00:00.11 | spootnick | Igbothom_III: change exten => _0.,1,Dial(SIP/${EXTEN}@astrasip-out) to exten => _0.,1,Dial(SIP/astrasip-out/${EXTEN}) |
00:00.19 | clyrrad | groogs, 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 |
00:00.36 | Igbothom_III | I had that earlier, didn't work, ppl suggested I change it to how it is now |
00:00.41 | Igbothom_III | I'll change it back, though |
00:01.04 | X-Rob | spootnick - you are aware that astrasip doesn't work very well? |
00:01.13 | X-Rob | in fact, they're probably the worst VoIP providers in .au? |
00:01.21 | supaigtr | Anyone know if the 411p card has better echo yet? Or better yet how to disable and use the KB1 software ecan? |
00:01.45 | spootnick | X-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 |
00:01.57 | asterisk99 | anyone using Asterisk on Gentoo ? zaptel won't load after reboot (yes, I followed install instructions) |
00:01.58 | X-Rob | spootnick - sorry, wrong person |
00:02.03 | Igbothom_III | spootnick; same result - no output on * console, engaged signal |
00:02.04 | X-Rob | was meant to be to Igbothom_III |
00:02.18 | spootnick | Igbothom_III: take that line out of the [general] context as well. put it it [astrasip-out] maybe |
00:02.31 | spootnick | Igbothom_III: ahm, hold on |
00:02.37 | Igbothom_III | X-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... |
00:02.53 | Igbothom_III | I spoke to Faktortel... |
00:02.57 | spootnick | Igbothom_III: see that [astrasip-out] thing you have in extensions.conf? |
00:03.02 | Igbothom_III | yup |
00:03.07 | spootnick | Igbothom_III: it's in the wrong place. put in in sip.conf |
00:03.09 | groogs | clyrrad: ... i thought you said you were using AMP? you should have an extensions.conf with from-internal in it |
00:03.11 | Igbothom_III | ok |
00:03.33 | spootnick | Igbothom_III: rename it to [astrasip], just to avoid confusion |
00:03.34 | groogs | if you're not using AMP then there IS no call waiting stuff defined, you have to do it all yourself |
00:03.47 | Igbothom_III | k |
00:04.18 | spootnick | Igbothom_III: then change the context of [astrasip] to, e.g.: [astrasip-out] |
00:04.26 | spootnick | sorry, context=astrasip-out |
00:04.28 | *** join/#asterisk Xbouncethis (n=egg@pcp01534754pcs.huntsv01.al.comcast.net) |
00:04.37 | spootnick | (in sip.conf) |
00:05.00 | kuku5 | groogs: that sucks - what if I park 10 calls - how will i know what is where? |
00:05.16 | spootnick | Igbothom_III: done it? |
00:05.35 | groogs | kuku5: huh? |
00:05.42 | Igbothom_III | yup |
00:05.45 | kuku5 | ( parked calls ) |
00:05.58 | Igbothom_III | exact same result |
00:05.59 | spootnick | Igbothom_III: now go back to extensions.conf, create a context [astrasip-out] and put that dial line i mentioned to you before |
00:06.04 | Igbothom_III | aha :) |
00:06.33 | groogs | am i missing like 50% of what is happening here or are people just not making sense? |
00:06.48 | spootnick | Igbothom_III: reload asterisk, and try dialing a number starting with 0 (as per your Dial line) |
00:07.14 | Igbothom_III | k |
00:07.30 | Igbothom_III | exact same result |
00:07.47 | spootnick | k, 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) |
00:07.53 | Igbothom_III | yup |
00:08.00 | X-Rob | groogs - kuku5 is speaking randomly. |
00:08.27 | groogs | ok |
00:08.54 | spootnick | Ariel_: any luck? i'll send you a pack of budweiser if that thing of having a free did work for me =) |
00:11.25 | kuku5 | groogs: 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 |
00:11.38 | spootnick | Igbothom_III: and? |
00:12.07 | Igbothom_III | getting there... conf in a few files - Xorcom Rapid distro splits 'em up into many .conf files :) |
00:12.17 | X-Rob | kuku5 - that's right. |
00:12.17 | spootnick | ah, ok |
00:12.30 | kuku5 | is there a way to fix that ? |
00:12.36 | X-Rob | no |
00:12.48 | X-Rob | the long answer is "possibly, with openpbx.org" |
00:12.55 | spootnick | Igbothom_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 |
00:12.57 | kuku5 | like extra hardware |
00:13.02 | kuku5 | sidecar or something |
00:13.32 | Igbothom_III | spootnick; 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!! |
00:14.15 | Igbothom_III | http://pastebin.ca/26014 |
00:14.23 | X-Rob | Igbothom_III - when you download asterisk, it comes with sample configuration files. |
00:14.23 | spootnick | hm, i never looked at it to write my confs. i based myself on the samples from 'make samples'. but ok, not to worry |
00:14.32 | groogs | kuku5: you mean you can see where it is with a Key system.. and asterisk is NOT a key system .. |
00:14.35 | Igbothom_III | not in this distro |
00:14.44 | X-Rob | as I said |
00:14.48 | X-Rob | when _you_ download asterisk... |
00:14.52 | groogs | kuku5: which is why when you park something, it tells you the extension its parked on |
00:14.54 | Igbothom_III | I've not got a single Linux distro here that wants to work properly without heavy massaging |
00:15.02 | spootnick | Igbothom_III: "exten => _0.,1,Dial(SIP/${EXTEN}@astrasip-out) to exten => _0.,1,Dial(SIP/astrasip-out/${EXTEN})" |
00:15.08 | spootnick | Igbothom_III: make sure you're using the second one |
00:15.16 | X-Rob | Igbothom_III - CEntOS+AMP works. Asterisk@Home is built for it. |
00:15.30 | groogs | besides, i don't see what the big deal is. people picking up random parked calls leads for an interesting day at the office |
00:15.32 | Igbothom_III | not 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 |
00:15.37 | n3u7 | something tells me that not loading these 20 or thirty app modules to get asterisk to run will cripple my ssetup |
00:15.45 | spootnick | Igbothom_III: in sip.conf, change "type=peer" to "type=friend" |
00:15.46 | X-Rob | Igbothom_III - CentOS works perfectly. |
00:15.48 | X-Rob | really. |
00:15.52 | X-Rob | It use it for all my asterisk machines. |
00:15.59 | Igbothom_III | I'd like to... |
00:16.06 | groogs | debian works fine here |
00:16.10 | Igbothom_III | Actually, I'd prefer FreeBSD |
00:16.11 | X-Rob | debians is good too. |
00:16.13 | spootnick | yeah, and so does gentoo, debian, and any other distro who calls itself a linux distro |
00:16.29 | DaPrivateer | Ok, 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) |
00:16.42 | Igbothom_III | well, want me to try compiling and pastte the legion error messages that each distro showed? :( |
00:16.48 | X-Rob | DaPrivateer - upgrade to CVS-HEAD for everything. |
00:16.49 | groogs | Igbothom_III: what are your 'issues'? you don't have required libraries to compile? thats not something thats magically fixed |
00:16.58 | groogs | unless maybe you install absolutely everything.. :p |
00:17.08 | n3u7 | lgbothom_III:in the last week I have tried ubuntu and SuSE9.3 |
00:17.10 | spootnick | Igbothom_III: it's either about old libraries installed by default, or by them not being there at all |
00:17.13 | DaPrivateer | X-Rob im running on BSD so thats probably not a good idea |
00:17.14 | DaPrivateer | hehe |
00:17.18 | spootnick | Igbothom_III: ok, got the last changes i mentioned? |
00:17.24 | n3u7 | now I'm onto sarge |
00:17.25 | Igbothom_III | and that is from distros downloaded in the last week! |
00:17.36 | n3u7 | only yhe apt get will install not the cvs |
00:17.43 | groogs | Igbothom_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 |
00:17.56 | Ariel_ | spootnick, sorry wife came home with dinner. |
00:18.06 | spootnick | Ariel_: lol, sorry man |
00:18.07 | kuku5 | groogs: 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) |
00:18.39 | groogs | kuku5: 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 |
00:18.54 | kuku5 | on what phone |
00:19.07 | spootnick | Ariel_: still got the address of that pastebin? if not, http://pastebin.ca/26012 |
00:19.14 | Ariel_ | looking at it now |
00:19.35 | spootnick | goddamn iax! that's something i never thought i would have any issues setting up |
00:19.54 | groogs | kuku5: spa-841s, analog connected to ATA's... |
00:19.57 | Ariel_ | spootnick, what is the error your getting? |
00:20.12 | kuku5 | does it "say" or does it "show on the lcd" |
00:20.12 | spootnick | Ariel_: chan_iax2.c:6629 socket_read: Rejected connect attempt from 204.13.233.114, who was trying to reach 's@' |
00:20.29 | groogs | say, as in SayNumber(71) |
00:20.38 | kuku5 | ok |
00:20.46 | spootnick | Ariel_: 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 |
00:21.00 | kuku5 | What happens when there are 10 calls all at once |
00:21.05 | groogs | kuku5: i also have FOP installed, and it shows me all the parked extensions |
00:21.08 | kuku5 | how does one put them all on hold |
00:21.18 | spootnick | Igbothom_III: so? |
00:22.28 | Ariel_ | spootnick, it's like my setup except I have my inbound type=user context name [878201000060] |
00:22.44 | groogs | kuku5: 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 |
00:22.59 | spootnick | Ariel_: um, you mean "type=user", and context what? |
00:23.37 | Ariel_ | I have two context one type=peer and and 2nd one called [878201000060] which is where I have type=user |
00:23.52 | kuku5 | hm |
00:23.58 | Ariel_ | context=from-pstn for me |
00:24.10 | Ariel_ | but your heading I had to change it to the number |
00:24.15 | alephcom | Anybody here have an experience with POE-Component-Client-Asterisk-Manager? I think I successfully built a manager driven realtime rating engine. :-) |
00:24.35 | Ariel_ | POE as power over ethernet? |
00:25.19 | kuku5 | can I purchase one POE adapter? |
00:25.21 | alephcom | portable multitasking and networking framework for Perl I'm not sure where the abreviation came from. |
00:25.31 | spootnick | Ariel_: and what's the context name where you have type=peer? |
00:25.37 | alephcom | kuku5: Sure, but I can't sell it. |
00:26.06 | Ariel_ | I have [goiax] |
00:26.22 | spootnick | Ariel_: both have context=from-pstn? |
00:26.26 | Igbothom_III | spootnick; 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... |
00:26.36 | kuku5 | alephcom: why not |
00:26.48 | spootnick | Igbothom_III: ok, now i don't know what that problem means. but now your dialplan knows what you're trying to do |
00:26.53 | Ariel_ | spootnick, I use in my setup context=from-pstn the context you send it to is up to you. |
00:27.11 | alephcom | kuku5: 'cause I don't sell them. The POE I was referring to is a networking framework for perl. |
00:27.28 | Igbothom_III | actually, sorry, that was a spurious error. |
00:27.35 | Ariel_ | 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 |
00:27.39 | kuku5 | what about your framework - im in need of a simple billing system |
00:27.47 | Igbothom_III | The changes resulted in the exact same result - no output on * console, engaged signal |
00:28.05 | spootnick | Igbothom_III: allright, shoot your current conf again in pastebin |
00:28.08 | alephcom | kuku5: I'm the author of ASTPP, I'll msg you privately. |
00:28.15 | Igbothom_III | ok |
00:28.34 | *** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com) |
00:28.38 | kuku5 | ok |
00:29.50 | clyrrad | Ariel_ 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? |
00:30.01 | spootnick | Ariel_: got it. great! it lacked the [87820100xxxxx] in iax.conf |
00:30.05 | supaigtr | Anyone know how to turn off the VPM in the 411p and use the software echo can? |
00:31.25 | X-Rob | spootnick - he obviously doesn't have the Dial* in the context his sip phone is in |
00:31.28 | X-Rob | check that |
00:31.37 | spootnick | X-Rob: that's what i'm guessing |
00:32.01 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
00:32.50 | Netgeeks | grrrr, stupid non-functional realtime |
00:33.07 | Igbothom_III | http://pastebin.ca/26015 |
00:34.00 | X-Rob | hah |
00:34.02 | X-Rob | no he doesn't |
00:34.06 | X-Rob | Igbothom_III - include => astrasip-out |
00:34.10 | X-Rob | in your default |
00:34.26 | X-Rob | duh |
00:34.31 | X-Rob | IT's a Pan. Duh. |
00:34.34 | X-Rob | *rawor* |
00:34.50 | spootnick | hold on, it's not only that |
00:34.50 | X-Rob | then extensions reload |
00:35.00 | Igbothom_III | IT is fine, * is my pain right now :) |
00:35.29 | spootnick | Igbothom_III: in sip.conf, set [astrasip] to "type=friend" |
00:35.46 | Igbothom_III | that goes in as "include => astrasip-out" in sip.conf? |
00:35.51 | spootnick | Igbothom_III: and "context=astrasip-out" |
00:36.08 | X-Rob | Igbothom_III |
00:36.17 | spootnick | Igbothom_III: AND, in extensions.conf, change "exten => _0.,1,Dial(SIP/astrasip-out/${EXTEN})" to "exten => _0.,1,Dial(SIP/astrasip/${EXTEN})" |
00:36.19 | X-Rob | your system doesn't know where to load that Dial astrasip-out |
00:36.31 | X-Rob | you need to include it in your default context |
00:36.35 | spootnick | Igbothom_III: don't bother including anything |
00:36.40 | X-Rob | context's refer to extensions.conf |
00:36.41 | Igbothom_III | k |
00:36.50 | spootnick | X-Rob: no he doesn't my default context doesn't include anything. in fact, it's empty |
00:37.06 | X-Rob | spootnick - look at his extensions.conf |
00:37.09 | spootnick | (i mean, there's a "wrong number dude" there |
00:37.20 | X-Rob | he's using a [default] context. |
00:37.37 | X-Rob | being that he hasn't put any other contexts there. |
00:37.42 | X-Rob | I'd assume that that is all he's got. |
00:37.47 | spootnick | X-Rob: i did. but for this to work (he being able to dial using astrasip), there's no need for that |
00:38.07 | spootnick | X-Rob: yeah, i understand. but in sip.conf, if the [astrasip] entry points to astrasip-out, it's fine |
00:38.09 | X-Rob | spootnick - hush. you're wrong 8) |
00:38.19 | X-Rob | spootnick - that's for _receiving_ calls |
00:38.21 | X-Rob | look at 501. |
00:38.23 | Igbothom_III | I'll wait ... :) |
00:38.32 | X-Rob | he doesn't have a 'context=' in there. That's his sip phone |
00:38.40 | X-Rob | that means it's going to load the [default] context. |
00:38.43 | spootnick | hahaha, man, wish when i was starting there was people fighting over the correct solution for my problem |
00:38.53 | X-Rob | the [default] context doesn't include the astrasip-out context. |
00:38.54 | Igbothom_III | lol |
00:38.55 | spootnick | spootnick: allright, do what X-Rob says |
00:38.59 | X-Rob | therefore, he can't dial anything starting with 0 |
00:39.00 | spootnick | ops |
00:39.05 | spootnick | Igbothom_III: allright, do what X-Rob says |
00:39.11 | *** join/#asterisk pussfeller (n=todd@12.150.129.170) |
00:39.17 | spootnick | Igbothom_III: and if it doesn't work, then do as i say |
00:39.23 | Igbothom_III | ok, but what was that again? :) |
00:39.27 | X-Rob | ig |
00:39.36 | X-Rob | underneat [default] in extensions.conf you've got a pile of include=> lines |
00:39.42 | Igbothom_III | X-Rob; can you please repost as it is mixed up in a lot of other things |
00:39.50 | Igbothom_III | yup |
00:40.00 | X-Rob | after the last of the include=> lines put |
00:40.04 | X-Rob | include => astrasip-out |
00:40.35 | X-Rob | uh |
00:40.36 | X-Rob | hang on |
00:40.41 | 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 |
00:40.48 | X-Rob | Whatever context you have the Dial(..) in. |
00:40.48 | Igbothom_III | hanging |
00:41.16 | X-Rob | Igbothom_III - I'm lookin gat 26013 |
00:41.21 | X-Rob | that says 'from-astrasip' |
00:41.24 | Igbothom_III | 26015 |
00:41.27 | spootnick | X-Rob: http://pastebin.ca/26015 |
00:41.36 | X-Rob | ok |
00:41.45 | X-Rob | yes. include => astrasip-out |
00:41.45 | spootnick | aha! that's why! |
00:41.47 | *** join/#asterisk mistral (i=mistral@jstevenson.plus.com) |
00:42.03 | X-Rob | then do an extensions reload |
00:42.18 | *** join/#asterisk Insanity5 (n=feaw@ip68-105-215-25.tl.dl.cox.net) |
00:42.21 | DaPrivateer | Does anyone know of any work on the ability to terminate a skype account on an asterisk pbx? |
00:42.26 | X-Rob | you also have to fix the dial string which is referencing astrasip-out |
00:42.29 | spootnick | Igbothom_III: actually, make that a 'reload |
00:42.36 | X-Rob | which is actually astrasip in your sip.conf |
00:42.41 | spootnick | if you changed anything in iax.conf. you'll need it |
00:42.44 | *** join/#asterisk MatsK (n=mk@99.80-202-83.nextgentel.com) |
00:42.54 | Igbothom_III | not changed iax.cong |
00:42.57 | X-Rob | Ariel_ - I agree. But he's just learning. |
00:43.07 | Ariel_ | X-Rob, yes your right. |
00:43.21 | Igbothom_III | "include => astrasip-out" in the extensions.conf [globals] section? |
00:43.25 | X-Rob | No |
00:43.27 | X-Rob | [default] |
00:43.27 | Ariel_ | if you change extensions.conf just do reload extensions |
00:43.31 | Igbothom_III | ok |
00:43.34 | Igbothom_III | changing now |
00:43.34 | spootnick | default 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 |
00:43.35 | X-Rob | your SIP phone is looking at [default] |
00:43.38 | spootnick | (dug) |
00:44.01 | X-Rob | your device can ONLY do things referenced from [default] |
00:44.09 | Ariel_ | no default does mean something anything that does not find a correct context falls there even un authorised calls |
00:44.11 | Igbothom_III | woohoo - getting further! |
00:44.33 | spootnick | Ariel_: isn't that actually what say in the [general] context of sip.conf? |
00:44.40 | X-Rob | spootnick - so. Perl and evil? |
00:44.47 | Insanity5 | Am I the only one that has had endless problems with terminatino providers? Echos, jitter, silence supression, to name a few. |
00:44.48 | spootnick | spootnick: yep |
00:44.48 | Igbothom_III | http://pastebin.ca/26016 |
00:44.58 | X-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 |
00:45.04 | spootnick | fuck sake, i'm going nuts |
00:45.08 | spootnick | X-Rob: yep |
00:45.31 | Igbothom_III | lol |
00:45.34 | Ariel_ | spootnick, in sip.conf default context can be changed but others like mgcp, h323, and iax.conf will fall throught to the default |
00:45.51 | pauldy | when you call dbput and dbget where are those values really stored? |
00:45.59 | spootnick | Ariel_: i always thought you could change it wherever... weird |
00:46.22 | Ariel_ | spootnick, try it. you will see what happens |
00:46.22 | X-Rob | Igbothom_III - you've been told about 20 times |
00:46.31 | X-Rob | change 'astrasip-out' to 'astrasip' in the dial plan |
00:46.32 | spootnick | pauldy: wherever you say it should be in res_mysql.conf |
00:46.51 | X-Rob | Dial command even |
00:47.09 | Ariel_ | Igbothom_III, you have a great sample extension setup on your system. it's located /usr/src/asterisk/configs/extensions.conf.sample |
00:47.12 | pauldy | spootnick, and if you don't have that file? |
00:47.23 | spootnick | pauldy: then create one. check voip-info.org for a sample |
00:47.23 | morale | where do voip providers get their PSTN numbers from or how do they get the routing from VoIP -> PSTN going? |
00:47.27 | Igbothom_III | Ariel_; nope, I don't |
00:47.54 | pauldy | spootnick, does * still maintain stateful info on its own? |
00:47.58 | Ariel_ | Igbothom_III, strange every normal setup does |
00:47.58 | X-Rob | Igbothom_III - you've been told about 20 times |
00:48.02 | X-Rob | change 'astrasip-out' to 'astrasip' in the dial command |
00:48.10 | X-Rob | that's 21 |
00:48.13 | pauldy | or is the db required to make it so that it does remember from one restart to the next |
00:48.13 | Igbothom_III | I read that, X-Rob and changing it now |
00:48.14 | file[laptop] | morale: either from another VoIP provider, or they have a PSTN connection |
00:48.18 | X-Rob | Now |
00:48.22 | X-Rob | you also want to drop the 0 from the start |
00:48.23 | Igbothom_III | I didn't ask to have that repeated then |
00:48.28 | spootnick | spootnick: ahm, goddamn it, that's right. res_mysql is something different. dbput and dbget work with Berkeley DB |
00:48.34 | X-Rob | which means changing ${EXTEN} to ${EXTEN:1} |
00:48.40 | spootnick | man, i'm out of here. i'm freaking |
00:48.49 | spootnick | pauldy: ahm, goddamn it, that's right. res_mysql is something different. dbput and dbget work with Berkeley DB |
00:49.01 | morale | file[laptop]: when you say a pstn connection, you mean a trunk from a pstn phone company? |
00:49.12 | Insanity5 | Anyone here use nufone? |
00:49.13 | pauldy | so somehwere there is a nice berkly style hash table that all the data is in |
00:49.15 | file[laptop] | morale: PRI, channelized T1, DS3, whatever |
00:49.25 | hcir | morale: PSTN numbers in the US are allocated by nanpa |
00:49.25 | tzanger | I use nufone |
00:49.32 | spootnick | pauldy: in /var/lib/asterisk/astdb |
00:49.39 | Igbothom_III | X-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) |
00:49.53 | tzanger | apparently NANPA is going to add a digit to area code and digit to extension |
00:49.54 | pauldy | nice thansk spootnick |
00:49.58 | tzanger | in the next 20 years |
00:50.16 | Igbothom_III | ok, changed the dialplan to "astrasip" and still fails |
00:50.17 | Ariel_ | tzanger, a digit like? |
00:50.18 | pauldy | next question can this be changed in 1.0.9 or future version of * |
00:50.22 | morale | thanks |
00:50.35 | Igbothom_III | diff error message tho - pasting now |
00:50.44 | vooduhal | Hello 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? |
00:51.02 | Ariel_ | use the h extension |
00:51.04 | hardwire | anybody use snom via poe? |
00:51.10 | vooduhal | Ariel_: Thank you. |
00:51.15 | Igbothom_III | http://pastebin.ca/26017 |
00:51.23 | Ariel_ | hardwire, I have but the 200 mainly |
00:51.50 | Insanity5 | Echo cancellation! help! I cant' bare it any more! |
00:52.08 | Insanity5 | Nufone to chicago area numbers echos a lot -- for my own voice -- and the other users can't hear it. How can I fix it? |
00:52.13 | tzanger | Ariel_: as in NXXNXXXXXX -> NXXXNXXXXXXX |
00:52.29 | tzanger | Insanity5: interesting |
00:52.30 | Ariel_ | tzanger, that sucks |
00:52.35 | tzanger | only to chicago? |
00:52.45 | vooduhal | I 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? |
00:52.48 | Insanity5 | tzanger - at least to 630 area code... I've tried 4 phone numbers |
00:52.49 | spootnick | pauldy: changed to what? |
00:52.53 | Insanity5 | tzanger - I only otherwise dial to idaho |
00:53.05 | Ariel_ | Igbothom_III, do you have setup in your sip called [astrasip] |
00:53.06 | tzanger | Insanity5: can you find any IVRs in that area I can call and test? |
00:53.06 | Insanity5 | tzanger - which sounds perfect, might I add. |
00:53.16 | Insanity5 | tzanger - how about a voice mail? Just talk to yourself |
00:53.21 | Igbothom_III | http://pastebin.ca/26015 is (basically) my config |
00:53.53 | Insanity5 | tzanger - call my fathers office - 630-887-8640. Press 0 to shut the gal up and start talking to yourself. |
00:53.58 | tzanger | Insanity5: :-) |
00:54.04 | tzanger | ok where the hell's my phone |
00:54.07 | tzanger | dammit |
00:54.14 | tzanger | you leave for a few days and the cat disappears your phone |
00:54.41 | spootnick | pauldy: 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 |
00:54.46 | Ariel_ | cat ate it |
00:54.47 | pauldy | spootnick, mysql/postgres |
00:55.01 | spootnick | pauldy: guess i answered your question before i understood it then |
00:55.04 | pauldy | thats what I'm looking for thanks spootnick |
00:55.09 | Insanity5 | tzanger - lol |
00:55.57 | hardwire | Ariel_: aby 360> |
00:56.01 | hardwire | any 360? |
00:56.07 | Igbothom_III | Ariel_; tho I ow have "include => astrasip" in default.conf |
00:56.13 | pauldy | kind 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 |
00:56.34 | Ariel_ | what is default.conf??/ |
00:56.34 | pauldy | and it isn't configurable via the web interface |
00:56.44 | pauldy | have to manualy go to each phone and dial *70 |
00:56.47 | Igbothom_III | it is a file loaded by extensions.conf |
00:56.51 | Ariel_ | Igbothom_III, you don't include a sip account |
00:56.52 | Insanity5 | tzanger - Find cat or phone yet? |
00:56.54 | Igbothom_III | effectively, it is in [default] |
00:57.07 | spootnick | Ariel_: he's using some sort of conf generator |
00:57.16 | Igbothom_III | nope, Xorcom Rapid 1.1 |
00:57.26 | Igbothom_III | I'm manually editing the conf files - no generator here |
00:57.29 | spootnick | Igbothom_III: and does it generate config files? |
00:57.29 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
00:57.33 | Igbothom_III | no |
00:57.40 | spootnick | my bad then |
00:57.49 | Igbothom_III | it has its defaults, which allow mw to call internally, the rest I'm adding manually |
00:57.59 | Ariel_ | Igbothom_III, locate extensions.conf.sample then do some reading there |
00:58.05 | *** join/#asterisk criptos (n=criptos@201.145.227.17) |
00:58.37 | Igbothom_III | for 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) |
00:58.50 | Igbothom_III | Ariel_; I said before that the file does not exist |
00:58.56 | spootnick | i could never stand AMP for what it generated. i can't imagine what this other thing does |
00:59.15 | Ariel_ | Igbothom_III, it does just not in the directory I posted |
00:59.26 | Ariel_ | rapid puts it in a different location |
00:59.31 | Insanity5 | spootnick - amp? |
00:59.39 | pauldy | spootnick, its at least headed in the right direction I think sure makes managing configs a breaze |
00:59.39 | spootnick | Insanity5: asterisk management portal |
00:59.44 | Insanity5 | ahh |
00:59.59 | pauldy | damb internet gremlins took my it |
01:00.00 | Ariel_ | spootnick, I use amp, normal setups, and others it's just a learning curve |
01:00.20 | spootnick | pauldy: 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 |
01:00.30 | Igbothom_III | the 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 |
01:00.49 | criptos | zap/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 :( |
01:00.57 | Igbothom_III | ok - so is there any issue running * (with a wcfxo card) on FreeBSD? |
01:01.00 | pauldy | true but thats how you get it into the hands of people on the other side of the bel curve |
01:01.05 | Igbothom_III | if I'm gonna download another distro... |
01:01.14 | Ariel_ | r should never be used. |
01:01.20 | Igbothom_III | or should I try Centos 4.x again? |
01:01.23 | pauldy | then true havok can be achieved |
01:01.39 | Ariel_ | Igbothom_III, CentOS |
01:01.55 | Igbothom_III | k, going there again :) |
01:02.24 | Ariel_ | rapid uses debian which if you know is a good distro as well |
01:02.35 | Igbothom_III | 4.2 worth it, or still beta? |
01:02.47 | criptos | ? |
01:02.51 | Ariel_ | Igbothom_III, I have not tried it yet. But it's not beta any more |
01:02.56 | Igbothom_III | k |
01:03.07 | Igbothom_III | so, should be fine :) |
01:03.15 | Ariel_ | besides the people on #centos are very helpful |
01:03.37 | Igbothom_III | just need the first 2 ISOs, or all? |
01:03.38 | pauldy | if your thinking of using centos for asterisk why not grab the asteriskat home package |
01:04.00 | Igbothom_III | pauldy; because I'd rather a STANDARD install, thanks to the issues I'm seeing here :) |
01:04.02 | Ariel_ | Igbothom_III, I have not tried it yet |
01:04.08 | Ariel_ | but you don't need xwindows |
01:04.18 | Igbothom_III | with 4.1, how many ISOs needed for *? |
01:04.25 | Igbothom_III | yeah, not wanting that on at all |
01:05.00 | Ariel_ | 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) |
01:05.41 | Igbothom_III | ok, downloading first 3 just in case, and will see what happens then. :) |
01:05.49 | Ariel_ | hello shido6 welcome |
01:05.52 | *** join/#asterisk wunderkin (n=wunderki@VDSL-130-13-234-137.PHNX.QWEST.NET) |
01:06.03 | Igbothom_III | Anything "interesting" needs to be done before installing * on it, and which version of * is the most stable? |
01:06.20 | Ariel_ | Igbothom_III, I think there is a network setup. Check with the guys at #centos |
01:06.27 | Igbothom_III | k |
01:06.38 | shido6 | ? |
01:06.40 | Delta34 | anybody using cisco 7960 phones? |
01:06.42 | Ariel_ | Igbothom_III, I use only for production 1.0.9.2 right now |
01:06.46 | shido6 | I have asterisk@home running as my firewall |
01:06.49 | shido6 | finally |
01:06.53 | Ariel_ | nice |
01:07.06 | Ariel_ | what did you put on it smoothwall or shorewall |
01:07.09 | Igbothom_III | Ariel_; ok, I'd not really like to use Beta in production anyway |
01:07.15 | twisted[asteria] | iptables! |
01:07.21 | Igbothom_III | m0n0wall is better than SmoothWall :) |
01:07.22 | twisted[asteria] | oh wait |
01:07.36 | Ariel_ | Igbothom_III, yes it is but it's based on fbsd |
01:07.41 | Igbothom_III | yup |
01:07.43 | Igbothom_III | :) |
01:07.47 | Ariel_ | can't really put it on a asterisk box |
01:07.48 | Igbothom_III | that's what I have here |
01:07.53 | Igbothom_III | on a Soekris net4501 |
01:07.54 | Igbothom_III | nice |
01:08.00 | twisted[asteria] | no, but you can put asterisk on an fbsd box ;) |
01:08.03 | Igbothom_III | does * work on fbsd yet? |
01:08.04 | twisted[asteria] | just without hardware |
01:08.10 | Igbothom_III | aha :( |
01:08.23 | kuku5 | Delta34: i am |
01:08.49 | Ariel_ | twisted[asteria], how are you doing tonight? |
01:09.02 | twisted[asteria] | Ariel_, pretty good... about to leave the office in the next 10 minutes or so |
01:09.10 | Ariel_ | nice |
01:09.20 | Ariel_ | putting in a late night (normal) |
01:09.28 | Delta34 | kuku5: what does your latency times show for sip show peers |
01:09.39 | Delta34 | do u qualify your phones? |
01:09.48 | twisted[asteria] | Ariel_, heh.. i'm usually here pretty long as of lately |
01:09.56 | Ariel_ | Delta34, what is your problem? |
01:10.32 | Delta34 | all my cisco phones show up as 70ms for 7.4 code, on 7.5 cisco code they show up as 140ms |
01:10.33 | n3u7 | well I finally got asterisk running on Sarge with an x100p card |
01:10.39 | n3u7 | *ghetto |
01:10.43 | Delta34 | this is all internal lan |
01:10.51 | n3u7 | can anyone help me set up voicemail? |
01:10.52 | Delta34 | for xten clients they show up at 8ms |
01:10.54 | kuku5 | i msg u |
01:11.06 | kuku5 | 70-140 ms |
01:11.10 | Ariel_ | Delta34, well then go back then send an email to cisco about it. |
01:11.12 | twisted[asteria] | n3u7, www.voip-info.org/wiki-Asterisk |
01:11.19 | twisted[asteria] | n3u7, or, voicemail.conf.sample |
01:11.29 | Igbothom_III | ok, 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 :) |
01:11.34 | shido6 | i made a script using iptables |
01:11.36 | n3u7 | thanks twisted |
01:11.36 | shido6 | no shorewall |
01:11.39 | vooduhal | Ok, 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? |
01:11.40 | shido6 | none of that crap |
01:11.55 | twisted[asteria] | vooduhal, if it's zap, just turn on faxdetect |
01:12.00 | Ariel_ | shido6, nice do you want to post the script? |
01:12.23 | *** part/#asterisk criptos (n=criptos@201.145.227.17) |
01:12.29 | twisted[asteria] | vooduhal, otherwise, google for nvfaxdetect |
01:12.35 | vooduhal | I 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. |
01:12.40 | vooduhal | K. |
01:12.46 | Ariel_ | wait(3) |
01:12.47 | *** join/#asterisk Administrator_ (n=sheva@host-200-94-47-83.block.alestra.net.mx) |
01:12.48 | shido6 | sure... now that I know my mouse was dying and not someone hacking into my mactel box |
01:12.51 | shido6 | :) |
01:12.58 | Ariel_ | nice |
01:13.16 | Ariel_ | shido6, thanks for the help last week. (nufone) |
01:13.39 | vooduhal | Ariel_: I thought wait discarded all audio passed on the channel? |
01:13.47 | Administrator_ | Desert Zarzamora?? |
01:14.05 | Ariel_ | ?????? |
01:14.22 | Ariel_ | vooduhal, I use wait all the time for fax detection |
01:15.06 | vooduhal | Ok. |
01:15.20 | n3u7 | wait:is there something I can do to dial into x100p to verify asterisk is working? |
01:15.50 | Ariel_ | n3u7, plug a phone line to it then dial that number |
01:17.24 | Katty | twisted[asteria]: there's spirit matter drifting about our universe |
01:17.49 | Administrator_ | I though you were another guy... |
01:17.54 | *** join/#asterisk docelm0 (n=docelm0@pool-70-110-66-127.tampfl.fios.verizon.net) |
01:17.57 | Katty | twisted[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) |
01:18.17 | docelm0 | whadup?! |
01:18.31 | Ariel_ | damm wilma is suppose to now come this way... aRgh |
01:18.50 | supaigtr | Better duck. |
01:18.55 | Administrator_ | see ya |
01:19.03 | X-Rob | fred will kick your arse if you mess with wilma. |
01:19.05 | docelm0 | this way? Ariel where r u? |
01:19.25 | Ariel_ | miami |
01:19.30 | docelm0 | TAMPA! |
01:19.48 | docelm0 | You could have probably guessed that by my hostname.. |
01:19.55 | Ariel_ | docelm0, your also in it's sights |
01:20.07 | docelm0 | BRING IT ON! |
01:20.17 | Ariel_ | sick puppy |
01:20.32 | supaigtr | Move away from shore ppl. |
01:20.43 | docelm0 | Dude I made it thru charlie.. And I was stuck in the middle of it in orlando |
01:20.52 | Ariel_ | supaigtr, haha |
01:21.14 | Ariel_ | 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) |
01:21.23 | docelm0 | How old are you? |
01:21.27 | docelm0 | True.. I wasnt here then |
01:21.45 | Ariel_ | docelm0, I might have kids your age. I am an old fart |
01:21.52 | docelm0 | Im 28 |
01:22.12 | Ariel_ | docelm0, my eldest is 22 hahaha. |
01:22.21 | docelm0 | ok maybe you are.. |
01:22.27 | Ariel_ | docelm0, but my youngest is 2 |
01:23.14 | Qorky | How 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. |
01:23.19 | n3u7 | :( |
01:23.21 | Qorky | erm. hope that made sense :) |
01:23.53 | docelm0 | Damn dude.. |
01:23.54 | Qorky | at the momment it just says asterisk is calling. |
01:24.10 | Ariel_ | setCallerID |
01:25.07 | Qorky | ;exten => s,1,setCallerID('Blah') ? |
01:25.53 | *** join/#asterisk spootnick (n=irc@CPE-144-133-126-245.nsw.bigpond.net.au) |
01:26.21 | Insanity5 | tzanger - 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) |
01:26.32 | Qorky | like thaqt ? |
01:27.23 | glomph | Looking 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. |
01:27.46 | spootnick | did anybody here ever used multiple switch statements (for realtime) in extensions.conf? (same context, like stacking extensions) |
01:28.17 | Qorky | sweet. it works :) |
01:28.19 | Ariel_ | exten => s,3,SetCIDName(Blah) |
01:28.23 | Qorky | thants Ariel_ |
01:28.44 | Ariel_ | what is vbuzzzer |
01:28.55 | glomph | vbuzzer.com SIP provider in Toronto |
01:29.13 | Ariel_ | ahh |
01:29.38 | glomph | I see postings that it works with asterisk, but not for me. (And I have set up many asterisk boxes successfully) |
01:29.40 | Ariel_ | post your settings for them on pasterbin.ca and give us your error and let us see what we can do |
01:30.09 | Ariel_ | you can do sip debug when you make a call to see what it's looking for |
01:30.47 | glomph | yeah, I think it is doing something funky with the ports. The data is not getting through. |
01:31.35 | Insanity5 | Nufone to chicago area numbers echos a lot -- for my own voice -- and the other users can't hear it. How can I fix it? |
01:31.46 | Ariel_ | try adding fromdomain= and fromuser= that sometimes helps with there using SER or BroadWorks setup |
01:32.13 | Ariel_ | Insanity5, hummmm how about other areas |
01:32.25 | Insanity5 | Ariel_ - nope |
01:32.35 | Insanity5 | Ariel_ - 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) |
01:32.49 | Ariel_ | it could be there servers |
01:32.54 | glomph | I 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) |
01:33.01 | Ariel_ | ask them support@nufone.net |
01:33.13 | Ariel_ | data-passing?? |
01:33.24 | Insanity5 | Ariel_ - I gave up on that address a while ago. Is there a way that I can locally compensate. |
01:33.25 | Insanity5 | ? |
01:33.29 | glomph | I mean the call sets up all right, but no audio |
01:33.48 | Ariel_ | Insanity5, sometimes they hang out here and others at #nufone |
01:34.02 | Ariel_ | glomph, that is another issue |
01:34.03 | *** join/#asterisk P-NuT (n=pnut_@fw.office.unitedip.net.au) |
01:34.03 | Insanity5 | Ariel_ - Is it possible to correct it myself? |
01:34.14 | Ariel_ | Insanity5, don't think so |
01:34.26 | Ariel_ | glomph, are you behind a nat/firewall? |
01:34.43 | glomph | NO |
01:35.03 | Ariel_ | do you have extenip=yourexternIP localnet=192.168.XXX.0/255.255.255.0 in sip.conf |
01:35.14 | Nugget | NAT blows goats. |
01:35.33 | Ariel_ | do you have a firewall on the asterisk box? |
01:35.44 | kuku5 | Insanity5: if no audio its a firewall issue |
01:36.04 | glomph | SIP blows goats. NAT just makes it more obvious |
01:37.17 | kb1_kanobe | however, I may have to roll back to last May, which has been solid as a rock. |
01:37.27 | X-Rob | kb1_kanobe - drumkilla did a whole pile of stuff a couple of days ago |
01:37.39 | X-Rob | roll back to when app_page was submitted |
01:37.46 | X-Rob | (eg, about a week?) |
01:37.48 | X-Rob | it's great there. |
01:37.58 | kb1_kanobe | It's wierd - there's no oops or anything. The machine just stops servicing the nics and the PRI. |
01:39.26 | Ariel_ | 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.... |
01:40.10 | glomph | start talking about linux and asterisk, she will fall asleep IMMEDIATELY |
01:40.39 | *** join/#asterisk dos000 (n=dos000@i216-58-60-251.cybersurf.com) |
01:40.57 | Ariel_ | 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) |
01:47.52 | ManxPower | ~docs |
01:47.54 | jbot | it 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 |
01:48.36 | ManxPower | no audio is usually a NAT or firwewall issue, but can also sometimes be a codec issue or a bindaddr= issue. |
01:49.00 | ManxPower | kb1_kanobe, no IRQ conflict, I assume. |
01:49.31 | *** join/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
01:49.54 | ManxPower | kb1_kanobe, I finally got a tellabs echo canceler working. |
01:50.29 | ManxPower | (yes, I was the one that many months ago sent you a message titled "Help me kb1_kenobi, you're my only hope!" |
01:50.32 | X-Rob | Ariel_ - how old is she? |
01:51.09 | Qorky | anyone 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. |
01:51.17 | Qorky | this is on a snom 360 |
01:52.57 | *** part/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net) |
01:53.23 | ManxPower | Qorky, many messages in the mailing lsit archive. |
01:53.26 | ManxPower | ~mailinglist |
01:53.28 | jbot | somebody 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) |
01:53.48 | X-Rob | Qorky - you need HINT's |
01:53.52 | spootnick | is there a way to call AbsoluteTimeout *during* a call to update it's remaining time? |
01:54.01 | X-Rob | HINT's tell asterisk which SIP/... relate to which 'number' |
01:54.31 | P-NuT | Hi all. |
01:54.40 | Delta34 | can cisco phones do the same, x-rob? |
01:54.41 | P-NuT | How do I make an offsite IAX extension? |
01:54.51 | P-NuT | can someone give me a brief rundown? |
01:55.44 | Qorky | hints eh |
01:55.50 | Qorky | i'll look into it. |
01:55.53 | Qorky | i can do it though? |
01:56.18 | Qorky | it can be done yeah ? |
01:56.18 | glomph | exten => 12345,1,Dial(IAX2/user:secret@servername/destination-number) |
01:56.27 | *** join/#asterisk mthem (i=merlintm@64.235.245.133) |
01:58.08 | X-Rob | Qorky - I'm doing in on my 360 now. |
01:58.11 | X-Rob | Use firmware 4.3 |
01:58.34 | Qorky | sweet :) |
01:58.43 | Qorky | good phone by the way. |
01:59.03 | Qorky | best 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) |
02:00.32 | tekati | Any 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 |
02:00.50 | glomph | you probably got a new kernel |
02:00.51 | Netgeeks | udev perhaps? |
02:01.06 | glomph | and the zaptel module won't load |
02:01.15 | tekati | I did get a new kernel and I recompiled the zaptel stuff. |
02:01.21 | tekati | udev? |
02:01.33 | Netgeeks | read the README.udev in zaptel source director |
02:01.39 | tekati | Thanks. |
02:01.45 | Netgeeks | it will tell you what to do |
02:01.56 | Delta34 | what 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) |
02:04.35 | tekati | Dont 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? |
02:04.36 | wunderkin | its 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) |
02:09.57 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
02:11.52 | *** join/#asterisk brent21 (n=brent21m@pcp0010945096pcs.arlngt01.va.comcast.net) |
02:12.37 | X-Rob | tekati - fedora core 1 is broken. |
02:12.41 | X-Rob | you have to use 3 at least. |
02:13.01 | X-Rob | Delta34 - it's not an application. Please, RTFW. |
02:13.15 | P-NuT | so to make an IAX extension, I make an IAX extension |
02:13.21 | P-NuT | in asterisk@home |
02:13.33 | P-NuT | and then I put exten => 12345,1,Dial(IAX2/user:secret@servername/destination-number) |
02:13.39 | P-NuT | in extensions.conf |
02:13.49 | P-NuT | and then I use this extension ofsite? |
02:13.54 | X-Rob | P-NuT - if you're using asterisk@home, you should be asking on #amportal |
02:14.10 | P-NuT | They are no help though. |
02:14.33 | X-Rob | No, especially when you don't ask. |
02:14.35 | P-NuT | All I want is an offsite extension be it SIP or IAX but I can't get any info on how to do it |
02:14.36 | X-Rob | you're not even on the channel. |
02:14.40 | brent21 | was 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? |
02:15.01 | P-NuT | I was X-Rob. |
02:15.03 | X-Rob | P-NuT - ask on #amportal. |
02:15.08 | P-NuT | ok.. *sigh* |
02:15.30 | X-Rob | Ahh there he is. |
02:15.42 | *** join/#asterisk philm (n=a@r43h15.res.gatech.edu) |
02:17.21 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
02:17.26 | kuku5 | Anyone know cpu usage for ulaw-*-ulaw ? |
02:17.43 | skyen | issue-less, me thinks. |
02:17.52 | kuku5 | as in? |
02:18.00 | skyen | low. i don't now |
02:18.18 | skyen | never even noticed it. not that i run any bigger sites, though. |
02:18.33 | skyen | is your system slow? |
02:18.35 | kuku5 | how many sim. calls? |
02:18.46 | kuku5 | no - trying to decide on hardware |
02:18.47 | skyen | 20 at max |
02:18.57 | bjohnson | if both sides use the same codec, should be damn little cpu usage |
02:19.00 | bjohnson | like a router |
02:19.24 | skyen | ah, that's usually the case here. |
02:20.31 | bjohnson | I didn't read how many calls kukucachu was expecting |
02:20.33 | Katty | someone find me an animated movie...specifically an artist's recreation of earth's gravitational field and how light is bent around it |
02:21.26 | *** join/#asterisk file (n=jcolp@mctnnbsa31w-142166095097.nb.aliant.net) |
02:21.27 | skyen | homework, hey? |
02:21.30 | Igbothom_III | or, watch more Simpsons :) |
02:21.38 | bjohnson | I've never seen such a thing |
02:21.58 | skyen | I never knew light was bent around planets |
02:22.09 | bjohnson | big mirrors |
02:22.10 | Katty | skyen: gravity bends light |
02:22.34 | Igbothom_III | Katty; absolutely |
02:22.43 | Igbothom_III | and planets bend gravity |
02:22.43 | Katty | if it's powerful enough. |
02:22.51 | Katty | it'll even eat it |
02:23.02 | Katty | sort of, anyway |
02:23.02 | bjohnson | yum |
02:23.08 | Igbothom_III | hence Black Holes |
02:23.11 | bjohnson | now that's an extreme diet |
02:23.14 | Katty | they're predicting some sort of space time bend around our planet |
02:23.22 | Igbothom_III | that emit some radiation, but consume a lot more :) |
02:23.24 | Katty | sort've like a blackhole bending everything around it's event horizon |
02:23.41 | mmlj4 | Gravity: it's not just a good idea, it's the law. |
02:23.52 | Igbothom_III | nah, gravity sucks |
02:23.54 | Katty | mmlj4: well, some things do break the law of gravity |
02:23.58 | Katty | mmlj4: dark energy. |
02:24.04 | Katty | mmlj4: it sorta holds our universe together |
02:24.06 | Igbothom_III | dark chocolate, too |
02:24.06 | bjohnson | EVIL ENERGY |
02:24.18 | mmlj4 | um, right, they just disproved dark matter |
02:24.35 | Katty | mmlj4: i've yet to read that. |
02:24.39 | bjohnson | I wish they'd disprove dark turkeu meat |
02:24.50 | bjohnson | hate that stuff |
02:24.53 | Katty | mmlj4: post url |
02:24.56 | Netgeeks | Off 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 |
02:25.10 | mmlj4 | don't have it, but it was recent... the last 2 weeks |
02:25.14 | Katty | Netgeeks: i'm well aware of what it, in theory, looks like. |
02:25.18 | bjohnson | and wrap it around a planet? |
02:25.21 | Netgeeks | Then take a small weight and drop it on the sheep |
02:25.31 | Katty | Netgeeks: what i want to see is the extend thing s are being bent around our planet |
02:25.36 | wundaboy | i just built asterisk and when i try and start it i get this: Ouch ... error while writing audio data: : Broken pipe |
02:25.41 | Katty | s/extend/extent |
02:25.44 | n3u7 | oh well |
02:25.50 | X-Rob | Netgeeks - bastard. Why are you throwing rocks at a sheep? |
02:25.54 | Katty | mmlj4: i will look for it. |
02:25.54 | n3u7 | x100p is not cooperating |
02:26.03 | skyen | X-Rob: haha, thinkin the same thing :) |
02:26.18 | Netgeeks | now, 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 |
02:26.27 | Netgeeks | the math isn't too difficult actually |
02:26.46 | Katty | math needs to draw me a pretty picture |
02:26.47 | X-Rob | Netgeeks - if a ball has zero weight, it has no inertia.How are you going to effect it? |
02:26.56 | X-Rob | affect it |
02:26.58 | Katty | X-Rob: mirrors |
02:26.59 | mmlj4 | Katty: http://rss.slashdot.org/Slashdot/slashdot/to?m=738 |
02:27.06 | Netgeeks | if you assume the mass to be a single point, the math is easy, you could plug numbers in for the mass of the earth... |
02:27.07 | Katty | X-Rob: light has no weight, yet it's moving and has mass |
02:27.15 | bjohnson | X-Rob: telepathy |
02:27.18 | X-Rob | Katty - uh. Weight == Mas. |
02:27.19 | X-Rob | s |
02:27.27 | Katty | X-Rob: no |
02:27.28 | arp2 | uhm |
02:27.28 | mmlj4 | um, weight ne mass |
02:27.30 | Katty | X-Rob: mass = energy |
02:27.32 | arp2 | weight is not mass |
02:27.34 | Netgeeks | Weight = Mass * gravity |
02:27.39 | bjohnson | no |
02:27.43 | mmlj4 | F = M * A |
02:27.46 | bjohnson | Weight = Mass * acceleration |
02:27.56 | arp2 | acceleration due to gravity |
02:28.04 | Netgeeks | *nod* arp2 |
02:28.10 | bjohnson | arp2: sometimes |
02:28.17 | X-Rob | that's what weight is |
02:28.22 | bjohnson | arp2: could be from other forces too |
02:28.34 | Katty | i'm still pondering this light event horizon thing. |
02:28.43 | X-Rob | if it has mass, it _does_ have weight. Unless you can find a 'null' part of the universe where there's no gravity from anywhere. |
02:28.47 | mmlj4 | that'll take forever |
02:28.48 | wundaboy | if my box does not have a sound card what kind of sound support do i need to have in my kernel to run asterisk? |
02:28.49 | X-Rob | which, I believe is impossible. |
02:28.56 | Netgeeks | yup, impossibel |
02:28.57 | arp2 | x-rob, thats fine |
02:28.58 | Katty | if 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 |
02:29.04 | Katty | there must be some point where the ray is ripped apart |
02:29.12 | Katty | half of it goes to the black hole, the other half keeps going |
02:29.17 | arp2 | but saying mass == weight, is just, incorrect :) |
02:29.18 | Katty | so, that would either mean a photon is rpiped in half |
02:29.18 | bjohnson | poor little light speck |
02:29.26 | X-Rob | arp2 - I was simplifing. I do apologise. |
02:29.35 | Katty | in which case both haves are somehow connected with this pilot wave theory |
02:29.43 | Katty | or the photon is stuck in place with equal forces pull on it |
02:29.56 | Katty | or it acts as a wave...the energy basis rather than a partical basis |
02:30.20 | bjohnson | I think it's just magic |
02:30.27 | Katty | in which case the light just dies on itself |
02:30.30 | *** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net) |
02:30.35 | Netgeeks | The problem katty is that a photon is both a wave and a particle at the same time |
02:30.37 | bjohnson | a big, cosmic Doug Hennesey |
02:30.43 | Katty | light doesn't have infinate energy. but what happens when a photon dies? |
02:30.53 | bjohnson | tiny funeral |
02:31.02 | Netgeeks | what do you mean by die? |
02:31.06 | Katty | Netgeeks: then what happens to it. |
02:31.19 | Katty | Netgeeks: it can't stay still |
02:31.20 | Netgeeks | when it crosses the event horizon? |
02:31.24 | Katty | it going against maxwell's law |
02:31.28 | Igbothom_III | it doesn't die but transform into a different form of energy/mass |
02:31.30 | X-Rob | as bjohnson said. There's a tiny funeral. |
02:31.42 | X-Rob | with other photon mourners |
02:31.54 | Igbothom_III | Maxwell Smart's law? Poor bugger died the other week |
02:31.56 | X-Rob | But they can never damn well figure out where the funeral is. |
02:32.03 | bjohnson | is that like .. a photn never dies, it just fades away |
02:32.05 | X-Rob | or how much the corpse weighs. |
02:32.24 | Igbothom_III | X-Rob; it doesn't weighmuch, it is very light :) |
02:32.32 | Katty | Netgeeks: 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 |
02:32.35 | bjohnson | exactly, a light particle |
02:32.37 | bjohnson | like lint |
02:32.40 | X-Rob | heh |
02:33.00 | Katty | Netgeeks: there is some point, whatever it's called, where there are equal forces pulling on that ray of light |
02:33.07 | bjohnson | blackhole = space belly button |
02:33.12 | Netgeeks | you cannot separate the wave/particle attributes of the photon.... it doesn't break apart. |
02:33.24 | bjohnson | I'm starting a new law, the belly button law |
02:33.27 | Katty | Netgeeks: it'strying to go it's normal speed, and the blackhole is apply equal negative forces. |
02:33.44 | Katty | Netgeeks: so, it cannot remain stationary according to maxwell's law |
02:33.45 | Netgeeks | The 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 |
02:33.53 | mmlj4 | armchair quantum physicists, hrmph |
02:34.03 | bjohnson | a piece of lint, travelling at the speed of lint, tries to traverse a belly button in space |
02:34.07 | bjohnson | what happens |
02:34.10 | X-Rob | Netgeeks - 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' |
02:34.15 | Katty | Netgeeks: thusly, it either acts as a particle or a wave (or, as you say, both at the same time) |
02:34.32 | Netgeeks | Ah! the gamma emmiters! |
02:34.32 | Katty | Netgeeks: when you rip apart a photon and send it seven miles apart from each other down fiberobtics, they're still linked |
02:34.58 | Katty | Netgeeks: 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 |
02:35.02 | X-Rob | Katty - that's called 'Spooky' entanglement. |
02:35.08 | X-Rob | deservedly sow. |
02:35.11 | X-Rob | so. |
02:35.20 | bjohnson | so if one goes on a date? |
02:35.24 | Netgeeks | Katty: I'm not familiar with "ripping apart photons" This probably is something that is new since I actually was a physicisy |
02:35.28 | X-Rob | bjohnson - the other one does _not_ get laid. |
02:35.30 | Katty | Netgeeks: what happens to a wave that's ripped apart? |
02:35.32 | Netgeeks | physicist, even |
02:35.48 | bjohnson | poor little photon bastard |
02:35.52 | Katty | Netgeeks: even if it /was/ stationary light, it would just...stop |
02:36.02 | Katty | Netgeeks: it wouldn't have infinate energy to stay..uhh... lit |
02:37.06 | Katty | and if it's a trackable conversion... |
02:37.15 | Katty | that means we can watch it, map it..and find the equal force horizon |
02:37.30 | Katty | that would at least let us /see/ it a little better |
02:37.30 | *** join/#asterisk erickj_az (n=erickj_a@wsip-68-98-222-74.ph.ph.cox.net) |
02:37.46 | mmlj4 | all these newfangled ideas are really strange to me... photons being ripped apart... who makes this stuff up? |
02:38.15 | Katty | or 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 |
02:38.20 | Katty | in theory, of course. |
02:38.29 | mmlj4 | the little guy just hangs at the event horizon, taking forever to make up his mind, does he fall in, or wriggle out |
02:38.32 | Netgeeks | Okay, I don't think I can help, as the ability to 'rip apart a photon' is beyond my comprehension..... |
02:38.33 | *** join/#asterisk Syncros (n=sysop@noc.routermonkey.net) |
02:38.44 | Katty | Netgeeks: moment, i'll get the experiment |
02:39.10 | Netgeeks | I finished my masters in physics in 88, and at that time, photons were not rippable apartable... hehe |
02:39.15 | mmlj4 | yes, 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) |
02:39.19 | twisted | wheeee |
02:39.40 | twisted | mv file /dev/bed |
02:39.42 | erickj_az | I 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. |
02:39.51 | twisted | er no wait |
02:39.55 | twisted | that would mean i made file into a bed |
02:40.00 | file | ooh twisted |
02:40.03 | twisted | cat file > /dev/bed |
02:40.03 | file | I didn't know you were like that! |
02:40.11 | drumkilla | w00t |
02:40.14 | twisted | sup dk |
02:40.41 | Katty | Netgeeks: http://www.cebaf.gov/news/internet/1997/spooky.html <- by rip in half i mean a pair. |
02:41.05 | Netgeeks | einsten's Spooky action at a distance problem? |
02:41.19 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
02:41.28 | Katty | the effect of a black hole on one photon /should/ have an affect on its escaped pair |
02:41.42 | twisted | turns out it just needed some air. |
02:41.49 | twisted | in the tires. |
02:41.58 | Katty | Netgeeks: or am i reading into that incorrectly? |
02:43.01 | Netgeeks | sec, reading the url you sent |
02:43.24 | Katty | Netgeeks: 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. |
02:43.45 | Katty | Netgeeks: 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 |
02:44.42 | Netgeeks | Yep, 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 |
02:45.24 | Netgeeks | The 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 |
02:45.30 | X-Rob | Personally, I think it's the FSM. |
02:45.30 | Katty | hmm, yet another interesting topic. string theories. |
02:45.40 | Netgeeks | You can't apply a problem in quantum physics to the einstein physics |
02:45.45 | X-Rob | He's touching the photons with his holy noodle. |
02:46.06 | Katty | Netgeeks: but even using quantum physics...breaking a photon into two equal pairs... |
02:46.18 | Netgeeks | so talking about how a photon reacts at an event horizon doesn't work |
02:46.20 | Katty | Netgeeks: there should be a way to watch the effects of a black hole on light |
02:46.25 | Katty | Netgeeks: why not? |
02:46.36 | Katty | Netgeeks: surely it doesn't drag the entire ray in ;) |
02:46.48 | Katty | Netgeeks: that concept is absurd. |
02:46.55 | Netgeeks | because the event horizon is a mathematical function of einstein physics |
02:47.07 | Katty | hmm. |
02:47.35 | Netgeeks | what 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) |
02:48.05 | Netgeeks | and do all your work in the quantum thoery world |
02:48.10 | *** join/#asterisk CANO-1982 (n=cano1982@201.255.51.6) |
02:48.16 | Netgeeks | and that is some very nasty math |
02:49.11 | CANO-1982 | y have a problem with my tdm400p |
02:49.26 | Netgeeks | String 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) |
02:49.26 | CANO-1982 | can y ask here? |
02:49.38 | Netgeeks | how the pair reacts the way it does |
02:50.10 | Netgeeks | a REALLY bad and basic analogy is this |
02:50.37 | CANO-1982 | I can´t make fxo module work properly |
02:50.39 | Netgeeks | if 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 |
02:51.05 | Netgeeks | now 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 |
02:51.14 | Netgeeks | now run a needle through the folded paper |
02:51.16 | CANO-1982 | have to plu and unplug the tephone cable every incoming call |
02:51.25 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
02:51.29 | Netgeeks | according to the 2d world people they see two dots |
02:51.48 | Netgeeks | when you twist or move the needs to the 2d world people both dots seem to be connected |
02:52.10 | Netgeeks | two completely separate entities to them seem to act in concert at a distance |
02:52.11 | CANO-1982 | I think dimensions is only a mathematical abtraction, only that |
02:53.00 | Netgeeks | The current evolution og string thoery i think contemplates 11 dimensions.... |
02:54.35 | Netgeeks | Quantum physics is a very disturbing field, IMHO.... |
02:55.31 | Netgeeks | You still there Katty? |
02:55.51 | Katty | Netgeeks: i made a phonecall. |
02:56.03 | blitzrage | evening all |
02:56.28 | Netgeeks | Are 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? |
02:56.40 | blitzrage | quantum physics r0x |
02:56.59 | blitzrage | Netgeeks: yes, I know that test |
02:57.11 | Katty | Netgeeks: no, i've not read that. |
02:57.14 | Katty | Netgeeks: but i shall. |
02:57.16 | blitzrage | Netgeeks: as soon as you observe it, it snaps into place, if not, then you get an interference wave |
02:57.20 | Netgeeks | basically it's this |
02:57.26 | Katty | Netgeeks: my phonecall cleared up a few things. |
02:57.53 | Netgeeks | if you shoot a single photon through a difracting slit and observe it as a wave, it creates a wave diffracting pattern |
02:58.05 | Katty | yes |
02:58.25 | Netgeeks | if you observe it as a particle, it strikes at some random location that wold be part of the diffraction pattern, but only one spot |
02:58.34 | blitzrage | Netgeeks / Katty: I *HIGHLY* recommend you get Brian Greene's, "The Fabric of the Cosmos: Space, Time, and the texture of Reality" |
02:58.51 | Netgeeks | if you shoot enough of them and observe them as particles, you recreate the diffraction wwave pattern |
02:58.58 | blitzrage | Netgeeks: actually, you can't "observe" the waveform |
02:58.59 | erickj_az | does 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 |
02:59.04 | file | Brian Greene is gooooood |
02:59.12 | erickj_az | or is that even possible? |
02:59.14 | file | erickj_az: contexts don't exist with SIP |
02:59.22 | blitzrage | erickj_az: thats only with IAX2 |
02:59.29 | file | http://www.pbs.org/wgbh/nova/elegant/ -> good to watch |
02:59.47 | erickj_az | What is the dial command for iax2, then? |
02:59.57 | Netgeeks | The 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 |
03:00.22 | Netgeeks | each and every time |
03:00.29 | blitzrage | file: great show |
03:00.47 | blitzrage | http://www.pbs.org/wgbh/nova/elegant/greene.html |
03:01.09 | Katty | Netgeeks: i think i need more base information |
03:01.17 | Katty | Netgeeks: i'm having a difficult time with that sinking in |
03:01.42 | erickj_az | I 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) |
03:01.56 | Netgeeks | Katty, 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 |
03:02.06 | Katty | Netgeeks: k |
03:02.09 | Netgeeks | you can even find it in Schroedinger's cat book |
03:02.18 | Katty | yay, schroedinger's cat! |
03:02.48 | erickj_az | Will DUNDi do it if I just put the SIP deives oin the same context on both servers? |
03:02.51 | Netgeeks | Now, 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.... |
03:03.00 | Netgeeks | this is what they did |
03:03.07 | blitzrage | that Brian Greene book I told you about is pretty much the only book you need to get started understanding the universe around us |
03:03.50 | Netgeeks | and 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 |
03:04.28 | Netgeeks | Thus, they concluded that the photons are connected somehow |
03:06.13 | blitzrage | file: I leave in 3 days |
03:06.17 | *** join/#asterisk Xbouncethis (n=egg@pcp01534754pcs.huntsv01.al.comcast.net) |
03:07.18 | *** part/#asterisk Xbouncethis (n=egg@pcp01534754pcs.huntsv01.al.comcast.net) |
03:08.01 | *** join/#asterisk dasuberdavid (n=egg@pcp01534754pcs.huntsv01.al.comcast.net) |
03:08.34 | Katty | Netgeeks: tunneling also interests me. |
03:08.35 | twisted | blitzrage, where are your photos? |
03:08.39 | *** join/#asterisk dos000_ (n=dos000@i216-58-59-142.cybersurf.com) |
03:09.02 | *** join/#asterisk websae (n=websae@CPE-24-167-204-30.wi.res.rr.com) |
03:09.29 | Netgeeks | I don't know if there is much to say about tunneling... |
03:09.55 | websae | can i use my broadvoice account to use my computer to dialup to the internet? |
03:09.57 | websae | is that possible? |
03:10.16 | Netgeeks | It's just the word given to the wierd fact that quantum entities can 'avoid' physical objects in thier way |
03:10.42 | Katty | Netgeeks: and the bigger the barrier, the quicker they seem to travel? |
03:10.47 | Katty | that doesn't really make sense either |
03:10.51 | websae | anyone have any ideas on that? |
03:11.03 | *** join/#asterisk wolfson` (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
03:11.05 | websae | using voip account to dialup to an internet connection? |
03:11.07 | *** join/#asterisk jsmith (n=jsmith@smithfam.dsl.xmission.com) |
03:11.34 | Katty | Netgeeks: has anyone tried piggybacking anything onto a photon.uhh, half? |
03:12.12 | Netgeeks | hehe, I've no clue Katty. You thinking of teleportation? |
03:12.20 | Katty | bank transactions, actually |
03:12.27 | Katty | no ecryption required |
03:12.39 | Katty | what one photon does, the other should, right? |
03:12.46 | Katty | faster than the speed of light |
03:12.47 | Netgeeks | Well, you could use photons to carry data, right? fiber optics does this |
03:13.10 | Katty | Netgeeks: would the photon break down? |
03:13.16 | Netgeeks | so you could in thoery using tunneling get data from one location to another instantly |
03:13.33 | Katty | Netgeeks: invent it. |
03:13.48 | Katty | Netgeeks: the speed of light is now slow. |
03:14.07 | Netgeeks | not that I'm aware of... from what I understand of tunneling, there is only a random chance that the quantum particle/wave will tunnel |
03:14.10 | Netgeeks | not all do |
03:14.15 | Netgeeks | so you would lose data |
03:14.27 | Netgeeks | or the data arriving would be garbled |
03:15.39 | Katty | :< |
03:15.53 | Katty | we must stablize photon daughters! |
03:16.01 | Katty | by finding something else that will tunnel with it |
03:16.08 | Katty | something stable |
03:16.17 | Katty | and osmoene bind them |
03:16.20 | Katty | somehow, i mean |
03:16.32 | Netgeeks | alas, 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) |
03:16.44 | pauldy | katty just use more than one and ask at&t if you can use their crc method and boom |
03:16.44 | Katty | :< |
03:17.20 | Katty | i still think there's a way to make space travel much faster usin gnothing but weak magnetic fields. |
03:17.45 | Katty | like 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) |
03:17.48 | saint_ | hi all |
03:18.00 | pauldy | Katty I thinkthe idea that light travels at differnt speeds through differnt mediums means ftl travel is possible to me at least |
03:18.10 | Netgeeks | I 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 |
03:18.12 | saint_ | I have a super question...: Is it possible to use a cell phone for an outgoing trunk ? |
03:18.29 | Katty | Netgeeks: hmm. |
03:18.34 | Katty | Netgeeks: query. |
03:18.47 | Katty | Netgeeks: has there been any advancements to space folding? |
03:18.58 | Katty | Netgeeks: like how they think the photons halves are just one, but on different planes |
03:19.33 | Katty | hop the needle for a ride, so to speak |
03:19.39 | Netgeeks | um, two thoeries address that, the expanding decision thoery and string thoery |
03:19.41 | *** join/#asterisk mcadory (n=mcadory@208.149.64.28) |
03:19.52 | Katty | meh, string theory still confuses me |
03:19.58 | kb1_kanobe | saint_: yes, though I beleive you need GSM hardware. I recall it's quite popular in europe. |
03:20.10 | saint_ | well... |
03:20.29 | Netgeeks | the 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 |
03:20.34 | X-Rob | saint_ - there's some hardware that lets you dock your mobile phone and it turns it into a FXS port. |
03:20.34 | saint_ | 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... |
03:20.39 | X-Rob | you can then get an FXO card and use that. |
03:20.50 | saint_ | In the USA it is all fucked up with the FCC ... |
03:21.03 | Netgeeks | string thoery is understood by less than maybe 50 people in the world.... I'm not one of them by a long shot |
03:21.19 | saint_ | 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 .. |
03:21.23 | Katty | Netgeeks: every action has an equal and oposite reaction |
03:21.33 | X-Rob | saint_ http://www.cellsocket.com/index.htm |
03:21.35 | Netgeeks | Thats newtonian physics |
03:21.39 | Katty | Netgeeks: for ever action in this universe, there is an equal and opposite universe to counteract it |
03:21.42 | saint_ | 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 ? |
03:21.52 | X-Rob | saint_ - HELLO? |
03:21.55 | X-Rob | *kick kick* |
03:22.00 | X-Rob | ARE YOU PAYING ATTENTION? |
03:22.06 | kb1_kanobe | saint_: 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. |
03:22.09 | X-Rob | http://www.cellsocket.com/index.htm |
03:22.13 | Netgeeks | Katty: ahh, hrm, is that another thoery out there? |
03:22.33 | X-Rob | bugger |
03:22.34 | X-Rob | it's broken |
03:22.40 | saint_ | cellsocket is not working .. |
03:22.42 | X-Rob | their website is broken, but they do exist. |
03:22.50 | Netgeeks | althogh that probably falls into the first alternate thoery I mentioned |
03:22.51 | Katty | Netgeeks: that would no doubt be a seperate existance |
03:22.56 | Netgeeks | wish I could remember it's name |
03:22.57 | X-Rob | That's what they do. You dock your mobile, and it give syou an FSX |
03:23.00 | Katty | Netgeeks: unless you're getting int some sort of anti-particle thingy |
03:23.17 | Katty | Netgeeks: for every particle this is an anti-particle pair |
03:23.28 | X-Rob | saint_ - google for 'em. |
03:23.37 | Katty | i love how our universe just likes creating things out of nowhere for us! |
03:23.39 | Katty | fun fun. |
03:24.01 | Netgeeks | Oh, 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 |
03:24.20 | Katty | Netgeeks: i could create fire, and the anti-fire. |
03:24.27 | Netgeeks | So you could create a photon, if you also created an anti-photon |
03:24.27 | Katty | Netgeeks: or move and anti-move an object |
03:25.08 | Katty | back to string theory |
03:25.16 | Katty | from what i understand there are two types of strings |
03:25.17 | Katty | open and closed |
03:25.36 | Katty | closed exist in our dimension because they're trapped...having both ends here.....sort like on a guitar. |
03:25.45 | Katty | where as closed particles are more of a loop...they have no beginning or end |
03:25.49 | Netgeeks | But 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 |
03:25.53 | Katty | so they can wander around and not bump into thing |
03:26.34 | Katty | Netgeeks: yes but the universe spews out particles and anti-particles at times. |
03:26.43 | Katty | Netgeeks: very briefly, anyway |
03:27.15 | Katty | Netgeeks: i would take that as something accidently got dumped into our dimension |
03:27.21 | Katty | Netgeeks: and to balance it out, something had to give |
03:27.38 | Katty | Netgeeks: which then forces it back to where it was...originally |
03:28.32 | Netgeeks | *speechless* |
03:29.11 | Katty | i guess that's something like a white hole |
03:29.24 | Katty | i don't know of anything else that would be a theoretical leak |
03:29.52 | Netgeeks | well, I have to head out now. Nice thoerizing with you |
03:29.58 | Katty | kthxbi |
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) |
03:36.31 | websae | curious |
03:36.44 | websae | anyone know about dialing an ISP over your VoIP connection with ATA adapter? |
03:36.46 | websae | is that possible? |
03:37.24 | kb1_kanobe | you mean PC-Modem-ATA-(voip)-ITSP-ISP ? |
03:37.57 | jsmith | Modem over VoIP won't work :-) |
03:38.08 | jsmith | (Well, maybe at 9600 baud, if your latency is *VERY* low) |
03:38.27 | kb1_kanobe | agreed. Only if there is no jitter and the bandwidth of the modem is less than the bandwidth of the codec (eg. fax over ulaw) |
03:39.20 | Nugget | even then, it's a crapshoot. :) |
03:39.54 | syle2 | why isn;t this working errrrr |
03:40.01 | Qorky | has the syntax changed ? |
03:40.01 | Qorky | asterisk*CLI> capi info |
03:40.01 | Qorky | Contr1: 2 B channels total, 2 B channels free. |
03:40.01 | Qorky | Contr2: 2 B channels total, 2 B channels free. |
03:40.07 | Qorky | i have 4 channels free |
03:40.14 | syle2 | if (mysql_real_query(&mysql, sqlcmd2, strlen(sqlcmd2))) { } |
03:40.14 | Qorky | <PROTECTED> |
03:40.17 | syle2 | else { |
03:40.21 | Qorky | but i cant dial anymore :( |
03:40.30 | Qorky | i just updated to the latest cvs-head |
03:40.44 | syle2 | int nrows; MYSQL_RES *result; result = mysql_store_result(&mysql); nrows = mysql_num_rows(result); |
03:40.55 | syle2 | pissin me off |
03:41.07 | syle2 | if(nrows == 0) { always returns true |
03:41.32 | syle2 | i see the query is getting called |
03:41.49 | syle2 | just 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) |
03:52.06 | websae | well 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) |
03:53.30 | glm2k | websae: you already have internet, just take out the Vo from the IP |
03:54.00 | Nugget | just get AOL. |
03:54.14 | dasuberdavid | lol |
03:54.16 | glm2k | hehe |
03:55.22 | Deedubb | laugh |
03:55.39 | Deedubb | ... I was bored, sorry |
03:57.11 | Qorky | <PROTECTED> |
03:57.21 | Qorky | any know a bit about it? |
03:58.50 | blitzrage | I use AOL. Its awesome. |
03:59.09 | file[laptop] | blitzrage: never say that again |
03:59.19 | websae | well 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-??? |
03:59.21 | blitzrage | file[laptop]: what... I'm serious. I just switched over a couple of days ago |
03:59.55 | file[laptop] | I think my ISP has all my e-mail on a time delay |
04:00.01 | ManxPower | websae, no |
04:00.04 | file[laptop] | and why am I responding to an e-mail bkw sent me... |
04:00.13 | ManxPower | You are trying to do IPoV |
04:00.22 | websae | yes |
04:00.30 | blitzrage | isn't modem PPP ? |
04:00.34 | blitzrage | but either way :) |
04:00.50 | *** join/#asterisk Insanity5 (n=feaw@ip68-105-215-25.tl.dl.cox.net) |
04:01.00 | Insanity5 | I 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) |
04:05.02 | yxa | if i scripted * to call 2 external numbers using 2 fxo, is there a way to bridge the 2 calls? |
04:05.58 | blitzrage | blind xfer? |
04:06.29 | jsmith | yxa: Yes, you could do it through call files or the manager interface. |
04:07.33 | file[laptop] | uh oh it's jsmith |
04:08.42 | file[laptop] | eep |
04:08.57 | yxa | jsmith i understand the call files part but what abt the ami part? |
04:09.20 | jsmith | You should be able to do the same thing over AMI |
04:09.45 | yxa | jsmith oh i want it totally scripted |
04:10.57 | jsmith | OK, gotta get some sleep -- long day tomorrow. |
04:11.41 | *** join/#asterisk glomph (n=black@c-24-18-145-249.hsd1.wa.comcast.net) |
04:12.30 | glomph | Asterisk rules, check out the story about the Gumstix-server. TINY |
04:12.41 | glomph | http://www.linuxdevices.com/news/NS7497964764.html |
04:13.01 | blitzrage | glomph: dude... that's SOOOOOO last week :) |
04:13.44 | glomph | Ouch! Lots of venomous eels in here |
04:14.17 | blitzrage | glomph: I like Windows as a desktop OS :) |
04:14.30 | glomph | I like Windows as a doorstop OS |
04:14.49 | blitzrage | glomph: 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 :) |
04:15.16 | glomph | Wish I could have made the journey |
04:15.17 | dasuberdavid | is kristian in here? |
04:17.37 | *** join/#asterisk damned (n=vpol@damned.vpol.org.ru) |
04:17.39 | Qorky | bah. i cant get it to work |
04:17.44 | tehdely | ass turd dicks |
04:17.51 | Qorky | No 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) |
04:19.18 | spootnick | has anybody ever used multiple switch (realtime) statements in the same dialplan? can't seem to be able to make second one work... |
04:19.34 | blitzrage | dasuberdavid: nah, he doesn't use IM/IRC |
04:19.43 | blitzrage | spootnick: I don't think you can |
04:19.56 | blitzrage | spootnick: but not positive |
04:20.03 | blitzrage | (having never used realtime) |
04:20.33 | spootnick | quote from voip-info.org. "And YES! You can have multiple switches and multiple family names using this method. " |
04:20.43 | spootnick | referring to having multiple switches |
04:21.50 | spootnick | i'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 |
04:22.03 | spootnick | but once there, if i try to |
04:22.15 | spootnick | but 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) |
04:39.53 | Insanity5 | Damit --- 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. |
04:40.06 | Insanity5 | :( |
04:40.15 | X-Rob | hah |
04:40.25 | X-Rob | you got 'em for $0? |
04:40.28 | Insanity5 | ya |
04:40.31 | X-Rob | sweet. |
04:40.46 | Insanity5 | They were free after mail in rebate last week (vonage units), and an unlock hack was on dslreports.com so I went for it. |
04:41.30 | Insanity5 | They'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. |
04:41.54 | blitzrage | Insanity5: wonder if that was the unlock thing that JunK-Y made |
04:42.09 | FuriousGeorge | blitzrage: how was astricon |
04:42.10 | Insanity5 | blitzrage - what is this? |
04:42.22 | blitzrage | FuriousGeorge: pretty damn good! |
04:42.27 | blitzrage | FuriousGeorge: and San Fran was even better |
04:42.33 | blitzrage | Insanity5: thats all I know :) |
04:42.44 | FuriousGeorge | blitzrage: did you end up using IPCOP for that thing |
04:42.56 | blitzrage | FuriousGeorge: nah... m0n0wall is 100 times better :) |
04:43.07 | FuriousGeorge | neverheard of it |
04:43.10 | FuriousGeorge | heard of smoothwall |
04:43.11 | blitzrage | FuriousGeorge: really?! |
04:43.13 | Insanity5 | What's a good 10 minute QOS hack? |
04:43.19 | Insanity5 | for a linux router? |
04:43.20 | blitzrage | FuriousGeorge: crazy... www.m0n0.ch |
04:43.28 | Insanity5 | so peer to peer doesn't kill downloads |
04:43.30 | blitzrage | Insanity5: www.m0n0.ch |
04:43.53 | Insanity5 | blitzrage - see http://www.dslreports.com/forum/remark,14450684?hilite=pap2+unlock |
04:43.55 | blitzrage | Insanity5: www.m0n0.ch <-- yep, go to the traffic shaper wizard, and click the appropriate check boxes |
04:44.07 | Insanity5 | blitzrage - But I can't use my current linux distro with that, can I? |
04:44.09 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
04:44.19 | blitzrage | Insanity5: what do you mean? Its a router/FW (runs off of CDrom) |
04:44.32 | Insanity5 | blitzrage - Well, I might want to run a web server or other goodies. |
04:44.34 | blitzrage | Insanity5: it sits outside of whatever you're using |
04:44.39 | blitzrage | Insanity5: so use another server |
04:44.40 | Insanity5 | on the same box |
04:44.41 | FuriousGeorge | i think call parking can be done better |
04:44.58 | Insanity5 | It's for residential use, for me and 2 roomies. |
04:45.21 | blitzrage | Insanity5: then good luck... m0n0wall was the only thing that didn't require a month of time to learn iptables / tcng |
04:45.26 | FuriousGeorge | i 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 |
04:45.43 | blitzrage | FuriousGeorge: then you want to talk to oej -- he made such a thing |
04:45.48 | FuriousGeorge | or just put a caller on hold, and someone can pull it from you |
04:45.57 | FuriousGeorge | oej? |
04:46.09 | blitzrage | Olle E. Johansson (the guy who does the SIP channel) |
04:46.27 | blitzrage | FuriousGeorge: oej at edvina dot net |
04:47.47 | FuriousGeorge | i think im gonna go make a request |
04:48.02 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
04:55.09 | FuriousGeorge | this would probably be pretty easy to implement. if i park a call it automatically goes to ${EXTEN}# (or something like that) |
04:55.11 | Qorky | hmm since i loaded now head and capi. i get this. |
04:55.11 | Qorky | Contr2: 2 B channels total, 2 B channels free. |
04:55.11 | Qorky | <PROTECTED> |
04:55.11 | Qorky | <PROTECTED> |
04:55.11 | Qorky | <PROTECTED> |
04:55.12 | Qorky | <PROTECTED> |
04:55.14 | Qorky | <PROTECTED> |
04:55.17 | Qorky | when just tryig to make a call. |
04:55.30 | Qorky | == ISDN1: CAPI Hangingup |
04:55.30 | Qorky | <PROTECTED> |
04:55.38 | *** join/#asterisk jeffik (n=Jeff@node-423a160a.mdw.onnet.us.uu.net) |
04:55.39 | Qorky | looks like its making several calls ? |
04:55.53 | FuriousGeorge | ~pb |
04:55.54 | jbot | it 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) |
05:00.12 | Deedubb | KitKat is good |
05:02.42 | Insanity5 | Is it possible to have asterisk ring multiple pots lines, and once one answers, stop ringing the others? IE: Ring my cell and home phone? |
05:04.57 | Insanity5 | I love how my isps dns server has a reverse entry for 172.16.0.50 :( |
05:05.06 | Insanity5 | Always messes with my home computer's hostname, hehe. |
05:05.33 | Katty | Deedubb: mew? |
05:05.35 | FuriousGeorge | Insanity5: thats the default behavior |
05:06.15 | FuriousGeorge | dial(zap/cell&zap/home) |
05:07.00 | *** part/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
05:09.29 | Insanity5 | FuriousGeorge - I know. but isps shouldn't poison their dns servers with that crap :) |
05:09.59 | FuriousGeorge | i 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) |
05:11.44 | Smi|k | Does anyone here have experience with asterisk / crm / shopping cart / google api? I really want to make an all-in-one turnkey solution |
05:13.05 | Insanity5 | FuriousGeorge - Ahh, ok. |
05:13.22 | Insanity5 | FuriousGeorge - 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? |
05:14.30 | FuriousGeorge | Insanity5: by default, fxo kewlstart signaling (i guess) detects the pickup on pots. if the voice mail picks up then it answered |
05:14.54 | Insanity5 | FuriousGeorge - I wondered how it detected the difference between a ring and say, a pbx generated ring. |
05:15.21 | Insanity5 | Is there a special signal sent over the line once its picked up? |
05:16.24 | FuriousGeorge | Insanity5: actually i dont know how it works. i think the kewlstart signalling (vs. loopstart) just detects a hangup |
05:16.47 | kuku5 | Is the 7940 a good secretary phone ? |
05:17.04 | kuku5 | I'm looking at the two buttons and im worried it wont be... |
05:18.49 | Smi|k | or any idea where I'd look for that matter? |
05:19.43 | Insanity5 | Hmmm, my spa-2002 is misbehaving. Random dropped packets, ping times spiking to 2000 ms on the local subnet, no dialtone. |
05:19.46 | Insanity5 | wtf |
05:20.47 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
05:21.08 | wunderkin | hopefully this will be the last time the modem gets put to sleep or unplugged |
05:21.21 | Qorky | using the new chan_capi (0.6) |
05:21.22 | Insanity5 | ? |
05:21.36 | Qorky | how can i define multiple incomingmsn's ? |
05:22.02 | Qorky | like? 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) |
05:32.15 | FuriousGeorge | Qorky: im not sure what ur trying to say |
05:32.28 | FuriousGeorge | incomingmsn? |
05:33.01 | Qorky | do you know you way around the capi.conf ? |
05:33.02 | FuriousGeorge | i guess its an isdn thing, which i dont know about |
05:33.11 | FuriousGeorge | no sorry bro |
05:33.17 | Qorky | no worries :) |
05:33.19 | Qorky | cheers anyway |
05:36.45 | Insanity5 | Will this record natively in ulaw (so no ulaw>wav>ulaw transcode) -- I'm trying to record a greeting: |
05:36.45 | Insanity5 | <PROTECTED> |
05:39.09 | Corydon76-home | I think the ulaw suffix is .ul |
05:39.38 | Insanity5 | [Description] |
05:39.38 | Insanity5 | <PROTECTED> |
05:39.55 | Corydon76-home | Have you tried it yet? |
05:39.58 | Insanity5 | So how do I get it to record in ulaw? do filename.ul? ulaw is what is used everywhere else in asterisk for ulaw. |
05:40.54 | blitzrage | AstriCon 2005 pictures: http://leifmadsen.com/gallery/astricon_2005 |
05:41.23 | Corydon76-home | Pretty pretty blue eyes |
05:41.39 | blitzrage | lol |
05:41.52 | blitzrage | surprisingly, I didn't get hit on that much at AstriCon :D |
05:42.05 | Corydon76-home | surprisingly |
05:42.06 | blitzrage | must 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) |
05:42.24 | orlok | Anybody know a place to get cisco sip images apart from cisco? |
05:42.25 | orlok | :) |
05:42.33 | Insanity5 | orlok - google? |
05:42.53 | Insanity5 | orlok - Or buy a service contract on one for 'overseas' off cdw.com -- it's like $10 or so. |
05:43.47 | orlok | Insanity5: yeah, i found lots of people asking the same thing |
05:43.53 | orlok | cisco shit me to tears |
05:44.08 | Dr_Ray | there is an ebay cd |
05:44.10 | orlok | they ship routers that cannot function as advertised, unless you upgrade the IOS |
05:44.32 | Dr_Ray | we 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) |
05:45.09 | Insanity5 | orlok - you're supposed to pay extra for that functionality too :). |
05:45.09 | Insanity5 | hehe. |
05:45.18 | orlok | what, working hardware? |
05:45.23 | Insanity5 | Dr_Ray - does your cco account get access to router ios images too? Just courious. |
05:45.29 | orlok | this is a VPN router with a crypto card |
05:45.35 | orlok | and they reboot every 15 minutes when you se the VPN |
05:45.36 | Dr_Ray | no, I believe I only get 7960 images |
05:45.43 | wunderkin | hehe, 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 |
05:45.52 | Dr_Ray | so I doubt I could get 7970 or 7940 images |
05:45.54 | Insanity5 | We have some on some Cisco pixes, and I can get everything. |
05:46.12 | Dr_Ray | I've never looked actually |
05:46.53 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-240-180.dsl.irvnca.pacbell.net) |
05:47.04 | Dr_Ray | I've thought of mirroring the cisco images with a chinese/russian web hosting company |
05:47.19 | Insanity5 | lol. I believe it's already been done. |
05:47.21 | Insanity5 | heh |
05:47.38 | Insanity5 | Are the cisco ip phones really that great? |
05:47.59 | blitzrage | Insanity5: yes |
05:48.12 | Dr_Ray | once you get sip 7.5 on it, it's like wonderfull |
05:48.21 | Dr_Ray | it was worth the $80 |
05:48.26 | Insanity5 | I mean, isn't it just like, it's another phone? |
05:48.29 | blitzrage | I have a 7960, a Polycom IP500 and a pair of Uniden UIP200's |
05:48.37 | blitzrage | Insanity5: you don't own one do you? :) |
05:48.40 | Insanity5 | Nope |
05:48.46 | Insanity5 | Do you use it for business use? |
05:48.48 | blitzrage | Insanity5: you'll understand when you get one :) |
05:48.53 | Dr_Ray | don't get me wrong, I like the budgetone 101.. |
05:48.55 | blitzrage | Insanity5: yes... I use it for conference calls and such |
05:49.03 | blitzrage | I hate grandstream stuff :) |
05:49.12 | Dr_Ray | but for my front desk staff, the cisco was a no brainer |
05:49.15 | Insanity5 | Home/soho or coporate use? |
05:49.31 | orlok | Insanity5: 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) |
05:49.44 | wunderkin | ah i did a show features and * is hangup by default doh |
05:50.07 | blitzrage | Insanity5: home run business (my bedroom) |
05:50.15 | e-Hernick | I command bits to move and they do. |
05:50.22 | blitzrage | Insanity5: but I do a lot of conference calls, and the Cisco phone just works great for it |
05:50.45 | Insanity5 | blitzrage - Ahh -- didn't think it would be too hard to do the ccalls asterisk side with the flahs button and an ata |
05:51.04 | Insanity5 | $18 software upgrade solution: http://cgi.ebay.com/Cisco-7960-7940-SIP-7-5-Firmware-Upgrade-CD_W0QQitemZ5819481213QQcategoryZ51204QQrdZ1QQcmdZViewItem |
05:51.07 | Dr_Ray | I 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 |
05:51.20 | sgorilla | thats hella expensive |
05:51.28 | sgorilla | has anyone used the aastra phones |
05:51.29 | Dr_Ray | well, it's been hella rock solid |
05:51.44 | sgorilla | what is the most rock solid phone? |
05:51.44 | Insanity5 | Don't they have some color ones now too? |
05:51.46 | *** join/#asterisk DougRoyer (i=doug@70-67.69-92-cpe.cableone.net) |
05:51.46 | blitzrage | I got my 7960 for free by running the Asterisk Documentation Project :) |
05:51.52 | sgorilla | cool |
05:52.07 | blitzrage | sgorilla: I think the 7960 is rock solid, along with the Polycom series of phones |
05:52.14 | Dr_Ray | when 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 |
05:52.38 | sgorilla | Dr_Ray: dont you need some other piece of hardware for the phones? |
05:52.43 | sgorilla | like a callmanager |
05:52.44 | Dr_Ray | he'd seen cisco, and he knew that cisco made networking/voip |
05:52.57 | Dr_Ray | sgorilla - no you can put a sip image on it |
05:52.59 | blitzrage | call mangler? nah |
05:53.11 | e-Hernick | I have analysed the structural integrity of the 7960 phone, and let me tell you that your feeble "rocks" are unworthy of the name. |
05:53.21 | sgorilla | sgorilla: you can use asterisk and cisco phones without having to buy anything else? |
05:53.25 | Dr_Ray | not trying to start a flamewar, but I personally don't like the polycoms |
05:53.28 | Insanity5 | hmmm - check this out: http://cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0900aecd8030e546.html |
05:53.39 | sgorilla | the polycoms i heard take a long time to boot up |
05:53.42 | Dr_Ray | the 7970 does not have a sip image |
05:53.46 | sgorilla | i dont want to wait 3 minutes for a phone to boot up |
05:53.57 | Dr_Ray | it takes the cisco 30-40 seconds to reboot |
05:53.58 | e-Hernick | Dr_Ray, what alternative conferencing speakerphones do you prefer ? |
05:54.00 | sgorilla | i want somethign that is simple and have good speakers |
05:54.04 | Dr_Ray | 7960 |
05:54.22 | Dr_Ray | my dislike of polycom is purely based on looks |
05:54.25 | Dr_Ray | not actual use |
05:54.30 | sgorilla | oh |
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) |
05:55.13 | Insanity5 | Free after mail in rebate vonage unlocked adapters work for me I guess :) |
05:55.16 | e-Hernick | I mean, surely the 7960 cannot compare to the Polycom SoundStation ? |
05:55.35 | e-Hernick | Ah, I see that you are a superficial person. |
05:55.37 | wunderkin | if 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 |
05:55.41 | sgorilla | Insanity5: can you hook up directly to vonage using sip |
05:55.46 | sgorilla | or do you use an fxo card to the router? |
05:55.57 | e-Hernick | He's using unlocked adapters. |
05:56.08 | sgorilla | whats a good simple phone for a business? |
05:56.12 | sgorilla | that is cheap |
05:56.14 | e-Hernick | He's using illegal hacking software to disable their coupling to Vonage. |
05:56.17 | Insanity5 | sgorilla - fxo, but I unlocked them. |
05:56.26 | Insanity5 | I disagree with illegal hacking software. |
05:56.34 | sgorilla | sgorilla: how did you unlock it? |
05:56.35 | e-Hernick | sgorilla, what about a Sipura and a good analog phone ? |
05:56.37 | Insanity5 | I bought the hardware, I'm entitled to it. Doctrine of first use. |
05:56.49 | sgorilla | Insanity5: but you could of violated the DMCA |
05:56.57 | Insanity5 | sgorilla - that is about it. |
05:57.04 | e-Hernick | I like the Plantronics CT12. |
05:57.06 | Insanity5 | sgorilla - But i'm using the software supplied with the phone. |
05:57.13 | e-Hernick | But it doesn't like WiFi. |
05:57.40 | sgorilla | Insanity5: its a phone that came with vonage? |
05:58.01 | e-Hernick | Though for around 200$ you can get a nice corded phone, a Plantronics CT12 and a SPA-2002 |
05:58.01 | Insanity5 | sgorilla - No, a phone I bought at staples that said vonage ready on the box. |
05:58.09 | Insanity5 | sgorilla - With no contract or service plan required. |
05:58.36 | Insanity5 | sgorilla - see http://www.dslreports.com/forum/remark,14450684?hilite=pap2+unlock |
05:58.38 | sgorilla | oh |
05:58.43 | sgorilla | cool, so they lost money on the phone |
05:58.50 | sgorilla | the probllem is there is no QoS |
05:59.05 | Insanity5 | sgorilla - Best handled at the router, but it does suck. |
05:59.11 | Insanity5 | The adapter marks the packets though. |
05:59.47 | sgorilla | Insanity5: what type of QoS do you do on the router? |
05:59.49 | e-Hernick | We 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. |
06:00.40 | Insanity5 | sgorilla - you ahve two choices... by port, or by the ToS flag on the IP packets hwich the spa-2000 can flag. |
06:01.06 | sgorilla | i am looking to do it by port |
06:01.10 | sgorilla | flagging the ToS bit |
06:01.14 | e-Hernick | Who here would dare defy America by illegally perverting a Vonage-ready adapter for use with open source software ? |
06:01.18 | sgorilla | then using iptables to mangle it |
06:01.28 | sgorilla | well i guess its already mangled |
06:01.36 | sgorilla | just need to figure out how to use it with tc |
06:01.42 | e-Hernick | Why would you mangle a packed that's already been marked as high-QoS ? |
06:01.49 | sgorilla | it seems that *bsd QoS is better |
06:01.57 | e-Hernick | htb3 rules |
06:02.01 | sgorilla | yeah i retracted my statement |
06:02.04 | sgorilla | whats htb3? |
06:02.07 | Insanity5 | You just have to set the router to give it higher priority. |
06:02.22 | e-Hernick | Okay, sgorilla, have you ever configured Linux QoS ? |
06:02.28 | sgorilla | yes |
06:02.32 | e-Hernick | sgorilla, you complain about tc and you don't even know about htb3 ? |
06:02.35 | sgorilla | but very inexperienced |
06:02.51 | sgorilla | i dont complain, just from reading over different mailling lists |
06:02.56 | sgorilla | pf vs tc |
06:02.57 | e-Hernick | Well then, I'll tell you what: Linux QoS is a pain to learn, but it's very powerful. |
06:03.09 | e-Hernick | pf certainly sounds rather easier to manage. |
06:03.11 | sgorilla | e-Hernick: i have ntop set up so I can see traffic flows |
06:03.19 | sgorilla | yeah it is easier, easier syntax |
06:03.31 | e-Hernick | I'll give you that, anything has easier syntax than tc |
06:03.46 | sgorilla | i am looking for a system to do bgp routing |
06:03.52 | sgorilla | and failover firewall/router |
06:04.01 | sgorilla | without going the cisco router because of cost |
06:04.04 | e-Hernick | zebra ? |
06:04.12 | sgorilla | yeah i was looking at that |
06:04.16 | e-Hernick | Wait, I think it has a new name. |
06:04.18 | sgorilla | actually its not zebra anymore |
06:04.26 | sgorilla | its quagga(sic) or something like that |
06:04.39 | e-Hernick | Right |
06:04.47 | sgorilla | openbsd has this really nice redundant routing protocol |
06:04.58 | e-Hernick | Well, BSD might be a good choice. |
06:05.00 | sgorilla | where you can put a crossover cable on two routers, and do a ha setup |
06:05.09 | sgorilla | i am much more familiar with linux though |
06:05.19 | e-Hernick | Yeah, same here. |
06:05.22 | Insanity5 | bgp with anything but cisco... buy a damn $200 cisco off ebay if you have the money to buy bgp capaable links. |
06:05.27 | e-Hernick | I've had an OpenBSD box for a while |
06:05.31 | e-Hernick | I didn't like it so much. |
06:05.40 | Insanity5 | 2600 3620 are good starts. |
06:05.59 | e-Hernick | A used cisco. That's a thought. |
06:06.03 | sgorilla | i am probably going to get some EOL cisco box to start playing around with |
06:06.07 | sgorilla | yeah definetly |
06:06.18 | Insanity5 | but you probably dont' need bgp :) |
06:06.20 | e-Hernick | Tell you what - you need redundancy ? Get both a cisco and a Linux box. |
06:06.22 | Insanity5 | hehe |
06:06.28 | sgorilla | i need to do vlan |
06:06.34 | sgorilla | why not bgp? |
06:06.35 | Insanity5 | vlan is layer 2 |
06:06.41 | e-Hernick | Why BGP ? |
06:06.42 | sgorilla | have two multihomed fiber connections |
06:06.45 | Insanity5 | cisco routers do routing ... layer 3 |
06:06.51 | e-Hernick | What kind of connections ? |
06:06.55 | sgorilla | cisco routers do layer 2 also |
06:06.58 | sgorilla | 100 mb |
06:07.02 | Insanity5 | sgorilla - And... you can't afford a bgp capable router but are buying big fiber pipes? |
06:07.06 | e-Hernick | Nice |
06:07.13 | sgorilla | haha i am not buying the pipes =) |
06:07.14 | Insanity5 | usually they don't come in 100 megabit unless they're fiber ethernet handoffs. |
06:07.27 | sgorilla | they are fiber ethernet handoffs |
06:07.29 | Insanity5 | Most of the time the only thing that comes in a multiple of 100 is ethernet. |
06:07.32 | e-Hernick | Yeah, you get two 100mbit pipes |
06:07.34 | sgorilla | 100 mb port |
06:07.47 | e-Hernick | Do you even have the right to run BGP on these pipes ? |
06:07.50 | Insanity5 | And very few people hand off the fiber at 100 megabit anymore unless distance is a concern. |
06:07.57 | e-Hernick | I mean, are you gonna get registration with your upstream routers ? |
06:08.08 | sgorilla | yeah |
06:08.10 | Insanity5 | What isp? |
06:08.12 | Insanity5 | cogentco? |
06:08.16 | sgorilla | time warner telecom |
06:08.17 | Insanity5 | metro area network? |
06:08.22 | Insanity5 | Is it in a data center? |
06:08.30 | e-Hernick | Is it in a university ? |
06:08.33 | sgorilla | newly built small datacenter |
06:08.35 | Insanity5 | Are you sure it's fiber? |
06:08.39 | sgorilla | yes |
06:08.40 | Insanity5 | Most likely fast ethernet handoff. |
06:08.45 | sgorilla | well starting out 20mb |
06:08.50 | Insanity5 | Well, that's pretty stupid. |
06:08.54 | e-Hernick | Yeah, most likely they have fiber coming in the builiding and you get ethernet. |
06:08.57 | Insanity5 | Not many people use fiber for data centers. |
06:09.00 | sgorilla | with pri |
06:09.06 | sgorilla | what do they use? |
06:09.23 | e-Hernick | So, you have two 100mbps links coming in your small datacenter, from two different providers ? |
06:09.25 | Insanity5 | there used to be a time when it was the only way to get gigabit speeds, but even now gigabit ethernet handoffs are reaidly available. |
06:09.45 | Insanity5 | e-Hernick - I kind of doubt they're terminated with fiber into his cage. |
06:09.50 | e-Hernick | So, are you trying to get equipment to route the whole datacenter ? |
06:09.51 | sgorilla | right now there is just 1 20 mb |
06:09.53 | Insanity5 | Someone else is probably doing that for him and handing them ethernet. |
06:10.01 | sgorilla | i am thinking about the future |
06:10.07 | sgorilla | it needs to be physicall diverse |
06:10.09 | e-Hernick | sgorilla, are you getting a mere datacenter cage or are you renting a unprepared small room |
06:10.23 | sgorilla | renting unprepared small room |
06:10.24 | e-Hernick | I mean, here in quebec city, most datacenters are actually in small rooms in the basement of the big hotels |
06:10.51 | sgorilla | what is good price for 20 mbit ethernet handoff per month |
06:10.55 | e-Hernick | Those hotels are full of datacenters and most of the fiber of the city comes there, and most of the big antennas are too |
06:11.05 | sgorilla | ok telecom hotels |
06:11.06 | Insanity5 | sgorilla - I seirosuly doubt you are getting fiber handoff from an ISP. |
06:11.08 | e-Hernick | sgorilla, depends where you are, depends what kind of QoS, if you get oversold.. |
06:11.14 | e-Hernick | sgorilla, what provider |
06:11.17 | sgorilla | Insanity5: why? |
06:11.18 | Insanity5 | sgorilla - You are getting ethernet. You probably won't have fiber unless it was an ATM link. |
06:11.20 | sgorilla | time warner telecom |
06:11.20 | e-Hernick | I mean, cogentco is cheap and dirty |
06:11.25 | Insanity5 | sgorilla - there is no good reason. |
06:11.25 | e-Hernick | time warner |
06:11.29 | e-Hernick | are they reselling from somebody |
06:11.30 | sgorilla | fiber runs on the telephone pole nearby |
06:11.50 | e-Hernick | sgorilla, and you're getting two fiber runs or just one ? |
06:11.58 | sgorilla | one starting out |
06:12.04 | e-Hernick | so you have just one link |
06:12.07 | Insanity5 | sgorilla - 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 |
06:12.09 | sgorilla | run another to make it physically diverse |
06:12.10 | e-Hernick | how are you gonna get redundancy |
06:12.32 | e-Hernick | Okay, so let's recap the situation. |
06:12.35 | sgorilla | not going to be redundancy at first |
06:12.48 | sgorilla | it has fiber on the pole next to it |
06:12.57 | sgorilla | going to a cisco 5650 |
06:13.01 | sgorilla | well not sure the number |
06:13.06 | e-Hernick | You'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. |
06:13.12 | Insanity5 | I think he's a little confused :) |
06:13.14 | sgorilla | its like an old cisco that handles fiber, with a ethernet handoff |
06:13.18 | Insanity5 | And he's buying non-existant routers. |
06:13.20 | e-Hernick | sgorilla, is that accurate ? |
06:13.35 | e-Hernick | He's probably getting the 5650 from time warner |
06:13.37 | sgorilla | tier-1 bandwidth i believe |
06:13.39 | sgorilla | where im at |
06:13.46 | sgorilla | yeah thats coming from time warner |
06:13.47 | Insanity5 | time warner is cogentco is crap. |
06:13.48 | e-Hernick | sgorilla, okay, is the rest accurate |
06:13.58 | *** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com) |
06:13.59 | Insanity5 | but, it's cheap b/w |
06:14.00 | e-Hernick | sgorilla, do you care much about traffic shaping ? |
06:14.01 | sgorilla | oh timewarner uses cogent? |
06:14.06 | sgorilla | yes i do |
06:14.08 | Insanity5 | nope, but it's only one step above them. |
06:14.09 | kuku5 | hm |
06:14.13 | e-Hernick | sgorilla, older ciscos are not the best choice |
06:14.15 | sgorilla | what is the best ? |
06:14.18 | Insanity5 | He shouldn't need to shape traffic in a data center. |
06:14.32 | kuku5 | well he has only 20mb |
06:14.32 | Insanity5 | Somethings wrong with your network allocations if this is the case. |
06:14.37 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
06:14.39 | sgorilla | why wouldnt you shape traffic? |
06:14.51 | e-Hernick | sgorilla, tell you what. Build two routers for your datacenter. One uses gentoo and one uses freebsd. You put them in a failover setup. |
06:15.10 | e-Hernick | That forces you to configure two routers identically. Then, you choose whichever router works best as primary, and the other one as failover. |
06:15.21 | e-Hernick | You get to use both pf and tc. That's a win/win situation. |
06:15.28 | sgorilla | you wouldnt use two identical for failover? |
06:15.29 | sgorilla | haha |
06:15.36 | sgorilla | what is cheap b/w? |
06:15.38 | e-Hernick | No, you want to different machines. |
06:15.40 | Insanity5 | bandwidth |
06:15.44 | e-Hernick | cheap bandwith is like, cogentco |
06:15.53 | Insanity5 | e-Hernick - time warner isn't much better. |
06:15.58 | sgorilla | 2000 - 20mbit a month |
06:16.07 | e-Hernick | 2000 is reasonable. |
06:16.16 | Insanity5 | that's not dirt cheap, but locatino location location is what matters. |
06:16.24 | Insanity5 | where is it? |
06:16.29 | sgorilla | includes full e1 - and 100 dids |
06:16.38 | sgorilla | well 2 mbit off the 20 for voice |
06:16.45 | sgorilla | not paying for the fiber install |
06:16.50 | e-Hernick | If 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 |
06:17.03 | sgorilla | yeah |
06:17.13 | sgorilla | i am thinking i should colo |
06:17.17 | e-Hernick | So you get a full e1 ? |
06:17.18 | e-Hernick | Nice. |
06:17.21 | sgorilla | and just get ports to connect to other datacenters |
06:17.24 | kuku5 | e-Hernick: have you done that ? |
06:17.37 | Insanity5 | sgorilla - 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 :) |
06:17.38 | sgorilla | then you can use other providers |
06:17.47 | sgorilla | Insanity5: true =) |
06:17.54 | e-Hernick | kuku5, yes, I've done that. My on-location servers route some data through my hub servers which have better connectivity |
06:18.09 | sgorilla | Insanity5: what do you need for a datacenter? |
06:18.17 | kuku5 | you have a direct link between them ? ( say no ) |
06:18.20 | sgorilla | it will have hvac with humidity control |
06:18.26 | sgorilla | but not n+1 hvac |
06:18.28 | e-Hernick | kuku5, no, I go through the internet |
06:18.32 | e-Hernick | Well, it's like the boat and ship distinction. |
06:18.35 | Insanity5 | sgorilla - What are you doing? data centers are generally points where people can purchase bandwidth? |
06:18.35 | sgorilla | just be able to put in mobile cooling |
06:18.48 | Insanity5 | sgorilla - As in, you order a cage and some b/w. usually run by telco's or major isps. |
06:18.49 | sgorilla | well its just a server farm |
06:18.49 | e-Hernick | You're describing a small computer room. A boat, not a ship. Not a datacenter. |
06:18.52 | kuku5 | that $100 idea is not so bad |
06:19.00 | kuku5 | didnt think of that |
06:19.01 | sgorilla | =) |
06:19.12 | e-Hernick | I mean, most businesses with a few employees have more power than that in their computer room and way more e1s than you. |
06:19.13 | kuku5 | what do you route ? |
06:19.18 | e-Hernick | Calling it a datacenter is pretentious. |
06:19.29 | sgorilla | ok what should i call it? |
06:19.34 | Insanity5 | sgorilla - server room? |
06:19.35 | kuku5 | a room |
06:19.38 | sgorilla | server room |
06:19.39 | sgorilla | ok |
06:19.41 | sgorilla | heh |
06:19.43 | wunderkin | the dungeon |
06:19.48 | Insanity5 | sgorilla - Now let's get back on track. |
06:19.53 | e-Hernick | I move Web traffic, IAX2, SIP, Jabber, rsync, bittorrent and more. |
06:20.07 | e-Hernick | Oh, lots of mail too. |
06:20.10 | sgorilla | ok |
06:20.14 | e-Hernick | I use front-end servers with excellent connectivity |
06:20.26 | Insanity5 | sgorilla - 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. |
06:20.40 | e-Hernick | They're providing the router. |
06:20.42 | sgorilla | not paying for the router termination |
06:20.47 | sgorilla | or the fiber installation |
06:20.58 | e-Hernick | He gets an ethernet port in a 5650 |
06:21.00 | sgorilla | or actually the internet connection itslef |
06:21.04 | Insanity5 | sgorilla - Often termination equipment is included. Then your isp is going to give you an ethernet cable off that router. |
06:21.10 | sgorilla | yeah |
06:21.11 | sgorilla | yes |
06:21.18 | Insanity5 | sgorilla - Your best bet is to use the router they provided for you. |
06:21.29 | sgorilla | but i dont think i have access to configure it |
06:21.30 | Insanity5 | sgorilla - I don't know what your doing that's special that needs further routing. |
06:21.36 | sgorilla | well nothing at first |
06:21.41 | sgorilla | just need some type of firewall |
06:21.43 | e-Hernick | Yeah, 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 |
06:21.45 | Insanity5 | sgorilla - Well that's betwen you and them. |
06:21.46 | sgorilla | well redundant firewalls |
06:21.48 | e-Hernick | Though that would be fun |
06:22.02 | Insanity5 | sgorilla - Well two cisco pix w/ failover cables. |
06:22.09 | Insanity5 | sgorilla - For any kind of large enterprise setup. |
06:22.11 | e-Hernick | "and remember, don't look into the fiber laser; it's live !" |
06:22.21 | sgorilla | haha |
06:22.30 | Igbothom_III | "Please don't look into the Laser with remaining eye" |
06:22.32 | kuku5 | i think you guys are overkilling this redundancy thing |
06:22.39 | Insanity5 | sgorilla - 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. |
06:22.43 | sgorilla | what is the difference between pix and linux |
06:22.49 | sgorilla | i want to be able to run snort with barnyard |
06:22.56 | brookshire | pix sucks |
06:22.57 | brookshire | :) |
06:22.57 | Insanity5 | snort doesn't need to be on the router. |
06:23.01 | drumkilla | brookshire: your mom |
06:23.02 | sgorilla | true |
06:23.03 | Insanity5 | brookshire - but at least it's reliable. |
06:23.06 | e-Hernick | sgorilla, why are you still looking for a solution ? I've given you the perfect solution already. |
06:23.17 | sgorilla | the pix? |
06:23.22 | brookshire | Insanity5: and very hackable (last month) |
06:23.27 | Insanity5 | sgorilla - 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. |
06:23.33 | e-Hernick | sgorilla, 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. |
06:23.39 | kuku5 | image_version : "P0S3-07-3-00" |
06:23.39 | kuku5 | FirmLoadID : "PC030301" |
06:23.39 | kuku5 | DSPLoadID : "PS03AT38" |
06:23.47 | e-Hernick | Now, whichever works best will become your primary, and the other will be the failover. |
06:23.49 | kuku5 | Does this mean it has old software? |
06:23.55 | kuku5 | (cisco 7940) |
06:24.03 | brookshire | Insanity5: we have a linux firewall.. it pushes our 10mbit fiber and 100mbit colo |
06:24.05 | sgorilla | e-Hernick: : that sounds like a good idea |
06:24.15 | brookshire | no problems yet and it's been there for a couple of months |
06:24.28 | drumkilla | that's because linux is hot |
06:24.42 | Insanity5 | brookshire - 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. |
06:24.51 | brookshire | drumkilla: and it's you're mom ;) |
06:24.57 | Insanity5 | brookshire - optimized code, risc processors, and processor switched routing just handle the nasty stuff so much better. |
06:25.05 | brookshire | "you're" must misspell it correctly ;) |
06:25.27 | sgorilla | i heard cisco pix can get unstable under DoS |
06:25.29 | e-Hernick | sgorilla, 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. |
06:25.35 | Insanity5 | sgorilla - I imagine any firewall can. |
06:25.36 | brookshire | insain: it runs digium.com ;) |
06:26.07 | Insanity5 | brookshire - but seriously, if you reboot that for 5 minutes at 3 am, just a minor inconvience. |
06:26.14 | Insanity5 | Might loose an order or something. |
06:26.22 | e-Hernick | sgorilla, 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. |
06:26.30 | sgorilla | ok |
06:26.42 | brookshire | Insanity5: nah |
06:26.45 | sgorilla | is there anytype of fastboot option with linux or bsd? |
06:26.46 | Insanity5 | I 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. |
06:26.54 | sgorilla | like be able to load up memory right away |
06:27.00 | sgorilla | like a laptop does with a snapshot |
06:27.16 | orlok | e-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) |
06:28.04 | Insanity5 | I'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. |
06:28.17 | brookshire | os anyways... i think the datacenter is more unreliable than the setup we have |
06:28.21 | sgorilla | i heard once you go over 100mb its not good |
06:28.37 | sgorilla | to use linux or vanilla bsd |
06:28.45 | wunderkin | brookshire: which one? |
06:28.54 | Insanity5 | sgorilla - You'll be applying kernel hacks at minimum... assuming that 100mb is composed of tons of individual flows. |
06:29.20 | brookshire | api-digital... they only have like 2 generators and 10 connections ;) |
06:29.25 | Insanity5 | sgorilla - 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. |
06:29.46 | brookshire | but cisco is overpriced |
06:29.51 | brookshire | and we can do it cheaper :) |
06:29.56 | Insanity5 | brookshire - Depends, I don't buy new cisco |
06:30.11 | brookshire | and we can do nifty stuff with it too.. because it's open |
06:30.13 | Insanity5 | brookshire - Ok, $230 on ebay, cisco 3620, add $50 for ethernet nic, and it's routing 15 megabit nicely. |
06:30.29 | e-Hernick | sgorilla, you have no need for BGP |
06:30.29 | Insanity5 | brookshire - I can't do that... reliabily... with linux... for that price. |
06:30.32 | brookshire | Insanity5: we're firewalling 100mbit |
06:30.38 | brookshire | +10 mbit fiber |
06:30.45 | drumkilla | with a beefy server :) |
06:30.53 | Insanity5 | brookshire - Well the fiber is likely not termianted into the linux box, right? |
06:30.56 | drumkilla | </stupid router war> |
06:31.00 | sgorilla | haha |
06:31.04 | e-Hernick | sgorilla, all you need is a simple router with failover. |
06:31.09 | sgorilla | yup |
06:31.10 | brookshire | Insanity5: it terminates to cat5 :) |
06:31.11 | e-Hernick | sgorilla, answer this: why do you need BGP ? |
06:31.14 | Insanity5 | brookshire - Somethings in the middle, handing it off to ethernet. |
06:31.18 | sgorilla | i dont need bgp |
06:31.21 | brookshire | but we do the routing |
06:31.25 | Insanity5 | e-Hernick - He probably doesnt' even have an AS number. |
06:31.26 | sgorilla | just something i was thinkg about for the future possibly |
06:31.30 | orlok | i hate cisco |
06:31.31 | drumkilla | blah blah blah blah ... |
06:31.32 | orlok | with a passion |
06:31.36 | e-Hernick | sgorilla, the future is the future. When you get there, make your choice. |
06:31.46 | sgorilla | true |
06:31.48 | brookshire | Insanity5: it's not a cisco.. nor a router |
06:32.00 | brookshire | just an ethernet converter |
06:32.01 | e-Hernick | sgorilla, the redundant router solution that I propose will cost you very little. How many network interfaces do you need in each router ? |
06:32.11 | Insanity5 | brookshire - 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 |
06:32.21 | Insanity5 | brookshire - And i'm quite suprised you need 100 for digium. |
06:32.50 | e-Hernick | Insanity5, 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. |
06:32.51 | Insanity5 | brookshire - And if you're already getting it in ethernet, you may onlyn eed a firewall, better/cheaper cisco choices for that. |
06:32.59 | sgorilla | e-Hernick: probably 2 ports |
06:33.23 | sgorilla | is it possible for switches to failover ? |
06:33.24 | e-Hernick | sgorilla, very well then. Do you know what kind of computer you want to build ? |
06:33.25 | X-Rob | pix -is- not cheaper than a 3620. |
06:33.36 | Insanity5 | e-Hernick - I just can't pay $2-3k in monthly internet costs, and not justify a damn $400 router in the budget, sorry. |
06:33.39 | e-Hernick | sgorilla, not with cheaper switches. You're going to have to trust the switches. |
06:33.40 | sgorilla | e-Hernick: probably a linux box 512 megs ram, celeron |
06:33.43 | Insanity5 | X-Rob - It isn't, but I was referring to 100 mb |
06:33.44 | brookshire | pix is a piece of crap.. i use to use one daily |
06:34.02 | sgorilla | what router do you suggest? |
06:34.10 | sgorilla | old cisco router from ebay? |
06:34.20 | sgorilla | i need sometype of switch that can do vlan atleast |
06:34.21 | brookshire | digium pci card :) |
06:34.29 | brookshire | at least for t1 |
06:34.29 | brookshire | hehe |
06:34.52 | Insanity5 | sgorilla - you can get a 24 port maanged vlan capapble cisco switch for $500 on ebay. |
06:35.48 | sgorilla | what model would that be? |
06:35.50 | X-Rob | 29xx switches are 'ok' |
06:36.00 | Insanity5 | 2940xl (make sure it's the 8 meg flash one) will work. |
06:36.04 | X-Rob | don't get a 19xx, they don't do 802.1q (eg, 'standard' vlans) |
06:36.11 | Insanity5 | the 4 meg flash doesn't do vlan |
06:36.18 | Insanity5 | that's the cheapeast option |
06:36.31 | Insanity5 | 3500 series are a better choice for something more modern, but you sound like you're cheap today. |
06:36.39 | sgorilla | yes |
06:36.48 | kuku5 | 2500 is doable |
06:36.59 | Insanity5 | ? |
06:37.00 | e-Hernick | sgorilla, 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. |
06:37.26 | e-Hernick | sgorilla, use the machines to do traffic shaping, routing, but also VPN and transparent proxying |
06:37.37 | sgorilla | ok |
06:37.42 | Insanity5 | trasnparent proxiyng... iick |
06:37.45 | sgorilla | run snort on a different box? |
06:37.46 | drumkilla | so how about that Asterisk |
06:37.49 | drumkilla | it's pretty cool, huh |
06:37.57 | sgorilla | yeah running asterisk also |
06:38.06 | brookshire | gentoo + digium pci card = one heck of a router :) |
06:38.08 | X-Rob | drumkilla - asterisk eh? I've heard something about that. |
06:38.08 | sgorilla | not sure how the t1 is going to be handed off |
06:38.10 | e-Hernick | You have two routers, one primary, one secondary |
06:38.14 | drumkilla | it's pretty awesome |
06:38.20 | drumkilla | it lets you make phones ring |
06:38.22 | Insanity5 | e-Hernick - the TCO for most companies just won't pan out for custom programming and distrobution creation. |
06:38.24 | e-Hernick | Your data logging and monitoring will run on the secondary |
06:38.31 | X-Rob | phones eh? Wow. Fancy that. |
06:38.35 | sgorilla | ok |
06:38.38 | X-Rob | There should be an IRC channel for talking about asterisk, you know. |
06:38.43 | sgorilla | haha |
06:38.48 | Igbothom_III | X-Rob; what for? :) |
06:38.52 | drumkilla | X-Rob: I agree, let's make one |
06:38.54 | brookshire | drumkilla: i wish chatrooms had a mute button ;) |
06:39.00 | Insanity5 | it's /ignore :) |
06:39.00 | sgorilla | ##asterisk |
06:39.02 | drumkilla | how about #asterisk-forrealthistime |
06:39.07 | e-Hernick | Insanity5, you know, I agree. But I like open source, and this is open source. |
06:39.14 | X-Rob | I was thinkging of #asterisk-no,really. |
06:39.20 | brookshire | Insanity5: but that's like 8 buttons.. i just need one |
06:39.24 | e-Hernick | Okay, so let's talk about asterisk |
06:39.30 | e-Hernick | What about we set up a conference call for the channel ? |
06:39.31 | brookshire | a big red one :) |
06:39.32 | sgorilla | how would the routers physically be hooked up? |
06:39.33 | *** join/#asterisk glomph (n=black@c-24-18-145-249.hsd1.wa.comcast.net) |
06:39.35 | wunderkin | X-Rob: you don't want to join me in #3,000? |
06:39.40 | drumkilla | let's play name a feature, and it gets in! |
06:39.43 | Insanity5 | e-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. |
06:39.49 | drumkilla | someone name a feature and I'll write it real quick |
06:39.49 | X-Rob | drumkilla - porn-on-demand! |
06:39.52 | *** part/#asterisk glomph (n=black@c-24-18-145-249.hsd1.wa.comcast.net) |
06:39.52 | brookshire | haha... /disco inferno |
06:39.58 | brookshire | best irc command ever! |
06:39.58 | e-Hernick | sgorilla, you put a switch between the routers and the 5650 |
06:40.05 | drumkilla | alright, to continue this discussion, join #2,000 |
06:40.08 | Insanity5 | e-Hernick - If you want to put the linux box behind the router for VPN, proxy, etc, I can see that. |
06:40.09 | e-Hernick | sgorilla, get an old cisco switch, those rule |
06:40.18 | sgorilla | ok |
06:40.20 | X-Rob | Heh |
06:40.21 | e-Hernick | Insanity5, Linux routers are way more flexible. |
06:40.31 | brookshire | e-Hernick: agreed :) |
06:40.36 | sgorilla | but that switch will be different from the server switch correcT? |
06:40.39 | drumkilla | kids just aren't curious anymore these days ... |
06:40.47 | e-Hernick | sgorilla, yes |
06:40.54 | sgorilla | like a small switch |
06:41.00 | sgorilla | what would you use in that case? |
06:41.07 | Insanity5 | e-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... |
06:41.21 | sgorilla | liike 4 ports |
06:41.25 | sgorilla | but reliable |
06:41.39 | e-Hernick | sgorilla, (the internet) --> (5650) --> (small, old cisco 100mb switch) --> (primary router and secondary router) --> (server switches) |
06:41.40 | sgorilla | linksys switch |
06:41.44 | sgorilla | bhaha that is cisco |
06:41.50 | e-Hernick | linksys is low-end. |
06:41.53 | sgorilla | ok cool |
06:42.16 | e-Hernick | One 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 |
06:42.19 | sgorilla | how about running snort |
06:42.29 | e-Hernick | snort can be run on either your primary and secondary routers |
06:42.33 | sgorilla | ok cool |
06:42.37 | e-Hernick | your secondary doesn't have to be entirely idle either |
06:42.44 | Insanity5 | snort can run anywhere on the network. |
06:43.00 | sgorilla | that sounds like a decent setup |
06:43.07 | e-Hernick | Yes, but he might have more than a single server switch |
06:43.10 | wunderkin | drumkilla: speaking of features... |
06:43.16 | e-Hernick | As a matter of fact, I would recommend at least two server switches. |
06:43.25 | sgorilla | why is that? for failover? |
06:43.26 | Insanity5 | e-Hernick - Now... if he's at this stage, let's see how many weeks this will take to set up. |
06:43.33 | e-Hernick | Put 4 network interfaces in each of the routers |
06:43.33 | Insanity5 | :) |
06:43.47 | e-Hernick | have two server switches |
06:43.47 | wunderkin | when 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 |
06:43.51 | sgorilla | ok i see |
06:43.57 | e-Hernick | you can have redundancy on the server level too |
06:44.11 | e-Hernick | or you can simply group off your servers |
06:44.16 | Insanity5 | For 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. |
06:44.21 | sgorilla | i guess each computer can have redundancy hooking up to two switches |
06:44.23 | Insanity5 | or asterisk for phone |
06:44.29 | sgorilla | atleast have the core servers run on two switches |
06:44.31 | sgorilla | for failover |
06:44.40 | e-Hernick | You don't necessarily need to configure failover at first |
06:44.45 | brookshire | AD is lamer than cisco |
06:44.51 | sgorilla | Active Directory? |
06:45.03 | Insanity5 | brookshire - But it works excellent for certain corporate situations, on a TCO basis. |
06:45.04 | brookshire | ldap! |
06:45.07 | X-Rob | Adal Dilaton? |
06:45.10 | e-Hernick | But 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 |
06:45.13 | brookshire | just use ldap |
06:45.29 | Insanity5 | brookshire - And hire 4 more people to manage the 500 w2k desktops? nope. |
06:45.31 | brookshire | you know how many applications support ldap? |
06:45.33 | sgorilla | ok that sounds good |
06:45.41 | e-Hernick | sgorilla, make sure to have a very good monitoring system that contacts you immediately if any problem sis detected |
06:45.41 | brookshire | how many *more* |
06:45.49 | sgorilla | e-Hernick: like nagios? |
06:45.50 | brookshire | ad is ldap you fool |
06:45.52 | e-Hernick | sgorilla, a part of this system must be installed outside your datacenter |
06:45.53 | brookshire | and free |
06:46.02 | Insanity5 | brookshire - But any ldap doesn't integrate into a win32 environemnt in the same manner. |
06:46.04 | e-Hernick | sgorilla, like nagios or zabbix yes |
06:46.06 | sgorilla | its a "server room" |
06:46.11 | brookshire | or if you want to pay.. but novell's directory |
06:46.11 | sgorilla | =) |
06:46.12 | e-Hernick | yes, it's a small server room |
06:46.16 | sgorilla | very small |
06:46.17 | brookshire | it's far superior ;) |
06:46.20 | e-Hernick | a datacenter is something bigger |
06:46.30 | Insanity5 | brookshire - novell's stuff is good and comes with the necessary tools (included) to keep TCO down. |
06:46.34 | sgorilla | never heard of zabbix |
06:46.41 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
06:46.57 | e-Hernick | sgorilla, 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 |
06:47.02 | Insanity5 | brookshire - 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) |
06:47.19 | Insanity5 | brookshire - Novell is quite cheaper, but sometimes a little bit more to maintain. |
06:47.27 | drumkilla | wunderkin: yes? |
06:47.34 | Insanity5 | brookshire - but at least it includes a lot in the licenseing, like their mail server. |
06:47.34 | shido6 | Novell? |
06:47.39 | sgorilla | zabbix screenshots look cool |
06:47.44 | shido6 | what Novell product is worth buying? |
06:47.45 | Insanity5 | shido6 - scroll up a but :) |
06:47.49 | Insanity5 | bit |
06:47.50 | e-Hernick | As for my two server switches, if I were on a buget, I'd go with one gigabit switch and one fast ethernet switch |
06:48.04 | sgorilla | i guess you want to collect information on all the computers |
06:48.05 | brookshire | shido6: their ldap stuff is unmatched |
06:48.08 | X-Rob | STP (Spanning Tree Protocol) is your friend, people. |
06:48.12 | X-Rob | Learn to love it. |
06:48.13 | sgorilla | and store them in a database, also do remote syslog monitoring |
06:48.14 | Insanity5 | e-Hernick - and why does he need gigabit? |
06:48.14 | shido6 | ok I know nothing about ldap |
06:48.24 | e-Hernick | sgorilla, yes, you want to check all servers and all services, as well as all connectivity |
06:48.31 | brookshire | ldap is like a glorifed nis |
06:48.32 | e-Hernick | sgorilla, if at all possible you want gigabit between the servers, because of backups and data transfers between servers. |
06:48.34 | Insanity5 | e-Hernick - Even with over 600 servers here running multi-gigabit sql queries, I don't need gigabit anywhere but the core. |
06:48.44 | sgorilla | also have a dialup to access a terminal |
06:48.49 | sgorilla | and be able to ssh on the network |
06:48.50 | wunderkin | drumkilla: ? did you see my question after that :D |
06:48.52 | kuku5 | Oct 20 01:15:14 WARNING[10101]: chan_zap.c:770 zt_open: Unable to open '/dev/zap/pseudo': No such device |
06:48.52 | kuku5 | Oct 20 01:15:14 ERROR[10101]: chan_zap.c:6731 chandup: Unable to dup channel: No such device |
06:48.52 | kuku5 | Oct 20 01:15:14 WARNING[10101]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device |
06:48.52 | kuku5 | Oct 20 01:15:14 WARNING[10101]: app_meetme.c:230 build_conf: Unable to open pseudo device |
06:48.53 | sgorilla | if the fiber goes down |
06:48.55 | kuku5 | What does this mean ? |
06:48.57 | Insanity5 | PASTEBIN.COM! |
06:49.00 | e-Hernick | If you are building a small network, gigabit is not much more expensive. |
06:49.00 | sgorilla | also be able to access serial port on cisco |
06:49.02 | kuku5 | ergh - sorry |
06:49.07 | e-Hernick | For large networks, gigabit is much more expensive per port |
06:49.14 | Insanity5 | e-Hernick - it is expensive for Cisco (quality) stuff. |
06:49.15 | shido6 | ztdummy in Makefile kuku5 if u have no digium gear |
06:49.22 | brookshire | kuku5: install ztdummy? |
06:49.28 | kuku5 | i thought i did |
06:49.35 | shido6 | recompile zaptel |
06:49.43 | e-Hernick | Insanity5, 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. |
06:49.44 | Insanity5 | kuku5 - modprobe both, remove them, readd |
06:49.52 | e-Hernick | I like the netgear gigabit switches. |
06:49.58 | kuku5 | modprobe ? |
06:50.12 | Insanity5 | e-Hernick - Yeah, non-name brand switch gigabit vs 100 megabit name brand at your core backbone. Not a good idea. |
06:50.14 | e-Hernick | They're fast, have vlan, management, jumbo frames and cost around 25$ per port of gigabit |
06:50.32 | Insanity5 | e-Hernick - and so much for running netdisco, or millions of other utilities that make a large netowrk amangeable throguh cisco switches. |
06:50.36 | sgorilla | i heard SMC are decent |
06:50.46 | kuku5 | dell are ok |
06:50.53 | Insanity5 | dell are rebranded JUNK |
06:50.55 | Insanity5 | Don't bother |
06:50.58 | e-Hernick | Insanity5, 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. |
06:51.19 | e-Hernick | But I'm very good at running small networks, and nowadays I deploy gigabit, and it's worth it. |
06:51.22 | Insanity5 | We have 50 switches, 10 dells, 40 ciscos. Guess what? 1 cisco failed, and 3 dells in the last 2 years. |
06:51.22 | e-Hernick | Not much more expensive. |
06:51.33 | shido6 | dood |
06:51.36 | shido6 | dont do it |
06:51.45 | sgorilla | Insanity5: using what router? |
06:51.46 | e-Hernick | You are running a large network, your considerations are very different. |
06:51.47 | Insanity5 | e-Hernick - But dual router and crappy switch? |
06:51.55 | Insanity5 | sgorilla - doesn't matter, they're switches. |
06:52.03 | sgorilla | i mean what switch |
06:52.06 | shido6 | buy the giga-hardware with mad reviews and can handle a a lot of packet per second |
06:52.13 | shido6 | packets |
06:52.14 | brookshire | not all switches are equal |
06:52.17 | Insanity5 | sgorilla - dell 48 ports, and one dell 24 port gigabit |
06:52.20 | shido6 | hell no they are not |
06:52.24 | brookshire | arp floop a cheap router and find out :) |
06:52.34 | brookshire | INSTANT HUB! |
06:52.37 | sgorilla | haha |
06:52.54 | Insanity5 | We finally just ebayed our dell switches. |
06:52.54 | sgorilla | dell should not be vulnerable to that |
06:52.55 | syle2 | where can i view the struct table for cdr? |
06:52.57 | sgorilla | that is old school stuff |
06:53.13 | Insanity5 | sgorilla - dell is rebranded garbage. and they're gigabit swtiches have trouble syncing to some nic cards, big time. |
06:53.24 | Insanity5 | sgorilla - Combine that with no way to get error counters on the interface, and you have problems. |
06:53.43 | wunderkin | syle2, README.cdr is that what you mean? |
06:54.10 | sgorilla | Insanity5: netflow looks good on cisco |
06:54.11 | Insanity5 | If you must have a cheap managable switch, intel 510T's are OOOOOLD buy reliable. |
06:54.15 | Insanity5 | and big. |
06:54.17 | sgorilla | you can integrate that with ntop |
06:54.21 | syle2 | no the actual c struct |
06:54.30 | wunderkin | ok |
06:54.38 | syle2 | hmm think i found it |
06:54.42 | Insanity5 | sgorilla - 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. |
06:54.42 | syle2 | --/usr/include/asterisk/cdr.h |
06:54.46 | drumkilla | grep "struct ast_cdr" include/asterisk/* |
06:54.47 | drumkilla | :-p |
06:54.55 | shido6 | I stil own and ude my old DS104 netgear hub on my local lan for xbox xfers |
06:55.02 | shido6 | its not plastic |
06:55.13 | syle2 | just making sure src and dst are chars hehe |
06:55.17 | sgorilla | what is a good progam for monitoring load, temperature, cpu usage on windows boxses? |
06:55.28 | sgorilla | shido6: you mod xboxes? |
06:55.35 | shido6 | :) |
06:55.37 | Insanity5 | sgorilla - Umm, snmp + your favorite front end? |
06:55.59 | Insanity5 | sgorilla - manually populate them in mrtg/rrdtool, or take the lazy way out and use cacti and/or big brother. |
06:55.59 | shido6 | blame Mark, I saw his xbox with asterisk on it and felt the need to screw M$ |
06:56.12 | wunderkin | drumkilla: 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 |
06:56.20 | kuku5 | MODULES=zaptel tor2 torisa wcusb wcfxo wcfxs \ |
06:56.20 | kuku5 | <PROTECTED> |
06:56.29 | kuku5 | I had this compiled already |
06:56.31 | brookshire | shido6: the pbxbox :) |
06:56.36 | shido6 | yeah |
06:56.59 | kuku5 | Any ideas? |
06:57.01 | sgorilla | haha xbox with asterisk |
06:57.11 | Insanity5 | hehe |
06:57.12 | sgorilla | that is nuts, i have modded about 30 xboxes |
06:57.20 | sgorilla | xbmc is really nice |
06:57.20 | Insanity5 | sgorilla - Want to fix mine? :) |
06:57.22 | kuku5 | Insanity5: what should i do with the /dev/za/pseudo |
06:57.30 | sgorilla | what is wrong with it? |
06:57.42 | sgorilla | i have fixed out xboxes from just changing out compenents |
06:57.43 | Insanity5 | kuku5 - I wasn't patying attention, but I fixed my ztdummy problems by removing and adding the proper module. |
06:57.45 | brookshire | kuku5: modprobe ztdummy ? |
06:58.04 | Insanity5 | kuku5 - It in turn will add the depentent modules. |
06:58.07 | sgorilla | i havent done any type of stuff with logic analyzers |
06:58.10 | syle2 | hmmmm |
06:58.16 | kuku5 | [root@mypbx zaptel]# modprobe ztdummy |
06:58.16 | kuku5 | [root@mypbx zaptel]# |
06:58.17 | Insanity5 | sgorilla - I have to hit the power button about 40 times to get it to power up. |
06:58.17 | syle2 | char src[AST_MAX_EXTENSION]; |
06:58.29 | Insanity5 | sgorilla - oncei t's on it's fine. It has a homebrew mod applied to it from many years ago. |
06:58.30 | brookshire | kuku5: mow try your conf |
06:58.31 | syle2 | anyone know what AST_MAX_EXTENSION is set to or where i can find it |
06:59.09 | kuku5 | WO! |
06:59.10 | kuku5 | WOW! |
06:59.13 | kuku5 | it worked ... WHY ? |
06:59.21 | brookshire | confs need to be in sync |
06:59.36 | brookshire | ztdummy uses the cpu to help with timing |
06:59.53 | kuku5 | ok - so now my paging works :) |
06:59.55 | Insanity5 | Somebody needs to make a stupid "timing device" that plugs into serial/usb/pci |
06:59.58 | Insanity5 | for dirt CHEAP |
06:59.59 | Insanity5 | :0 |
07:00.04 | kuku5 | I have a big problem with this 7940 |
07:00.07 | Insanity5 | It can't be that hard of a problem. |
07:00.12 | brookshire | Insanity5: it's called a x100p |
07:00.13 | Insanity5 | kuku5 - send it to me. |
07:00.13 | X-Rob | Insanity5 - you're nts. |
07:00.13 | kuku5 | I ahve no clue how this will work for a secretary |
07:00.15 | X-Rob | nuts |
07:00.20 | kuku5 | I mean - 2 extensions |
07:00.21 | X-Rob | X100p |
07:00.23 | X-Rob | It's like $2. |
07:00.24 | X-Rob | get over it. |
07:00.37 | Insanity5 | X-Rob - I could use an ebay clone, don't know how well it will work though. |
07:00.39 | X-Rob | However, it's _more_ accurate to use RTC |
07:00.48 | kuku5 | Which phones you guys use for secretaries? |
07:00.54 | Insanity5 | X-Rob - I thought ztdummy was the worst option. |
07:00.56 | X-Rob | you don't care about the audio quality, you just want timing. |
07:01.02 | brookshire | kuku5: we love polycom :) |
07:01.13 | brookshire | they have a neat one with addable extensions :) |
07:01.18 | brookshire | block |
07:01.19 | X-Rob | Insanity5 - RTC is the _best_ option on 2.6 kernels |
07:01.20 | brookshire | thingys |
07:01.31 | kuku5 | can you transfer with one push ? |
07:01.35 | brookshire | mmmhmm |
07:01.41 | kuku5 | or do you ahve to hit transfer, and then the line |
07:01.43 | Insanity5 | 2.6.8-24.5-default #1 |
07:01.47 | Insanity5 | hmmm ok so I guess I'm good. |
07:01.55 | brookshire | they can also be programed with xml |
07:02.05 | brookshire | say like.. placed in a certain spot on your website |
07:02.10 | kuku5 | yeh |
07:02.16 | kuku5 | but thats not needed |
07:02.23 | Insanity5 | Are there cheap cisco phones (old ones) that like, work only with mgcp? Or have tehy all been firmware flashed over? |
07:02.25 | brookshire | it's good for redoing extensions though |
07:02.25 | kuku5 | I need a girl to answer 5 calls withing 15 sec |
07:02.36 | brookshire | on like 400 phones |
07:02.37 | brookshire | at one time |
07:02.38 | brookshire | :D |
07:02.48 | Insanity5 | kuku5 - You need an automated interactive girl attendant lol. |
07:02.53 | kuku5 | what do you mean redoing ? |
07:03.08 | kuku5 | Insanity5: no... if you have a good system you can do it |
07:03.18 | kuku5 | but you need to be able to TRANSFEr a call with 1 button |
07:03.41 | Insanity5 | kuku5 - Difficult without being rude to people (3 seconds to ask them who they want? I can't say hello that quick). |
07:03.43 | e-Hernick | That's correct. You don't need a girl to simply answer the phone for you. |
07:03.49 | e-Hernick | You need a girl to record a very nice IVR |
07:03.59 | kuku5 | no |
07:04.07 | kuku5 | you need to answer the phone - period |
07:04.13 | Insanity5 | Who wants to make me an intro prompt? |
07:04.27 | brookshire | allison can |
07:04.31 | Insanity5 | I need some girl to make me a press 1 for blah thing. |
07:04.33 | brookshire | why do you buy one? |
07:04.36 | drumkilla | ~thevoice |
07:04.42 | drumkilla | ~allison |
07:04.43 | jbot | [allison] The IVR Voice, http://theivrvoice.com/ and http://thevoice.digium.com/ |
07:04.44 | *** join/#asterisk memic (n=memic@chicago089.server4free.de) |
07:04.44 | Insanity5 | I need 7 words, that's it :) |
07:04.50 | memic | wow |
07:05.00 | memic | how to limit ring time to 10 sec? Dial(SIP/memic) .. ? |
07:05.11 | brookshire | Insanity5: it's like $12 |
07:05.15 | brookshire | from digium |
07:05.16 | memic | Dial(SIP/memic,10) seems not to work |
07:05.24 | Koshatul | memi: isn't it Dial(SIP/memic,10) ? |
07:05.26 | brookshire | i'm sure you can afford it with all that expensive cisco gear you own |
07:05.31 | kuku5 | these sidecars are expensive ! |
07:05.34 | Insanity5 | brookshire - It's for my personal use :) |
07:05.48 | Koshatul | memi: try Dial(SIP/memic,10,) ? |
07:05.54 | drumkilla | you can use the Record dialplan applicaiton to make your own prompts |
07:05.57 | drumkilla | if you don't care about quality |
07:06.15 | Insanity5 | drumkilla - That's what I'm going to do. It can't sound... that bad :) |
07:06.18 | memic | k |
07:06.22 | drumkilla | it works ... |
07:06.37 | Insanity5 | It's just... Press 1 for Me, Press 2 for joe (roomate #1) press 3 for bob (roomate #2). |
07:06.43 | Insanity5 | like 7 words, literally. |
07:06.52 | Koshatul | Insanity5: you can record that yourself easily |
07:07.01 | e-Hernick | Getting a good IVR recording can take some work. |
07:07.07 | memic | Koshatul problem is phone is ringing but my second phone is not |
07:07.11 | drumkilla | all of those words are there except for the names ... |
07:07.17 | drumkilla | and there are even some names, but not sure if those are included |
07:07.20 | Koshatul | Insanity5: actually, i got an old girlfriend to do the IVR for my company, she has a great voice :) |
07:07.20 | e-Hernick | If you're not used to making recordings, you might have trouble getting the right quality. |
07:07.22 | drumkilla | checkout asterisk-sounds from cvs |
07:07.25 | drumkilla | and check out what is in there |
07:07.28 | e-Hernick | You need to use a good microphone among other things |
07:07.33 | memic | (SIP/phone1,7,&SIP/phone2,7,) |
07:07.41 | e-Hernick | A phone won't do for quality recordings. |
07:07.42 | memic | but only phone1 is rining |
07:07.52 | Insanity5 | drumkilla - I'm sure the prerecorded words are out there, except for the names. |
07:07.56 | drumkilla | memic: wrong syntax |
07:07.57 | Koshatul | memic: nah, it's Dial(SIP/phone1&SIP/phone2,7) |
07:08.07 | memic | Koshatul thx |
07:08.08 | memic | will try |
07:08.25 | Koshatul | i'm starting to hate engin, i can't seen to get DID working |
07:08.32 | drumkilla | g'night |
07:08.36 | e-Hernick | Insanity5, 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 |
07:08.36 | Koshatul | night |
07:08.46 | brookshire | b'killa ;) |
07:08.54 | brookshire | killad |
07:08.54 | drumkilla | brookshire: !!!!!!! |
07:09.02 | Koshatul | Insanity5: you can use audiology and cut and paste celebrity voice as well |
07:09.03 | Insanity5 | e-Hernick What else would you do? |
07:09.15 | Koshatul | my friend did that for his answering machine, he cut and pasted star trek voices |
07:09.17 | Insanity5 | e-Hernick - For a basic 3 roomate setup? |
07:11.35 | memic | drumkilla & Koshatul thx everything is working now.. |
07:11.38 | memic | :) |
07:11.42 | e-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." |
07:12.04 | Insanity5 | What is festival? |
07:12.08 | Insanity5 | e-Hernick - lol |
07:12.23 | memic | i have strange messages in asterisk log |
07:12.24 | memic | Oct 20 10:53:03 WARNING[12460]: Unable to forward frame |
07:12.32 | Insanity5 | e-Hernick - That would be good, and scare more telemarkers away. You think simple having the IVR will get you off most lists? |
07:12.42 | memic | after zap driver reload problem was gone.. |
07:12.57 | Insanity5 | Oct 20 02:12:52 WARNING[17966]: app_festival.c:350 festival_exec: festival_client: connect to server failed |
07:13.00 | wunderkin | can 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 : ( |
07:14.14 | Insanity5 | wunderkin - As long as it isn't optioned out, it should play with Recorder. |
07:14.23 | e-Hernick | Insanity5, 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) |
07:14.46 | e-Hernick | My IVR also filters based on callerid |
07:15.01 | brookshire | psk! |
07:15.02 | e-Hernick | Most 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. |
07:15.11 | brookshire | oh.. different psk |
07:15.21 | brookshire | my bad :) |
07:15.21 | Insanity5 | e-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. |
07:15.29 | wunderkin | Insanity5: as long as what isn't? recorder? :) |
07:15.49 | Insanity5 | wunderkin - show application recorder. It's in there about supressing the tone. As long as that's not set, you'll be ok. |
07:16.01 | e-Hernick | Yeah, the predictive dialers. They get my IVR, so they have to follow the procedure I lay out for them. |
07:16.22 | Insanity5 | e-Hernick - Well I figured they'd just hang up immediately and give up on the ivr. |
07:16.23 | Insanity5 | hehe. |
07:16.27 | e-Hernick | Prior business contacts get sent to a voicemail. |
07:16.37 | brookshire | there is a nifty trap app out there somewhere |
07:16.39 | e-Hernick | But they rarely get to me. |
07:16.50 | e-Hernick | The trap app is a joke. |
07:16.54 | brookshire | telemarketers come in and get caught in a loop |
07:17.01 | brookshire | e-Hernick: i know |
07:17.06 | e-Hernick | The way my app is set up, they get told exactly how to make a business proposal to me. |
07:17.11 | brookshire | you can have whitelists though :) |
07:17.27 | Insanity5 | Seems like too much hassle... over a "telemarketer... go away" greeting message. |
07:17.35 | wunderkin | Insanity5: im not sure what an application would have to do with options, and there is no application called recorder.. im doing an attended transfer |
07:17.39 | e-Hernick | But it's fun to make an IVR. |
07:17.45 | e-Hernick | It's your voice. |
07:18.16 | sgorilla | ok im back |
07:18.54 | *** part/#asterisk Deedubb (n=Deedubb@S010600055d22c57f.vf.shawcable.net) |
07:18.55 | Insanity5 | hehe |
07:19.35 | Insanity5 | wunderskin - I thought you were asking why it didn't beep when you recorded. I meant show application record, sorry. |
07:19.58 | wunderkin | gotta be something simple, ive had this working before but totally different setup.. im not doing recording and yeah thats what i thought you meant |
07:20.18 | sgorilla | e-Hernick: is it one of those telemarketer torture ivr setups? |
07:20.43 | Insanity5 | Do they even listen of just hang up? |
07:21.01 | e-Hernick | Well, my logs show that some of them have listened. |
07:21.11 | e-Hernick | Though, I'm gonna make a new version in the next few days. |
07:21.16 | e-Hernick | I'm going to make it bilingual this time, too. |
07:21.20 | e-Hernick | Right now it's only in french-canadian. |
07:21.29 | Insanity5 | e-Hernick - How do you know that they were telemarkers? |
07:21.29 | Insanity5 | hehe |
07:21.52 | wunderkin | e-Hernick: oh yeah, one of those "for spanish press 9" things huh? haha i love those, how ignorant |
07:22.24 | e-Hernick | Insanity5, I don't, but it took me a few minutes to record and program in AEL. |
07:22.37 | Insanity5 | I had a phone number once of a humours IVR -- it was some company of "monkeys".... |
07:22.48 | e-Hernick | Yeah. It's "For english, press nine" |
07:22.52 | Insanity5 | had options to accomodate all the most annoying things about IVRS and telephones. |
07:22.56 | e-Hernick | Default is french-canadian. |
07:23.01 | *** join/#asterisk djin (n=djin@213-132-172-4.multikabel.nl) |
07:23.03 | Insanity5 | hehe. |
07:23.42 | Insanity5 | Yawn... I hate compiling crap on pentium 3's. |
07:24.19 | e-Hernick | I'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. |
07:25.30 | wunderkin | <PROTECTED> |
07:25.45 | *** join/#asterisk Miggidy (i=user@dsl-202-72-180-171.wa.westnet.com.au) |
07:26.21 | djin | I need the run * in realtime , what would be the most stable release to go for (CVS of 1.2b1)? |
07:26.26 | Insanity5 | e-Hernick - better than recording straight to ulaw with no transcoding? |
07:26.28 | wunderkin | ah zap49 is me.. hmm |
07:26.38 | e-Hernick | Insanity5, significantly |
07:26.41 | sgorilla | they need a ivr with simon says with dial tone |
07:26.49 | Insanity5 | e-Hernick - how so? Doesn't make sense. |
07:27.22 | e-Hernick | Sure it does. |
07:27.29 | e-Hernick | I'm recording at 16-bit 48khz |
07:27.38 | e-Hernick | I'm processing the audio at that resolution |
07:27.41 | e-Hernick | and then I downsample |
07:27.57 | *** join/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net) |
07:28.07 | mover | morning |
07:28.12 | mover | :-/ |
07:28.46 | Insanity5 | how do you process to increase quality? |
07:28.50 | X-Rob | oooh |
07:28.52 | Insanity5 | I mean you can't make something of nothing. |
07:29.03 | X-Rob | schlockmercenary has become part of blank label comics |
07:29.04 | mover | who has got t38-bits get working and if so, how? :-) |
07:29.09 | X-Rob | and *hah* he's got star billing! |
07:29.11 | X-Rob | woo howard! |
07:29.39 | e-Hernick | Insanity5, http://www.l3i.ca/accueil.wav |
07:30.00 | e-Hernick | This is an example of a file in my IVR |
07:30.46 | e-Hernick | I tell you, without some audio processing you won't get that kind of quality. |
07:31.26 | sgorilla | why do you need audio processing? |
07:31.43 | e-Hernick | You don't need it, but it increases the subjective quality of the sound. |
07:31.49 | e-Hernick | Which is good. |
07:32.03 | Insanity5 | how 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) |
07:32.39 | sgorilla | i wonder how that sounds over a phone network |
07:32.45 | sgorilla | compared to listening to the actual wav |
07:33.05 | wunderkin | figured it out |
07:33.47 | e-Hernick | well, call 8457382479 |
07:34.02 | *** part/#asterisk newmember (i=user@S010600036d1139fb.cg.shawcable.net) |
07:34.23 | X-Rob | 8675309 |
07:34.52 | Insanity5 | Yikes, foreign language :0 |
07:35.01 | e-Hernick | but it sounds good |
07:35.17 | Insanity5 | It does, but it's also hard to tell if you don't know the language. |
07:35.22 | e-Hernick | better than if I had recorded with a cheap mic and done no processing |
07:35.26 | Insanity5 | hehe |
07:35.27 | e-Hernick | well, the next version will be bilingual |
07:35.32 | kuku5 | Did anyone use the gxp 2000 phones? |
07:36.15 | *** join/#asterisk Aze` (n=aze@85-18-136-114.ip.fastwebnet.it) |
07:36.51 | kuku5 | I cant get ulaw to work on them |
07:36.54 | kuku5 | the best is gsm |
07:36.57 | kuku5 | which sucks |
07:37.41 | orlok | Does anybody know how to unlock a cisco ip phone? |
07:37.57 | kuku5 | yah |
07:38.15 | orlok | ahh |
07:38.17 | kuku5 | via keypad? |
07:38.17 | orlok | **# apparently |
07:38.22 | Aze` | Anyone has last florz's patch ? |
07:39.03 | orlok | kuku5: apparently yeah |
07:42.18 | kuku5 | which version |
07:43.22 | X-Rob | kuku5 - they're good phone |
07:43.44 | X-Rob | s |
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) |
07:55.41 | puzzled | morning oej |
07:55.50 | oej | Morning |
07:56.35 | djin | I need the run * in realtime , what would be the most stable release to go for (CVS of 1.2b1)? |
07:57.11 | djin | oh, did I mention it's on a production machine? ;) |
07:57.17 | sgorilla | haha |
07:57.41 | sgorilla | what do you mean by realtime? |
07:57.58 | djin | Uuuh, asterisk realtime? |
07:58.01 | sgorilla | like QNX? |
07:58.26 | djin | Nope, running SIP and IAX from a DB instead of config files. |
07:58.41 | djin | or at least thus users |
07:58.43 | oej | beta 1 is rather old, so go for CVS head |
07:58.48 | sgorilla | ok |
07:58.54 | sgorilla | go with cvs head |
07:58.56 | sgorilla | test it in a lab |
07:59.05 | djin | I will :) |
07:59.07 | sgorilla | then rollback if neccessary if you catch some bugs |
07:59.09 | sgorilla | or fix the bugs =) |
07:59.32 | djin | depends on the bugs |
08:00.01 | djin | oej, will there be an Astricon Europe next year? |
08:00.26 | djin | Or is that something you don't want to think about at this time . . |
08:00.27 | syle2 | extconfig.conf , do i have that just because i;m in cvs or installed asterisk-addons? or is this not in stable? |
08:01.01 | syle2 | just wondering if cvs stable has extconfig.conf |
08:01.03 | oej | Some kind of Astricon Europe will be happening |
08:01.14 | oej | Trying to find a proper format for it and a proper location |
08:01.36 | djin | Cool, looking forward to visting again. |
08:01.49 | djin | Might I suggest Amsterdam ;) |
08:02.26 | djin | Gives a new meaning to the Golden Pub, I guess. |
08:02.40 | oej | Well, a bit too many distractions... Was thinking Karesuando. No one will find anything to do up there, apart from participating in the conference |
08:02.58 | djin | true :) |
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) |
08:07.20 | shido6 | am I still connected here |
08:08.33 | kippi | is there away i can find out what part of my SIP registation is failing? |
08:10.04 | Insanity5 | watch asteirsk log |
08:10.06 | Insanity5 | easiest way |
08:10.14 | Insanity5 | in console, set verbose to like 3 |
08:10.23 | Insanity5 | asterisk -cvvvp starts it this way. |
08:13.21 | Koshatul | has 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) |
08:18.32 | kippi | I 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' |
08:21.33 | RoyK | kippi: that's usually just a bad password or so....... |
08:24.34 | kippi | if I look in /etc/asterisk/extensions.conf I should have a SIP setting for extension 6696? |
08:28.18 | kippi | ok, 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) |
08:39.58 | Koshatul | what could be causing a "SIP/2.0 404 Not Found" when i have incoming calls ? |
08:42.42 | Aze` | Anyone has last florz's patch pls (mirror is down)? |
08:43.39 | kippi | anyone 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) |
08:45.51 | Aze` | kippi> /etc/asterisk/sip_additional.conf |
08:46.09 | Aze` | example |
08:46.38 | loud | Koshatul, user cannot be located by the server. |
08:47.38 | Aze` | kippi> /etc/asterisk/extensions_additional.conf |
08:47.51 | Aze` | kippi> ls -l /etc/asterisk7 ? |
08:50.15 | kippi | I am getting this error now |
08:50.17 | kippi | Asterisk Dynamic Loader Starting: |
08:50.17 | kippi | <PROTECTED> |
08:50.17 | kippi | <PROTECTED> |
08:51.04 | kippi | fixed |
08:51.06 | syle2 | is an area code always 3 digits? |
08:51.10 | syle2 | even international/ |
08:52.00 | syle2 | crap i think some are like 7 |
08:52.26 | Dr_Ray | woo... 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) |
08:53.28 | syle2 | what a pain in the ass |
08:53.45 | syle2 | how am i suppose to select an area code when they aren;t same number of digits |
08:53.51 | syle2 | from database |
08:54.59 | Koshatul | loud: thanks, i found the problem, my dialplan was "exten => s,1,Dial(${ENGINEXT},60,Tt" instead of "exten => 0731111111,1,Dial(${ENGINEXT},60,Tt)" |
08:55.13 | Koshatul | i didn't realise it wouldn't just use the s, exten |
08:56.24 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
08:56.49 | syle2 | whats the internation format? |
08:56.53 | syle2 | international |
08:57.46 | iDunno | 00<countrycode><numberwithoutleading0> |
08:57.50 | iDunno | (usually) |
08:58.10 | syle2 | isn;t it 001 then country code |
08:58.19 | syle2 | err |
08:58.20 | syle2 | 011 |
08:58.25 | kippi | how easy is CDR to get working? |
08:58.55 | syle2 | i'm coding cdr software lol |
08:59.01 | iDunno | 001 would be america, wouldn't it? |
08:59.18 | syle2 | no it would be calls from canada/NA to overseas i think |
08:59.33 | iDunno | 'oh' |
08:59.36 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
08:59.47 | syle2 | same if you were in europe |
08:59.52 | iDunno | from the UK I'm fairly sure it's 00<countrycode><numberwithoutleading0> |
08:59.52 | syle2 | 011 back to canada/US |
09:00.17 | taec_ | From the UK + Ireland, it is. |
09:00.17 | cjk | hi,did anyone get the Playback application working in features.conf (DYNAMIC_FEATURES) |
09:00.29 | taec_ | 001 dials the states from the UK. |
09:00.44 | kippi | if 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? |
09:00.51 | iDunno | I'm glad I'm not going mad :) |
09:01.03 | taec_ | :) |
09:02.13 | syle2 | well how about the phone number itself |
09:02.25 | syle2 | is it always 7? |
09:02.30 | syle2 | maybe i can parse backwards |
09:02.36 | syle2 | for teh country code |
09:03.52 | *** join/#asterisk shanky (n=shanky@238.Red-80-33-29.staticIP.rima-tde.net) |
09:03.58 | shanky | good afternoon |
09:04.15 | diamon | Or morning, as the TZ may be! |
09:05.04 | shanky | just 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 |
09:08.49 | Gibster | 11:00 -!- ShintaLT [sdfsdfsdf@sg-749.telkomadsl.co.za] has joined #wdsl |
09:10.38 | kippi | is there a command to list all registered sip connections |
09:15.03 | loud | kippi, as in users or carriers. |
09:18.55 | kippi | users |
09:19.10 | loud | sip show peers |
09:22.30 | *** join/#asterisk Newbie___ (i=me@60.48.53.104) |
09:23.16 | Newbie___ | 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) |
09:24.58 | shido6 | yeah |
09:25.06 | shido6 | Background |
09:25.49 | Newbie___ | shido6: i am playing a dial tone exten => s,3,Playtones(350+440) |
09:26.35 | shido6 | and you expect something back? |
09:26.38 | kippi | I have my two sip phones connected, but if I try and ring either phone from the other phone it dosn't ring at all |
09:27.05 | Newbie___ | expect user to enter DTMF after the tone is played |
09:27.08 | shido6 | when you receive dtmf what do you want to do with those tones? |
09:27.34 | Newbie___ | stop the tones and read the dtmf |
09:27.39 | shido6 | and do what |
09:28.01 | Newbie___ | and then dial the desired number |
09:29.34 | shido6 | what number |
09:29.46 | shido6 | is there a pattern |
09:29.47 | shido6 | ? |
09:29.53 | shido6 | that all the numbers might match |
09:29.54 | shido6 | ? |
09:31.00 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
09:31.44 | cjk | damn 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..... |
09:32.10 | shido6 | Newbie___: |
09:32.12 | shido6 | http://pastebin.ca/26037 |
09:33.52 | rob112 | Hi can anyone help with dsp call progress? |
09:36.35 | kippi | it says the two lines have connected but the other phone never rang. Anyone got any ideas? |
09:37.05 | shido6 | kippi: just a sip.conf / extensions.conf snafu |
09:37.15 | shido6 | users and peers |
09:37.31 | shido6 | use a user to make a call and a peer to dial a phone |
09:39.50 | *** join/#asterisk pashah (n=pashah@ns.itconnection.ru) |
09:39.55 | pashah | hello |
09:40.11 | pashah | will digium's te110p work in pcix slot? |
09:40.11 | RoyK | olleh |
09:40.21 | RoyK | guess so |
09:40.28 | RoyK | i'm using te410p in one |
09:40.44 | pashah | RoyK: thanks |
09:40.46 | pashah | cheers! |
09:43.55 | kippi | shindo6: so where do I need to look to make the changes? |
09:45.34 | RoyK | kippi: read the docs |
09:45.39 | RoyK | ~docs |
09:45.40 | jbot | hmm... 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 |
09:46.07 | RoyK | kippi: http://pastebin.ca/26038 |
09:46.28 | kippi | ok |
09:58.18 | syle2 | could i get some feature requests plz |
09:58.27 | syle2 | for asterisk billing system |
10:02.44 | *** join/#asterisk Tjief (n=Tjief@r0x0r.dk) |
10:05.00 | iDunno | right - 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) |
10:05.21 | X-Rob | linphone apparentl |
10:05.22 | X-Rob | y |
10:05.44 | MatsK | iDunno: Try X-Lite for linux |
10:06.02 | Aze` | Anyone has last florz's patch for zaphfc ? |
10:07.00 | rob112 | Hi does anyone know the best way to detect tones over IAX2 ALAW ? |
10:07.55 | iDunno | linphone appears to suck ;) |
10:08.03 | iDunno | MatsK: hmm. is there source? |
10:09.10 | *** join/#asterisk squirrelv5 (n=squirrel@203.131.188.170) |
10:09.30 | squirrelv5 | hi there to all |
10:11.22 | MatsK | iDunno: I don't know |
10:12.40 | iDunno | doesn'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) |
10:16.59 | ful|work | hey |
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) |
10:29.09 | MozartsGhost | ~dialtone |
10:29.16 | MozartsGhost | bleh :/ |
10:30.12 | *** join/#asterisk shanky (n=shanky@238.Red-80-33-29.staticIP.rima-tde.net) |
10:30.15 | shanky | hi again |
10:30.24 | shanky | just 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 |
10:35.32 | MozartsGhost | anybody know how to detect ringtone on a pstn line, off a Digium TDM400P card ? |
10:35.51 | MozartsGhost | want to detect dialtone before it tries to dial. |
10:37.08 | MozartsGhost | can't seem to find any available docs/info on the subject. |
10:37.24 | MozartsGhost | guess thats a no brainer then hey. |
10:37.25 | MozartsGhost | oh well. |
10:38.33 | MozartsGhost | ek gan vandag my account kry hoopelik. |
10:38.44 | MozartsGhost | ewps, wrong window. |
10:38.53 | X-Rob | MozartsGhost - can't be done. Sorry. |
10:39.34 | MozartsGhost | k, 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) |
10:48.29 | trym | It 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? |
10:49.48 | X-Rob | your nat is broken? |
10:49.54 | RoyK | trym: well. decreasing the register timeout might be a good idea :) |
10:50.12 | RoyK | X-Rob: nat ususally needs far lower register timeout anyway |
10:50.26 | X-Rob | RoyK - I hate nat. lots. |
10:50.50 | RoyK | X-Rob: well. we have a few thousand customers, most of them behind nat..... |
10:50.52 | trym | royk: 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? |
10:50.55 | RoyK | you learn to deal with it |
10:51.17 | RoyK | trym: from sip.conf |
10:51.17 | RoyK | ;maxexpiry=3600 ; Max length of incoming registration we allow |
10:51.17 | RoyK | ;defaultexpiry=120 ; Default length of incoming/outoing registration |
10:51.28 | RoyK | play with those |
10:51.34 | trym | thanks |
10:51.47 | RoyK | trym: hvor er du kobla opp? |
10:51.51 | trym | ip24 |
10:51.54 | RoyK | ah |
10:51.54 | trym | :/ |
10:52.16 | RoyK | trym: kom til asterisk-no |
10:52.21 | trym | ja husker deg.. du jobbet i briiz før |
10:52.23 | RoyK | eller #ip24 |
10:52.26 | trym | så ble de "slått sammen" right ? |
10:52.31 | RoyK | vi spiste ip24..... |
10:52.35 | trym | haha |
10:52.38 | trym | nice |
10:55.33 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-43-69.44-151.net24.it) |
10:55.44 | kippi | I 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 |
10:59.42 | RoyK | kippi: pastebin the output and the config |
10:59.48 | RoyK | ~pastebin |
10:59.49 | jbot | pastebin 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) |
11:04.42 | kippi | http://pastebin.ca/26043 |
11:04.59 | gordonjcp | hey folks |
11:05.15 | e-Hernick | hey folk-man |
11:05.33 | gordonjcp | does anyone in here have any experience of the Astralis X101P card? |
11:06.32 | e-Hernick | isn't that a cheap 20$ digium ripoff card ? |
11:07.39 | gordonjcp | seems to be |
11:08.07 | gordonjcp | e-Hernick: worth bothering with, or not? |
11:08.53 | e-Hernick | well, it's never going to give very good results, but it's a fun toy for the price |
11:09.15 | gordonjcp | why would it be any different to an X100P? |
11:10.02 | e-Hernick | cheap amps. cheap caps. cheap everything. no support. |
11:10.24 | e-Hernick | And the X100P isn't a very good card to start with. |
11:10.32 | cjk | hi, 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 |
11:10.56 | gordonjcp | e-Hernick: tbh, I doubt there's much of a difference between the components. |
11:11.22 | e-Hernick | well then, it should work just fine |
11:11.28 | e-Hernick | what's wrong with yours ? |
11:11.52 | gordonjcp | e-Hernick: mmm. For 3 quid I can hardly go wrong, can I? |
11:12.24 | e-Hernick | well, it can sound so bad as to be useless |
11:13.08 | kippi | has 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) |
11:13.47 | X-Rob | kippi - you have two '1's |
11:13.59 | kippi | ah |
11:14.08 | bintut | where can i read the basics of asterisk? its concepts and definitions? the bandwidth requirement? difference between sip and aix? |
11:14.16 | X-Rob | ~docs |
11:14.18 | jbot | i 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 |
11:15.10 | X-Rob | kippi - I also get the feeling that you're using asterisk@home |
11:15.22 | kippi | nope |
11:15.45 | kippi | asterisk@home just works |
11:15.47 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
11:16.20 | bintut | jbot: ok. thanks.. |
11:16.37 | bintut | what's the usual codecs being used in asterisk? |
11:16.56 | X-Rob | bintut - ulaw/alaw, gsm, ilbc, g729 etc. |
11:17.06 | X-Rob | bintut - 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) |
11:17.16 | X-Rob | kippi - you're not using A@H? |
11:17.23 | kippi | nopw |
11:17.28 | kippi | I have AMP installed |
11:17.33 | t0ke | hello |
11:17.35 | X-Rob | OK. |
11:17.42 | X-Rob | If you're using AMP, you don't want to mess with extensions.conf |
11:17.49 | X-Rob | you use exensions_custom.conf |
11:17.50 | t0ke | anyone know if there is any limitation about E1 cards to use with Asterisk free? |
11:18.17 | X-Rob | kippi - and amp questions should be on #amportal |
11:18.23 | X-Rob | Bug me over there. |
11:19.10 | bintut | X-Rob: i'm checking the site.. thanks.. :) |
11:23.38 | t0ke | anyone know if there is any limitation about E1 cards to use with Asterisk free? |
11:24.38 | power1 | Im 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? |
11:26.50 | bintut | how will i know how much bandwidth will i need for my voip setup? |
11:26.58 | X-Rob | ~bandwidth |
11:27.00 | jbot | rumour 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. |
11:27.00 | bintut | that is, voip+pbx? |
11:27.18 | X-Rob | bintut - all this, and more, is available on voip-info.org |
11:27.19 | X-Rob | really. |
11:27.23 | X-Rob | there's a pile of information there |
11:27.24 | iDunno | hmm. iaxcomm kinda works. |
11:27.32 | X-Rob | try googleing for 'site:voip-info.org bandwidth' |
11:27.53 | bintut | X-Rob: yeah. but i don't understand it really the bandwidth calculator? |
11:28.11 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
11:28.23 | bintut | X-Rob: in your opinion, for a 10 local lines, how much bandwidth will you allocate? |
11:28.56 | X-Rob | bintut - 1mbit. |
11:28.58 | bintut | let's say, how much bandwidth will you shape for each line? |
11:29.09 | X-Rob | Because I'd be using ulaw. |
11:29.15 | X-Rob | alaw even |
11:29.32 | bintut | X-Rob: you mean, 100kbps per line? |
11:29.37 | X-Rob | yep |
11:29.45 | bintut | oh! that's big enough |
11:29.58 | bintut | what are the other options besides "alaw" |
11:30.00 | bintut | ? |
11:30.20 | X-Rob | show codecs |
11:30.36 | X-Rob | bintut - it would be easier if you grabbed a spare PC, downloade asterisk@home 2.0beta[something] and have a play |
11:31.56 | bintut | X-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) |
11:37.05 | power1 | Im 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? |
11:37.45 | MatsK | <power1> check /var/log/messages for info |
11:37.51 | tzafrir_laptop | hmm, app_segfault from bristuff just proved itself very useful for the case of asterisk is run withwith -p |
11:38.33 | t0ke | anyone have more one T1/E1 with Asterisk? |
11:38.39 | tzafrir_laptop | Though a segfault is known to be a, well, feature, of some other parts of asterisk in some extreme circumstances |
11:38.47 | bintut | thanks guys.. gtg now.. |
11:39.00 | gordonjcp | ls |
11:39.58 | power1 | MatsK, looks normal except for this "Oct 20 19:31:29 asterisk1 init: Id "s0" respawning too fast: disabled for 5 minutes" |
11:41.00 | power1 | I 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.. |
11:41.13 | power1 | you think this is a hardware problem. |
11:41.17 | tzafrir_laptop | power1, why the &*^*^%&%^ would asterisk be started directly from /etc/inittab ? |
11:41.40 | MatsK | <power1> What card is it, is it a cloned X100P ? |
11:41.49 | power1 | tzafrir_laptop, this is a clean install. and thats what in the logs...you tell me? |
11:42.09 | power1 | MatsK, Digium TDM400P FXO/FXS Card |
11:42.18 | tzafrir_laptop | "asterisk1" is the hostname, then |
11:42.24 | MatsK | ok |
11:42.28 | tzafrir_laptop | So the message is unrelated to asterisk, I gather |
11:42.33 | power1 | tzafrir_laptop, yip thats right |
11:42.42 | X-Rob | tzafrir_laptop - that's saying there's a mgetty on ttys0 that's trying to talk to the modem and failing |
11:42.43 | tzafrir_laptop | grep ^s0 /etc/inittab |
11:42.57 | X-Rob | ignore that |
11:43.21 | power1 | tzafrir_laptop, s0:12345:respawn:/sbin/agetty -i -h -L 9600 ttyS0 vt100 |
11:44.05 | tzafrir_laptop | sed -i -e 's/^s0:/#&/' /etc/inittab && telinit -q # assuming there aren't typos and mistakes here |
11:44.40 | tzafrir_laptop | This should help you to ignore that... |
11:44.41 | power1 | I 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. |
11:45.45 | MikeJ[Laptop] | morning |
11:46.56 | power1 | could this be a problem " Registered tone zone 0 (United States / North America)" I am in Africa? |
11:51.46 | MatsK | <power1> You are talking about two problems! Hangup problem and kudzu detection. |
11:51.50 | t0ke | Anyone have Asterisk with more one E1/T1 interfaces or know if there is any limitation about number of E1/T1 supported? |
11:52.15 | MatsK | <power1> Have you updated kuzu ? |
11:52.59 | Ahrimanes | t0ke: i'd say number of cards supported is mostly a performance issue and pci bus slot issue |
11:53.08 | power1 | MatsK, 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. |
11:53.45 | t0ke | Ahrimanes: someone comment me that I needed to buy Asterisk Business Edition if I wanted more 120 channels |
11:53.59 | syle2 | where can i get a list of every area code in the world |
11:54.20 | power1 | MatsK, 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? |
11:54.23 | MatsK | syle2: e164.org |
11:54.34 | t0ke | Ahrimanes: talked me about limitation between commercial and free license |
11:54.51 | t0ke | Ahrimanes: I dont know if it is true |
11:54.57 | syle2 | looking at the page, where? |
11:54.59 | Ahrimanes | t0ke: i doubt there's such a limit.. but then again, most servers probably are better of with around 120 channels perfromance wise |
11:55.04 | Ahrimanes | t0ke: i dont think it's there |
11:55.17 | t0ke | ok..tnks |
11:55.54 | MatsK | <power1> No, but try update the system (yum update) |
11:55.55 | t0ke | Ahrimanes: maybe then it will be more depurate for supporting more 120 channels |
11:56.07 | Ahrimanes | t0ke: depurate? |
11:56.17 | MozartsGhost | who |
11:56.20 | MozartsGhost | ewps. |
11:56.27 | t0ke | Ahrimanes: optimized |
11:56.57 | e-Hernick | Well, 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. |
11:57.00 | Ahrimanes | t0ke: that might well be, asterisk business edition is based on cvs code newer than 1.0.9 and optimized by digium |
11:57.45 | e-Hernick | And we're talking below 10$ per channel |
11:57.53 | t0ke | Ahrimanes: I understand |
11:58.04 | t0ke | Ahrimanes: tnks by your feedback |
11:58.16 | Ahrimanes | e-Hernick: many things are good ideas, but bottomline ends up with what decision management makes |
11:58.20 | Ahrimanes | t0ke: np :) |
11:58.34 | e-Hernick | Sure, but the decision is based upon the recommendations they receive. |
11:58.46 | t0ke | Ahrimanes: do you know what about perfomance between Digium and Sangomas cards? |
11:58.55 | power1 | MatsK, 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? |
11:59.14 | Ahrimanes | t0ke: 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) |
11:59.49 | t0ke | Ahrimanes: yeah..and surely will have best support for Asterisk |
12:00.08 | supa_thygar | power1 where in Africa ? |
12:00.46 | kippi | has anyone got madplay to work with asterisk for music on hold? |
12:00.58 | Ahrimanes | t0ke: i should think so yes :) |
12:01.07 | power1 | supa_thygar, Johannesburg South Africa |
12:01.16 | supa_thygar | Pretoria SA here |
12:01.18 | Ahrimanes | t0ke: allthough sangoma do seem to take pride in asterisk support |
12:01.37 | power1 | supa_thygar, kewl <grin> |
12:02.05 | power1 | supa_thygar, hows ur asterisk skills? |
12:02.12 | t0ke | Ahrimanes: Sangoma is going out one new card next monday with echo cancellation and it is 126 ms..really nice |
12:02.27 | supa_thygar | well not good but we have a system up and running |
12:02.37 | e-Hernick | The digium card with EC is only 16 ms when running all channels, right ? |
12:02.38 | t0ke | Ahrimanes: I received an email from Sangoma yesteday explained it. |
12:02.46 | Ahrimanes | t0ke: oh nice, how many ports? |
12:02.52 | t0ke | e-Hernick: yes |
12:02.56 | t0ke | Ahrimanes: quad |
12:03.04 | supa_thygar | power1 : i have a problem with asterisk dipping but thats about it |
12:03.05 | Ahrimanes | t0ke: ok, any pricing info? |
12:03.12 | t0ke | Ahrimanes: and will go out next year 8 ports |
12:03.26 | t0ke | Ahrimanes: one sec |
12:03.30 | Ahrimanes | t0ke: :) |
12:04.02 | t0ke | Ahrimanes: look this: "If you are seeking hardware based echo cancellation, the A104d- T1/E1/J1 |
12:04.03 | supa_thygar | power1: you can use ZA in the zone |
12:04.34 | power1 | supa_thygar, did u have a problem with cellular phones calling in and asterisk not dropping the fx0 channel when the person hangs up? |
12:04.36 | t0ke | Ahrimanes: for resellers or isp...$1890 |
12:04.39 | *** join/#asterisk damned (n=vpol@prior.lanck.net) |
12:04.53 | e-Hernick | ouch, a104d is 2700$ list |
12:04.57 | supa_thygar | power1 nope |
12:04.57 | power1 | MatsK, its found about 15 packages it want to upgrade...should i let it? |
12:05.11 | t0ke | e-Hernick: yes..in list it is price |
12:05.16 | Ahrimanes | t0ke: oh.. that's not too bad |
12:05.17 | RoyK | e-Hernick: what is that d for? |
12:05.17 | power1 | MatsK, sorry for the juvenile questions..ive never used centos before ..i only know gentoo. |
12:05.24 | e-Hernick | the d is for echo cancellation |
12:05.31 | Ahrimanes | no, for dsp |
12:05.36 | power1 | supa_thygar, <grin> I allways get the weird and wonderfuls! |
12:05.37 | supa_thygar | power1 : i know what is wrong |
12:05.48 | power1 | supa_thygar, im all ears |
12:05.55 | t0ke | Ahrimanes: http://www.sangoma.com/products/p_aft-104-specs.htm |
12:05.56 | Bonzai009 | hi all |
12:06.04 | Ahrimanes | t0ke: yes, found it :) |
12:06.07 | t0ke | ;) |
12:06.12 | supa_thygar | power1 in one of the config files there is a setting that you can reverse polarity put it on yes |
12:06.15 | Ahrimanes | t0ke: know the price for te411p digium card? |
12:06.37 | power1 | supa_thygar, what does that do? |
12:06.58 | t0ke | Ahrimanes: I have price for it here in Spain..in EUR...go to convert it to dollars if you want |
12:07.00 | supa_thygar | it makes the fx0 drop the line if a cellphone hangs up |
12:07.03 | power1 | supa_thygar, and where do i put ZA in the zone? |
12:07.10 | Bonzai009 | is 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 |
12:07.28 | Ahrimanes | t0ke: just euro is fine :) |
12:07.32 | supa_thygar | power1 : in zaptel.conf |
12:07.38 | power1 | supa_thygar, thanks....so u must have had that problem ...if you know how to sort it out. |
12:07.39 | t0ke | Ahrimanes: 2100 Eur |
12:07.56 | supa_thygar | power1 yea just could not remember |
12:08.11 | Bonzai009 | power1 i had to put it in my indications.conf as well |
12:08.23 | Ahrimanes | t0ke: wow.. ~$700 price diff |
12:08.24 | supa_thygar | power1 : i have a problem with clarity and dipping at the moment |
12:08.33 | Ahrimanes | t0ke: maybe one should try out a sangoma card.. :) |
12:08.50 | Bonzai009 | has any one had issues with asterisk on xeon servers?.. |
12:08.59 | t0ke | Ahrimanes: 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 |
12:09.18 | Bonzai009 | particularely were the server just hangs or the pri just freezes up |
12:09.18 | t0ke | Ahrimanes: then to have 4 cpus with one E1 on each one |
12:09.23 | power1 | supa_thygar, is dipping kinda like volume fluctuation? |
12:09.33 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
12:09.47 | Bonzai009 | power1 kinda like telkom trying to provide a 1st world service hehehe |
12:10.08 | supa_thygar | power1 yea |
12:10.34 | supa_thygar | power1: it's like somtimes the calls are 100% and then all the sudden it turns into crap |
12:10.39 | power1 | Bonzai009, LOL, good old hellkom......did you see telkom dropped the lawsuit against hellkom.co.za |
12:10.55 | Ahrimanes | t0ke: 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 |
12:11.00 | Bonzai009 | power1 yeah :> makes me happy now we need a sno to come in :| |
12:11.10 | power1 | supa_thygar, weird....are u running asterisk @ home..or normal version on top of a distro? |
12:11.27 | Bonzai009 | any one had issues on xeon servers with asterisk 1.0.9 ?.. |
12:11.31 | supa_thygar | power1 running it at my office |
12:11.40 | Bonzai009 | my pri keeps freezing on xeon |
12:11.47 | Bonzai009 | all ports lock up until i reboot |
12:11.49 | power1 | Bonzai009, yeah...ASAP....did u read the icasa finding on telkom ADSL practices? |
12:11.53 | Bonzai009 | hellkom tested the line |
12:12.12 | supa_thygar | power1: using 2 TDM400p |
12:12.29 | Bonzai009 | power1 yeah what a rip off R2k a month for a 30gb adsl cap |
12:12.49 | supa_thygar | power1: running normal version |
12:12.58 | Bonzai009 | power u have a link to that icasa adsl practice. |
12:13.54 | Bonzai009 | power1 u on adsl ?.. |
12:14.01 | power1 | Bonzai009, 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> |
12:14.22 | supa_thygar | power1 lol :) |
12:14.32 | power1 | supa_thygar, what distro u running it on ( gentoo bsd etc)? |
12:14.40 | supa_thygar | power1: @least some ne is winning |
12:14.43 | Bonzai009 | power1 they dont that to me as well self install and they never sent me a bill hehehe |
12:14.49 | supa_thygar | normal dist on gentoo |
12:14.54 | Bonzai009 | power1 can i private u quickly?.. |
12:16.09 | power1 | Bonzai009, sure go 4 it |
12:16.46 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
12:17.37 | power1 | supa_thygar, do i do that reverse polarity in the file zapata.conf? |
12:17.59 | supa_thygar | power1: cant remember which file |
12:18.01 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
12:18.18 | supa_thygar | power1: let me check and see if i can find it |
12:18.53 | power1 | supa_thygar, thanks man! |
12:21.56 | kippi | has 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 |
12:23.09 | kippi | on voicemail to I don't hear the voice asking for the password |
12:23.42 | _m_ | Do you have any zaptel based timing source? |
12:24.51 | kippi | i have a TE110P card installed but there is no line in it at the mo |
12:25.32 | _m_ | That shouldn't matter. |
12:25.32 | synthetiq | <PROTECTED> |
12:25.34 | Bonzai009 | any 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) |
12:31.02 | Bonzai009 | any one had issues on xeon servers with isdn pri card where all ports just freeze up untill you reboot asterisk |
12:31.20 | Ahrimanes | reboot asterisk or the server? |
12:31.38 | *** join/#asterisk coppice (n=chatzill@219.199.17.210.dyn.pacific.net.hk) |
12:31.43 | Bonzai009 | Ahrimanes done that give it 10 to 15 minutes and it freezes |
12:31.47 | Bonzai009 | its a genuine intel board |
12:32.24 | Ahrimanes | Bonzai009: i meant.. you said until you reboot asterisk.. do you mean reboot the server or restart asterisk? |
12:32.46 | Bonzai009 | Ahrimanes reboot the whole server even riped power outa ups |
12:33.14 | Ahrimanes | Bonzai009: ah ok.. hm havent had a problem like that |
12:33.28 | Bonzai009 | its plain random can be any time but normaly with in 15 minute max |
12:33.32 | Bonzai009 | hmmm |
12:33.48 | Bonzai009 | maybe 1.2 will fix the problem any idea when tis going to be stable?.. |
12:34.09 | Ahrimanes | not really, rsn i hope :) |
12:34.58 | Bonzai009 | Ahrimanes 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) |
12:35.23 | Ahrimanes | Bonzai009: well many in here say it's better than stable.. but i do tend to stick to stable as well :) |
12:35.26 | e-Hernick | Hey, 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) |
12:38.26 | supa_thygar | Any 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) |
12:38.37 | Inkubot | good morning |
12:39.01 | Inkubot | i've got a cisco ata 186 and when i make a call it takes about 15 seconds, any ideas why is that ? |
12:39.15 | Bonzai009 | supa_thygar check the codecs on your phones and in your iax.conf and sip.conf |
12:39.20 | supa_thygar | o 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) |
12:40.39 | supa_thygar | Bonzai009 is the iax.conf not just if you connect to a nother asterisk server ? |
12:42.02 | supa_thygar | Bonzai009 what codecs are you using ? |
12:43.44 | Bonzai009 | supa_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 |
12:44.18 | supa_thygar | Bonzai009 i'm not using it via the internet |
12:44.34 | Bonzai009 | supa_thygar well just incase u were i sugested the iax lol |
12:44.46 | power1 | supa_thygar, whats your phones using " ulaw" ? |
12:45.01 | supa_thygar | power1: yep ulaw it is |
12:45.09 | power1 | Bonzai009, hmmm i think my phones only supports sip...what a bummer. |
12:45.39 | iDunno | IAX is better for over the internet as it's, like, 1 port and everything travels over it... |
12:45.42 | iDunno | SIP is a nasty mess. |
12:45.50 | kippi | for moh do i need to add anything extra for sip? |
12:46.03 | Inkubot | no kippi |
12:46.16 | iDunno | (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) |
12:46.22 | power1 | iDunno, 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) |
12:46.28 | lehel | hello there |
12:46.45 | iDunno | power1: we're using SIP phones with a local asterisk server |
12:47.07 | iDunno | which seems to work for us, so far. |
12:47.25 | supa_thygar | Any 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 |
12:47.49 | kippi | any ideas why I can't hear anything? is there a * number I can dial you see if it is working or not? |
12:48.26 | Inkubot | kippi musiconhold=default in zapata.conf |
12:48.51 | Inkubot | and some mp3's in /var/lib/asterisk/mohmp3 |
12:48.57 | mmlj4 | kippi: 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) |
12:49.36 | Bonzai009 | iDunno lol i know sip gave me gray hairs so i upgraded my phones out there to iax |
12:49.39 | Inkubot | damn cisco phone, take so long to register |
12:50.17 | Bonzai009 | I'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) |
12:51.22 | supaigtr | Hello. |
12:53.58 | kippi | musiconhold=default is in zapata |
12:56.38 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
12:59.42 | Katty | mew. |
13:00.47 | Katty | RoyK: :< |
13:01.56 | Inkubot | i resolve the problem.. extension+# and dials |
13:03.14 | diamon | So, 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) |
13:04.04 | gordonjcp | diamon: stanavoice? |
13:04.07 | Ariel_ | Morning everyone |
13:04.07 | Katty | diamon: lots. |
13:04.15 | Katty | hihi, Ariel_ *hug* |
13:04.23 | Ariel_ | hello Katty |
13:04.54 | iDunno | morning Katty :) |
13:07.16 | diamon | gordonjcp: Cool, I'll look them up... |
13:07.27 | mutilator | mm good breakfast today, reheated breadsticks |
13:07.34 | diamon | Katty: I'm just fishing for names people suggest. Usually hurts less that way. |
13:08.29 | diamon | If nothing else, sometimes I get names to avoid, which is almost as good. :) |
13:08.40 | Katty | k |
13:08.46 | Katty | hihi, iDunno |
13:10.16 | RoyK | iPod, and iChat, and iDunno? |
13:12.52 | *** join/#asterisk Pikoro (n=webmaste@db.sunny-net.ne.jp) |
13:13.30 | Pikoro | i got a wierd problem here.. ;D |
13:13.53 | Pikoro | i can make outbound sip calls, but cannot receive incoming sip calls |
13:14.07 | Pikoro | and internal calls don't work |
13:14.21 | Pikoro | i've been working on this for like 2 weeks now :D |
13:15.26 | supa_thygar | Any 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 |
13:15.37 | Ariel_ | Pikoro, are you using AMP or some other predone system? |
13:15.44 | *** join/#asterisk CANO-1982 (n=cano1982@201.255.50.206) |
13:15.45 | Pikoro | Ariel_, no |
13:15.49 | Pikoro | tried it though |
13:16.23 | iDunno | RoyK: I am in no way affiliated with Apple ;) |
13:16.24 | Pikoro | i am positive that all I'm missing is some kind of internal route.. like it can't find the context or something |
13:16.39 | Ariel_ | Pikoro, When you call extension to extension what does the CLI say in sip debug on |
13:16.44 | Pikoro | all i get is a busy signal |
13:16.44 | iDunno | RoyK: I was iDunno loooong before they started prefixing things with i, dammit ;) |
13:17.25 | Pikoro | well, in my client i get a 486: Busy Here |
13:17.51 | Pikoro | extension is reporting busy and has no voicemail |
13:18.23 | Pikoro | what sucks is that at any one time, I have had all 3 parts working |
13:18.34 | Pikoro | had internal and from pstn working but no outgoing |
13:18.39 | *** join/#asterisk hadi57 (n=al_moghr@83.136.8.206) |
13:18.48 | Pikoro | so i tried to fix outgoing calls and now no incoming calls work |
13:18.49 | RoyK | iDunno: sue them :) |
13:19.15 | Pikoro | i'm about ready to just pay someone to log in and fix it for me :\ |
13:19.17 | Ariel_ | Pikoro, post your dial section in pastebin.ca for us to take a look |
13:19.39 | Pikoro | ok, the conf files i have now were created with amp but i no longer have amp installed |
13:19.46 | nextime | anyone with the latest spandsp 0.0.2pre21 on a debian sid with libtiff 3.7.4 and a TE400P quad pri digium card? |
13:19.49 | Pikoro | dial section from extensions.conf? |
13:19.58 | coppice | iDunno: they used eXXXX, then iXXXX. They'll soon move on to oXXXX and leave you alone |
13:20.00 | lehel | Pikoro, from CLI |
13:20.27 | iDunno | heh - true, true ;) |
13:21.47 | Pikoro | damnit... i can't copy and paste from this stupid dos window.. scrolling it gets everything all screwed up |
13:21.52 | Pikoro | i 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) |
13:24.44 | lehel | paste it |
13:25.02 | Inkubot | asterisk doesn't recognize the incoming ID |
13:25.09 | tzanger | coppice: heh |
13:25.11 | Inkubot | any ideas ? |
13:25.20 | tzanger | the iDunno... haha |
13:25.29 | tzanger | followed by the iDontGivaShit |
13:25.58 | *** join/#asterisk ast_freak (n=jesse@hades-out.universalsystems.net) |
13:25.58 | Pikoro | http://pastebin.ca/26050 |
13:26.43 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
13:26.52 | RoyK | coppice: iirc the eMac came out _after_ the iMac, so perhaps aMac comes next..... |
13:26.55 | RoyK | AA Mac |
13:28.26 | Pikoro | bMac |
13:29.01 | Ariel_ | Pikoro, your not sending the correct info to dialparties |
13:29.38 | Pikoro | ok, where should I go to fix that? |
13:29.42 | Pikoro | extensions.conf no? |
13:29.47 | *** join/#asterisk nexis (n=nexis@12-207-56-108.client.mchsi.com) |
13:30.19 | Pikoro | anyplace I can grab some known good conf files and just change the registration string, etc...? |
13:30.40 | Pikoro | ok, that was just an internal call |
13:31.03 | Pikoro | shall i paste an internal call now/ |
13:31.15 | Pikoro | err.. incoming |
13:31.35 | nexis | Pikoro, 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) |
13:31.46 | Pikoro | nexis, that's what i did :D |
13:33.29 | Pikoro | http://pastebin.ca/26051 <- a call coming in from the PSTN |
13:33.35 | Pikoro | or not coming in rather.. heh |
13:33.45 | Ariel_ | 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 |
13:34.19 | tzafrir_laptop | AMP's trunks setup is not very helpful. Still requires much docs reading |
13:34.21 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
13:34.30 | Pikoro | ok |
13:34.36 | tzafrir_laptop | In fact, I found it even more confusing than hand-configuring asterisk |
13:34.41 | Pikoro | i wanted to do this without amp :D |
13:34.48 | tzafrir_laptop | At least for a simple setup |
13:34.49 | Pikoro | i went back to the default conf files |
13:34.54 | Ariel_ | Pikoro, then remove them all start with the samples |
13:34.55 | Pikoro | this should be a simple setup |
13:35.23 | Pikoro | I did that one time and couldn't get anything to work... |
13:35.30 | Pikoro | :( |
13:35.35 | Pikoro | i'll give it another shot |
13:35.43 | Pikoro | time to go home now, it:s almost 11pm |
13:35.54 | Ariel_ | Pikoro, print them out then follow the logic that they have. |
13:36.01 | Pikoro | k |
13:36.11 | Pikoro | believe it or not, it IS all starting to make sence :D |
13:36.20 | Pikoro | at least I'll know what I'm doing when i'm done |
13:36.21 | Pikoro | heh |
13:36.21 | Ariel_ | you need to do some reading as to rules and use macro which will make things easy to setup. |
13:36.57 | nexis | is it possable to set flags with a call file like with the dial string, such as H or L |
13:37.21 | Ariel_ | 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 |
13:38.55 | Ariel_ | 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) |
13:42.29 | *** join/#asterisk ful|work (n=fulgas@213.58.130.46) |
13:43.03 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
13:43.07 | *** join/#asterisk CANO-1982 (i=alejandr@201.255.50.206) |
13:44.19 | Ariel_ | 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) |
13:46.00 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
13:46.51 | Pikoro | i got it |
13:46.53 | Pikoro | been reading it |
13:47.05 | Pikoro | problem is there is nothing about sip in there really |
13:47.08 | Pikoro | just iax |
13:47.17 | CANO-1982 | I have a problem with my TDM400p board |
13:47.18 | Pikoro | it very breifly covers sip |
13:47.18 | *** join/#asterisk Los415 (n=los415@ssf-office.corp.race.com) |
13:47.20 | Inkubot | mi asterisk in some calls doesn't hungup and the lines is busy for a time. |
13:47.48 | Inkubot | any idea ? |
13:47.52 | CANO-1982 | I have to unplug and plug again each time y have an incoming call on my FXO module |
13:47.59 | CANO-1982 | any idea? |
13:48.38 | syle2 | feature requests on billing system? |
13:48.52 | CANO-1982 | Ive tried aswerpolarity.., callooprogres, busydetect, hanguponpolarity.., loopstart ang kewlstart |
13:49.13 | snitt | Inkubot: search for disconnect provision |
13:49.30 | Los415 | syle being able to re rate calls |
13:49.37 | *** join/#asterisk astoria (n=haydenth@66.235.201.217) |
13:50.06 | syle2 | what do you mean? |
13:50.17 | Inkubot | thnks snitt |
13:50.18 | CANO-1982 | Inkubot, have you tried unpluging the phone cable? |
13:50.20 | Inkubot | google now |
13:50.39 | Inkubot | yeps.. but thats it is an ugly solution |
13:50.40 | Pikoro | well, talk to ya'll tomorrow |
13:50.53 | Los415 | like if a call gets rated at a certain rate |
13:50.54 | *** part/#asterisk Striker`Work (n=striker@lunar-linux/developer/pdpc.bronze.striker) |
13:50.56 | Los415 | and customer calls |
13:51.21 | CANO-1982 | well, its a start |
13:51.38 | syle2 | well what i was thinking was this.... |
13:51.38 | Los415 | need to re rate calls to |
13:51.47 | CANO-1982 | I cant try teh groundstart signaling |
13:51.51 | CANO-1982 | maybe that works |
13:52.19 | syle2 | editable 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 |
13:52.32 | syle2 | if you wanted |
13:52.58 | Los415 | hrmmm |
13:53.05 | Los415 | billing is a nightmare |
13:53.15 | Los415 | i even hate looking at the mile long code |
13:53.25 | Los415 | it's so complex |
13:53.28 | syle2 | you;d hate to see c code i have so far doing this |
13:53.36 | Los415 | we wrote some but it's some complex stuff |
13:53.53 | CANO-1982 | Inkubot, have you tried the things that I told before? |
13:54.06 | syle2 | is that what you mean by re-rate though? |
13:54.11 | Inkubot | when the lines is busy |
13:54.14 | Inkubot | i restart asterisk |
13:54.19 | Inkubot | and just work fine.. |
13:54.29 | Inkubot | when i call from a external number |
13:54.38 | Inkubot | the ivr answer.. but if i end the call.. |
13:54.40 | Katty | where do you find porabolas? |
13:54.43 | Los415 | well like we have had customers call in say this call was rated higher |
13:54.46 | Katty | magnetic fields? |
13:54.47 | Inkubot | the ivr keeps going and then end in the vm |
13:54.54 | Los415 | so we had to re rate the calls to a differnt rate |
13:55.02 | Los415 | which magically changes there invoice up ect. |
13:55.02 | Inkubot | i really don't know why is that |
13:55.11 | syle2 | owww |
13:55.14 | *** join/#asterisk _santiago_ (n=santiago@208.195.215.158) |
13:55.25 | syle2 | for that month only? |
13:55.39 | Los415 | it could be anything |
13:55.41 | Los415 | hehe |
13:56.04 | *** join/#asterisk gambolputty (n=gambolpu@72.240.242.4) |
13:56.05 | tzanger | Katty: some women? |
13:56.31 | tzanger | depends though, it ranges from x=y^2 to x=0.01y^2 |
13:56.47 | CANO-1982 | well, 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) |
13:57.24 | tzanger | Actually 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 |
13:58.48 | CANO-1982 | have you tried that?Inkubot? |
13:58.57 | iDunno | tzanger: are you *sure* it was the dog? |
13:59.14 | tzanger | iDunno: 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) |
14:01.32 | QbY | anyone 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) |
14:06.55 | spiekey | howdy #asterisk! |
14:08.23 | supaigtr | sup |
14:08.50 | ful|work | anyone got this type of error -> http://pastebin.ca/26053 ? |
14:09.37 | supaigtr | Thats a warning. |
14:09.45 | Starcode | I have a problem with a iax <-> iax connection between 2 servers. |
14:10.10 | ful|work | sure |
14:10.18 | ful|work | nevetheless smth must be wrong |
14:10.52 | Starcode | In iax.conf is [server_a] type=friend; username=abc; secret=pass; host=10.2.2.2; context=iax |
14:11.40 | Inkubot | CANO-1982: i don't try that |
14:11.41 | Starcode | If go a Dial(IAX2/server_a/${EXTEN:2},30,r} I can see: |
14:12.09 | Starcode | "Executing Dial("Zap/1-1", "IAX2/server_a/1130|30|r") in new stack |
14:12.17 | Starcode | Called server_a/1130 |
14:12.21 | Katty | tzanger: mew? |
14:12.26 | Katty | tzanger: what are we talking about again? |
14:12.37 | Starcode | Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW |
14:12.39 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
14:12.57 | Starcode | Timestamp: 00012ms SCall: 16384 DCall: 00000 [10.2.2.2:0] |
14:13.11 | Katty | tzanger: parabolic orbits? |
14:13.25 | Starcode | But no ip packet is really sent to 10.2.2.2... i testet it with tcpdump |
14:13.35 | sivana | anyone know what a supervised transfer event looks like for the manager api? |
14:13.42 | Starcode | In the other direction, everything works. |
14:14.12 | Starcode | I 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) |
14:14.29 | tzanger | Katty: you said parabolas |
14:14.33 | synthetiq | fuck |
14:14.36 | tzanger | I said some women have impressive parabolas |
14:14.41 | astoria | parabola=dangerous amazon snake |
14:14.42 | synthetiq | im getting now foudn errors on ym sip phones |
14:14.47 | synthetiq | and calls cant go out |
14:14.50 | synthetiq | but they can come in |
14:14.56 | synthetiq | what could be the problem? |
14:15.01 | synthetiq | i change dnothing in the dial plan |
14:15.10 | Katty | tzanger: i found orbits (some of them) to be a porabola |
14:15.22 | tzanger | I know nothing about astronomy |
14:15.28 | Katty | tzanger: except it's a porabola, so it's not really an 'orbit' |
14:15.33 | Katty | just more like a swing-about |
14:15.48 | tzanger | well yeah if it escapes the gravity well it's not an orbit anymore |
14:15.58 | tzanger | there are circular and elliptical orbits |
14:16.05 | tzanger | but orbits by definition are closed shapes |
14:16.26 | Katty | there's also hyperbola based orbits |
14:16.35 | Katty | i'm reading all about such things right now (= |
14:16.42 | tzanger | I thought hyperbolas weren't closed |
14:16.49 | Katty | they're not |
14:16.55 | Katty | they're still considered an orbit though |
14:17.00 | Inkubot | CANO-1982: msg |
14:17.03 | tzanger | really... how the hell does it go around ? |
14:17.29 | Starcode | this is the asterisk-channel, isn't it? |
14:17.30 | Starcode | ;-) |
14:17.58 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
14:18.08 | tzanger | Starcode: nope |
14:18.08 | Katty | tzanger: it doesn't |
14:18.16 | tzanger | Katty: hmm interesting |
14:18.24 | spiekey | i bought a "octoBRI PCI ISDN" from junghanns.net. Do i only have to plug my telco cable into my FXO port of the card? |
14:18.41 | Katty | tzanger: if you take the earth's orbit 18.5 m/s |
14:18.53 | Katty | well the velocity around the sun |
14:19.04 | Katty | and kicked it up to 26 miles a second |
14:19.11 | tzanger | you're gettin me all hot, talkin astrophysics |
14:19.22 | Katty | it'd go shooting off into space in a parobolic orbit |
14:19.26 | sivana | anyone 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) |
14:19.49 | iDunno | only till it found something else with enough gravity, though ;) |
14:20.10 | Katty | tzanger: it's not a trajectory, because those end, somewhere. |
14:20.28 | Katty | a porabola, in theory...never ends. |
14:20.46 | gordonjcp | no, that's an ellipse, isn't it? |
14:21.04 | Katty | gordonjcp: the earth is on an elliptical orbit right now, yes. |
14:21.19 | gordonjcp | I thought a parabola wasn't closed |
14:21.38 | Starcode | Does someone have a working IAX <-> IAX setup? |
14:21.40 | tzanger | a parabola isn't |
14:21.44 | iDunno | Katty: 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 ;) |
14:21.46 | tzanger | hence why I don't think it's an orbit |
14:21.50 | supaigtr | Starcode: Define working? |
14:21.54 | gordonjcp | ah, but it seems you *can* have a parabolic orbit |
14:21.56 | tzanger | you have circular, elliptical or irregular orbits |
14:22.17 | Katty | iDunno: you're getting off the topic. |
14:22.33 | iDunno | Katty: I thought that was the idea ;) |
14:22.48 | Katty | tzanger: some comments have a parabolic orbit |
14:22.58 | *** join/#asterisk gambolputty (n=gambolpu@72.240.242.4) |
14:22.59 | Katty | tzanger: they swing about the sun and keep on going |
14:23.01 | tzanger | Katty: interesting, my comments usually stay put :-) |
14:23.05 | Starcode | When 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 |
14:23.09 | Katty | teehee! |
14:23.31 | Starcode | Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW |
14:23.31 | gordonjcp | http://en.wikipedia.org/wiki/Orbit |
14:23.36 | Starcode | Timestamp: 00012ms SCall: 16384 DCall: 00000 [10.2.2.2:0] |
14:23.48 | gordonjcp | ahaa! if you launch something at escape velocity, then it is a parabolic orbit |
14:23.55 | Katty | tzanger: http://www-history.mcs.st-andrews.ac.uk/history/Java/Parabola.html |
14:23.57 | gordonjcp | that 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) |
14:24.49 | lcars | hi folks..what's the proper way for debugging a 'Requested context didn't get merged' error message ? |
14:25.00 | supaigtr | Starcode: 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? |
14:25.40 | Starcode | No, because my server B is SIP/IAX only and trunk need Zap-driver for some reasons |
14:25.52 | Katty | if you have two stars that go past each other, and their gravitational fields interact, then they usually end up having a hyperbolic trajectory |
14:26.04 | supaigtr | Yea. PRI -> * -> IAX - IAX * -> SIP? |
14:26.13 | Starcode | WARNING[229391]: chan_iax2.c:5460 build_user: Unable to support trunking on user 'server_b' without zaptel timing |
14:26.32 | Starcode | Yes |
14:26.52 | supaigtr | I have the same thing working more or less. |
14:27.02 | Starcode | But when calling on pri *A does not send ip packets to *B |
14:27.14 | spiekey | PSTN stands for what? |
14:27.18 | supaigtr | pastbin your iax and extension where the dialing happens. |
14:27.24 | Katty | purple soy tuna nuggets |
14:27.41 | spiekey | Katty: i am kind of serious ;) |
14:27.41 | supaigtr | Pissing Standing Top Neutral |
14:27.53 | Katty | spiekey: so? doesn't mean i have to be |
14:28.02 | Katty | spiekey: you could always google you know |
14:28.19 | supaigtr | Public Switched Telephone Network |
14:28.23 | tzanger | ~pstn |
14:28.25 | jbot | rumour has it, pstn is Pubic Switched Telephone Network, or "please stop the nonsense" |
14:28.25 | spiekey | thanks |
14:28.52 | synthetiq | guys 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 |
14:29.12 | supaigtr | ds3 voip? GW or *? |
14:29.20 | synthetiq | * |
14:29.33 | supaigtr | ds3 with * how? |
14:29.43 | spiekey | PSTN is the line/cable/port to my telco, right? |
14:29.49 | synthetiq | i pump my voip out bound thru a ds3 |
14:30.04 | Katty | iDunno: 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) |
14:30.12 | supaigtr | Yea what * hardware does that at this moment? |
14:30.26 | supaigtr | They were flung once. |
14:30.37 | iDunno | Katty: well, true, but asteroids kinda do, and they're only a little bit smaller ;) |
14:30.39 | Katty | not..exactly. |
14:30.42 | Starcode | [general] |
14:30.42 | Starcode | bandwidth=low |
14:30.42 | Starcode | disallow=lpc10 |
14:30.42 | Starcode | jitterbuffer=no |
14:30.42 | Starcode | tos=lowdelay |
14:30.42 | tzanger | synthetiq: 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. |
14:30.43 | Starcode | [server_b] |
14:30.45 | Starcode | type=friend |
14:30.46 | Katty | supaigtr: they didn't fling, the grew outward. |
14:30.46 | [TK]D-Fender | Sangoma has a DS3 card. |
14:30.47 | Starcode | username=server_b |
14:30.49 | Starcode | secret=pass |
14:30.58 | Starcode | host=10.116.168.250 |
14:30.58 | tzanger | Starcode: don't do that |
14:30.59 | Starcode | context=main-routing-table |
14:31.01 | Starcode | permit=0.0.0.0/0.0.0.0 |
14:31.01 | Katty | iDunno: ye, comments flinging about teh sun |
14:31.02 | tzanger | Starcode: use pastebin |
14:31.05 | supaigtr | Ouch!!! |
14:31.06 | tzanger | ~pastebin |
14:31.07 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
14:31.08 | synthetiq | im under pressure thats hwy im having toruble typing |
14:31.32 | Katty | iDunno: but if you can graph a parabola using the usual x^2-x+1 thingy |
14:31.37 | tzanger | synthetiq: 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. |
14:31.44 | Katty | iDunno: then you can map out if they're going to smack into anything on the way |
14:32.14 | Katty | iDunno: and, in fact, it'd be a rather easy graph |
14:32.16 | iDunno | Katty: you need to calculate the gravitational fields around things that they're passing to get it accurate, though ;) |
14:32.39 | iDunno | Katty: though, probably don't need to be *that* accurate ;) |
14:32.45 | Katty | that's what GRACE is for ;) |
14:32.50 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
14:32.51 | supaigtr | synthetiq: Explain your setup? |
14:33.14 | Katty | i'm waiting for GRACE to give us a good idea of earth's gravitational field |
14:33.24 | Katty | gravitation being my favorite topic. |
14:33.38 | synthetiq | it jsut randomly stopped letting us from makign out bound calls |
14:33.39 | Katty | curious as to how much space-time bend we might be dragging. |
14:33.42 | synthetiq | interneal calls are fine |
14:34.02 | supaigtr | synthetiq: What hardware do you have doing TDM? |
14:34.21 | synthetiq | te411p |
14:34.40 | supaigtr | Ok so what does zttool tell you? |
14:34.48 | synthetiq | i rebooted the machines |
14:34.54 | *** join/#asterisk fiXXXerMet (n=Kyle@ip67-154-236-201.z236-154-67.customer.algx.net) |
14:35.02 | synthetiq | its not the zap cards, because i cant make calls via IP outbound |
14:35.16 | fiXXXerMet | Hi 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? |
14:35.37 | supaigtr | Ok so pastbin your dialplan. |
14:35.42 | tzanger | synthetiq: never reboot a machine unless you know what's going on |
14:35.53 | iDunno | Katty: I think anti-gravity is more fun... |
14:36.04 | iDunno | Katty: involves less crashing to earth with a bump, for a start :) |
14:36.13 | supaigtr | Espcially a 411p it may never boot back. :) |
14:36.31 | Katty | iDunno: not to me :) |
14:36.36 | tzanger | synthetiq: so you have a DS3 coming into an M13 to break it out into individual T1s, which you have going in to TE411 cards. |
14:36.46 | vader-wrk | anyone know where to get some super deals on cisco ip phones? other than ebay? |
14:36.49 | tzanger | synthetiq: and incoming calls work just fine, which means your switch setup and everything is set up |
14:37.35 | tzanger | synthetiq: 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 |
14:38.09 | tzanger | not 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) |
14:39.45 | Starcode | In the dailplan the only interesting thing for iax <-> iax is in [call-groups]: exten => _391130,4,Dial(IAX2/server_b/${EXTEN:2},30,r) |
14:39.52 | RussC | Does any one have experince with polycom SoundPoint IP 301's? |
14:40.03 | supaigtr | RussC: sure |
14:40.08 | astoria | RussC: I have experience, with 300s,500s, and 600s |
14:40.21 | supaigtr | synthetiq: Is your dialplan tring to send out over TDM? |
14:41.03 | RussC | Well 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 |
14:41.04 | synthetiq | icomign calls are on separate ptri |
14:41.17 | shido6 | ZzzzZZZ |
14:41.19 | vader-wrk | do any of you use softphones at your places? |
14:41.24 | synthetiq | outgoign calls go either thru t1 or IP |
14:41.24 | astoria | RussC: Does teh Asterisk Debug CLI show anything? |
14:41.34 | astoria | RussC: Is it registering? |
14:41.35 | synthetiq | voice IP or data DS3 (logn dist) |
14:41.37 | RussC | no it does not |
14:41.58 | astoria | I suggest your run ethereal or something and see what is happening to the packets. |
14:42.05 | RussC | Astoria: that is what has lead me to belive that the phone and server are not talking |
14:42.11 | ian_k | fiXXXerMet - use freeworlddialup if you just want to test |
14:42.24 | astoria | RussC: well, obviously.. what does :sip show peers show? |
14:42.26 | RussC | Astoria: I will try that |
14:42.40 | RussC | Astoria: shows them as unknown |
14:43.07 | astoria | RussC: what is in your sip.conf? |
14:43.22 | supaigtr | synthetiq: Your not answering the question. Can you verify that * is trying to send the calls? Do you see ISDN errors? |
14:43.24 | astoria | Paste your sip.conf and your sip show peers to pastebin and ill look at it. |
14:44.05 | RussC | ok thanks |
14:44.30 | RussC | Astoria: what is a pastebin? |
14:44.44 | clyrrad | http://pastebin.ca |
14:44.48 | vader-wrk | do any of you use softphones at your places? |
14:44.49 | astoria | ~pastebin |
14:44.50 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
14:45.05 | RussC | ok thanks all |
14:45.13 | *** join/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au) |
14:45.19 | fiXXXerMet | Can anyone recommend a VoIP provider for testing? Something cheap? |
14:45.28 | ian_k | fiXXXerMet freeworlddialup - free |
14:45.28 | astoria | fiXXXerMet: asterlink or nufone.. |
14:45.39 | supaigtr | voipjet |
14:45.43 | fiXXXerMet | I have Verizon Voicewing and a PAP2 device, but I doubt they support this. |
14:45.43 | ian_k | fiXXXerMet free for toll-free numbers anyway |
14:45.45 | fiXXXerMet | Thank you both |
14:47.33 | RussC | Astoria: http://pastebin.ca/26059 |
14:48.45 | iCEBrkr | So, does fax detection work with VoIP calls? :P |
14:48.50 | iCEBrkr | Not that I care to send a fax |
14:48.58 | astoria | RussC: yeah, it looks like their not communicating correctly.. Your config looks OK.. I'd fire up ethereal and see where your packets are going. |
14:49.05 | iCEBrkr | I'm just wondering if Asterisk will land in the 'fax' extension after dial() |
14:49.28 | RussC | Astoria: thank you is there anything I should look for inpaticular? |
14:49.42 | astoria | RussC: No, but I'd look for packets coming from the IP of your phone.. |
14:49.56 | astoria | RussC: just see where they are going.. |
14:50.11 | RussC | Astoria: 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) |
14:50.41 | RussC | Astoria: 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? |
14:51.10 | supaigtr | voipjet,nufone |
14:51.16 | astoria | RussC: sure, but that's probably not your problem here. |
14:51.21 | _Sam-- | im using teliax, just wondering about other options...thanks |
14:51.48 | RussC | Astoria: I would just like to see it is correctly setup :-) |
14:51.58 | astoria | RussC: sure, just pastebin it. |
14:52.18 | RussC | ok |
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 |
14:52.59 | vader-wrk | sam whats your voip setup like internally? |
14:53.03 | vader-wrk | ie. phones, etc. |
14:53.06 | RussC | Astoria: http://pastebin.ca/26061 thanks again |
14:53.13 | vader-wrk | im 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 |
14:53.23 | vader-wrk | i wanna build an asterisk box with some softphones |
14:53.47 | _Sam-- | it should be a pretty straight forward project these days |
14:54.01 | astoria | RussC: 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.. |
14:54.09 | vader-wrk | my system is going to have to intergrate with a good chunk of old analog phones |
14:54.22 | vader-wrk | anyone have experience with that? |
14:54.22 | _Sam-- | you will probably want those iaxy thingies |
14:54.40 | astoria | RussC: In your extension lines, rather than having 03 or 04, use the incoming number that your carrier is sending you, like your DID. |
14:54.42 | vader-wrk | becuase we have alot of places that don't have network cable run to them |
14:54.45 | vader-wrk | only telephone lines |
14:54.54 | RussC | Astoria: I had it that way and it did not work I forgot to change it back thanks. Other than that? |
14:55.09 | astoria | RussC: Yeah, it's OK, other than that... |
14:55.14 | vader-wrk | sam what phones are you using? |
14:55.21 | astoria | RussC: 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 |
14:55.34 | vader-wrk | ya but you can use channel banks for the analog phones |
14:55.55 | astoria | I won an iaxy at cluecon and the guy never sent it to me! what a jerk! |
14:55.56 | vader-wrk | atleast thats what im gathering |
14:56.16 | vader-wrk | the ADIT 600 looks like what i would use |
14:56.19 | vader-wrk | with an E1 connection |
14:56.23 | astoria | i heart that adit 600 |
14:56.29 | iCEBrkr | Is there some reason why I can't call my home voicepulse number with my work voicepulse number?? Two seperate accounts |
14:56.29 | vader-wrk | you have an ADIT 600? |
14:56.34 | iCEBrkr | Weird |
14:56.49 | astoria | Yes. I have an adit 600. |
14:56.58 | astoria | Well, I don't own it.. I lease it from XO. |
14:57.08 | vader-wrk | do you use it to power analog phones? |
14:57.38 | astoria | No, 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) |
14:57.45 | vader-wrk | oh |
14:58.05 | _Sam-- | it does say the adit supports 6 pots |
14:58.07 | vader-wrk | i was looking to pick up like two 24 port models with FXS cards |
14:58.51 | astoria | adit is pricey, but plays very well with asterisk! |
14:58.57 | _Sam-- | how many regular analog phones are you looking to connect? |
14:59.07 | vader-wrk | not sure the exact number |
14:59.10 | *** join/#asterisk n0where (n=kc@s199048.ppp.asahi-net.or.jp) |
14:59.18 | astoria | You might want to look at Rhino callbanks too.. |
14:59.19 | vader-wrk | but it's going to be in the ballpark of probably 40-50 phones |
14:59.30 | vader-wrk | i found some adit 600s on ebay cheap |
14:59.54 | vader-wrk | not 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) |
15:00.15 | astoria | Looking on ebay now - not that cheap... |
15:01.05 | _Sam-- | how do those rhino channel banks interface with asterisk? |
15:01.20 | vader-wrk | yesterday they had some ADIT 600s with cards and stuff for under 500 |
15:01.22 | astoria | Via T1 |
15:01.28 | astoria | Damn... |
15:01.32 | vader-wrk | adit 600 can connect through ethernet |
15:01.36 | vader-wrk | instead of t1 |
15:01.40 | vader-wrk | so you don't need a card |
15:01.41 | _Sam-- | so you would plug the rhino channel bank into like, a digium type card? |
15:01.45 | vader-wrk | just a network connection |
15:01.49 | vader-wrk | ya |
15:02.00 | astoria | vader-wrk: SIP? |
15:02.08 | vader-wrk | i dunno |
15:02.28 | astoria | Hmmm. I'm curious to know which protocol it uses.. |
15:03.02 | vader-wrk | ya not sure |
15:03.13 | vader-wrk | im just sooo leary about doing this system |
15:03.24 | astoria | Why? |
15:03.25 | vader-wrk | because it's soo many phones,extensions, etc. and quipment |
15:03.29 | vader-wrk | equipment |
15:03.46 | vader-wrk | and trying to intermix analog and digital |
15:03.58 | *** join/#asterisk gambolputty (n=gambolpu@72.240.242.4) |
15:04.03 | astoria | Well, 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) |
15:04.22 | jake1932 | looks 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 |
15:04.51 | astoria | I dont know what his existing setup is, but you'd get * .. |
15:04.57 | vader-wrk | well 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 |
15:04.57 | jake1932 | (with a CMG card) |
15:05.00 | astoria | Sure, you can do alot more.. |
15:05.03 | vader-wrk | for construction reasons |
15:05.16 | astoria | Yeah, then when you move, drop only cat5, that's what we did. |
15:05.22 | [TK]D-Fender | Channel banks are an iffy thing I find. Seems better to use ATA's (more hardphone functionality) |
15:05.48 | jake1932 | <PROTECTED> |
15:06.00 | [TK]D-Fender | Plus you have to factor in the T1 card. |
15:06.18 | [TK]D-Fender | Sipura's are great, and AudioCodes is #1. |
15:06.28 | astoria | All those ATAs would make my head spin.. |
15:06.30 | jake1932 | my Sipura sucks |
15:06.38 | astoria | I heart Sipura. |
15:06.39 | vader-wrk | ya well where we wanted to move the people we have cat5 but no telephone |
15:06.55 | vader-wrk | so we figured that we would integrate this we the current pbx system or replace the pbx system with this |
15:07.02 | vader-wrk | or a combination of both |
15:07.03 | [TK]D-Fender | Depends if you use a high density gateway like the AudioCodes rackmount stuff. 1 box. |
15:07.18 | vader-wrk | use this and then phase it from temp solution to full solution |
15:07.22 | glomph | Keeping a separate wiring plant just for phones is archaic |
15:07.31 | astoria | Yeah, good point [TK] |
15:07.33 | [TK]D-Fender | Or if you want even better functionality try Citel's digital phone_. SIP gateway. |
15:07.54 | jake1932 | from what I've been hearning - channel bank + T1 is the way to go |
15:08.19 | vader-wrk | ya alot of places in the building have computers now but not all |
15:08.25 | vader-wrk | so that becomes a problem |
15:08.29 | vader-wrk | if we roll out IP phones |
15:08.36 | vader-wrk | because some places won't have cat5 |
15:08.36 | astoria | Are 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-Fender | jake1932 : 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 |
15:09.05 | astoria | jake1932: actually, I like TKs idea better... I don't have any experience with SIP gateways though. |
15:09.21 | RussC | Astoria: could you look at http://pastebin.ca/26062 please |
15:09.33 | vader-wrk | we have a wireless system |
15:09.41 | [TK]D-Fender | Norstar phones on a Citellink gateway would eb a cool alternative so as to avoid transformers for your analog phones, etc. |
15:09.43 | vader-wrk | but those wireless phones would be booookuuu bucks |
15:09.57 | _Sam-- | no, wireless softclients for the desktops |
15:10.03 | RussC | Astoria: I am now getting this error, I setup the phones to register with the server. Oops |
15:10.11 | vader-wrk | we were thinking that too |
15:10.12 | jake1932 | [TK]D-Fender: mine is real simple - SPA3000 FXO = big echo - no matter what I do |
15:10.16 | vader-wrk | softphones |
15:10.21 | vader-wrk | but i haven't found a good client |
15:10.26 | [TK]D-Fender | jake1932 : Really? I'm about to get one for home..... |
15:10.26 | vader-wrk | and 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 |
15:10.38 | jake1932 | [TK]D-Fender: maybe it'll work for you |
15:11.02 | [TK]D-Fender | jake1932 : I can only hope. what phones are you using apart from the FXS on it? |
15:11.35 | jake1932 | tried 3 - cisco 7960, analog + DTA310, SPA841 |
15:11.43 | [TK]D-Fender | jake1932 : and is its all phones? |
15:11.44 | _Sam-- | vader-wrk: sipps: http://ww2.nero.com/sippstar/eng/SIPPS_What_Is_SIPPS.html |
15:11.46 | jake1932 | yep |
15:11.58 | jake1932 | now i'm totally VOIP and no echo |
15:12.11 | jake1932 | (but network issues occasionally cause blips) |
15:12.16 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
15:12.18 | [TK]D-Fender | jake1932 : wow.... Sipura & Cisco pretty much tell you that something serious needs to be fixed. Done a F/W upgrade on it? |
15:12.24 | jake1932 | yep |
15:12.29 | jake1932 | was at the newest |
15:12.36 | jake1932 | the phones are fine |
15:13.07 | astoria | RussC: you need to fix your exten=>03 in your [03] context... 03 doesn't exist.. |
15:13.13 | funxion | anyone know the best way to configure an e1 from an digium card to a cisco |
15:13.19 | astoria | RussC: wait.. |
15:13.38 | [TK]D-Fender | jake1932 : :/ Ugh. I'd still like to ditch analog altogether, but we're still a little while away from Dry-line DSL |
15:14.08 | jake1932 | yep - I can't do dry DSL - either - I just fwd my honme number for now |
15:14.18 | astoria | RussC: Your dial cmd is jacked up... |
15:14.19 | jake1932 | still cheaper |
15:14.41 | RussC | Astoria: ok |
15:14.43 | jake1932 | (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) |
15:15.41 | JASON-0 | Hello, 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) |
15:15.52 | zobia | Hello everyone |
15:15.58 | zobia | i have strange issue |
15:16.34 | zobia | when 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) |
15:17.53 | jpm_SD | JASON-0, The Maintance tab is Asterisk @ Home specific.. it's not part of AMP |
15:18.20 | jpm_SD | and as far as I know there is no easy way to add it... |
15:18.41 | RussC | Astoria: what might I do to fix it? |
15:19.23 | astoria | RussC: Hmmmm... Can you just copy that single command from your extensions.conf in here? |
15:19.26 | astoria | hold on |
15:19.46 | fiXXXerMet | Has 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. |
15:19.52 | RussC | Astoria: exten=>03,1,dial(SIP/03,20) |
15:20.29 | astoria | RussC: what are you trying to do with that?? When someone calls in, ring 03? |
15:20.55 | astoria | Also, it should be dial(SIP/thecontextyouhaveinsip.conf) |
15:21.13 | JASON-0 | jpm_SD: Thanks |
15:21.14 | zobia | Hello any one knows about the press repeat problem? |
15:21.24 | RussC | Astoria: ok so what is the ,20 |
15:21.31 | astoria | so, pete and his brother repeat were on a boat |
15:21.40 | jake1932 | oh no |
15:21.47 | astoria | Well, the time to ring the extension. |
15:22.04 | RussC | Astoria: alright |
15:22.50 | astoria | bless you child |
15:22.50 | jake1932 | bless you |
15:22.50 | Katty | astoria: i'm not a child. |
15:22.50 | *** join/#asterisk bintut (n=bintut@202.128.40.243) |
15:22.50 | Katty | astoria: but thanks anyway. |
15:22.56 | jake1932 | gazoontight |
15:23.09 | RussC | Astoria: so it needs to match the line context= in the sip.conf? |
15:23.09 | jpm_SD | Online alergies... scary thought. |
15:23.12 | jake1932 | <PROTECTED> |
15:23.32 | Katty | jpm_SD: dimensional based allergies. |
15:23.39 | jpm_SD | Timestamps make my skin itchy |
15:23.59 | astoria | RussC: No. I needs to match the context (the stuff in the brakcets) |
15:24.11 | jake1932 | it's especially scary considering virus control on the net has not been great |
15:24.32 | RussC | ohh I see |
15:24.42 | *** join/#asterisk skrusty (i=muad@xdev.net) |
15:24.46 | NewSole2 | I just replaced all my Digium junk with Sangoma hardware.. selling the digium cheap... got 10 TE410P Cards.... Any takers Best offer |
15:25.08 | skrusty | anyone know why cdr_odbc would say it's updated the table, but doesn't?! using odbc->tds->mssql |
15:25.10 | astoria | NewSole2: how much want for one? |
15:25.43 | NewSole2 | I said best offer |
15:25.51 | astoria | 50 bucks |
15:25.55 | jpm_SD | Got 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) |
15:26.02 | jpm_SD | Analog too.. :) |
15:26.07 | jake1932 | <PROTECTED> |
15:26.14 | jpm_SD | jake1932, yup |
15:26.17 | jake1932 | <PROTECTED> |
15:26.26 | jpm_SD | jake1932, Negitive |
15:26.42 | Katty | what goes well with mushroom soup? |
15:26.51 | jake1932 | salt? |
15:26.58 | astoria | A roast beef sandwich. |
15:27.04 | jpm_SD | Katty, Toast? |
15:27.24 | Katty | astoria: anything else? |
15:27.36 | file | NewSole2: how about a signed autograph of myself? |
15:27.36 | jake1932 | maybe i should unload my digium stuff too before this sag stuff becomes mainstream |
15:27.40 | zobia | Hello who know why the dtmf tones are double always? |
15:27.42 | sivana | Katty: rice |
15:27.46 | astoria | Katty: no |
15:27.46 | NewSole2 | current bid is a Night with the Wife.... |
15:27.46 | RussC | Astoria: ok the phones can call its self but not each other lol |
15:27.47 | Katty | sivana: oooh, good idea. |
15:28.02 | Katty | sivana: what sort of protein goes well with rice and mushroom soup? |
15:28.21 | astoria | Well, in each of the phone's contexts (in extensions) you need to put an entry if you dial the other phone.. |
15:28.26 | sivana | Katty: hrm.. you need enough to counter the rice and soup :) |
15:28.33 | tzanger | rice and mushrooms? I'd throw some chicken in there but I know that's not what you're thinking |
15:28.40 | sivana | hehe |
15:28.43 | jpm_SD | jake1932, Not having the 1000 Interrupt polls per second is a nice bonus too. |
15:28.48 | Katty | sivana: well, i'm plotting an enchilada recipe |
15:28.55 | Katty | sivana: where the 'enchilada sauce' is mushroom soup |
15:29.00 | sivana | Katty: maybe some man-made chicken meat? |
15:29.01 | jake1932 | jpm_SD: when are they expected? |
15:29.16 | Katty | sivana: possibly. |
15:29.18 | *** join/#asterisk h3x0r (n=h3xor@64.192.116.16) |
15:29.25 | sivana | hrm.. not sure.. |
15:29.30 | Katty | sivana: that and a can of corn should compliment nicely.....i'd think |
15:29.32 | jpm_SD | jake1932, Doug said 3 weeks when I spoke to him at Astricon. |
15:29.37 | Katty | or maybe rice, corn, and refried beans |
15:29.41 | sivana | ya |
15:29.48 | sivana | and pepper |
15:29.50 | sivana | :) |
15:30.17 | jake1932 | best offer |
15:30.26 | astoria | 2o bucks |
15:30.41 | RussC | Astoria: sorry I dont understand. I have 03 and 04 the only two in my setup, in sip.conf and extensions.conf |
15:31.14 | astoria | You need to have an exten=>OTHEREXTENSIONNUMBER with a dial cmd in eachother's extensions.. |
15:31.19 | jake1932 | hey you offered the other guy 50 |
15:31.37 | astoria | Well, supply clearly exceeds demand now. |
15:31.39 | RussC | Astoria: ok, I see |
15:31.56 | jake1932 | true |
15:31.59 | astoria | RussC: i mean "eachother's contexts" |
15:32.18 | bjohnson | 21 bucks |
15:32.41 | jake1932 | increments of $2.50 at least |
15:32.44 | astoria | bjohnson, stop bidding. we don't need to bid. there are digium cards a plenty! |
15:33.14 | bjohnson | I don't want the card, just my cut for driving up the price |
15:33.34 | astoria | bjohnson: do you sell insurance ? :) |
15:34.45 | jake1932 | actually - 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? |
15:35.16 | jake1932 | maybe someone else will have better luck |
15:35.42 | RussC | Astoria: outstanding thank you so much for your help |
15:36.20 | astoria | RussC: no problem |
15:36.38 | *** join/#asterisk jsaunders (i=js@S01060060971c5817.va.shawcable.net) |
15:37.20 | jsaunders | Is 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? |
15:37.48 | file | jsaunders: SIP phone, or what |
15:37.58 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
15:37.59 | jsaunders | yes, sip ata] |
15:38.05 | file | it's up to the ATA to do it |
15:38.15 | file | Asterisk merely sends a "there's messages waiting" packet to it |
15:38.18 | jsaunders | reheally |
15:38.19 | file | the ATA decides what to do with it. |
15:38.21 | *** part/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net) |
15:38.22 | jsaunders | Thanks for the tip. :) |
15:38.35 | file | SIP devices do a lot of stuff themselves. |
15:38.40 | jsaunders | Does asterisk send this packet automagically? Or is it a setting? |
15:38.44 | [TK]D-Fender | Hey 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) |
15:38.50 | astoria | My server plays "Beleieve it or not" on the speakers when a voicemail is left. |
15:38.57 | file | jsaunders: you have to tell it the mailbox in sip.conf, it's not psychic |
15:39.05 | mishehu | bah. |
15:39.05 | jsaunders | file: Heheh |
15:39.10 | jsaunders | file: Tnx mang. |
15:39.11 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
15:39.13 | astoria | mishehu, how you doin? |
15:39.27 | file | jsaunders: you now owe me one DJ Sammy MP3 |
15:39.35 | fiXXXerMet | Has 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. |
15:39.36 | file | or your soul |
15:39.36 | iDunno | file: dammit! why isn't is psychic! |
15:39.38 | mishehu | astoria: getting killed here. 3 classes at college, and trying to run a business! heh |
15:40.09 | astoria | mishehu: yikes! I hear ya. |
15:40.29 | file | iDunno: I wrote a psychic module for SER when I was sitting at a conference |
15:40.39 | astoria | Hey, anyone here one of the organizers of cluecon, back in July. |
15:40.44 | file | astoria: hi. |
15:40.58 | astoria | file: cylogistics never sent me my iaxy I won! |
15:41.14 | file | astoria: weren't you supposed to talk to them? if not I have that guy's business card here |
15:41.29 | astoria | file: yeah, I have it and i've emailed him a bunch of times with no reply |
15:41.39 | file | tried phone? |
15:41.41 | supaigtr | Anyone have trouble ticket software recommendations? |
15:41.45 | astoria | file: so i am going to start emailing him naked pictures |
15:41.52 | astoria | file: no, all i have is his email addy |
15:41.54 | zobia | please help with my dtmf tones double problem |
15:42.05 | BladeRunner05 | How can I use mohmp3 ? I have to write code in extensions.conf ? or what ? |
15:42.11 | file | astoria: see privmsg |
15:42.28 | astoria | thanks file |
15:42.58 | mishehu | BladeRunner05: you have to pay $$ to the RIAA, you pirate. |
15:43.08 | mishehu | since we all know that only pirates use mp3 files. |
15:43.08 | Ahrimanes | lol |
15:43.31 | mishehu | and l33t pirates use aac or ogg. |
15:44.47 | *** join/#asterisk _tekati_ (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
15:44.51 | BladeRunner05 | its not a commercial music |
15:45.06 | BladeRunner05 | How can suggest me ? |
15:48.35 | bintut | what'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? |
15:49.57 | mmlj4 | bintut: there's a page on the wiki about asterisk dimensioning, that might help |
15:50.17 | bintut | mmlj4: 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. |
15:50.53 | mmlj4 | yes: on the wiki |
15:50.54 | _m_ | Any ideas? |
15:51.03 | astoria | bintut: a 486 at least. |
15:51.34 | *** part/#asterisk n0where (n=kc@s199048.ppp.asahi-net.or.jp) |
15:52.06 | bintut | astoria: yeah, i know at least 486.. if you will be deploying the above specs, what will you choose? |
15:52.47 | astoria | bintut: i'm not sure. EDO ram is so expensive these days. |
15:52.53 | *** join/#asterisk gaspiz (i=gaspi@86.34.6.164) |
15:53.13 | gaspiz | hi there, can you hepl a rookie? |
15:53.29 | *** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
15:53.32 | astoria | bintut: 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) |
15:53.44 | gaspiz | not so rookie |
15:54.06 | *** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
15:54.14 | cjk | hi, 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 |
15:54.41 | bintut | astoria: where's the wiki? sorry, i'm a newbie.. :(( |
15:54.49 | astoria | bintut: www.voip-info.org |
15:54.59 | gaspiz | I have a problem when I set up sip users |
15:55.09 | gaspiz | when I set there names numbers it's ok |
15:55.20 | *** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3685577.sympatico.ca) |
15:55.32 | gaspiz | when I put names i can't place or receive calls |
15:55.45 | gaspiz | so: name=1001 it's ok |
15:55.52 | gaspiz | name= gasparz |
15:55.57 | gaspiz | problem... |
15:56.22 | *** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus) |
15:56.49 | gaspiz | ideas? |
15:57.07 | bintut | astoria: ok. thanks.. :) |
15:57.08 | astoria | what file is this in?? |
15:57.31 | gaspiz | I use realtime for sip users |
15:57.42 | funxion | did I read wrong but I see hat in the new 1.2 cvs head there is support for t38 pass through? |
15:57.45 | jsaunders | file: I gots rizapp if yer lookin' for it, no DJ Sammy though. Jew want somethin'? |
15:57.57 | jsaunders | file: I'll send ya a few of my faves, np. |
15:58.16 | file | jsaunders: nah I'm in a DJ Sammy mood |
15:58.34 | jsaunders | file: Hmm, never heard of him. I'll have to snag some. Any faves? |
15:58.49 | file | Why... Rise Again... The Boys of Summer |
15:59.01 | *** join/#asterisk _Thor (i=CS@user-vc8fl7n.biz.mindspring.com) |
15:59.10 | iDunno | (but that covers a lot of the time, really) |
15:59.31 | jsaunders | file: denamrk? What genre? Oh, techo/rave kinda? |
15:59.37 | file | trance-sorta |
15:59.45 | jsaunders | file: Sweet! You like Royksopp? |
15:59.47 | _Thor | hello, 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> |
16:00.24 | gaspiz | anyone using letters for sip usernames? |
16:00.26 | *** join/#asterisk dstruct (i=dstruct@unaffiliated/dstruct) |
16:00.27 | funxion | has anyone tried T38 passthrough in new CVS head |
16:00.28 | dstruct | yo |
16:00.36 | bintut | guys, what type of phone do you suggest for a 50 ends/local phones: an analog, ip phone or softphone? |
16:00.40 | dstruct | anyone 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... |
16:00.44 | file | jsaunders: can't say I do |
16:01.01 | jsaunders | file: Hmm, because you've never listened to them? Or because you simply aren't partial to their music? |
16:01.09 | astoria | bintut: 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) |
16:01.26 | t0ke | hello |
16:01.42 | dstruct | astoria: heheh |
16:02.06 | file | jsaunders: never listened |
16:02.08 | astoria | You'll probably be dropping at least 10 grand on equipment... |
16:02.13 | joelsolanki | Hello 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 ? |
16:02.19 | skrusty | anyone any good with cdr_odbc? :) |
16:02.45 | jsaunders | file: Reheally... dude! Do yourself a favor and grab 'em. You will enjoy, promise. |
16:02.56 | jsaunders | file: btw, dj sammy's pretty tight. Thanks for the heads up. |
16:03.47 | file | aye |
16:04.53 | funxion | what genre is htat |
16:05.28 | CANO-1982 | I have a problem with my TDM400p board |
16:05.32 | CANO-1982 | I have to unplug and plug again each time y have an incoming call on my FXO module |
16:05.39 | CANO-1982 | any idea? |
16:05.42 | CANO-1982 | Ive tried aswerpolarity.., callooprogres, busydetect, hanguponpolarity.., loopstart ang kewlstart |
16:06.08 | supa_thygar | CANO-1982 private me i'll tell you |
16:07.37 | jsaunders | funxion: rave/trance |
16:07.59 | funxion | like house? |
16:08.35 | funxion | what genre within rave/trance |
16:08.44 | gaspiz | <PROTECTED> |
16:08.54 | jsaunders | funxion: 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. |
16:09.03 | funxion | tru |
16:09.23 | funxion | I like the slower IDM |
16:09.58 | jsaunders | funxion: 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 |
16:16.58 | mutilator | anyone have java 1.4.2 or older installed that can compile an applet for me? |
16:20.10 | bintut | anyone here can i ask about hardware configurations not being suggesting to hire consultants because i'm here to learn not to look for consultants? |
16:21.23 | skrusty | bintut: what's up? |
16:21.28 | *** part/#asterisk jsaunders (i=js@S01060060971c5817.va.shawcable.net) |
16:21.49 | cybertank | ``````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````` |
16:21.50 | cybertank | ``````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````` |
16:21.55 | cybertank | ``````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````` |
16:21.59 | SarahEmm | uhh... |
16:22.00 | cybertank | ```````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````````` |
16:22.01 | SarahEmm | cybertank? |
16:22.02 | skrusty | heh :/ |
16:22.18 | cybertank | keyboard problem :) |
16:22.19 | SarahEmm | bintut: sure |
16:22.19 | skrusty | lets hope he just fell asleep while working :) |
16:22.22 | skrusty | hehe :) |
16:22.23 | SarahEmm | cybertank: lol |
16:22.25 | RoyK | ~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) |
16:25.18 | jac]Z[oby | hi |
16:25.42 | jac]Z[oby | anyone knows where to set the MSN to which the bristuffed zaptel thingy listens so? |
16:25.45 | jac]Z[oby | to |
16:26.07 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
16:27.20 | bintut | skrusty: 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? |
16:29.02 | SarahEmm | bintut: 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) |
16:29.07 | skrusty | server 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 |
16:29.22 | SarahEmm | bintut: as for capacity, i have no idea.. there's a wiki article on it tho |
16:29.27 | skrusty | SarahEmm: was a longer question than i expected :) |
16:29.48 | SarahEmm | bintut: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
16:29.52 | SarahEmm | skrusty: me too :) |
16:30.08 | bintut | SarahEmm: honestly, i don't know those terms you've mentioned like t1/e1, pri, channels, etc.. sorry.. |
16:30.20 | skrusty | bintut: then you really need to learn them |
16:30.22 | astoria | and you're deploying a 50 node network? |
16:30.39 | SarahEmm | bintut: then you need to learn about telco. |
16:30.43 | skrusty | you need to have a good look through the wiki, start at the beginning! :) |
16:30.51 | SarahEmm | bintut: before deploying a big system, you need to learn a Lot about this :) |
16:31.07 | astoria | It's really easy to mess up if you don't have everything planned out right the first time. |
16:31.19 | skrusty | and it can cost a lot more if you get it wrong :D |
16:31.20 | bintut | skrusty: 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 |
16:31.35 | skrusty | right, then that's your starting point |
16:31.35 | SarahEmm | bintut: well, you should learn those terms then :) |
16:31.37 | bintut | SarahEmm: i'm already on the asterisk dimension site |
16:31.58 | skrusty | decide on codec's etc and read up on what hardware you'll need to interconnect with telco's in your area (if required) |
16:32.12 | bintut | SarahEmm: i know computer networking terms (some) but this asterisk/voip thingie is different.. :( |
16:32.29 | skrusty | bintut: can i suggestion whatis.com! |
16:32.37 | jac]Z[oby | anyone knows about bristuff and msn settings |
16:32.39 | jac]Z[oby | ?? |
16:32.41 | SarahEmm | bintut: it uses telco terms, because it's a telco system. you need to learn about telco if you're deploying a telco system. |
16:32.41 | skrusty | it should bring you upto speed quickly if you know a term you want to look for |
16:32.53 | skrusty | e.g. if you don't know what E1 means, look it up on there |
16:32.59 | skrusty | or 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) |
16:33.57 | skrusty | ``follow the wiki`` :) |
16:34.11 | twista | hey there ;) |
16:34.15 | bintut | ok.. |
16:34.16 | skrusty | hi |
16:34.34 | skrusty | bintut: it'll make it far simpler for us to help you, if you know that extra little bit! |
16:35.09 | skrusty | and on that note... hometime! |
16:36.03 | *** part/#asterisk gaspiz (i=gaspi@86.34.6.164) |
16:36.04 | 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) |
16:36.05 | *** join/#asterisk mm_pt (n=mm_pt@a83-132-248-28.cpe.netcabo.pt) |
16:36.38 | mm_pt | hi, any one could help me with xlite register in asterisk? |
16:36.48 | iDunno | *without* using Dial(SIP/phoneblah&SIP/phoneblip&SIP/phoneblop) |
16:37.01 | skrusty | an agi? :) |
16:37.16 | iDunno | skrusty: that was almost the conclusion that I'd come to :/ |
16:37.24 | skrusty | :] |
16:37.32 | skrusty | l8r people! |
16:37.34 | iDunno | AGI can return to the dialplan, right? ;) |
16:37.51 | FuriousGeorge | iDunno: there are call groups for calling out w/ in the same technology |
16:37.55 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
16:38.03 | twista | doesnt anybody have any hints regarding my problem? :S |
16:38.26 | iDunno | FuriousGeorge: it's an incoming call, and I want it to dial the available phones, really. |
16:38.38 | SarahEmm | twista: what's the problem? |
16:38.38 | mm_pt | any one could help me on registering x lite in asterisk |
16:38.45 | twista | <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) |
16:38.53 | iDunno | it's not a big problem, because at the moment I can just specify them all in a long line seperated by & ;) |
16:39.00 | bintut | SarahEmm, 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? |
16:39.25 | astoria | bintut: You don't NEED one for any of those features. |
16:39.26 | FuriousGeorge | iDunno: 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 |
16:39.40 | twista | because 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 |
16:39.54 | FuriousGeorge | ive never implemented a queue, but it cant be hard |
16:40.02 | iDunno | FuriousGeorge: tried the Dial(SIP/1) earlier, that doesn't work ;) |
16:40.04 | bintut | SarahEmm, 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? |
16:40.07 | astoria | bintut: 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) |
16:40.15 | FuriousGeorge | iDunno: you have to set the groups up first |
16:40.28 | mm_pt | SarahEmm 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 |
16:40.35 | iDunno | FuriousGeorge: pickup and call groups where specified in the sip.conf |
16:40.41 | bintut | astoria: yeah, i will.. i just want to get some suggestion and later on, i should know.. |
16:40.46 | SarahEmm | mm_pt: i now have 5 people asking me questions |
16:40.50 | astoria | mm_pt: look on the wiki |
16:40.57 | SarahEmm | mm_pt: i'll try to help in a few minutes, but just ask your question here and someone might answer |
16:40.59 | iDunno | anyways - home time :) |
16:41.11 | FuriousGeorge | you 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 |
16:41.14 | SarahEmm | twisted[asteria]: why can't they talk directly? |
16:41.22 | mm_pt | Thanks for answering me! Sarah Emm ! |
16:41.28 | mm_pt | i could wait |
16:41.31 | FuriousGeorge | twista: yeah, how come they cant talk |
16:41.33 | SarahEmm | err |
16:41.42 | SarahEmm | i didn't say wait, i just said ask your question here and someone might be able to help |
16:41.48 | mm_pt | ok |
16:41.58 | twista | well thats the scenario my company wants me to work on ;) |
16:42.01 | jac]Z[oby | how does the bristuffed active ACER ISDN card know on which msn to listen to? |
16:42.04 | astoria | mm_pt: http://www.voip-info.org/wiki/view/Asterisk+phone+xten+xlite |
16:42.08 | twisted[asteria] | SarahEmm, huh? |
16:42.14 | mm_pt | thanks astoria |
16:42.16 | SarahEmm | twisted[asteria]: gah wrong person sorry |
16:42.16 | twista | like a situation where the asterisk server has two interface but is not "allowed" to route between them |
16:42.27 | FuriousGeorge | twista: the scenario has to coincide with reality. why, hypotheticallh cant they talk |
16:42.35 | twisted[asteria] | SarahEmm, heh, k |
16:42.39 | SarahEmm | twista: so the situation doesn't actually exist? |
16:42.49 | SarahEmm | twista: so it's not allowed to route between them, but it can proxy between them? |
16:42.55 | SarahEmm | sounds like you want a SIP proxy, not * |
16:43.07 | twista | kind of, yes |
16:43.15 | twista | but it also handles the connection to the psdtn |
16:43.17 | twista | pstn |
16:43.27 | drumkilla | twista and twisted, how cute |
16:43.29 | FuriousGeorge | twista: if its a nat issue you can set externip and localnet on the * server |
16:43.47 | FuriousGeorge | and two clients that couldnt talk directly can now be friends with * |
16:44.01 | FuriousGeorge | or "friends in* *" |
16:44.12 | SarahEmm | twista: so there's no NAT, but they're on seperate subnets then? and they're not allowed to talk directly to eachother? |
16:44.36 | FuriousGeorge | in that case u'd use a vpn right? |
16:44.51 | twista | yes...imagine one interface with 192.168.0.0/24 and one with 172.16.0.0/16 (just as an example) |
16:44.53 | astoria | bintut: I wouldn't tell that to your boss... |
16:45.02 | FuriousGeorge | or use an IAX client and tunnel the 4569 port (i think it is) |
16:45.10 | twista | well a vpn would be the easy way that'll work in any case |
16:45.53 | SarahEmm | semihere for a few, on a call |
16:46.04 | twista | but 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) |
16:46.12 | bintut | astoria: :) |
16:46.21 | FuriousGeorge | if 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? |
16:46.48 | twista | hmm? |
16:47.16 | SarahEmm | twista: ahh.... i'm not sure offhand on that... |
16:47.34 | twista | FuriousGeorge 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) |
16:48.01 | twista | so right now the phones ring (because asterisk takes care of the sip-part) but of course the rtp packets get lost |
16:48.07 | FuriousGeorge | twista: two interfaces=two ethX? |
16:48.08 | Juxt | hello |
16:48.15 | Juxt | can someone explain how asterisk -p option works? |
16:48.25 | Juxt | the pseudo-realtime thing |
16:48.31 | Juxt | does it actually improve performance? |
16:48.37 | *** join/#asterisk power1 (n=marktren@rndf-146-4-251.telkomadsl.co.za) |
16:48.45 | twista | FuriousGeorge yes...right now its eth0 and eth0:0 for testing purposes |
16:49.07 | twista | oh and i switched to the cvs head today (from 1.0.9) |
16:49.59 | FuriousGeorge | hmm, i have no idea how that would work from an * config perspective, but maybe you need SER |
16:50.08 | FuriousGeorge | ask the guys in #SER |
16:50.19 | power1 | can 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? |
16:50.51 | FuriousGeorge | power1: does a@h use kewlstart signalling for fxo? |
16:51.18 | twista | FuriousGeorge hmm i read a lot of stuff about proxys, ser etc. today but thought i probably could avoid having to setup additional software |
16:51.44 | FuriousGeorge | twista: i can't say for certain that you cant, but ive jest never considered it |
16:51.47 | power1 | FuriousGeorge, ummm I dont know, how do i check this? |
16:52.04 | FuriousGeorge | power1: nano -w /etc/asterisc/zapata.conf |
16:52.12 | FuriousGeorge | then paste that on pastebin.ca |
16:52.14 | FuriousGeorge | ~pb |
16:52.16 | jbot | i 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) |
16:52.46 | FuriousGeorge | twista: also will need /etc/zaptel.conf |
16:53.19 | FuriousGeorge | i mean power1: |
16:53.27 | twista | does 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 |
16:54.04 | FuriousGeorge | what sort of nat problem? |
16:54.17 | mm_pt | any one who had experience problems regestering xlite with asterisk? call not aproved... |
16:54.52 | FuriousGeorge | mm_pt: what if it has registered and your dialplan isnt setup |
16:54.53 | znoG | if i want to connect to a h323 gateway, i just need h323 gateway functionality on Asterisk, not gatekeeper.. right? |
16:55.36 | mm_pt | in winxp computer it doesn register, i've configured sip.conf and extensions.conf |
16:55.41 | FuriousGeorge | znoG: i think you need a gatekeepser, im not certain. i think i remember reading somewhere that "Asterisk cannot function ax the gatekwwper,atm" |
16:55.46 | twista | FuriousGeorge 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) |
16:55.50 | mm_pt | i was just trying echo *45 |
16:56.14 | power1 | FuriousGeorge, http://pastebin.ca/26068 |
16:56.16 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
16:56.16 | twista | but i'll figure that out by myself...or at least i'll try before i'll ask about that stuff again here |
16:56.25 | FuriousGeorge | as 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 * |
16:56.39 | twista | yep, thats what i expected |
16:56.40 | FuriousGeorge | right ports forearded* |
16:57.06 | mm_pt | is there any channel for freshman with asterisk?:) |
16:57.15 | mm_pt | i don't want to bother hete |
16:57.16 | mm_pt | here |
16:58.45 | astoria | mm_pt: read stuff on the wiki.. thats the best way.. |
16:58.46 | twista | oh but FuriousGeorge, any experience (or at least anything about) the nat modules for netfilter? ( http://www.iptel.org/sipalg/ ) |
16:58.49 | FuriousGeorge | mm_pt: its called www.voip-info.org |
16:58.49 | mm_pt | or 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? |
16:59.19 | FuriousGeorge | twista: i use IP Cop and its been all i need so far |
16:59.28 | power1 | FuriousGeorge, thanks, did u get my pastebin? |
16:59.49 | twista | oehm does it come with those modules? (or just the regular stuff?) |
16:59.58 | FuriousGeorge | power1: u gotta post the link for me here |
17:00.32 | power1 | FuriousGeorge, i did look above |
17:00.41 | FuriousGeorge | my bad :) |
17:00.50 | power1 | FuriousGeorge, <grin> |
17:01.07 | FuriousGeorge | power1: there must be more |
17:01.10 | FuriousGeorge | scroll down |
17:02.06 | power1 | FuriousGeorge, nope thats it 48 lines |
17:02.38 | power1 | FuriousGeorge, u sure u r not looking for /etc/zaptel.conf |
17:03.33 | FuriousGeorge | power1: oh i see |
17:03.59 | FuriousGeorge | power1: i guess i gotta see zapata-auto.conf and zapata_additional.conf which are being included |
17:04.08 | FuriousGeorge | # isnt a comment |
17:04.20 | *** join/#asterisk damned (n=vpol@damned.vpol.org.ru) |
17:04.38 | power1 | FuriousGeorge, ok lemme paste em. |
17:04.55 | [TK]D-Fender | repeat of my earlier question : |
17:04.58 | [TK]D-Fender | Hey 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) |
17:05.27 | SarahEmm | you can NAT SIP, it's just a pain... |
17:05.32 | FuriousGeorge | [TK]D-Fender: i dunno, dot use'em myself. have you looked at iaxtel |
17:05.55 | FuriousGeorge | file[laptop]: are you around? |
17:06.00 | twista | oh 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-Fender | FuriousGeorge : I need a softphone, not termination |
17:06.23 | FuriousGeorge | [TK]D-Fender: i meant to say iaxphone, i think its called |
17:06.37 | power1 | FuriousGeorge, heres the 1 http://pastebin.ca/26074 |
17:06.38 | [TK]D-Fender | SarahEmm : I'm worried about double-NAT scenarios..... |
17:06.39 | twista | and all by using & parsing the existing config, without templates or profiles which overwritte your existing configuration |
17:06.43 | brimstone | iaxcomm [TK]D-Fender |
17:07.13 | FuriousGeorge | brimstone: thanks |
17:07.22 | brimstone | FuriousGeorge: np |
17:07.25 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
17:07.39 | power1 | FuriousGeorge, zapata_additional.conf is empty... |
17:07.40 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
17:08.01 | FuriousGeorge | power1: 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-Fender | I tried IAXCOMM, and it was kinda flakey |
17:10.11 | brimstone | [TK]D-Fender: you might be better off with setting up an Asterisk box inside both networks and a IAX trunk between them |
17:10.13 | jarrod | weeeeee |
17:10.21 | brimstone | then you won't have NAT issues and can use SIP phones locally |
17:11.06 | [TK]D-Fender | brimstone : An idea if I wasn't going to use it for salespeople's laptops..... |
17:11.38 | brimstone | if people are going out in the field, may i recommend an IAXy ? |
17:12.26 | power1 | FuriousGeorge, the second paste is of zapata-auto.conf |
17:12.51 | FuriousGeorge | power1: what number? |
17:13.02 | *** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net) |
17:14.02 | power1 | FuriousGeorge, here is zaptel.conf http://pastebin.ca/26076 |
17:14.39 | power1 | FuriousGeorge, zapata-auto.conf is number 26074 |
17:14.43 | FuriousGeorge | power1: hmm, all that work for nothing, you are using kewlstart signalling, which is supposed to detect a remote hangup |
17:14.53 | FuriousGeorge | is this happening incoming outgoing or both |
17:15.16 | power1 | FuriousGeorge, what field in what of the conf files denotes kewlstart signaling.. |
17:15.39 | FuriousGeorge | fignalling fx*ks |
17:15.48 | FuriousGeorge | signalling=fx*ks |
17:16.18 | power1 | FuriousGeorge, 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. |
17:16.44 | FuriousGeorge | power1: i had a problem once where it wouldnt pickup |
17:16.50 | FuriousGeorge | what version of the driver are you using? |
17:16.54 | FuriousGeorge | 1.9.2 is latest |
17:17.19 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
17:17.22 | FuriousGeorge | someone 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 |
17:17.23 | brimstone | 1.0.9.2 |
17:17.27 | FuriousGeorge | so thats how i check |
17:17.34 | FuriousGeorge | brimstone: do'h thanks |
17:17.51 | brimstone | FuriousGeorge: no problem, got to earn this hostmask some how |
17:18.02 | FuriousGeorge | twista: listen to brimstone :) |
17:18.21 | *** join/#asterisk loick (n=loick@APuteaux-151-1-34-29.w82-120.abo.wanadoo.fr) |
17:18.36 | power1 | FuriousGeorge, im using the standard a@h iso that runs on top of centos..i dunno? |
17:20.02 | morale | w |
17:20.31 | FuriousGeorge | does it answer the line and call out? |
17:21.28 | FuriousGeorge | twista: i mean: does it pickup the line when you call out |
17:21.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
17:21.57 | copantl | can asterisk be interconnected to a pstn via C7? |
17:23.27 | FuriousGeorge | !C& |
17:23.32 | FuriousGeorge | ~C7 |
17:23.50 | power1 | FuriousGeorge, 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) |
17:24.06 | FuriousGeorge | g is the group |
17:24.15 | FuriousGeorge | so 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-Fender | brimstone : Iaxy is a nifty idea, but they need WiFi access, etc. |
17:25.47 | brimstone | ah |
17:25.47 | [TK]D-Fender | mc |
17:26.22 | brimstone | could just use a crossover cable and ICS on the winderz laptopo |
17:26.30 | brimstone | but 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) |
17:28.01 | cripito | hi |
17:28.37 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
17:31.57 | cripito | there is any way to totally disable the cdr? |
17:32.15 | cripito | CDR logging: enabled |
17:32.15 | cripito | CDR mode: simple |
17:32.26 | skyen | got secrets, hey? |
17:32.32 | cripito | no :) |
17:32.43 | skyen | sorry, I dunno |
17:32.50 | cripito | i 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? |
17:38.29 | cripito | i found the way... |
17:41.25 | [Lamer] | cripito: what is it? |
17:48.53 | astoria | i love technology |
17:49.02 | astoria | i've been chatting with babes all day |
17:49.20 | lehel | nice astoria;) |
17:50.00 | funxion | u mean old men pretending to be babes |
17:50.07 | snitt | lol |
17:50.41 | *** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3685577.sympatico.ca) |
17:52.33 | iCEBrkr | Crap, my cover is blown |
17:52.58 | funxion | so has anyone tried t38 pass through on new cvs head |
17:53.59 | supaigtr | t.38 is in head? |
17:54.09 | funxion | according to wiki |
17:54.15 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
17:54.31 | funxion | CVS as of Oct. 2005 now has very limited and experimental T.38 pass-through support for SIP (bug/patch 5090). |
17:54.55 | supaigtr | Hmm. |
17:55.12 | funxion | so i take it no one has tried it |
17:55.15 | supaigtr | funxion: Have you tested it? |
17:55.21 | supaigtr | I could try it. |
17:55.33 | funxion | I plan to upgrade later today |
17:55.46 | supaigtr | I'll try it real quick. |
17:55.58 | astoria | <PROTECTED> |
17:56.03 | supaigtr | funxion: what device are you sending to? |
17:56.03 | astoria | oh shit |
17:56.09 | astoria | i did not mean to paste that! |
17:56.14 | funxion | cisco 2620xm |
17:56.18 | astoria | well, please don't steal my license code. |
17:56.21 | funxion | lol |
17:56.26 | *** join/#asterisk darkskiez (n=darkskie@host-84-9-237-137.bulldogdsl.com) |
17:56.39 | astoria | well, that's going to get archived.. |
17:56.40 | Katty | astoria: i'll get right on that ;) |
17:56.44 | supaigtr | funxion: 5300 and MaxTNT |
17:56.48 | astoria | and i'm going to get a nasty letter from pdf lib |
17:56.58 | mmlj4 | use free software, no license issues :-) |
17:57.10 | Katty | astoria: you could contact the archivers. |
17:57.23 | astoria | who are the archivers? |
17:57.33 | Katty | the Powers That Be |
17:57.52 | jontow | :o |
17:58.03 | astoria | jesus. today is not my day.. |
17:58.12 | jontow | jbot is the archiver, iirc |
17:58.19 | astoria | i might as well just post my social security number here too.. |
17:58.26 | astoria | who runs jbot? |
17:58.26 | jontow | and if you /whois jbot |
17:58.31 | supaigtr | That would be great lemme get a pin. |
17:58.35 | jontow | you'll see that its TimRiker's bot.. so.. if you /whois TimRiker |
17:58.39 | jontow | ;) |
17:58.43 | astoria | thanks jontow |
17:58.46 | jontow | np |
17:58.56 | jontow | don't know if anyone else archives it |
17:58.56 | twista | FuriousGeorge? sry...i was afk... well of course it does not pick up the line |
17:58.59 | jontow | but worth a shot. |
17:59.39 | supaigtr | :) disable_vpm + KB1 = nice 411p..... |
17:59.47 | twista | the phones ring but they send their rtp packets straight into a wall |
18:01.04 | jontow | i will admit though, from a quick google, looks like its more than just jbot archiving |
18:01.12 | astoria | Oh well... |
18:02.19 | SarahEmm | voip handsets really aren't designed to be used with acoustic couplers. |
18:02.23 | astoria | oh well, it's an old license code anyway.. |
18:03.12 | supaigtr | SarahEmm: I haven't seen one of those in years... |
18:03.16 | SarahEmm | heh :) |
18:03.45 | *** join/#asterisk damned (n=vpol@prior.lanck.net) |
18:04.05 | supaigtr | SarahEmm: 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) |
18:04.31 | SarahEmm | supaigtr: you're using polycoms with acoustic couplers? for what |
18:04.32 | SarahEmm | ? |
18:05.00 | supaigtr | They were testing the hearing impaired devices. 300 baud modem like thingys. |
18:05.01 | Ahrimanes | acoustic coupler = aid for hearing impaired? |
18:05.08 | *** join/#asterisk Exomorph (i=Exomorph@216.251.134.2) |
18:05.15 | jontow | http://www.digitallyunique.com/koupler.html?src=247 |
18:05.22 | jontow | they still make them |
18:05.26 | *** join/#asterisk dstruct (i=dstruct@unaffiliated/dstruct) |
18:05.30 | yxa | does * generally run better with 2.4 or 2.6 kernels? |
18:05.48 | *** join/#asterisk RickTick (n=rpulido_@71.196.17.112) |
18:05.53 | cripito | 2.6 works better for me |
18:05.59 | SarahEmm | supaigtr: ahh. that's what i'm using it for too. the polycom handset on my TTY's cups for testing :) |
18:06.05 | supaigtr | 2.6 and 2.4 here works fine either way. |
18:06.17 | SarahEmm | Ahrimanes: no, acoustic coupler is cups you put the handset of a phone on and it 'listens' and 'talks' on the phone for data use |
18:06.28 | Exomorph | <PROTECTED> |
18:06.29 | Ahrimanes | SarahEmm: ah ok |
18:06.37 | jontow | sarah :))) |
18:06.38 | yxa | supaigtr is 2.6 faster than 2.4? or rather makes better use of cpu? |
18:06.39 | SarahEmm | Exomorph: it's on the digium store |
18:06.41 | SarahEmm | hihi jontow |
18:06.53 | jontow | awesome project, that is |
18:06.54 | supaigtr | yxa: ?? never tested? |
18:06.56 | SarahEmm | Exomorph: store.digium.com |
18:07.01 | SarahEmm | jontow: me? |
18:07.09 | jontow | yes, the tty code |
18:07.14 | SarahEmm | ahh :) |
18:07.15 | Exomorph | SarahEmm: Ahhhh... I was just on the main site, and didn't see it. Thanks. |
18:07.18 | supaigtr | SarahEmm. * has TTY code in it. Support? |
18:07.32 | RickTick | Anyone care to comment on using Quintum 4 port FXS/FXO gateway with asterisk? |
18:08.28 | SarahEmm | supaigtr: 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 |
18:08.40 | SarahEmm | i'm working on a basic set of apps to do stuff that you can do with sounds, but with text |
18:08.52 | SarahEmm | so you could have an IVR-like menu system for TTY users |
18:09.02 | supaigtr | Ahh. Thats cool. I wonder if it would meet requirements eventually of softswitch market? |
18:09.25 | supaigtr | Yea. We are trying to support 3 rd party and 411 type stuff for TTY users. |
18:09.33 | kuku5 | What do you guys think of hp switches ? |
18:09.48 | jontow | kuku5; i have good luck with the procurve 2524's |
18:09.52 | supaigtr | kuku5: they are cute. |
18:10.00 | kuku5 | I used them |
18:10.02 | SarahEmm | supaigtr: ahh nifty :o) |
18:10.09 | kuku5 | but I dont know if they are good ;) they work fine for me |
18:10.14 | kuku5 | Maybe someone had problems |
18:10.19 | SarahEmm | supaigtr: 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 |
18:10.36 | supaigtr | kuku5: Have had some ports go bad but other than that ok. |
18:10.40 | astoria | tty is for the hearing impaired right? |
18:10.48 | astoria | Is 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) |
18:11.20 | supaigtr | SarahEmm: Realtime interface? the stuff we use know is a vb.net interface into a 250,000 based BLVI system. |
18:11.20 | kuku5 | What parts? |
18:11.34 | SarahEmm | astoria: it's a text telephone. for hard of hearing, d/Deaf, speech impaired users. |
18:11.54 | SarahEmm | supaigtr: realtime interface? whatcha mean? |
18:12.06 | supaigtr | TTY is a very cool application in terms of the PSTN and VOIP. |
18:12.12 | SarahEmm | astoria: that's relay, where an operator relays in between voice and TTY users |
18:12.25 | astoria | oh yeah.. relay is the fun one.. |
18:12.38 | SarahEmm | fun? |
18:12.40 | Katty | you know what's really fun? |
18:12.44 | Katty | string theory. |
18:12.45 | Katty | now /that/ is fun |
18:13.08 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
18:13.19 | astoria | yeah, 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.. |
18:13.30 | SarahEmm | *sighs* |
18:13.34 | supaigtr | We 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. |
18:13.55 | Katty | heh 'solution' |
18:13.55 | *** join/#asterisk Xen^ (i=linux@202.5.131.110) |
18:13.58 | Xen^ | hello all |
18:14.08 | Katty | i'm always amused at the lack of information that 'solution' seems to create |
18:14.09 | astoria | i always feel bad for the relay operators, they have a pretty crummy job.. |
18:14.16 | Xen^ | do i need to install zaptel in order to use MOH ? |
18:14.46 | supaigtr | solution = $250,000 software and hardware combo that basically sucks |
18:15.11 | astoria | Xen^: no |
18:15.14 | supaigtr | solution = AKA nortel |
18:15.22 | SarahEmm | astoria: mew? |
18:15.23 | Xen^ | astoria : but its not working :( |
18:15.34 | Xen^ | getting this error local_ast_moh_start: No class: default |
18:15.35 | astoria | SarahEmm: mew? |
18:15.36 | jontow | personally.. i'd love to be a relay operator |
18:15.46 | astoria | you would?? |
18:15.51 | jontow | the fact that you can't really get in trouble unless you DON'T say what they want you to |
18:15.54 | jontow | hehe |
18:15.57 | astoria | lol, that's true! |
18:17.01 | supaigtr | SarahEmm: Has anyone turned tty into IM on SIP? |
18:17.01 | Xen^ | can some one please help :( |
18:17.29 | astoria | Is there any output? |
18:17.30 | SarahEmm | supaigtr: that's the next step for me i think. but it'll be more difficult, TTY is character oriented, IM is block oriented. |
18:17.41 | Xen^ | astoria : yes |
18:17.51 | astoria | Xen^: what does it say? |
18:17.51 | Xen^ | 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) |
18:18.40 | supaigtr | nortel has a char limit type system. Its really strange. Uses operator messaging which is NORTEL only. Really hard to deal with. |
18:19.02 | Xen^ | astoria : this is what i get when i press hold res_musiconhold.c:864 local_ast_moh_start: No class: default |
18:19.06 | supaigtr | TTY device send char limit message pops us sends char limit again message pops up. |
18:19.07 | SarahEmm | supaigtr: hrm? *confused* |
18:19.16 | SarahEmm | i don't understand.. |
18:19.30 | JASON-0 | Every 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: |
18:19.31 | astoria | what do you have in your musiconhold.conf?? |
18:19.34 | JASON-0 | Every 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: |
18:19.39 | JASON-0 | oops |
18:19.40 | Xen^ | astoria : wait |
18:20.10 | astoria | You need to have an entry like this: default => quietmp3:/var/lib/asterisk/mohmp3 |
18:20.32 | Xen^ | astoria : yeah but musiconhold.conf is different :) |
18:20.43 | Xen^ | astoria : check this http://pastebin.ca/26078 |
18:21.10 | astoria | Xen^: are you running cvs-head? |
18:21.15 | Xen^ | yupz |
18:22.13 | astoria | Xen^: 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. |
18:22.51 | Xen^ | but if you see my config it have it |
18:22.52 | Xen^ | :( |
18:22.57 | Xen^ | check this http://pastebin.ca/26078 |
18:23.00 | astoria | well, that's a context. |
18:23.10 | Xen^ | umm |
18:23.31 | Xen^ | default => quietmp3:/var/lib/asterisk/mohmp3 ? |
18:23.58 | astoria | Well, try making a context named [classes] |
18:24.03 | astoria | and putting that in it... |
18:24.06 | astoria | see if it does anything. |
18:24.11 | Xen^ | ok |
18:24.12 | astoria | Are you using the default .conf? |
18:24.17 | Xen^ | yupz |
18:26.32 | supaigtr | SarahEmm: 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 |
18:27.48 | SarahEmm | blvi? |
18:28.03 | SarahEmm | ahh okay re: blocks |
18:29.07 | supaigtr | blvi is just an operator line. |
18:29.14 | supaigtr | You 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) |
18:30.20 | MuppetMaster | Hello |
18:30.27 | MuppetMaster | Interesting presentation posted from Astricon here: http://www.jivesoftware.org/community/servlet/JiveServlet/download/47-16175-105681-2242/astricon-presentation-cleaned-2.ppt |
18:30.50 | rob112 | hey all, anyone up on dsp call progress? |
18:31.06 | SarahEmm | gaaah |
18:31.16 | SarahEmm | i get three lines of code on an app written and find more bugs int he core * code |
18:31.21 | SarahEmm | supaigtr: ahh okay |
18:31.22 | harryvv | I 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. |
18:31.24 | astoria | Xen^: try installing zaptel.. i'm not positivie, but perhaps it requires zaptel timing. |
18:31.38 | SarahEmm | harryvv: that's likely what they did, yeah |
18:31.53 | Xen^ | <astoria> : well thats fix now but now i get no sound |
18:31.53 | harryvv | okay |
18:32.11 | Xen^ | i just edit /etc/asteisk/zapata.conf |
18:32.28 | Xen^ | -- Started music on hold, class 'default', on SIP/1-2c71 |
18:33.13 | harryvv | btw, 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 |
18:33.38 | harryvv | number switch from local carrier to sip provider. Aka, get rid of the local clec |
18:35.04 | SarahEmm | supaigtr: you don't happen to know of any ring-signalling devices that support SIP or IAX do you? |
18:36.54 | supaigtr | SarahEmm: ring-signalling ? |
18:37.31 | cripito | i love this... 92 zap calls not a single drop today |
18:37.37 | supaigtr | U mean like a garage ringer? |
18:39.11 | SarahEmm | supaigtr: any kind of device that's just a ringer :) |
18:39.15 | harryvv | cripto, ive never had a drop on local calls |
18:39.21 | SarahEmm | ideally with visual ringing built in too, but that could be modified in |
18:39.39 | supaigtr | We cheat and use a sipura or a iaxy with a viking or other PSTN type device. |
18:39.55 | supaigtr | I 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) |
18:40.01 | ian_k | SarahEmm - take a shoebox... toss a sip.iax phone inside.. when it rings, just pretend it's a ringing shoebox. |
18:40.10 | SarahEmm | heh, yeah.... |
18:40.19 | JASON-0 | Anyone see anything wrong with this.. |
18:40.20 | JASON-0 | exten => s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} != ${RGPREFIX}]?4:3) |
18:40.22 | SarahEmm | supaigtr: that's what everyone says... |
18:40.28 | SarahEmm | i think i'll design/build a real device. |
18:40.31 | cripito | harryvv is 92 concurrent call from a dialer (predictive) connecting agents. b/f than today.. lots of drops |
18:40.47 | ian_k | JASON-O yeah.. looks like something only asterisk @home could generate.. :) |
18:40.55 | supaigtr | SarahEmm: the viking boxes are big enough to take an IAXY board and integrate into the box. |
18:41.10 | cripito | an entire wct405 :) |
18:41.32 | *** part/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net) |
18:41.43 | JASON-0 | ian_k: yes, I'm using Asterisk@Home |
18:41.51 | JASON-0 | my logs complain about that line.. |
18:42.02 | ian_k | Jason-o - what is the complaint? |
18:42.15 | JASON-0 | Oct 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: |
18:42.28 | JASON-0 | with an arrow under the ! |
18:43.00 | ian_k | is RGPREFIX defined? |
18:43.25 | SarahEmm | supaigtr: they're just audio tho right? |
18:43.52 | harryvv | I 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) |
18:44.32 | GeneG | Hi 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) |
18:44.36 | supaigtr | SarahEmm: They have strobe, relay,audio interface and audio only devices |
18:44.52 | SarahEmm | supaigtr: ahh. :) |
18:45.00 | JASON-0 | ian_k: There is no reference to RGPREFIX prior to this line in that file |
18:45.40 | JASON-0 | ian_k: I dont really know if thats causing a problem.. but every extention I dial gives me a busy signal |
18:45.45 | ian_k | define at as something random and try it again |
18:46.50 | ian_k | harryvv - what codec will you be using? will this be a soft pbx, or will you use digium zap hardware? |
18:47.18 | ian_k | JASOn-O it probably has a lot to do with it, depending on what priority 3 and 4 is for that exten |
18:48.14 | harryvv | ian, well the idea would be to replace the phones so will need local access of telus. it would need some cards for local access. |
18:48.32 | harryvv | no T-1 that I am aware of is invoved but I need to look further into it. |
18:48.51 | harryvv | there is very low call volume involved. |
18:49.01 | harryvv | at any one time. |
18:50.20 | ian_k | will the asterisk box be transcoding channels? |
18:50.35 | ian_k | (connecting different codecs) |
18:51.04 | harryvv | all ulaw |
18:51.07 | JASON-0 | ian_k: I dont know how to define.. |
18:51.35 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:51.52 | ian_k | harryvv you could run that on a low-end box easily |
18:52.33 | harryvv | sure |
18:52.40 | harryvv | but i want dependability |
18:52.55 | harryvv | this server will be located in the next province. |
18:52.57 | SarahEmm | what cards are you using harryvv? that may affect your decision too |
18:52.59 | ian_k | get a dependable low-end box. :) |
18:53.24 | harryvv | SarahEmm well the idea is thay want to save on long distance calls. |
18:53.32 | harryvv | But still need local access. |
18:53.39 | SarahEmm | err... 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) |
18:54.19 | harryvv | sara, no T-1 involved to my knowladge. So say a X100p or equal. mabey two per box. |
18:54.23 | SarahEmm | uhh |
18:54.27 | SarahEmm | they only need to make two calls at once? |
18:54.27 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
18:54.28 | ian_k | JASON-O: RGPREFIX=XYZ |
18:54.32 | SarahEmm | err |
18:54.33 | SarahEmm | one call |
18:54.37 | SarahEmm | plus leave one line for incoming |
18:54.44 | SarahEmm | (and you don't want to use an x100p really) |
18:55.05 | SarahEmm | if you really only do need one call out and one call in at a time, get a BRI |
18:55.08 | SarahEmm | but that sounds low |
18:55.18 | harryvv | its a 20 phone system and at what i was told 4 calls at any one time. So mabey 6-8 incomming calls. |
18:55.28 | harryvv | err |
18:55.35 | harryvv | incomming/outgoing calls I mean |
18:56.49 | ian_k | harryvv a 500mhz system could run a pure sip/iax network with ulaw |
18:57.35 | astoria | I run a webserver/voicemail system on a 500mhz and it runs fine under lots of webtraffic and moderate voip traffic.. |
18:57.41 | astoria | it's even on the public internet. |
18:58.18 | harryvv | ian i know |
18:58.57 | *** part/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4159391.sympatico.ca) |
18:58.59 | harryvv | so 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) |
19:04.39 | SarahEmm | harryvv: okay, so you need a lot more than 2 x100ps then :) |
19:04.59 | SarahEmm | if 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 |
19:05.39 | Inv_arp | SarahEmm: what technology would be best for 8 lines? |
19:06.14 | harryvv | yes I know |
19:06.26 | harryvv | and this is only one site |
19:06.29 | *** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
19:06.30 | harryvv | the main site |
19:06.32 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
19:06.35 | harryvv | where the pbx is located |
19:06.45 | harryvv | now |
19:07.28 | harryvv | I need to look behind this site to see if its T-1 or not. |
19:09.22 | greekman | anyone know if you can force irqs on the te410p? |
19:10.57 | SarahEmm | Inv_arp: probably a PRI with 8 channels active. |
19:11.12 | SarahEmm | you could also go with 4 BRIs... or get termination via IAX |
19:11.14 | SarahEmm | or SIP or whatever |
19:12.40 | astoria | I would go with a partial T1 if it was mission-critical, otherwise i'd just use SIP termination.. |
19:13.08 | astoria | greekman: 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) |
19:18.30 | harryvv | What cards can except more then one pstn connection ? |
19:19.04 | astoria | The Quad TDM cards, with four FXO modules, can. |
19:19.11 | astoria | So can channel banks. |
19:20.18 | SarahEmm | as can the dual/quad T1/E1 cards |
19:21.38 | *** join/#asterisk rocket (n=rocket@gentoo/developer/rocket) |
19:22.08 | vader-wrk | any of you guys use softphones? |
19:23.16 | vader-wrk | hehe |
19:23.21 | vader-wrk | guess no one likes softphones |
19:23.23 | vader-wrk | :) |
19:23.34 | harryvv | what model |
19:23.52 | vader-wrk | hi harryvv |
19:24.35 | harryvv | tdm40b |
19:24.41 | harryvv | hi vadar |
19:24.54 | vader-wrk | im starting to get a tighter plan together for this system |
19:25.21 | vader-wrk | but 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 |
19:25.35 | vader-wrk | which i think will be a major headache and cause the asterisk system to look bad |
19:25.49 | inspired | is NoCDR a good idea before a timeout (t)? |
19:25.58 | vader-wrk | they wanna seamlessly intergrate asterisk with the old pbx |
19:26.05 | vader-wrk | and i don't know if thats going to go smoothly |
19:26.20 | jarrod | hey how do i play back a variable to a customer so they hear the digits |
19:27.09 | jarrod | like Playvariable(${var}) and var=1111111111 |
19:27.09 | inspired | a 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) |
19:28.48 | harryvv | if a company has multiple phone numbers comming into there site does that nessesarly mean thay dont have a T-1 and only pstn connections? |
19:29.11 | harryvv | Want to know this so I wont have to dig though the telco closet and do a physical check. |
19:29.18 | SarahEmm | harryvv: no. could be either. |
19:29.20 | SarahEmm | you need to dig ;) |
19:29.22 | harryvv | This would be for a small site |
19:29.24 | SarahEmm | or look at their bills. |
19:29.25 | harryvv | okay |
19:29.27 | syzygyBSD | any ideas why I would be getting a "post-install wcte11xp failed" when I try to do a `modprobe wcte11xp`? it worked yesterday |
19:29.37 | SarahEmm | what's in dmesg? |
19:29.41 | harryvv | What would there bills say? |
19:29.46 | syzygyBSD | SarahEmm: bootup messages |
19:29.58 | harryvv | I dont think it would show the type of connections would it? |
19:31.16 | SarahEmm | harryvv: it might |
19:31.18 | SarahEmm | it should.. |
19:31.35 | SarahEmm | syzygyBSD: uhh.. yeah, i know. i meant related to the post-install failure :) |
19:32.03 | syzygyBSD | lol.. oh... |
19:32.31 | syzygyBSD | nothing about that module in it |
19:33.05 | syzygyBSD | first 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) |
19:35.36 | syzygyBSD | looks 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 |
19:36.08 | syzygyBSD | wanted to see what it said the first time, if i remove all the modules then modprobe them again everythign works |
19:36.37 | mcf3782 | I 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. |
19:37.16 | mcf3782 | But 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. |
19:37.30 | SarahEmm | mcf3782: your dialplan isn't set up right, likely. |
19:37.33 | SarahEmm | can you pastebin your extensions.conf? |
19:37.34 | mcf3782 | Any ideas where I should start looking for config issues? |
19:37.51 | *** join/#asterisk fifer (n=sirfifer@207.202.227.161) |
19:38.33 | fifer | Any Aastra 480i users here (This is not a call for help, rather the opisite) |
19:38.41 | syzygyBSD | mcf3782: what does the console say? |
19:38.44 | SarahEmm | mcf3782: extensions.conf... can you pastebin it? |
19:39.27 | _Thor | Oct 20 11:59:18 WARNING[580]: chan_oh323.c:4014 oh323_gk_check: Gatekeeper discovery failed. |
19:39.27 | _Thor | <PROTECTED> |
19:39.31 | mcf3782 | SarahEmm, I will go 'sanitize' a copy of my extensions.conf file and do that and let you know when it's there. |
19:39.36 | mutilator | yays |
19:39.42 | SarahEmm | okie |
19:39.47 | mutilator | got a few te405's in today |
19:39.48 | Delta34 | any 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 | _Thor | Hello, 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> |
19:40.17 | mcf3782 | syzygyBSD: The console doesn't say anything for an inbound attempt.. even running in vvvvvv mode. |
19:40.31 | syzygyBSD | only 6 v's? |
19:40.33 | syzygyBSD | lol |
19:40.37 | mcf3782 | :) |
19:40.51 | vader-wrk | any of you guys use softphones? |
19:40.59 | syzygyBSD | vader-wrk: I have one |
19:41.07 | vader-wrk | what client do you use? |
19:41.11 | syzygyBSD | x-lite |
19:41.19 | vader-wrk | how well does it work? |
19:41.23 | syzygyBSD | don't really use it though |
19:41.40 | mcf3782 | I finally got the x-lite softphone to work with my asterisk server. What an adventure that was. |
19:41.46 | syzygyBSD | works well enough, couple issues a friend said he had with it, but mainly phonebook issues |
19:41.59 | vader-wrk | does the phonebook link to asterisk? |
19:42.13 | syzygyBSD | lol, ya, took a while till i found a good configuration FAQ |
19:42.21 | syzygyBSD | vader-wrk: I dont' think so |
19:42.25 | vader-wrk | does it sound ok? |
19:42.54 | mcf3782 | the voice quality in my limited use/tests so far has been acceptable. |
19:45.03 | syzygyBSD | sound 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) |
19:46.02 | davidnicol | who can point me to some /dev/phone ioctl documentation? |
19:47.00 | *** join/#asterisk pussfeller (n=todd@12.150.129.170) |
19:47.14 | twista | re |
19:47.27 | mcf3782 | /m SarahEmm http://pastebin.ca/26090 |
19:47.37 | iDunno | guys: what IP handsets do you recommend? |
19:47.43 | mcf3782 | ok. that didn't work like I thought it would. |
19:47.49 | mcf3782 | I hate learning a new client |
19:48.07 | twista | oh another question...what kind of hardware do i need to connect asterisk to an analog port of a local pbx? |
19:48.15 | mcf3782 | SarahEmm - you have a pastebin link. :) |
19:48.19 | SarahEmm | iDunno: i'm a grrl, but i use a polycom 501... i may not be the best person to ask, but it suits my needs :) |
19:48.34 | twista | one of those ports where you usually connect e.g. a fax machine to |
19:48.34 | iDunno | features that I'd like: programmable buttons, tiltable, clear screens, and easy to use ;) |
19:48.35 | SarahEmm | twista: a TDM400 with one FXO port per port on the PBX |
19:48.38 | syzygyBSD | stupid modules.conf |
19:48.53 | SarahEmm | iDunno: okay, the polycoms don't have tiltable screens, but i've not seen a need to tilt them... |
19:48.54 | syzygyBSD | I will run the commands that need to run, don't do it for me |
19:48.59 | mcf3782 | My extensions.conf is.. a work in progress.. I'm still at the early stages of building this system. :) |
19:49.07 | twista | thx SarahEmm |
19:49.22 | twista | i'll look it up in a second |
19:49.27 | SarahEmm | mcf3782: what context does your sip.conf drop the call into? |
19:50.25 | twista | SarahEmm: is just the TDM400 supported or are there alternatives on the market? |
19:50.26 | davidnicol | does this channel have an answerbot? (hi, I'm new here) |
19:50.46 | loud | yes, type ~docs |
19:50.56 | davidnicol | ~docs |
19:50.57 | jbot | [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 |
19:51.08 | iDunno | SarahEmm: hmmm - there seems to be a few issues with the polycom 501, though - like slow boot and horrible configuration interface... |
19:51.22 | iDunno | otherwise, it fits most other parts ;) |
19:51.53 | mcf3782 | SarahEmm: [kphone] |
19:52.11 | *** join/#asterisk nobell (n=jdegraff@70.103.228.158) |
19:52.25 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
19:53.05 | SarahEmm | iDunno: horrible config interface? huh? |
19:53.12 | SarahEmm | and yeah, they boot slow. but how often do you reboot? |
19:53.32 | SarahEmm | twista: it's the only one i know of. there's actually the single-port x100p too, but it's not what you want. |
19:53.57 | SarahEmm | mcf3782: uhh, the *context* is kphone? |
19:53.58 | twista | erm no...one channel is not that much at all |
19:54.02 | SarahEmm | you don't have a kphone context. |
19:54.30 | mcf3782 | Hmm |
19:54.34 | davidnicol | http://www.linuxjournal.com/node/4468/print appears to be what one gets. |
19:54.59 | mcf3782 | *studies config files some more* |
19:56.38 | SarahEmm | mcf3782: you need a kphone context then, with whatever extensions you want the phone to be able to dia |
19:56.38 | SarahEmm | l |
19:57.10 | iDunno | SarahEmm: well, when reconfiguring things, lots ;) |
19:57.22 | mcf3782 | What if I want that phone to be able to dial anything that's configured on the PBX? :) |
19:57.29 | iDunno | I like the look of the GrandStream GXP-2000 :) |
19:58.12 | iDunno | we'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. |
19:58.29 | iDunno | I've just pointed him at the GXP-2000, so I might have a new phone to play with soon \o/ |
19:58.32 | SarahEmm | *blinks* |
19:58.37 | SarahEmm | the budgetone 101 is not bad? |
19:58.51 | SarahEmm | mcf3782: then you set up a context with all the extensions. |
19:59.16 | mcf3782 | Sounds like I have a lot of typing to do. :) |
19:59.17 | iDunno | it's not bad. it's got an easy to use config front end, and it can grab setup from tftp or http... |
19:59.35 | iDunno | (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) |
20:00.13 | iDunno | the 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. |
20:00.52 | SarahEmm | mcf3782: well, you can use patterns :) |
20:01.09 | SarahEmm | iDunno: so 'it's not bad' means 'it works' ;) |
20:01.19 | twista | iDunno: we just ordered an elmeg 290 phone |
20:01.22 | twista | which is also not bad |
20:01.24 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
20:01.24 | syzygyBSD | dirty te110p takes longer to modprobe then any of the other cards, so it has to be last, or a delay build in |
20:01.27 | SarahEmm | i dunno, personally i thought the 101 was pretty bad. plus mine is now broken... |
20:01.32 | paryl | whenever i have calls go to voicemail, the greeting is never played, it just plays the standard greeting. any idea why that would happen? |
20:01.34 | SarahEmm | i've been really happy with the olycoms |
20:01.46 | twista | but as far as i know its the same as the snome 2xx-something |
20:01.50 | twista | *snom |
20:02.01 | harryvv | idunno, what phone? |
20:02.02 | hans | for those not on -dev, http://hans.fugal.net/src/radp (create dialplans in ruby) |
20:02.14 | twista | and it wasnt too expensive |
20:02.21 | mcf3782 | SarahEmm: Can you give me an example so I have a place to start? |
20:03.01 | SarahEmm | mcf3782: i.e. _9XXXNXXXXXX,1,Dial(...) |
20:03.15 | SarahEmm | mcf3782: that'll match 10 digit local numbers prefixed with a 9 |
20:03.49 | mcf3782 | Ahh! Ok! *light goes off* I think I'm starting to get it now. :) |
20:04.27 | iDunno | SarahEmm: well, yes, it's not bad means that it actually talks to asterisk, and has reasonable sound :) |
20:04.41 | paryl | in one of my extensions i have "exten => 6001,2,Voicemail(u6001)"... shouldn't that play the unavailable greeting? |
20:04.53 | davidnicol | specifically, 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? |
20:05.19 | davidnicol | That is, can Asterisk pretend to be a /dev/phone rather than using a /dev/phone as a hardwware? |
20:06.06 | davidnicol | does openh323 have an IRC channel? I guess I'll ask their mailing list. |
20:06.39 | gordonjcp | can anyone recommend a SIP phone where the message indication light works? |
20:06.57 | gordonjcp | I want to compare what goes on with it to what's happening with my Avaya 4602 |
20:07.08 | iCEBrkr | <PROTECTED> |
20:07.08 | iCEBrkr | <PROTECTED> |
20:07.10 | iCEBrkr | errr. |
20:07.12 | iCEBrkr | WTF |
20:07.26 | iCEBrkr | I can call my 216 number, but not my 727 number? |
20:07.51 | iDunno | twista: ohh, that doesn't look bad :) |
20:13.34 | twista | well it works for me ;) |
20:13.52 | twista | but i actually had some problems with the grandstram budgettone 101 for a while |
20:14.05 | twista | well actually they still exist |
20:14.33 | twista | i don't remember how it exactly was but i think the elmeg phone was able to call the grandstream without problems |
20:14.53 | twista | but the other way round you ended up with something like a 'star trek spacehsip background noise' |
20:14.59 | twista | sounded quite funny |
20:15.38 | twista | and you couldn't here the guy on the other end of the line (wasnt able to figure out the problem yet) |
20:15.43 | twista | here = hear |
20:16.02 | twista | anyways...i'm off for the bed..hope that helps iDunno ;) |
20:16.10 | twista | n8 everyone |
20:17.29 | vader-wrk | do 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) |
20:18.40 | nmsclera | In a small-office scenario, is it wise to use Zap PRI Channels and something akin to AstFax as a fax machine replacement? |
20:18.48 | nmsclera | and if not, what's the most elegant solution aside from a separate line and fax machine for outbound faxing from asterisk? |
20:20.38 | tzanger | I have very very few issues with Tx/RxFax |
20:21.23 | Ahrimanes | tzanger: does Tx/RxFax require you to have pstn in the receiving asterisk or should it work over iax trunks? |
20:22.19 | tzanger | Ahrimanes: fax over IP is inherently tricky |
20:22.22 | tzanger | unless you use t.38 |
20:22.51 | tzanger | well t.37/t.38 |
20:22.59 | *** join/#asterisk Weezey (i=Weezey@206.210.109.229) |
20:23.00 | Ahrimanes | tzanger: ok.. any semi-usable solutions with t.38 currently available? |
20:23.15 | tzanger | but if you're IP link is rock-solid (hint: it's not) it works just fine |
20:23.31 | SarahEmm | bbiab. |
20:23.31 | tzanger | not 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) |
20:24.24 | e-Hernick | There has couple thousand $ in bounty for t.38 support but nobody is taking it |
20:24.37 | e-Hernick | If you want to do fax handling with *, don't count on t.38 |
20:24.51 | Ahrimanes | tzanger: well i'm 4 hops from the other * |
20:24.51 | e-Hernick | You could run * to do store-and-forward though. It fully receives the faxes and then resends them |
20:24.58 | nmsclera | tzanger: 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 |
20:25.10 | Weezey | Is 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. |
20:25.24 | Ahrimanes | tzanger: but does it require anything at the pstn terminating asterisk? |
20:25.32 | nmsclera | tzanger: or have I just not read the wiki thouroughly enough? |
20:25.40 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
20:26.14 | tzanger | nmsclera: I believe my solution's already on voip-info |
20:26.27 | tzanger | Ahrimanes: no, I take a PRI and the fax DID I send to rxfax |
20:26.39 | tzanger | rxfax receives, converts to PDF and scp's to my server |
20:27.14 | tzanger | I am 1 IAX2 hop from my PRI on a dedicated VOIP link |
20:27.32 | tzanger | actually I don't go to rxfax now, we have two real fax machines we send and receive from |
20:27.58 | Ahrimanes | ok, 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) |
20:28.10 | paryl | when 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) |
20:28.23 | kippi | hey |
20:28.24 | e-Hernick | paryl, the lock file. |
20:28.34 | tzanger | e-Hernick: hahah |
20:28.49 | tzanger | paryl: I'd check app_voicemail.c and see what it's trying to do |
20:28.52 | tzanger | offhand I don't know |
20:29.16 | e-Hernick | paryl, you say it works when you're not debugging ? |
20:29.25 | kippi | is there way to see if ztdummy is loaded? |
20:29.44 | paryl | yeah, 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) |
20:30.00 | Ahrimanes | kippi: lsmod ? |
20:30.06 | *** join/#asterisk pbd (n=plancomm@12.144.118.37) |
20:30.07 | e-Hernick | paryl, so everything works fine but you get a warning message ? |
20:30.22 | pbd | Greetings, all. |
20:30.32 | e-Hernick | Greetings, pbd. |
20:30.36 | paryl | right... 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) |
20:31.07 | e-Hernick | paryl, well, you say that voicemail works fine. does anything fail to work ? |
20:31.36 | paryl | not so far, afaik |
20:31.37 | kippi | if I have a TE100P card should i need to load ztdummy? |
20:31.47 | paryl | but it's obviously a concern |
20:31.48 | Ariel_ | I would not worry about all the warnings and notices There not an error. |
20:31.55 | Ahrimanes | kippi: dummy is if you dont have hardware |
20:32.04 | Ariel_ | kippi, no |
20:32.18 | kippi | if I don't have a line going into it, could this effect it? |
20:32.29 | e-Hernick | the only warning you want to look out for is "Unable to create lock file: Preparing to restore directory tree / to clean state" |
20:32.42 | e-Hernick | If you get it, you must exit asterisk within 10 seconds |
20:32.43 | pbd | kippi: No, the timing is done on PCI bus interrupts only. Connection is not necessary. |
20:33.17 | kippi | ok, one last question, is there away to make sure the card is installed probley? |
20:33.35 | e-Hernick | Well, if the card works with asterisk, then you know it's installed properly. |
20:34.17 | kippi | how can I tell if the card is working with asterisk? |
20:34.36 | e-Hernick | Can you get a line running into the card ? |
20:35.04 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
20:35.05 | kippi | not for another few days, and my voicemail isn't working, and i think it could be because of the card |
20:35.08 | paryl | something 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 |
20:35.12 | syzygyBSD | has 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 |
20:35.33 | e-Hernick | How would a disconnected card break your voicemail ? |
20:35.49 | e-Hernick | It don't make no sense. |
20:36.21 | sgorilla | ok it makes sense |
20:36.31 | e-Hernick | HOW? |
20:36.39 | e-Hernick | speak, gorilla man ! |
20:36.40 | kippi | because of the timings, if the card isn't working then I need to get it working so the timing is working |
20:36.53 | sgorilla | don't make no (double negative) is positive |
20:37.11 | e-Hernick | I've been watching firefly again. |
20:37.13 | sgorilla | you debug that thread? |
20:37.22 | e-Hernick | It's affected my language. |
20:37.36 | e-Hernick | kippi, what you need is to do is plug your card into either a live line or an ATA |
20:37.52 | sgorilla | kippi: you are saying the zaptel card is broken |
20:37.58 | sgorilla | but its using the kernel module for timing |
20:37.59 | e-Hernick | kippi, lest you do that, there can be no hope for proper debugging. |
20:38.02 | sgorilla | sense its broken |
20:38.05 | sgorilla | it messes stuff up |
20:38.10 | sgorilla | is this correct? |
20:38.12 | pbd | Kippi: You should be able to read the dmesg log for any problems with the Zaptel card. |
20:38.19 | cripito | mmmmm zaptel and voicemail related? |
20:38.22 | pbd | Or do a cat /proc/interrupts, and see it listed there. |
20:38.22 | e-Hernick | He's not saying anything, I'm not sure he knows what's going on |
20:38.23 | sgorilla | you should put gdb on it |
20:38.31 | sgorilla | that is true |
20:38.37 | *** join/#asterisk clive- (n=pirch@ndn-165-130-127.telkomadsl.co.za) |
20:38.48 | e-Hernick | kippi, you could just disable the zaptel card and see if it fixes your VM |
20:38.57 | sgorilla | set a breakpoint where it starts recording the messages |
20:38.57 | e-Hernick | kippi, take it out of the box if you want to be sure |
20:39.00 | cripito | or just bring ztdummy |
20:39.06 | sgorilla | e-Hernick: thats what i would do |
20:39.11 | sgorilla | possibly easiest fix |
20:39.37 | kippi | if I install ztdummy will that effect it latter when I put a line into the card? |
20:39.50 | sgorilla | haha |
20:39.55 | sgorilla | yes it probably will |
20:39.55 | e-Hernick | If 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. |
20:39.59 | sgorilla | if you are having problems |
20:40.05 | sgorilla | just physically take out the card |
20:40.12 | sgorilla | and reinstall asterisk |
20:40.15 | sgorilla | go from there |
20:40.20 | sgorilla | once you get voicemail working |
20:40.24 | sgorilla | install the card, see if it works |
20:40.32 | sgorilla | then compile the module, load it, see if it works |
20:40.38 | sgorilla | start from least complex to more complex |
20:40.40 | kippi | ok |
20:40.47 | e-Hernick | You can also recompile your kernel while offering the sacrifice of a blue-eyed kitten to the God of Digital Telephony. |
20:41.44 | kippi | ok cool |
20:41.48 | kippi | thanks for that |
20:42.07 | e-Hernick | Hey, do we have an IAX2 conference channel |
20:42.10 | e-Hernick | like #openpbx does |
20:43.23 | *** join/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net) |
20:43.34 | Bentley | I 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? |
20:44.11 | e-Hernick | It depends on what your definition of "truth" is. |
20:44.42 | Bentley | well, opposite of false |
20:44.56 | e-Hernick | " |
20:44.56 | e-Hernick | You 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." |
20:45.03 | e-Hernick | http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
20:45.33 | Bentley | ahh, thx |
20:46.11 | jarrod | whats a good commercial softswitch to use |
20:46.24 | e-Hernick | Asterisk Business Edition |
20:46.30 | jarrod | besides asterisk |
20:46.43 | e-Hernick | OpenPBX |
20:46.49 | FuriousGeorge | includes will catch dialpatterns in the included context so long as you put it before the catchall patter (1NXXNXXXXXX) |
20:46.51 | jarrod | not free |
20:47.12 | jarrod | for a provider platform |
20:47.21 | FuriousGeorge | right? |
20:47.35 | e-Hernick | My new fork of *, which I call "Expensive Commercial PBX Championship Edition" |
20:47.51 | e-Hernick | What about SER ? |
20:48.23 | sgorilla | "The Truly Open Source PBX" TM |
20:48.29 | kippi | do many ISP/VoIP companys use asterisk? |
20:48.31 | FuriousGeorge | iow 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 |
20:48.42 | e-Hernick | kippi, one hundred billion |
20:48.43 | jarrod | kippi: not the very successful ones |
20:48.54 | kippi | how comes? |
20:49.05 | sgorilla | cool offshot of asterisk |
20:49.12 | sgorilla | i never liked the asterisk license |
20:49.15 | kippi | is there a better system to use to build voip server? |
20:49.15 | GXTi | sometimes companies just need something from scratch |
20:49.17 | e-Hernick | The very successful ones use a combination of * and SER |
20:49.28 | GXTi | but in most cases asterisk is more than sufficient |
20:49.40 | kippi | anyone got a link for SER? |
20:49.42 | sgorilla | i looked into contributing some patches to asterisk |
20:49.51 | *** part/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net) |
20:49.52 | sgorilla | but you had to jump through all these hoops to do it |
20:49.57 | e-Hernick | Yes, * is very good, and combined with SER it's even better. But for smaller-scale setups, there is little need for SER. |
20:50.13 | kippi | ah |
20:50.15 | GXTi | sgorilla: could be worse |
20:50.21 | cripito | and u have to be real real big to put sip in from an asterisk server |
20:50.23 | GXTi | sgorilla: apple just throws out patches and redoes them from scratch |
20:50.59 | NetSkier | e-Hernick: Where is the tradeoff point, where adding SER makes things work better? |
20:51.17 | jarrod | asterisk does not perform well on the ip centrex provider platform |
20:51.37 | e-Hernick | Well, the tradeoff point is when r > 0.42, where r is the SER-requisiteness factor of your * installation. |
20:51.37 | jarrod | where things like FOP is the only presence utility available |
20:51.52 | jarrod | and scalability/redundancy is key |
20:52.01 | cripito | :) that depends ;) expecially if u have u own tool that can replace FOP |
20:52.11 | jarrod | yes cripito :) |
20:52.33 | *** join/#asterisk clint_ (n=clint@snap.helixsystems.com) |
20:52.58 | cripito | i had my own tool for manage the asterisk.. and an asterisk system managing 320 sip phones... without big issues in that part |
20:53.13 | cripito | so i am thinking that the rule of ser not always apply |
20:53.25 | cripito | at least u need more than a super huge system to need the ser |
20:53.35 | rocket | I 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 |
20:54.45 | jarrod | just create an extension that dumps to that context |
20:54.45 | e-Hernick | rocket, what about you install a softphone ? |
20:55.00 | dstruct | anyone here doing a pure meetme server? |
20:55.31 | rocket | e-Hernick: I did .. and I can dial ext 200 for example .. maybe I am just confusing myself .. :/ |
20:55.47 | rocket | e-Hernick: but how do i get the autoattendant to answer? |
20:56.04 | e-Hernick | Have you gotten the autoattendant to answer any phone at all ? |
20:56.12 | e-Hernick | Or are you still trying to do that ? |
20:56.29 | *** join/#asterisk jeremywhiting (n=jeremy@71-37-101-103.slkc.qwest.net) |
20:56.39 | rocket | none at all .. still trying to do that .. basically I am really trying to learn it and understand it .. |
20:56.47 | e-Hernick | Ah, I see. |
20:56.57 | cripito | ibye the way FOP is not a bad tool anyway |
20:57.01 | e-Hernick | Well, if you're starting with *, I might recommend that you learn AEL instead of extensions.conf |
20:57.05 | e-Hernick | With * 1.2.0-beta1 |
20:57.22 | cripito | and i known asteria have his own too |
20:57.24 | e-Hernick | I figure you're not going to go in production very soon, and the AEL is like 100 times nicer than extensions.conf |
20:58.33 | FuriousGeorge | http://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 |
20:58.40 | rocket | e-Hernick: ok .. I guess I was just messing with asterisk at home .. but it sounds like I need to go more low level |
20:58.50 | e-Hernick | rocket, what do you mean more low level ? |
20:59.18 | e-Hernick | rocket, asterisk is a very programmable system, and that's the strength of *. |
20:59.45 | rocket | well 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 |
21:00.17 | harryvv | how many here have integrated a legacy pbx with asterisk pbx? |
21:00.17 | e-Hernick | rocket, 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 |
21:00.17 | rocket | if that makes sense .. |
21:00.26 | e-Hernick | rocket, 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) |
21:00.32 | e-Hernick | rocket, but you've got to read about extensions.conf and how to program for it |
21:00.39 | kippi | is there away you can send txt/email messages to your phone on asterisk? |
21:00.41 | e-Hernick | harryvv, I've integrated a norstar with * using sipuras |
21:01.13 | rocket | e-Hernick: ok .. thanks .. I know I am missing something there so I will read more .. thanks for the pointer |
21:01.51 | harryvv | e-hernick, is there any web sites on how to integrate lagacy pbxs with asterisk? |
21:03.49 | FuriousGeorge | ok pop quiz |
21:03.55 | FuriousGeorge | everybody ready? |
21:03.58 | FuriousGeorge | http://pastebin.ca/26108 |
21:04.05 | FuriousGeorge | just shout out when you know the answer |
21:04.29 | *** join/#asterisk fanguin (n=user@p548F3EF0.dip.t-dialin.net) |
21:05.28 | FuriousGeorge | anyone? i thought that would be pretty basic. i can't test because the system is in use right now |
21:05.43 | fanguin | there is an "application: " line in callfiles to run an application. is it also possible to execute two apps? |
21:08.00 | FuriousGeorge | blitzrage: wake up im asking you a question :) |
21:09.36 | Ariel_ | 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) |
21:10.07 | stickyhorsey | sup folks |
21:10.11 | Ariel_ | It will match the ld settings first before the local one |
21:10.24 | harryvv | Are 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. |
21:11.06 | cripito | there is something in the wiki... but not customer level |
21:11.22 | FuriousGeorge | Ariel_: 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 |
21:12.44 | cripito | why 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) |
21:13.18 | Ariel_ | wow strange got knocked off |
21:13.30 | FuriousGeorge | i 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) |
21:13.46 | cripito | the old legacy pbx do that |
21:13.52 | FuriousGeorge | so if you wanna "pull" a call from me just transfer to ${MYEXTEN}# or something |
21:14.06 | sgorilla | kippi: yes |
21:14.07 | stickyhorsey | anyone here have time to help me figure out why i cant get app_backticks installed? |
21:14.28 | cripito | kippy lsmod | egrep ztdummy ;) |
21:14.35 | e-Hernick | there are a great many ways to integrate with a legacy pbx |
21:14.44 | e-Hernick | there is no "one size fits all" |
21:15.16 | sgorilla | e-Hernick: how about digital legacy pbx? |
21:15.26 | sgorilla | don't you need to reverse engineer it? |
21:15.38 | sgorilla | to get it to work with asterisk |
21:15.38 | e-Hernick | sgorilla, what difference does that make? You're going to interface through analog anyway. |
21:15.46 | e-Hernick | I'm integrating with a digital norstar |
21:16.00 | e-Hernick | Through the ATA and the line ports |
21:16.28 | cripito | yeap.. |
21:16.36 | oliverqg | hi 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? |
21:17.00 | sgorilla | e-Hernick: i though to do some of the funciontlity of the pbx |
21:17.19 | sgorilla | e-Hernick: like transfer lines, etc etc, like using the legacy phones functionality |
21:17.28 | e-Hernick | sgorilla, what kind of PBX |
21:17.36 | oliverqg | sorry I meant asterisk box.. |
21:18.27 | sgorilla | e-Hernick: nortel |
21:18.37 | sgorilla | toshiba |
21:19.07 | sgorilla | how would you reverse engineer those? |
21:19.18 | sgorilla | with an A-D card? |
21:20.38 | cripito | i think that he connected analogic... i did that with mine any way too |
21:21.01 | cripito | i just make a crossover T1 btw then |
21:21.45 | sgorilla | whats a crossover T1? |
21:22.09 | cripito | the 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 |
21:22.24 | harryvv | Im asuming that a T-1 is the same cable that is a dce/dte cable hooked to a cisco router? |
21:23.14 | FuriousGeorge | my 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 |
21:23.41 | FuriousGeorge | there has to be a way, but i cant picture it |
21:23.51 | e-Hernick | It depends on the size of your target system |
21:24.08 | *** join/#asterisk p0lar69 (n=kvirc@155.101.179.19) |
21:24.23 | e-Hernick | And on a great many other factors. |
21:25.00 | cripito | http://www.cisco.com/en/US/products/hw/routers/ps233/products_tech_note09186a00800a3f09.shtml#topic2 |
21:25.03 | e-Hernick | sgorilla, what kind of nortel are you looking at ? |
21:25.15 | e-Hernick | They make systems of pretty much every shape and size |
21:25.30 | e-Hernick | their 250000 extension systems are nice |
21:25.42 | e-Hernick | require a lot of wiring |
21:25.44 | cripito | :D that's too way big for me :P |
21:26.03 | e-Hernick | asterisk is pretty useless at these scales too |
21:26.18 | cripito | almost everything is uselles at these scales too |
21:26.23 | harryvv | scales? |
21:26.26 | harryvv | what scales |
21:26.33 | FuriousGeorge | fish scales |
21:26.34 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com) |
21:26.39 | e-Hernick | a PBX with 1/4 million phones |
21:26.39 | cripito | 250K extensions |
21:26.45 | harryvv | good grief |
21:26.56 | harryvv | that would take banks and banks of pbxes |
21:27.06 | e-Hernick | well |
21:27.10 | e-Hernick | think pentagon |
21:27.14 | harryvv | yea |
21:27.25 | cripito | i am trying to figure out how many asterisk servers u need for that :D |
21:27.27 | FuriousGeorge | one hundred dual opterons? |
21:27.27 | e-Hernick | or, think your average big-city CO |
21:27.29 | harryvv | I suspect thay have several pbx rooms. |
21:27.31 | FuriousGeorge | dual core? |
21:27.45 | e-Hernick | FuriousGeorge, nah, these systems are probably running antiquated 233mhz cpus |
21:28.03 | cripito | yup |
21:28.05 | FuriousGeorge | e-Hernick: i was thinking in terms of Asterisk hardware requirements |
21:28.17 | cripito | but they are totally segmented or break in modules |
21:28.26 | sgorilla | e-Hernick: any nortel, i guess, not any particular model |
21:28.50 | sgorilla | what is the most stable pbx? |
21:29.09 | sgorilla | that has hardware failover |
21:29.25 | sgorilla | not sure how you would have failover when you have a pri card that is connected |
21:29.41 | sgorilla | could you physically make a switch that goes between two asterisk boxes |
21:29.51 | sgorilla | and if one dies |
21:29.59 | sgorilla | it would physically switch the connection for the pri |
21:30.15 | FuriousGeorge | lets 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? |
21:31.20 | FuriousGeorge | ??? |
21:31.40 | cripito | <--- is back to his parked calls ;) |
21:31.57 | davidnicol | packet-switching does failover much more gracefully than circuit-switching |
21:32.42 | e-Hernick | with circuit switching, failover usually means you have a hot spare standing by with instructions on how to connect it |
21:32.54 | e-Hernick | when 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) |
21:33.21 | e-Hernick | it's not going to automagically failover |
21:33.44 | QbY | there was a project a while back that did queue reporting.. anyone know the name? (shows calls, hold times, etc..) |
21:33.50 | FuriousGeorge | how can i catch a goto(context,1,${EXTEN}) that doesnt have a matching extension |
21:33.51 | cripito | not really |
21:34.00 | cripito | think that u have 2 pri |
21:34.07 | cripito | the telco will do the failover for u |
21:34.14 | FuriousGeorge | does it come back on priority n+100? |
21:34.23 | e-Hernick | sure, if you have multiple circuits |
21:34.59 | cripito | if 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) |
21:35.08 | cripito | 2 twin system |
21:35.15 | cripito | so if 1 goes down |
21:35.23 | cripito | the another start working |
21:35.31 | cripito | even your phones can deal with that |
21:35.42 | cripito | the mayority of the new sip phones have a failover sip server |
21:36.07 | cripito | at least the decent ones ;) |
21:36.35 | Jabroni | guys anyone have problems with IVR that it stops after like 2 seconds of playing, then the line hangs up |
21:37.00 | Jabroni | http://pastebin.com/400310 |
21:37.03 | cripito | check the priorities |
21:37.06 | Jabroni | thats that it executes |
21:38.28 | Jabroni | its a remote sipura trunk, i checked the disconnect message on the sipura page, and it says voip call ended |
21:39.34 | cripito | -- Executing Hangup("SIP/797-9a7e", "") <---- i bet u this will disconnected the call for the sipura |
21:39.49 | Jabroni | the last priority asignted to the S extension is the one from NvBackgroundDetect() |
21:40.16 | cripito | nothing else from there? NVback.. is not jumping to any other? |
21:40.25 | Jabroni | yeah.. but there is no reason for a hangup |
21:40.31 | Jabroni | i just have a h,1,hangup() |
21:42.05 | Jabroni | i updated the pastebin with the extensions context |
21:43.34 | *** join/#asterisk Talnakh (n=Talnakh@217.22.177.17) |
21:43.50 | cripito | what happen if talk? |
21:44.05 | cripito | instead of a dtmf? |
21:44.30 | Jabroni | uhm.. im just listening for dtmf |
21:44.47 | Jabroni | u think its something related to nvbackgrounddetect()? |
21:45.04 | cripito | small test... put talk, 1, <something> and see what happen |
21:45.07 | Talnakh | hi 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? |
21:46.14 | Jabroni | testing now |
21:46.32 | Jabroni | k its doing tht |
21:46.36 | Jabroni | its the talk |
21:47.47 | Jabroni | woot |
21:47.49 | Jabroni | worked :) |
21:47.54 | Jabroni | i added the t flag |
21:47.57 | Jabroni | to nvbackgrounddetect |
21:48.46 | Jabroni | Ok.. another question... how asterisk manages the anwser detection for outgoing calls ?? |
21:49.27 | Jabroni | it has to be with the indications.conf ? |
21:50.06 | *** join/#asterisk buddah (n=djbrianc@67.110.253.128) |
21:50.07 | cripito | u mean answer machine detection? of just the answer? |
21:50.17 | buddah | anyone ever have a problem with musiconhold, that you get no sound? |
21:50.18 | Jabroni | just the anwser |
21:50.29 | Jabroni | some calls just get dropped after 3 minutes |
21:50.30 | buddah | when i test it, it just does start musiconhold, then stop musiconhold right away |
21:50.31 | Jabroni | checked the CDR |
21:50.42 | cripito | check this DIALSTATUS |
21:50.52 | Jabroni | and the disconnect reason is that it thinks that it was not anwsered |
21:51.52 | cripito | i told too soon :) |
21:53.13 | Jabroni | the dialstatus var is used for the dialplan right ? |
21:53.20 | cripito | yup |
21:53.51 | Jabroni | but how would i use it ?? I mean.. on my dialout-pstn context, i have Dial(ZAP/4-1/123123123) |
21:54.02 | Jabroni | after 3 mintures... boom.. the calls get dropped |
21:54.13 | Jabroni | but random |
21:55.11 | cripito | mmm interesting |
21:55.18 | buddah | anyone that might be able to help with musiconhold? |
21:55.47 | Jabroni | yup.. ive read bout the indications.conf.... but from what ive read that for detection for the incoming calls, hangup tones |
21:55.59 | Jabroni | never seen anything releated to detecting detecting outgoing anwsered calls |
21:56.16 | Jabroni | and using digium cards.. (hate those cheaps X100p clones :p) |
21:57.06 | Talnakh | hi 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? |
21:57.09 | cripito | :) xfo cards or tdm? |
21:57.17 | Talnakh | xfo |
21:57.26 | Jabroni | TDM |
21:58.02 | cripito | was for jabroni. i think that the issue for sharing interrupts could be for the tdm cards only |
21:58.37 | cripito | at 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) |
21:59.08 | edonkey | hello! |
21:59.21 | cripito | nice.. can we download something nice? |
21:59.30 | edonkey | i am getting the error: misdn_cfg_get: Invalid call to misdn_cfg_get! Port number 0 is not valid. |
21:59.32 | edonkey | <PROTECTED> |
21:59.34 | edonkey | any idea why? |
22:00.49 | *** join/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz) |
22:00.52 | patpatnz | hi guys |
22:00.52 | cripito | jabroni so u have a pri for just testing? |
22:00.54 | cripito | nice |
22:01.01 | cripito | i always wanted that |
22:01.13 | patpatnz | is there a way to stop * from proxying the RTP for a call going from SIP->H.323 |
22:01.14 | patpatnz | ? |
22:01.59 | *** join/#asterisk |cleric| (n=dacleric@p5482B4A9.dip0.t-ipconnect.de) |
22:02.18 | Jabroni | ive already set up 4 asterisk boxes :p |
22:02.36 | Jabroni | but those are issues that it have been popin' from my install.... |
22:02.50 | Jabroni | after fixing that I need to check how to get the call pickup workin' |
22:03.28 | cripito | that should be easy.. i still wanna my own pri for just testing :) |
22:03.43 | cripito | anyone known a asterisk solution provider in colorado? |
22:03.50 | Jabroni | i tried on my bro asterisk |
22:03.52 | Jabroni | and no luck :( |
22:04.11 | Jabroni | with the default implementation I can pickup a Zap call ringing a SIP channel on another sip channel right? |
22:04.32 | patpatnz | anyone? |
22:04.52 | cripito | i think that callgroup and pickupgroup should be enough |
22:05.02 | cripito | at least it work for me. |
22:05.23 | Jabroni | u added that on the sip.conf right ? |
22:05.35 | cripito | :/ i have realtime sip... but yes. |
22:07.00 | cripito | or someone... |
22:07.12 | hypa7ia | cripito: i know people who do in CA / NV |
22:07.22 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
22:07.23 | Jabroni | what kinda solution u need cripito? |
22:07.43 | cripito | not.. i wanna make a small demo for those guys.. i am not buying... |
22:07.51 | Jabroni | ooh |
22:07.53 | cripito | the idea is sell |
22:07.55 | Jabroni | sell |
22:07.56 | Jabroni | hehe |
22:07.56 | Jabroni | yeah |
22:08.12 | Jabroni | btw.. for what do u use the group=1 ? |
22:08.14 | cripito | www.cripiland.com/screenshots/manager4.jpg |
22:09.27 | ender | The requested URL /screenshots/manager4.jpg was not found on this server. |
22:10.21 | cripito | ploading |
22:10.22 | cripito | wait |
22:10.40 | cripito | http://www.cripiland.com/screenshots/manager1.jpg |
22:10.55 | *** join/#asterisk corry1000 (n=corne@ndn-165-149-251.telkomadsl.co.za) |
22:12.20 | Jabroni | ok 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 ? |
22:12.35 | Jabroni | with *8<extension> |
22:12.49 | cripito | http://www.cripiland.com/screenshots/manager4.jpg |
22:13.10 | cripito | even with *8 only |
22:13.14 | ender | cripito: Java? |
22:13.18 | cripito | c# |
22:13.29 | ender | oh |
22:13.51 | cripito | not documentation yet .. :((( |
22:13.57 | vader-wrk | cripito what are you using to manage asterisk? |
22:14.27 | *** part/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz) |
22:14.28 | Jabroni | i just get a busy tone on the phone |
22:14.34 | corry1000 | has anybody heart of a dss module or a snom dss module. i would really Appreciate |
22:14.34 | corry1000 | <PROTECTED> |
22:14.36 | cripito | http://www.cripiland.com/screenshots/manager3.jpg |
22:14.47 | cripito | vader i install astmanproxy |
22:14.52 | cripito | and build my own too |
22:14.53 | cripito | tool |
22:15.13 | sgorilla | looks good |
22:15.20 | sgorilla | what does it use as a database backend |
22:15.42 | cripito | as 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) |
22:16.05 | cripito | i hate parked calls ..... |
22:16.18 | tim27 | any here use voctel ??? |
22:16.28 | rayvd | and i hate parking at malls! >:o |
22:16.47 | cripito | :D na.. that depends on the hr... |
22:16.58 | rayvd | well it rhymed :) |
22:17.06 | cripito | try to park there at 11:59pm |
22:17.07 | cripito | ;) |
22:17.16 | cripito | u have the entire parking lot for u :P |
22:17.53 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
22:18.10 | sgorilla | cripito: you running that in house? |
22:18.34 | cripito | i am connected to a vpn to the site that i use for testing sgorilla. |
22:18.51 | Jabroni | cripito if I dial *8 should I be getting some output on the asterisk console? |
22:19.19 | cripito | when u make a reload extension did u make a reload res_features.so jabroni? |
22:19.49 | cripito | sgorilla the server for those screen shots is in atlanta.. |
22:19.52 | cripito | i am in colorado |
22:19.58 | Jabroni | just done it .. forgot to do it |
22:19.59 | Jabroni | hehe |
22:20.17 | sgorilla | cool |
22:20.21 | sgorilla | what type of vpn? |
22:20.41 | cripito | it sucks beleive me.... |
22:20.48 | Jabroni | no workie |
22:20.48 | Jabroni | oh wait.. do i have to include something? |
22:20.48 | Jabroni | in the context? |
22:20.59 | ender | can * 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) |
22:21.13 | cripito | no. jabroni. |
22:21.14 | wundaboy | i 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) |
22:21.32 | Jabroni | cuz the sip invites do are reaching asterisk |
22:21.34 | Jabroni | with *8 |
22:22.15 | cripito | :) basically it should work out of the box as soon u put the pickupgroup and callgroup |
22:22.42 | Jabroni | btw my base asterisk is an asterisk@home install |
22:22.42 | wundaboy | what does this mean? |
22:22.44 | wundaboy | WARNING[13012]: loader.c:440 load_modules: Loading module format_au.so failed! |
22:22.47 | iDunno | as long as it's enabled... |
22:22.54 | iDunno | in features.conf (IIRC) |
22:22.55 | *** join/#asterisk Katty (n=katrina@68-112-15-110.dhcp.cpgr.mo.charter.com) |
22:23.28 | tim27 | any have a clue why i can receive caller id from a voctel did using asterisk@home |
22:23.56 | cripito | wundaboy.. :) no big deal add in modules.conf noload=>format_au.so ... :) i don't think to many ppl use au anyway :D |
22:24.04 | tim27 | is it voctel that dont support it or it a config prob ??? |
22:24.49 | Jabroni | cripito what do u have in your features.conf file ? |
22:24.58 | cripito | 1 sec |
22:25.38 | wundaboy | cripito: roger |
22:26.09 | cripito | http://pastebin.ca/26116 |
22:26.31 | Jabroni | thats a 1.20 insatll? |
22:26.33 | wundaboy | chan_features.so failed! |
22:26.34 | Jabroni | or 1.0.9? |
22:26.50 | cripito | :D chan_features? |
22:26.55 | wundaboy | do i need chan_features? |
22:27.02 | cripito | yes u need.. |
22:27.07 | wundaboy | how do i fix it |
22:27.14 | tim27 | any have a clue ??? |
22:27.14 | FuriousGeorge | i have about three hundred area code exchange combinations that are considered local that i have to integrate into my dialplan |
22:27.17 | cripito | i could be mistaked but res_features is for 1.2.x |
22:27.21 | FuriousGeorge | im not looking forward to it |
22:27.32 | cripito | but chan_features is from 1.0.X? |
22:27.34 | wundaboy | this is how i installed asterisk (im nub to asterisk) 'emerge asterisk' |
22:27.47 | cripito | gentoo... |
22:27.48 | wundaboy | cause i know you all <3 gentoo.... |
22:27.53 | tim27 | anyone here know if voctel support CALLER ID |
22:28.20 | cripito | i prefer FC |
22:28.22 | cripito | sometimes |
22:28.30 | wundaboy | cripito: im trying to learn asterisk, what is the easiest way to install asterisk? |
22:28.42 | wundaboy | or best |
22:28.53 | tim27 | wundaboy: install asterisk@home |
22:29.00 | Jabroni | i started with asterisk@home |
22:29.08 | gordonjcp | cd /home/pkgsrc/comms/asterisk && make && make install |
22:29.13 | cripito | the easier way? have someone install tha for u :D and later explain |
22:29.13 | wundaboy | whats asterisk@home? |
22:29.15 | Jabroni | but onces you want to start doing some more complex stuff |
22:29.16 | gordonjcp | worked for me |
22:29.23 | Jabroni | u have to dump a@h |
22:29.36 | gordonjcp | what's "u" |
22:29.36 | cripito | i don't like a@h |
22:29.37 | gordonjcp | ? |
22:29.44 | cripito | you = u |
22:29.55 | gordonjcp | why not just type "you" then? |
22:29.56 | tim27 | a@h is good to become familiar |
22:30.04 | tim27 | you can check how the conf file |
22:30.18 | tim27 | are modified when you make a change |
22:30.21 | tim27 | in the web interface |
22:30.25 | cripito | custom. |
22:30.26 | wundaboy | is asterisk difficult to setup? |
22:30.38 | tim27 | with a@h |
22:30.44 | tim27 | very easy |
22:30.55 | tim27 | you grand mother can do it |
22:31.08 | gordonjcp | I doubt it |
22:31.12 | tim27 | wundaboy: what kind of setup you want |
22:31.12 | cripito | even without the home... if u client is not doing crazy things... |
22:31.15 | tim27 | soho ??? |
22:31.26 | gordonjcp | I don't think my grandmother could anyway |
22:31.26 | *** join/#asterisk jets (n=jets@guardian.pmt.org) |
22:31.30 | wundaboy | i have one phone, and hopefully it will be growing to more. |
22:31.32 | wundaboy | i want voicemail |
22:31.33 | gordonjcp | so hard to get ADSL installed in a coffin |
22:31.35 | wundaboy | a menu system |
22:31.36 | cripito | i can't wait to 1.2.x comes stable |
22:31.42 | tim27 | it depend how smart is your grand mother |
22:31.54 | tim27 | a@h |
22:31.56 | tim27 | will fit |
22:31.58 | wundaboy | which version of * should i get? |
22:31.59 | tim27 | perfectly |
22:31.59 | tim27 | this |
22:32.06 | wundaboy | 1.2? |
22:32.06 | wundaboy | 1.0? |
22:32.27 | cripito | if is for learn.. i suggest 1.2 |
22:32.34 | wundaboy | yes learning |
22:32.44 | tim27 | TIM is wondering if VOCTEL support caller id |
22:32.49 | wundaboy | which files do i need? |
22:32.56 | tim27 | for incoming call |
22:33.35 | tim27 | or anyway know a VOIP provider with CAN/US 800 DID that support caller id |
22:33.48 | tim27 | anyway=anyone |
22:33.52 | cripito | is for learn? try to cvs www.asterisk.org/download i think that the emerge of geentoo brings the 1.0.x |
22:33.53 | tim27 | :) |
22:34.11 | wundaboy | so i have a random question |
22:34.19 | wundaboy | is 2.9c/minute too much? |
22:34.28 | cripito | yes |
22:34.44 | sgorilla | 2 c a minute min |
22:34.56 | tim27 | anyone can help ??? |
22:35.09 | sgorilla | nufone |
22:35.13 | wundaboy | i need a service that will give me a 503 areacode, and never go down |
22:35.21 | wundaboy | any suggestions? |
22:35.23 | sgorilla | us toll free did |
22:35.24 | file[laptop] | never go down? |
22:35.27 | *** join/#asterisk gambolputty (n=gambolpu@72.240.242.4) |
22:35.30 | wundaboy | rarely |
22:35.33 | tim27 | sgorilla: do they have can toll free did |
22:35.34 | cripito | there is some ppl that offer even less than 2 |
22:35.35 | file[laptop] | that's better |
22:35.38 | wundaboy | and definetly never go down 8-5 |
22:35.38 | sgorilla | 99.99999999999% uptime |
22:35.50 | sgorilla | tim27: yes |
22:35.57 | cripito | sgorilla working in nufune? |
22:36.00 | file[laptop] | gotta remember with things though that this all travels over the public internet. |
22:36.01 | sgorilla | no |
22:36.10 | tim27 | greg ??? working on it ??? |
22:36.24 | sgorilla | working in, on ?? |
22:36.24 | tim27 | what are their rate |
22:36.27 | tim27 | nuphone ??? |
22:36.32 | tim27 | on |
22:36.43 | sgorilla | dunno i think around 2 cents a minute last time i checked |
22:36.54 | tim27 | but they charge for did ??? |
22:37.07 | *** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net) |
22:37.07 | tim27 | someone from nufone on this channel ??? |
22:37.16 | file[laptop] | stop doing that tim27... one question mark is good enough |
22:38.48 | wunderkin | heh !!! |
22:39.07 | file[laptop] | omg !!! ??? wtf |
22:39.25 | jets | lets get drunk at commit to CVS !!! |
22:39.40 | wunderkin | yey !!!oneoneoneone |
22:40.45 | ender | file[laptop]: can * play sounf files with more than 8000mhz ? |
22:40.51 | ender | er 8000hz? |
22:41.05 | file[laptop] | not really, no... everything in asterisk is 8000 |
22:41.41 | ender | ok, thanks. |
22:42.08 | *** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net) |
22:42.57 | cripito | file mono 8000mhz right? |
22:43.28 | file[laptop] | yes |
22:43.51 | cripito | brb |
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) |
22:45.19 | ender | thats what I'm converting to. |
22:45.50 | ender | I 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) |
22:48.38 | lesouvage | ender: maybe this link can be of help http://www.telenovela-world.com/~spade/linux/howto/MP3-CD-Burning/audio.html |
22:48.55 | cripito | telenovela? |
22:48.56 | cripito | :D |
22:49.28 | ender | thats some pretty basic info I already knew. |
22:49.40 | ender | my files are being converted, I was just trying to make sure I had the best possible quality result. |
22:49.43 | jpm_SD | can someone give me a quick list of possible reasons for getting a 403 Forbidden on a SIP registery? |
22:49.48 | vader-wrk | hmmm |
22:49.57 | vader-wrk | i must be doing something wrong i can't get a sip client registerd |
22:50.17 | vader-wrk | i setup the client in the sip.conf and i setup a context in the extensions.conf |
22:50.25 | vader-wrk | i can dial out with the softphone |
22:50.31 | vader-wrk | but i can't recieve anything |
22:51.37 | *** join/#asterisk kippi (n=kippi@cpc4-hatf3-6-0-cust33.lutn.cable.ntl.com) |
22:51.38 | kippi | <PROTECTED> |
22:51.46 | cripito | i 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) |
22:53.47 | cripito | varder the phone is in the same net that the asterisk? |
22:53.52 | tehdely | hello friends |
22:53.55 | vader-wrk | ya |
22:54.02 | lesouvage | jmp_SD: I think the password doesn't match. (secret=blablabla) |
22:54.06 | tehdely | has anyone ever wanted to take their fancy zaptel hardware |
22:54.10 | tehdely | and chuck it under a lawnmower |
22:54.11 | cripito | check the extension in extension.conf or ael |
22:54.16 | vader-wrk | ael? |
22:54.23 | tehdely | and shred the shitty no-good click-click-pop-click-pop FXO modules in a melee of destruction and anger |
22:54.25 | jpm_SD | lesouvage, Hrmm... ok. |
22:54.25 | cripito | 1.2.0 |
22:54.29 | tehdely | because i sure do ! |
22:54.38 | cripito | extensions.conf will be enough |
22:54.49 | vader-wrk | criptio this is what i followed |
22:54.50 | vader-wrk | http://www.astmasters.net/stuff/X-Lite-and-Asterisk.pdf |
22:55.15 | tehdely | seriously, i have had it up to here :( |
22:55.24 | tehdely | if anyone here has successfully solved audio quality problems on a tdm400p |
22:55.29 | tehdely | that weren't irq-related |
22:55.31 | tehdely | please let me know |
22:55.42 | cripito | what kind of audio issues tehdely? |
22:55.53 | tehdely | cripito: click click pop click click crackle click pop click |
22:56.05 | tehdely | clicking/crackling is so bad that half of the dtmf tones are messed up |
22:56.06 | tehdely | constant misdialing |
22:56.14 | cripito | timing issues? |
22:56.15 | tehdely | i have two tdm400p, 4 fxo each, in a dell poweredge 2850 |
22:56.19 | tehdely | they are on their own irqs entirely |
22:56.21 | tehdely | acpi is off |
22:56.23 | cripito | ah fxo |
22:56.28 | tehdely | zttest return sconsistent 99.98% - 100% |
22:56.36 | tehdely | but they sound absolutely terrible. unusably terrible |
22:56.48 | tehdely | i already RMAed a pair from digium because they thought i had bad cards afte rhearing the crackling |
22:56.51 | tehdely | but it is just as bad with the new ones |
22:57.17 | cripito | timing issues? |
22:57.25 | tehdely | why do you reckon it's a timing issue? |
22:57.34 | tehdely | it doesn't sound like timing issues i've heard on other transports (sip, etc.) |
22:57.40 | tehdely | it just sounds like shitty D-A converters |
22:57.42 | tehdely | or something else i'm overlooking |
22:57.47 | cripito | could be too |
22:58.15 | tehdely | i'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 |
22:58.23 | tehdely | and voip-info said it didn't have the issues with tdm cards previous dell models had |
22:58.35 | tehdely | and every FAQ tells me to keep tweaking things until zttest returns nominal figures... well it does :/ |
22:59.07 | cripito | all the teaks that i known are related to tdm cards |
22:59.11 | cripito | not fxo :D |
22:59.20 | tehdely | it is a tdm card |
22:59.22 | tehdely | tdm400p |
22:59.26 | cripito | ahhh |
22:59.27 | cripito | ok |
22:59.35 | cripito | 4 fxo |
22:59.37 | tehdely | yes |
23:00.11 | cripito | i still don't like dells for asterisk.. bad precedence.... |
23:00.15 | tehdely | so 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, |
23:00.27 | cripito | how many cards? |
23:00.27 | tehdely | two |
23:00.33 | cripito | try first with 1 |
23:00.36 | cripito | let see what happen? |
23:00.45 | tehdely | hmm maybe i will |
23:00.51 | tehdely | do you reckon there's some issue due to having two of them? |
23:01.08 | cripito | there is anything about this around.... at least not that i known. |
23:01.15 | cripito | i have 3 in a ibm server working ok |
23:01.32 | cripito | even sharing interrupts |
23:01.36 | cripito | and magically it works |
23:02.06 | *** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543623pcs.mainf01.in.comcast.net) |
23:02.10 | tehdely | :( |
23:02.15 | cripito | try one first see if work |
23:02.16 | tehdely | what os are you running |
23:02.33 | cripito | cvs head. |
23:02.40 | tehdely | no i mean os |
23:02.45 | tehdely | also i am running -stable |
23:02.55 | tehdely | i tried -head and it was worse |
23:02.55 | tehdely | heh |
23:03.20 | Vco | any 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.. |
23:03.34 | Vco | and when i hangup the sip phone, it immediatley rings again |
23:03.47 | cripito | fc 3 |
23:03.55 | tehdely | cripito: i was thinking of giving fedora a shot |
23:04.01 | tehdely | i despise red hat, and i'm using debian stable right now |
23:04.05 | tehdely | but i know digium supports fc |
23:04.17 | cripito | :) fc is very freindly.... |
23:04.20 | Vco | slackware runs pretty reliably as well |
23:04.23 | cripito | friendly |
23:04.25 | tehdely | friendly to everyone but a system administrator, cripito |
23:04.31 | cripito | :D |
23:04.35 | tehdely | there is nothing that irks me more than buggering around with sysv init and byzantine package systems |
23:04.37 | tehdely | oh wait, that's debian too |
23:04.38 | tehdely | life is hell :( |
23:04.55 | cripito | u don't need 2 many things from fc to run asterisk |
23:05.01 | cripito | so u can cap a lot the OS |
23:05.15 | tehdely | is your box smp, btw |
23:05.21 | n3u7 | tehdely I couldn't get the cvs asterisk to install on Sarge |
23:05.25 | n3u7 | or Ubuntu |
23:05.29 | n3u7 | or SuSE |
23:05.34 | cripito | no... |
23:05.35 | tehdely | n3u7: what build errors were you getting |
23:05.36 | tehdely | i can help you |
23:05.47 | cripito | i have smp servers but with wct4xxp |
23:06.24 | tehdely | i wish we would just get a pri and be done with this POTS nonsense |
23:06.25 | tehdely | heh |
23:06.38 | tehdely | if one card works and two don't, i guess i'll just shuffle the other card over into another server |
23:06.42 | tehdely | and have the two talk IAX |
23:06.49 | tehdely | that is such overkill though :( |
23:07.02 | cripito | that could be a solution.... try 1 first |
23:07.37 | cripito | especially with 2 dell 2850 :D |
23:07.44 | *** join/#asterisk aitnemed (i=daedalus@wsip-68-15-202-117.ok.ok.cox.net) |
23:07.45 | lesouvage | tehdely: 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. |
23:08.01 | cripito | i change u one for my zoekris :))))) |
23:08.15 | tehdely | lesouvage: is that a livecd by any chance |
23:08.23 | tehdely | because i would definitely test out a livecd distro to see if it improves things |
23:08.26 | tehdely | i'm not very keen on reinstalling at this point |
23:08.30 | tehdely | unless i am positive debian is the rpoblem |
23:08.53 | cripito | tehdely try 1 card first i don't beleive that the distro is the issue |
23:08.54 | lesouvage | tehdely: 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) |
23:09.30 | tehdely | lesouvage: 'install' is where i'm balking :P |
23:09.48 | tehdely | of 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 |
23:09.55 | tehdely | they made me use debian based on the results of a google search .... |
23:10.09 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
23:10.11 | tehdely | did i mention this is an enterprise pbx and it needs to be running on monday? :P |
23:10.34 | Cresl1n | :-D |
23:10.45 | tehdely | Cresl1n: mine is very non-enterprise thanks to the two tdm400ps in it |
23:11.02 | file | meep |
23:11.02 | tehdely | and the constant crackling and noise |
23:11.31 | cripito | tehdely PM and boss always do that |
23:11.41 | cripito | baby steps make the work.. don't you think? |
23:11.47 | nmsclera | Getting a 423 from one of our sip providers, anyone know a workaround? |
23:11.48 | tehdely | cripito: yes |
23:11.56 | lesouvage | Tehdely: 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. |
23:11.59 | tehdely | but they want me to document every step i take and justify every decision with a spreadsheet |
23:12.09 | tehdely | yet every time i succeed in geting something to work it's because of 20 hours of hacking and trial+error |
23:12.14 | tehdely | i should document my google searches and irclogs |
23:12.43 | tehdely | Thursday, October 20th: Typed 'tdm400p crackling dell' into google |
23:12.46 | tehdely | Found a mailing list post |
23:12.49 | tehdely | Searched frantically for replies |
23:13.09 | tehdely | Thursday, October 20th, 8 PM: Started drinking |
23:13.15 | cripito | welcome to the club... :))) |
23:13.19 | tehdely | haw |
23:13.24 | lesouvage | Tehdely: start downloading, this explaining is not going to solve your problem. |
23:13.31 | cripito | :) |
23:13.35 | cripito | try 1 card first |
23:13.37 | tehdely | will do |
23:13.40 | tehdely | i reckon i'll run back to work |
23:13.43 | tehdely | and pull the damn thing out of the rack |
23:13.47 | cripito | if this work.. u have 1/2 % of the job done |
23:14.10 | wunderkin | 0.5% isn't much of an accomplishment :D |
23:14.16 | cripito | as far you don't hit the pbx with an axe |
23:14.26 | cripito | is really something less than 50% ;) |
23:14.55 | Vco | that is, assuming the scope of wht you are trying to accomplish isn't being changed... |
23:14.59 | Vco | every f'ing day.. |
23:15.00 | Jabroni | cripito 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 ? |
23:15.07 | tehdely | Vco: thankfully at least that part was frozen |
23:15.14 | tehdely | they just haven't given me any of the resources and time i need to accomplish it :) |
23:15.31 | cripito | let 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 |
23:15.37 | cripito | now ... no drop calls |
23:16.32 | n3u7 | the problem with this channel is that the little arterisk user has no hope |
23:16.34 | cripito | must said there is nothing to pickup |
23:16.40 | Vco | ewww...and it's a compaq |
23:16.50 | tehdely | beer goggles, man |
23:16.52 | tehdely | thought it was an xserve |
23:17.01 | nmsclera | how does one increase the registration expiration period for an upstream sip provider? I keep getting a 423 SIP response from our asterisk box |
23:17.36 | nmsclera | rather, ON the asterisk box for that registration |
23:17.39 | lesouvage | tehdely: when a motherboard is underpowered it's behaving strange. I once waste an evening because of this. |
23:17.52 | tehdely | lesouvage: underpowered, eh? |
23:17.54 | cripito | n3u7 sometimes gets tricky to get answer |
23:17.57 | tehdely | what sort of symptoms would that cause |
23:18.12 | cripito | :) in a dell 2850? from dell....? |
23:18.24 | cripito | could be..... |
23:18.31 | cripito | but i don't think soo... |
23:18.36 | cripito | anyway |
23:18.52 | Vco | wtf |
23:18.56 | cripito | if 1 cards work.. that is 1 of the things too look |
23:19.04 | tehdely | cripito: yeah, i really should've ruled that out a while ago |
23:19.07 | cripito | but he don't known if 1 card works |
23:19.09 | lesouvage | tehdely: I mean with a power supply that doesn't supply enough power but enough to keep it running. |
23:19.09 | Vco | getting everyone/user is busy stuff on some incomgin DIDs |
23:19.18 | tehdely | lesouvage: i doubt that's the case here |
23:19.32 | tehdely | this server is designed to hold like 8 hard drives, 4 cpus, etc |
23:19.37 | tehdely | i have 2 hds, 2 cpus, and 3 pci cards |
23:19.40 | tehdely | i reckon it's doing fine for juice |
23:19.43 | cripito | exactly |
23:20.11 | cripito | anyway... as i said before |
23:20.17 | cripito | i will try 1 card before |
23:20.24 | cripito | anything else |
23:20.26 | tehdely | yeah, that's the #1 thing i will check out |
23:20.30 | lesouvage | thedely: how is your xorcom rapid download going? |
23:20.37 | cripito | :))) |
23:20.39 | tehdely | lesouvage: it's not, because i'm not at work where i would need to download it |
23:20.42 | tehdely | but i will try it |
23:20.53 | tehdely | i just wish it was a livecd, instead of forcing me to erase my entire debian install |
23:20.54 | tehdely | that is sort of a bummer |
23:20.55 | tehdely | :/ |
23:21.01 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
23:21.39 | cripito | i am very happy with my fc 3 |
23:21.57 | tehdely | i have a sort of deep, primal hatred of all things red hat |
23:22.03 | cripito | :)))) |
23:22.04 | tehdely | but at this point i think i hate all linuxes equally |
23:22.08 | tehdely | so i might give it a shot |
23:22.08 | tehdely | heh |
23:22.30 | alephcom | tehdely: amen |
23:22.36 | lesouvage | tehdely: 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. |
23:22.52 | cripito | i agree |
23:23.12 | Vco | hmm.... |
23:23.15 | *** join/#asterisk escribzz (n=escribzz@71.36.229.227) |
23:23.16 | cripito | basically u just need the 1th cd and a good internet |
23:23.18 | cripito | ;) |
23:23.18 | tehdely | you know |
23:23.24 | cripito | and yum the rest |
23:23.25 | tehdely | if it turns out |
23:23.29 | tehdely | another distro solves the problem |
23:23.41 | tehdely | i can smear it like a steaming turd in mr project manager's face |
23:23.50 | tehdely | "YOU HAVE TO USE DEBIAN AND ONLY DEBIAN. I SEARCHED ON DISTROWATCH AND THEY SAY IT'S GOOD!" |
23:24.07 | cripito | as i said....i don't think that the distro is the issue here |
23:24.09 | cripito | but could be |
23:24.14 | tehdely | i'm praying it is ;) |
23:24.16 | tehdely | hehehehe |
23:24.27 | Nugget | Linux is poo. |
23:24.31 | tehdely | Nugget: Fact. |
23:24.41 | escribzz | Anyone have a SER config that is deisgned to do rtp proxy and pass the sip registration and data off to an asterisk box? |
23:24.49 | tehdely | well, i take that back. it's ok for the desktop |
23:24.50 | escribzz | for nat |
23:24.53 | n3u7 | i thought it would be a fun project to get this x100p to work |
23:24.53 | tehdely | but i hate it on servers :P |
23:25.06 | n3u7 | now it's like 30 hrs plus and I'm obsessed |
23:25.46 | rayvd | escribzz: i have one that forwards all things SIP to the asterisk box and does rtp proxy |
23:26.12 | cripito | :)) |
23:27.07 | escribzz | Rayvd: 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) |
23:27.52 | cripito | <--- is back to his parked calls |
23:27.54 | Katty | i have news! |
23:27.56 | Katty | !! |
23:28.03 | cripito | :D |
23:28.05 | Katty | i've hit level 3 scramble in the 4d rubik's cube. |
23:28.05 | n3u7 | oh noes:Sarge installed without sound support |
23:29.39 | *** join/#asterisk marc324 (n=marc3234@206-248-135-84.dsl.teksavvy.com) |
23:30.51 | lesouvage | tehdely: 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) |
23:33.54 | wunderkin | i 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 |
23:42.41 | wunderkin | i 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) |
23:44.38 | Ariel_ | why |
23:44.45 | wunderkin | me? |
23:45.00 | *** join/#asterisk autobus (n=linux@80.172.14.4) |
23:45.20 | hardwire | mutilator: happy b-day |
23:45.22 | wunderkin | why what |
23:50.17 | autobus | any person from spain? |
23:50.44 | alephcom | I'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) |
23:52.08 | cripito | autobus did u need someone from spain or that speak spanish? |
23:52.44 | tehdely | cripito: if it's the latter, he won't be able to tell you |
23:53.01 | cripito | i speak boths :P |
23:53.10 | wunderkin | alephcom, 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) |
23:54.45 | wunderkin | ariel must like to just ask why |
23:55.20 | alephcom | wunderkin: Hey, thanks a LOT! I will try that. |
23:56.26 | wunderkin | hmm :/ |