you have multiple circuits
21:34.59cripitoif u need a failover.. then this is not too expensive
21:35.05*** join/#asterisk Jabroni (n=Jabroni@red-corp-201.143.46.226.telnor.net)
21:35.08cripito2 twin system
21:35.15cripitoso if 1 goes down
21:35.23cripitothe another start working
21:35.31cripitoeven your phones can deal with that
21:35.42cripitothe mayority of the new sip phones have a failover sip server
21:36.07cripitoat least the decent ones ;)
21:36.35Jabroniguys anyone have problems with IVR that it stops after like 2 seconds of playing, then the line hangs up
21:37.00Jabronihttp://pastebin.com/400310
21:37.03cripitocheck the priorities
21:37.06Jabronithats that it executes
21:38.28Jabroniits a remote sipura trunk, i checked the disconnect message on the sipura page, and it says voip call ended
21:39.34cripito-- Executing Hangup("SIP/797-9a7e", "") <---- i bet u this will disconnected the call for the sipura
21:39.49Jabronithe last priority asignted to the S extension is the one from NvBackgroundDetect()
21:40.16cripitonothing else from there? NVback.. is not jumping to any other?
21:40.25Jabroniyeah.. but there is no reason for a hangup
21:40.31Jabronii just have a h,1,hangup()
21:42.05Jabronii updated the pastebin with the extensions context
21:43.34*** join/#asterisk Talnakh (n=Talnakh@217.22.177.17)
21:43.50cripitowhat happen if talk?
21:44.05cripitoinstead of a dtmf?
21:44.30Jabroniuhm.. im just listening for dtmf
21:44.47Jabroniu think its something related to nvbackgrounddetect()?
21:45.04cripitosmall test... put talk, 1, <something> and see what happen
21:45.07Talnakhhi all. i have asterisk pbx with three X101p clones. one of the cards is sharing interrupt with network card. what might be the negative consequence of that?
21:46.14Jabronitesting now
21:46.32Jabronik its doing tht
21:46.36Jabroniits the talk
21:47.47Jabroniwoot
21:47.49Jabroniworked :)
21:47.54Jabronii added the t flag
21:47.57Jabronito nvbackgrounddetect
21:48.46JabroniOk.. another question... how asterisk manages the anwser detection for outgoing calls ??
21:49.27Jabroniit has to be with the indications.conf ?
21:50.06*** join/#asterisk buddah (n=djbrianc@67.110.253.128)
21:50.07cripitou mean answer machine detection? of just the answer?
21:50.17buddahanyone ever have a problem with musiconhold, that you get no sound?
21:50.18Jabronijust the anwser
21:50.29Jabronisome calls just get dropped after 3 minutes
21:50.30buddahwhen i test it, it just does start musiconhold, then stop musiconhold right away
21:50.31Jabronichecked the CDR
21:50.42cripitocheck this DIALSTATUS
21:50.52Jabroniand the disconnect reason is that it thinks that it was not anwsered
21:51.52cripitoi told too soon :)
21:53.13Jabronithe dialstatus var is used for the dialplan right ?
21:53.20cripitoyup
21:53.51Jabronibut how would i use it ?? I mean.. on my dialout-pstn context, i have Dial(ZAP/4-1/123123123)
21:54.02Jabroniafter 3 mintures... boom.. the calls get dropped
21:54.13Jabronibut random
21:55.11cripitommm interesting
21:55.18buddahanyone that might be able to help with musiconhold?
21:55.47Jabroniyup.. ive read bout the indications.conf.... but from what ive read that for detection for the incoming calls, hangup tones
21:55.59Jabroninever seen anything releated to detecting detecting outgoing anwsered calls
21:56.16Jabroniand using digium cards.. (hate those cheaps X100p clones :p)
21:57.06Talnakhhi all. i have asterisk pbx with three X101p clones. one of the cards is sharing interrupt with network card. what might be the negative consequence of that?
21:57.09cripito:) xfo cards or tdm?
21:57.17Talnakhxfo
21:57.26JabroniTDM
21:58.02cripitowas for jabroni. i think that the issue for sharing interrupts could be for the tdm cards only
21:58.37cripitoat least i have a similar system at home without any issue ;) and is my home pbx.. and no complains from my wife or mother in law... and they talk a LOT!!
21:59.04*** join/#asterisk edonkey (n=edonkey@p549D19B2.dip0.t-ipconnect.de)
21:59.08edonkeyhello!
21:59.21cripitonice.. can we download something nice?
21:59.30edonkeyi am getting the error: misdn_cfg_get: Invalid call to misdn_cfg_get! Port number 0 is not valid.
21:59.32edonkey<PROTECTED>
21:59.34edonkeyany idea why?
22:00.49*** join/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz)
22:00.52patpatnzhi guys
22:00.52cripitojabroni so u have a pri for just testing?
22:00.54cripitonice
22:01.01cripitoi always wanted that
22:01.13patpatnzis there a way to stop * from proxying the RTP for a call going from SIP->H.323
22:01.14patpatnz?
22:01.59*** join/#asterisk |cleric| (n=dacleric@p5482B4A9.dip0.t-ipconnect.de)
22:02.18Jabroniive already set up 4 asterisk boxes :p
22:02.36Jabronibut those are issues that it have been popin' from my install....
22:02.50Jabroniafter fixing that I need to check how to get the call pickup workin'
22:03.28cripitothat should be easy.. i still wanna my own pri for just testing :)
22:03.43cripitoanyone known a asterisk solution provider in colorado?
22:03.50Jabronii tried on my bro asterisk
22:03.52Jabroniand no luck :(
22:04.11Jabroniwith the default implementation I can pickup a Zap call ringing a SIP channel on another sip channel right?
22:04.32patpatnzanyone?
22:04.52cripitoi think that callgroup and pickupgroup should be enough
22:05.02cripitoat least it work for me.
22:05.23Jabroniu added that on the sip.conf right ?
22:05.35cripito:/ i have realtime sip... but yes.
22:07.00cripitoor someone...
22:07.12hypa7iacripito: i know people who do in CA / NV
22:07.22*** part/#asterisk mkrufky (n=mk@68.160.103.77)
22:07.23Jabroniwhat kinda solution u need cripito?
22:07.43cripitonot.. i wanna make a small demo for those guys.. i am not buying...
22:07.51Jabroniooh
22:07.53cripitothe idea is sell
22:07.55Jabronisell
22:07.56Jabronihehe
22:07.56Jabroniyeah
22:08.12Jabronibtw.. for what do u use the group=1 ?
22:08.14cripitowww.cripiland.com/screenshots/manager4.jpg
22:09.27enderThe requested URL /screenshots/manager4.jpg was not found on this server.
22:10.21cripitoploading
22:10.22cripitowait
22:10.40cripitohttp://www.cripiland.com/screenshots/manager1.jpg
22:10.55*** join/#asterisk corry1000 (n=corne@ndn-165-149-251.telkomadsl.co.za)
22:12.20Jabroniok if i add callpickup=1 and callgroup=1 to all my sip clients, suppostly i can pickup call from all phones on all phones right ?
22:12.35Jabroniwith *8<extension>
22:12.49cripitohttp://www.cripiland.com/screenshots/manager4.jpg
22:13.10cripitoeven with *8 only
22:13.14endercripito: Java?
22:13.18cripitoc#
22:13.29enderoh
22:13.51cripitonot documentation yet .. :(((
22:13.57vader-wrkcripito what are you using to manage asterisk?
22:14.27*** part/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz)
22:14.28Jabronii just get a busy tone on the phone
22:14.34corry1000has anybody heart of a dss module or a snom dss module. i would really Appreciate
22:14.34corry1000<PROTECTED>
22:14.36cripitohttp://www.cripiland.com/screenshots/manager3.jpg
22:14.47cripitovader i install astmanproxy
22:14.52cripitoand build my own too
22:14.53cripitotool
22:15.13sgorillalooks good
22:15.20sgorillawhat does it use as a database backend
22:15.42cripitoas soon i finish the DAMN parked calls today.. is ready for alpha deploiment ...
22:16.00*** join/#asterisk tim27 (n=tim27@97-70.dr.cgocable.ca)
22:16.05cripitoi hate parked calls .....
22:16.18tim27any here use voctel ???
22:16.28rayvdand i hate parking at malls! >:o
22:16.47cripito:D na.. that depends on the hr...
22:16.58rayvdwell it rhymed :)
22:17.06cripitotry to park there at 11:59pm
22:17.07cripito;)
22:17.16cripitou have the entire parking lot for u :P
22:17.53*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
22:18.10sgorillacripito: you running that in house?
22:18.34cripitoi am connected to a vpn to the site that i use for testing sgorilla.
22:18.51Jabronicripito if I dial *8 should I be getting some output on the asterisk console?
22:19.19cripitowhen u make a reload extension did u make a reload res_features.so jabroni?
22:19.49cripitosgorilla the server for those screen shots is in atlanta..
22:19.52cripitoi am in colorado
22:19.58Jabronijust done it .. forgot to do it
22:19.59Jabronihehe
22:20.17sgorillacool
22:20.21sgorillawhat type of vpn?
22:20.41cripitoit sucks beleive me....
22:20.48Jabronino workie
22:20.48Jabronioh wait.. do i have to include something?
22:20.48Jabroniin the context?
22:20.59endercan * play more than a 8000mhz wav pcm file?
22:21.01*** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net)
22:21.13cripitono. jabroni.
22:21.14wundaboyi just compiled asterisk and now im having troubles!
22:21.20*** join/#asterisk Frank999 (i=frank999@ppp-70-227-85-31.dsl.sfldmi.ameritech.net)
22:21.32Jabronicuz the sip invites do are reaching asterisk
22:21.34Jabroniwith *8
22:22.15cripito:) basically it should work out of the box as soon u put the pickupgroup and callgroup
22:22.42Jabronibtw my base asterisk is an asterisk@home install
22:22.42wundaboywhat does this mean?
22:22.44wundaboyWARNING[13012]: loader.c:440 load_modules: Loading module format_au.so failed!
22:22.47iDunnoas long as it's enabled...
22:22.54iDunnoin features.conf (IIRC)
22:22.55*** join/#asterisk Katty (n=katrina@68-112-15-110.dhcp.cpgr.mo.charter.com)
22:23.28tim27any have a clue why i can receive caller id from a voctel did using asterisk@home
22:23.56cripitowundaboy.. :) no big deal add in modules.conf noload=>format_au.so ... :) i don't think to many ppl use au anyway :D
22:24.04tim27is it voctel that dont support it or it a config prob ???
22:24.49Jabronicripito what do u have in your features.conf file ?
22:24.58cripito1 sec
22:25.38wundaboycripito: roger
22:26.09cripitohttp://pastebin.ca/26116
22:26.31Jabronithats a 1.20 insatll?
22:26.33wundaboychan_features.so failed!
22:26.34Jabronior 1.0.9?
22:26.50cripito:D chan_features?
22:26.55wundaboydo i need chan_features?
22:27.02cripitoyes u need..
22:27.07wundaboyhow do i fix it
22:27.14tim27any have a clue ???
22:27.14FuriousGeorgei have about three hundred area code exchange combinations that are considered local that i have to integrate into my dialplan
22:27.17cripitoi could be mistaked but res_features is for 1.2.x
22:27.21FuriousGeorgeim not looking forward to it
22:27.32cripitobut chan_features is from 1.0.X?
22:27.34wundaboythis is how i installed asterisk (im nub to asterisk) 'emerge asterisk'
22:27.47cripitogentoo...
22:27.48wundaboycause i know you all <3 gentoo....
22:27.53tim27anyone here know if voctel support CALLER ID
22:28.20cripitoi prefer FC
22:28.22cripitosometimes
22:28.30wundaboycripito: im trying to learn asterisk, what is the easiest way to install asterisk?
22:28.42wundaboyor best
22:28.53tim27wundaboy: install asterisk@home
22:29.00Jabronii started with asterisk@home
22:29.08gordonjcpcd /home/pkgsrc/comms/asterisk && make && make install
22:29.13cripitothe easier way? have someone install tha for u :D and later explain
22:29.13wundaboywhats asterisk@home?
22:29.15Jabronibut onces you want to start doing some more complex stuff
22:29.16gordonjcpworked for me
22:29.23Jabroniu have to dump a@h
22:29.36gordonjcpwhat's "u"
22:29.36cripitoi don't like a@h
22:29.37gordonjcp?
22:29.44cripitoyou = u
22:29.55gordonjcpwhy not just type "you" then?
22:29.56tim27a@h is good to become familiar
22:30.04tim27you can check how the conf file
22:30.18tim27are modified when you make a change
22:30.21tim27in the web interface
22:30.25cripitocustom.
22:30.26wundaboyis asterisk difficult to setup?
22:30.38tim27with a@h
22:30.44tim27very easy
22:30.55tim27you grand mother can do it
22:31.08gordonjcpI doubt it
22:31.12tim27wundaboy: what kind of setup you want
22:31.12cripitoeven without the home... if u client is not doing crazy things...
22:31.15tim27soho ???
22:31.26gordonjcpI don't think my grandmother could anyway
22:31.26*** join/#asterisk jets (n=jets@guardian.pmt.org)
22:31.30wundaboyi have one phone, and hopefully it will be growing to more.
22:31.32wundaboyi want voicemail
22:31.33gordonjcpso hard to get ADSL installed in a coffin
22:31.35wundaboya menu system
22:31.36cripitoi can't wait to 1.2.x comes stable
22:31.42tim27it depend how smart is your grand mother
22:31.54tim27a@h
22:31.56tim27will fit
22:31.58wundaboywhich version of * should i get?
22:31.59tim27perfectly
22:31.59tim27this
22:32.06wundaboy1.2?
22:32.06wundaboy1.0?
22:32.27cripitoif is for learn.. i suggest 1.2
22:32.34wundaboyyes learning
22:32.44tim27TIM is wondering if VOCTEL support caller id
22:32.49wundaboywhich files do i need?
22:32.56tim27for incoming call
22:33.35tim27or anyway know a VOIP provider with CAN/US 800 DID that support caller id
22:33.48tim27anyway=anyone
22:33.52cripitois for learn? try to cvs www.asterisk.org/download  i think that the emerge of geentoo brings the 1.0.x
22:33.53tim27:)
22:34.11wundaboyso i have a random question
22:34.19wundaboyis 2.9c/minute too much?
22:34.28cripitoyes
22:34.44sgorilla2 c a minute min
22:34.56tim27anyone can help ???
22:35.09sgorillanufone
22:35.13wundaboyi need a service that will give me a 503 areacode, and never go down
22:35.21wundaboyany suggestions?
22:35.23sgorillaus toll free did
22:35.24file[laptop]never go down?
22:35.27*** join/#asterisk gambolputty (n=gambolpu@72.240.242.4)
22:35.30wundaboyrarely
22:35.33tim27sgorilla: do they have can toll free did
22:35.34cripitothere is some ppl that offer even less than 2
22:35.35file[laptop]that's better
22:35.38wundaboyand definetly never go down 8-5
22:35.38sgorilla99.99999999999% uptime
22:35.50sgorillatim27: yes
22:35.57cripitosgorilla working in nufune?
22:36.00file[laptop]gotta remember with things though that this all travels over the public internet.
22:36.01sgorillano
22:36.10tim27greg ??? working on it ???
22:36.24sgorillaworking in, on ??
22:36.24tim27what are their rate
22:36.27tim27nuphone ???
22:36.32tim27on
22:36.43sgorilladunno i think around 2 cents a minute last time i checked
22:36.54tim27but they charge for did ???
22:37.07*** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net)
22:37.07tim27someone from nufone on this channel ???
22:37.16file[laptop]stop doing that tim27... one question mark is good enough
22:38.48wunderkinheh !!!
22:39.07file[laptop]omg !!! ??? wtf
22:39.25jetslets get drunk at commit to CVS !!!
22:39.40wunderkinyey !!!oneoneoneone
22:40.45enderfile[laptop]: can * play sounf files with more than 8000mhz ?
22:40.51enderer 8000hz?
22:41.05file[laptop]not really, no... everything in asterisk is 8000
22:41.41enderok, thanks.
22:42.08*** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net)
22:42.57cripitofile mono 8000mhz right?
22:43.28file[laptop]yes
22:43.51cripitobrb
22:44.47*** join/#asterisk zotz (n=zotz@24.231.36.100)
22:45.10*** join/#asterisk Los415 (n=los415@adsl-68-121-100-114.dsl.pltn13.pacbell.net)
22:45.19enderthats what I'm converting to.
22:45.50enderI was handed files in 44.1khz AIFF, and I"m using sox to convert them down to 8khz wav
22:46.36*** join/#asterisk shidan (n=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com)
22:47.18*** join/#asterisk cripito (n=ncripito@ip67-154-143-190.z143-154-67.customer.algx.net)
22:48.38lesouvageender: maybe this link can be of help http://www.telenovela-world.com/~spade/linux/howto/MP3-CD-Burning/audio.html
22:48.55cripitotelenovela?
22:48.56cripito:D
22:49.28enderthats some pretty basic info I already knew.
22:49.40endermy files are being converted, I was just trying to make sure I had the best possible quality result.
22:49.43jpm_SDcan someone give me a quick list of possible reasons for getting a 403 Forbidden on a SIP registery?
22:49.48vader-wrkhmmm
22:49.57vader-wrki must be doing something wrong i can't get a sip client registerd
22:50.17vader-wrki setup the client in the sip.conf and i setup a context in the extensions.conf
22:50.25vader-wrki can dial out with the softphone
22:50.31vader-wrkbut i can't recieve anything
22:51.37*** join/#asterisk kippi (n=kippi@cpc4-hatf3-6-0-cust33.lutn.cable.ntl.com)
22:51.38kippi<PROTECTED>
22:51.46cripitoi think that this could help you also ender http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk
22:52.59*** part/#asterisk oliverqg (n=oliverqg@dsl081-096-215.den1.dsl.speakeasy.net)
22:53.47cripitovarder the phone is in the same net that the asterisk?
22:53.52tehdelyhello friends
22:53.55vader-wrkya
22:54.02lesouvagejmp_SD: I think the password doesn't match. (secret=blablabla)
22:54.06tehdelyhas anyone ever wanted to take their fancy zaptel hardware
22:54.10tehdelyand chuck it under a lawnmower
22:54.11cripitocheck the extension in extension.conf or ael
22:54.16vader-wrkael?
22:54.23tehdelyand shred the shitty no-good click-click-pop-click-pop FXO modules in a melee of destruction and anger
22:54.25jpm_SDlesouvage, Hrmm... ok.
22:54.25cripito1.2.0
22:54.29tehdelybecause i sure do !
22:54.38cripitoextensions.conf will be enough
22:54.49vader-wrkcriptio this is what i followed
22:54.50vader-wrkhttp://www.astmasters.net/stuff/X-Lite-and-Asterisk.pdf
22:55.15tehdelyseriously, i have had it up to here :(
22:55.24tehdelyif anyone here has successfully solved audio quality problems on a tdm400p
22:55.29tehdelythat weren't irq-related
22:55.31tehdelyplease let me know
22:55.42cripitowhat kind of  audio issues tehdely?
22:55.53tehdelycripito: click click pop click click crackle click pop click
22:56.05tehdelyclicking/crackling is so bad that half of the dtmf tones are messed up
22:56.06tehdelyconstant misdialing
22:56.14cripitotiming issues?
22:56.15tehdelyi have two tdm400p, 4 fxo each, in a dell poweredge 2850
22:56.19tehdelythey are on their own irqs entirely
22:56.21tehdelyacpi is off
22:56.23cripitoah fxo
22:56.28tehdelyzttest return sconsistent 99.98% - 100%
22:56.36tehdelybut they sound absolutely terrible.  unusably terrible
22:56.48tehdelyi already RMAed a pair from digium because they thought i had bad cards afte rhearing the crackling
22:56.51tehdelybut it is just as bad with the new ones
22:57.17cripitotiming issues?
22:57.25tehdelywhy do you reckon it's a timing issue?
22:57.34tehdelyit doesn't sound like timing issues i've heard on other transports (sip, etc.)
22:57.40tehdelyit just sounds like shitty D-A converters
22:57.42tehdelyor something else i'm overlooking
22:57.47cripitocould be too
22:58.15tehdelyi'm megabummed because i bought this server on the recommendatino of digium, since they mention in several places it is fully tested with asterisk and their hardware
22:58.23tehdelyand voip-info said it  didn't have the issues with tdm cards previous dell models had
22:58.35tehdelyand every FAQ tells me to keep tweaking things until zttest returns nominal figures... well it does :/
22:59.07cripitoall the teaks that i known are related to tdm cards
22:59.11cripitonot fxo :D
22:59.20tehdelyit is a tdm card
22:59.22tehdelytdm400p
22:59.26cripitoahhh
22:59.27cripitook
22:59.35cripito4 fxo
22:59.37tehdelyyes
23:00.11cripitoi still don't like dells for asterisk.. bad precedence....
23:00.15tehdelyso far i have moved the cards to a differnet slot to get them off of a shared irq (both of them are on their own), disabled acpi, downgraded to stable zaptel,
23:00.27cripitohow many cards?
23:00.27tehdelytwo
23:00.33cripitotry first with 1
23:00.36cripitolet see what happen?
23:00.45tehdelyhmm maybe i will
23:00.51tehdelydo you reckon there's some issue due to having two of them?
23:01.08cripitothere is anything about this around.... at least not that i known.
23:01.15cripitoi have 3 in a ibm server working ok
23:01.32cripitoeven sharing interrupts
23:01.36cripitoand magically it works
23:02.06*** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543623pcs.mainf01.in.comcast.net)
23:02.10tehdely:(
23:02.15cripitotry one first see if work
23:02.16tehdelywhat os are you running
23:02.33cripitocvs head.
23:02.40tehdelyno i mean os
23:02.45tehdelyalso i am running -stable
23:02.55tehdelyi tried -head and it was worse
23:02.55tehdelyheh
23:03.20Vcoany ideas why incoming calls from a DID via sip, ring to the phone, i answer the phone, but get a busy signal, but the calling party still gets ringing..
23:03.34Vcoand when i hangup the sip phone,  it immediatley rings again
23:03.47cripitofc 3
23:03.55tehdelycripito: i was thinking of giving fedora a shot
23:04.01tehdelyi despise red hat, and i'm using debian stable right now
23:04.05tehdelybut i know digium supports fc
23:04.17cripito:) fc is very freindly....
23:04.20Vcoslackware runs pretty reliably as well
23:04.23cripitofriendly
23:04.25tehdelyfriendly to everyone but a system administrator, cripito
23:04.31cripito:D
23:04.35tehdelythere is nothing that irks me more than buggering around with sysv init and byzantine package systems
23:04.37tehdelyoh wait, that's debian too
23:04.38tehdelylife is hell :(
23:04.55cripitou don't need 2 many things from fc to run asterisk
23:05.01cripitoso u can cap a lot the OS
23:05.15tehdelyis your box smp, btw
23:05.21n3u7tehdely I couldn't get the cvs asterisk to install on Sarge
23:05.25n3u7or Ubuntu
23:05.29n3u7or SuSE
23:05.34cripitono...
23:05.35tehdelyn3u7: what build errors were you getting
23:05.36tehdelyi can help you
23:05.47cripitoi have smp servers but with wct4xxp
23:06.24tehdelyi wish we would just get a pri and be done with this POTS nonsense
23:06.25tehdelyheh
23:06.38tehdelyif one card works and two don't, i guess i'll just shuffle the other card over into another server
23:06.42tehdelyand have the two talk IAX
23:06.49tehdelythat is such overkill though :(
23:07.02cripitothat could be a solution.... try 1 first
23:07.37cripitoespecially with 2 dell 2850 :D
23:07.44*** join/#asterisk aitnemed (i=daedalus@wsip-68-15-202-117.ok.ok.cox.net)
23:07.45lesouvagetehdely: you can try xorcom-rapid (www.xorcom.com ) It has pretty good hardware recognition and autoconfiguration. It's comletely preconfigured and ready to use after installation. At least you can check the hardware.
23:08.01cripitoi change u one for my zoekris :)))))
23:08.15tehdelylesouvage: is that a livecd by any chance
23:08.23tehdelybecause i would definitely test out a livecd distro to see if it improves things
23:08.26tehdelyi'm not very keen on reinstalling at this point
23:08.30tehdelyunless i am positive debian is the rpoblem
23:08.53cripitotehdely try 1 card first i don't beleive that the distro is the issue
23:08.54lesouvagetehdely: No it's a download of 300 mb. It installs in just a couple of minutes.
23:09.26*** join/#asterisk toddf (n=toddf@ns0.fries.net)
23:09.30tehdelylesouvage: 'install' is where i'm balking :P
23:09.48tehdelyof course i would have tried 30 distros already if it weren't for my boss and his 'project manager' lackey breathing down my neck and trying to force my hand
23:09.55tehdelythey made me use debian based on the results of a google search ....
23:10.09*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
23:10.11tehdelydid i mention this is an enterprise pbx and it needs to be running on monday? :P
23:10.34Cresl1n:-D
23:10.45tehdelyCresl1n: mine is very non-enterprise thanks to the two tdm400ps in it
23:11.02filemeep
23:11.02tehdelyand the constant crackling and noise
23:11.31cripitotehdely PM and boss always do that
23:11.41cripitobaby steps make the work.. don't you think?
23:11.47nmscleraGetting a 423 from one of our sip providers, anyone know a workaround?
23:11.48tehdelycripito: yes
23:11.56lesouvageTehdely: Give it a shut, its preconfigured with sip extensions and iax extension and it has a not so fancy ooking but very fucntional menu to diagnose the system and the hardware.  It may be solve your problem withing an hour and it is based on debian.
23:11.59tehdelybut they want me to document every step i take and justify every decision with a spreadsheet
23:12.09tehdelyyet every time i succeed in geting something to work it's because of 20 hours of hacking and trial+error
23:12.14tehdelyi should document my google searches and irclogs
23:12.43tehdelyThursday, October 20th:  Typed 'tdm400p crackling dell' into google
23:12.46tehdelyFound a mailing list post
23:12.49tehdelySearched frantically for replies
23:13.09tehdelyThursday, October 20th, 8 PM:  Started drinking
23:13.15cripitowelcome to the club... :)))
23:13.19tehdelyhaw
23:13.24lesouvageTehdely: start downloading, this explaining is not going to solve your problem.
23:13.31cripito:)
23:13.35cripitotry 1 card first
23:13.37tehdelywill do
23:13.40tehdelyi reckon i'll run back to work
23:13.43tehdelyand pull the damn thing out of the rack
23:13.47cripitoif this work.. u have 1/2 % of the job done
23:14.10wunderkin0.5% isn't much of an accomplishment :D
23:14.16cripitoas far you don't hit the pbx with an axe
23:14.26cripitois really something less than 50% ;)
23:14.55Vcothat is, assuming the scope of wht you are trying to accomplish isn't being changed...
23:14.59Vcoevery f'ing day..
23:15.00Jabronicripito no luck with the call pickup :( when i dial either the *8 or the command i define on the features.conf nothing happens (nothing gets displayed on the console) ive seen that sometimes it says that if you try to pickup a call and there is no call at least it should be outputing that there is no chanel to anwser, right ?
23:15.07tehdelyVco: thankfully at least that part was frozen
23:15.14tehdelythey just haven't given me any of the resources and time i need to accomplish it :)
23:15.31cripitolet me put it this way... last week manxpower and twisted help me with a pbx with 92 concurrent calls and an dual xeon that was dropping calls like hell
23:15.37cripitonow ... no drop calls
23:16.32n3u7the problem with this channel is that the little arterisk user has no hope
23:16.34cripitomust said there is nothing to pickup
23:16.40Vcoewww...and it's a compaq
23:16.50tehdelybeer goggles, man
23:16.52tehdelythought it was an xserve
23:17.01nmsclerahow does one increase the registration expiration period for an upstream sip provider?  I keep getting a 423 SIP response from our asterisk box
23:17.36nmsclerarather, ON the asterisk box for that registration
23:17.39lesouvagetehdely: when a motherboard is underpowered it's behaving strange. I once waste an evening because of this.
23:17.52tehdelylesouvage: underpowered, eh?
23:17.54cripiton3u7 sometimes gets tricky to get answer
23:17.57tehdelywhat sort of symptoms would that cause
23:18.12cripito:) in a dell 2850? from dell....?
23:18.24cripitocould be.....
23:18.31cripitobut i don't think soo...
23:18.36cripitoanyway
23:18.52Vcowtf
23:18.56cripitoif 1 cards work.. that is 1 of the things too look
23:19.04tehdelycripito: yeah, i really should've ruled that out a while ago
23:19.07cripitobut he don't known if 1 card works
23:19.09lesouvagetehdely: I mean with a power supply that doesn't supply enough power but enough to keep it running.
23:19.09Vcogetting everyone/user is busy stuff on some incomgin DIDs
23:19.18tehdelylesouvage: i doubt that's the case here
23:19.32tehdelythis server is designed to hold like 8 hard drives, 4 cpus, etc
23:19.37tehdelyi have 2 hds, 2 cpus, and 3 pci cards
23:19.40tehdelyi reckon it's doing fine for juice
23:19.43cripitoexactly
23:20.11cripitoanyway... as i said before
23:20.17cripitoi will try 1 card before
23:20.24cripitoanything else
23:20.26tehdelyyeah, that's the #1 thing i will check out
23:20.30lesouvagethedely: how is your xorcom rapid download going?
23:20.37cripito:)))
23:20.39tehdelylesouvage: it's not, because i'm not at work where i would need to download it
23:20.42tehdelybut i will try it
23:20.53tehdelyi just wish it was a livecd, instead of forcing me to erase my entire debian install
23:20.54tehdelythat is sort of a bummer
23:20.55tehdely:/
23:21.01*** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au)
23:21.39cripitoi am very happy with my fc 3
23:21.57tehdelyi have a sort of deep, primal hatred of all things red hat
23:22.03cripito:))))
23:22.04tehdelybut at this point i think i hate all linuxes equally
23:22.08tehdelyso i might give it a shot
23:22.08tehdelyheh
23:22.30alephcomtehdely: amen
23:22.36lesouvagetehdely: just plug in a spared hd. It's just for the try. If it works you can erase the current system without your fingers crossed.
23:22.52cripitoi agree
23:23.12Vcohmm....
23:23.15*** join/#asterisk escribzz (n=escribzz@71.36.229.227)
23:23.16cripitobasically u just need the 1th cd and a good internet
23:23.18cripito;)
23:23.18tehdelyyou know
23:23.24cripitoand yum the rest
23:23.25tehdelyif it turns out
23:23.29tehdelyanother distro solves the problem
23:23.41tehdelyi can smear it like a steaming turd in mr project manager's face
23:23.50tehdely"YOU HAVE TO USE DEBIAN AND ONLY DEBIAN.  I SEARCHED ON DISTROWATCH AND THEY SAY IT'S GOOD!"
23:24.07cripitoas i said....i don't think that the distro is the issue here
23:24.09cripitobut could be
23:24.14tehdelyi'm praying it is ;)
23:24.16tehdelyhehehehe
23:24.27NuggetLinux is poo.
23:24.31tehdelyNugget: Fact.
23:24.41escribzzAnyone have a SER config that is deisgned to do rtp proxy and pass the sip registration and data off to an asterisk box?
23:24.49tehdelywell, i take that back.  it's ok for the desktop
23:24.50escribzzfor nat
23:24.53n3u7i thought it would be a fun project to get this x100p to work
23:24.53tehdelybut i hate it on servers :P
23:25.06n3u7now it's like 30 hrs plus and I'm obsessed
23:25.46rayvdescribzz: i have one that forwards all things SIP to the asterisk box and does rtp proxy
23:26.12cripito:))
23:27.07escribzzRayvd: The problem I'm having is I have multiple public ips and of coarse asterisk only supports one I was going to listen on the other interface and pass the data off to asterisk. The one I made works ok but since my customers are all behind nat I have one way audio..... Do you think yours will work?
23:27.12*** join/#asterisk DougRoyer (i=doug@70-67.69-92-cpe.cableone.net)
23:27.52cripito<--- is back to his parked calls
23:27.54Kattyi have news!
23:27.56Katty!!
23:28.03cripito:D
23:28.05Kattyi've hit level 3 scramble in the 4d rubik's cube.
23:28.05n3u7oh noes:Sarge installed without sound support
23:29.39*** join/#asterisk marc324 (n=marc3234@206-248-135-84.dsl.teksavvy.com)
23:30.51lesouvagetehdely: xorcom has a configuration service that sends you a .deb. file that you can install. This can save you a lot of typing and time and will have your pbx up and running on monday.
23:30.54*** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-83-251.tu.ok.cox.net)
23:33.54wunderkini cant find a way to associate the original channel and the channel used when doing an atxfer :(  i need the variables from the original channel
23:42.41wunderkini saw importvar but i need to find out the other channel name first :P
23:44.21*** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl)
23:44.38Ariel_why
23:44.45wunderkinme?
23:45.00*** join/#asterisk autobus (n=linux@80.172.14.4)
23:45.20hardwiremutilator: happy b-day
23:45.22wunderkinwhy what
23:50.17autobusany person from spain?
23:50.44alephcomI'm trying to dial via the local channel but it is bridging the calls and cutting out astcc.agi.  Does anybody see anything wrong with this string?   Local/201@home|30|HL/n(3420000:60000:30000)
23:52.08cripitoautobus did u need someone from spain or that speak spanish?
23:52.44tehdelycripito: if it's the latter, he won't be able to tell you
23:53.01cripitoi speak boths :P
23:53.10wunderkinalephcom, your /n is in the wrong spot, it should be after @home
23:54.34*** part/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
23:54.45wunderkinariel  must like to just ask why
23:55.20alephcomwunderkin:  Hey, thanks a LOT!  I will try that.
23:56.26wunderkinhmm :/

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