| 00:00.09 | Mercestes | if it wasn't for <tab> in asterisk I wouldn't be able to do anything useful. |
| 00:00.09 | Eagleray | Mercestes: it's usually 'module command argument1 argument2 argumentN' |
| 00:00.15 | Mercestes | ....same goes for Cisco IOS really |
| 00:00.29 | flush | hrm |
| 00:00.32 | flush | how do i reload dialplan ? |
| 00:00.39 | Mercestes | flush, Dialplan reload |
| 00:00.49 | Mercestes | without the caps |
| 00:00.59 | flush | doesnt work man.. |
| 00:01.07 | flush | no command named dialplan |
| 00:01.29 | Mercestes | are you in the asterisk cli? |
| 00:01.37 | fujin | ael reload |
| 00:01.38 | flush | i ran command # asterisk -r |
| 00:01.42 | fujin | dunno what the non-ael version is |
| 00:02.22 | Eagleray | fujin: neither do I.... closest I can find is this: http://www.google.com/search?q=define:ael |
| 00:02.32 | fujin | wha? |
| 00:02.43 | fujin | Eagleray: what the hell are you talking about? |
| 00:03.10 | Mercestes | http://www.voip-info.org/wiki-Asterisk+settings+Broadvoice |
| 00:03.12 | Eagleray | fujin: pointing out that I had no idea what the hell ael was, and that google didn't know either :p |
| 00:03.14 | Mercestes | oops |
| 00:03.17 | flush | only thing that shows about dialplan is command "dialplan show" |
| 00:03.19 | flush | what i do with this ? |
| 00:03.19 | Mercestes | info.org/wiki/view/Asterisk+AEL |
| 00:03.30 | Eagleray | Mercestes: cheers |
| 00:03.31 | Mercestes | flush, you type dialplan show |
| 00:03.51 | flush | and how i understand why asterisk wont make the phone ring when a call comes in |
| 00:03.58 | Mercestes | I don't know what AEL has to do with your asterisk version. |
| 00:03.59 | fujin | Eagleray: Asterisk Extension Language |
| 00:04.03 | flush | and why i get the fast busy dial tone when i hit a key |
| 00:04.17 | Eagleray | Mercestes: info.org doesn't know |
| 00:04.18 | flush | zttool shows "alarms: OK: |
| 00:04.18 | fujin | it's like C-ish for extensions.conf |
| 00:04.22 | Eagleray | fujin: thanks :D |
| 00:04.24 | Mercestes | Eagleray, Yea, I can't copy paste tonight. |
| 00:04.59 | Mercestes | flush, what is the output of your CLI when you try to make a call with core set verbose 37? |
| 00:05.28 | Eagleray | fujin: that explains why I thought it looked like PHP..... that also has C-like syntax |
| 00:05.38 | fujin | ael is awesome |
| 00:05.41 | Mercestes | funny....I thought it looked like java. |
| 00:05.48 | Eagleray | Mercestes: no dots |
| 00:05.50 | fujin | it looks like a number of languages |
| 00:05.56 | fujin | well, no OO = no dots |
| 00:06.04 | TJNII | Can you do a dialplan hint where the status is the status of a voicemail box? |
| 00:06.06 | fujin | therea ren't any complex data structures in AEL either. |
| 00:06.14 | TJNII | Basically a MWI? |
| 00:06.23 | Eagleray | fujin: no OO? Ouch.... |
| 00:06.48 | Eagleray | fujin: external scripts FTW :D |
| 00:07.02 | fujin | nah, they're shit too |
| 00:07.03 | Mercestes | TJNII: Yea, the MWI is pretty much "it." Is there something else you are trying to do? |
| 00:07.13 | fujin | no need for OO |
| 00:07.19 | fujin | not for this anyway |
| 00:07.40 | fujin | you don't need to say "new class to handle an incoming call", cause a call is just lexicographical, it goes from priority 1 to n |
| 00:07.44 | Mercestes | TJNII: You could always setup multiple mailboxes I suppose to pick up multiple MWIs. I don't think hint extensions has a status for VM mailboxes. |
| 00:07.53 | TJNII | hmmm |
| 00:08.10 | Mercestes | TJNII: FOP is nice. |
| 00:08.19 | TJNII | Yea, the documentation on dialplan hints seems a bit sketchy. |
| 00:08.31 | Mercestes | There really isn't alot to write... |
| 00:09.13 | Mercestes | If the user gets on the phone...it shows a red light..if they hang up, the red light goes away...atleast...that's the theory. |
| 00:09.53 | Eagleray | fujin: hmm..... I guess I'm not really used to writing simple apps. Most of the programming I do is PHP webapps, using OO |
| 00:10.10 | Eagleray | fujin: am learning perl though.... so far I like what I see |
| 00:10.34 | Eagleray | fujin: maybe I should quit all the offtopic babbling..... I'm probably disturbing the zen-like peace of the channel |
| 00:10.42 | fujin | PHP is a very bad example of OO (imo) |
| 00:10.47 | fujin | hell, PHP is just very bad |
| 00:10.48 | fujin | eof |
| 00:10.56 | Eagleray | fujin: why do you htink I'm learning perl :p |
| 00:11.14 | Eagleray | fujin: that said, PHP5actually does a halfway proper job of OO |
| 00:11.24 | Eagleray | fujin: and PHP6 should finish the job |
| 00:11.53 | Mercestes | I guess I shouldn't mention asp.net... |
| 00:12.07 | Eagleray | Mercestes: indeed not.... evil stuff |
| 00:12.10 | fujin | asp.net (C#) good stuff |
| 00:12.16 | fujin | forget VB, jscript etc |
| 00:12.29 | Eagleray | fujin: apart from needing .net or mono frameworks to run... |
| 00:12.49 | Eagleray | fujin: VB/JS don't count, they're client-side |
| 00:12.50 | Mercestes | and the microsoft licensing.... |
| 00:12.57 | Eagleray | Mercestes: not for mono |
| 00:12.58 | Mercestes | but other than that....:) |
| 00:13.06 | fujin | Eagleray: not at all, you can write asp.net in vbscript/jscript |
| 00:13.06 | *** part/#asterisk SpaceBass (n=SP@pool-72-84-196-212.rcmdva.fios.verizon.net) |
| 00:13.19 | fujin | anyway, c# is very nice ;[ |
| 00:13.22 | Mercestes | Yea, IDE has vb.net, C#.net and jscript. |
| 00:13.29 | Eagleray | fujin: you can? That seems unbelievably pointless..... |
| 00:13.34 | Mercestes | I coded mostly in VB.Net but...I learned a little C# in my time. |
| 00:13.38 | Eagleray | fujin: both languages are truly horrible |
| 00:13.43 | fujin | oh indeed |
| 00:13.45 | fujin | some swear by them |
| 00:14.31 | Mercestes | Eagleray, yea, but...that's the beauty of it...you don't even have to know how to program... |
| 00:14.41 | Mercestes | just hit a letter and it'll start spitting out valid code for you. |
| 00:21.40 | Eagleray | sorry guys, keyboard batteries went flat at the wrong moment |
| 00:21.47 | Eagleray | Mercestes: good point :p |
| 00:25.03 | russellb | rickross: g722 is fixed! |
| 00:25.10 | russellb | rickross: give it a try when you have a chance |
| 00:25.18 | russellb | rickross: took me most of the afternoon ... it was quite a pain :) |
| 00:31.56 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584628.dsl.bell.ca) |
| 00:32.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:33.15 | *** part/#asterisk Eagleray (n=erayd@osiris.erayd.net) |
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| 00:33.52 | *** join/#asterisk d00gster (n=doughant@bas1-toronto44-1177789678.dsl.bell.ca) |
| 00:34.11 | seanbright-home | actros1840: what version of asterisk are you running? |
| 00:34.45 | actros1840 | 1.4.7 |
| 00:34.55 | seanbright-home | hold |
| 00:35.20 | russellb | $ ./changes_since asterisk 1.4.7 |
| 00:35.20 | russellb | Changes since asterisk Version 1.4.7/ - svn revision 74254 |
| 00:35.20 | russellb | 1059 |
| 00:35.33 | russellb | just fyi :) |
| 00:35.55 | d00gster | folks, is it possible to have bluetooth on my * server so that when I arrive home, my cellphone attaches to the bluetooth and when I get a call on my cell, * answers and puts the call through my inbound rules |
| 00:36.58 | SteveTotaro | yes |
| 00:37.08 | SteveTotaro | chan_mobile |
| 00:42.08 | d00gster | thanks SteveTotaro |
| 00:46.48 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a48c0612f841fd92) |
| 00:48.50 | *** join/#asterisk mindCrime (n=chatzill@rrcs-70-63-129-50.midsouth.biz.rr.com) |
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| 01:00.33 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:00.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:01.09 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.105.210) |
| 01:01.31 | *** join/#asterisk BCS-Satori (n=BCS-Sato@204.52.131.22) |
| 01:03.49 | *** join/#asterisk Katty (n=The@adsl-75-8-89-202.dsl.stlsmo.sbcglobal.net) |
| 01:04.26 | Katty | allo. |
| 01:10.59 | Katty | so quiet. |
| 01:11.49 | seanbright-home | *burp* |
| 01:11.58 | Katty | nice one. |
| 01:12.15 | Katty | errr: you around? |
| 01:12.15 | seanbright-home | mmmm, chipotle |
| 01:12.26 | Katty | seanbright-home: i had mexican for dinner. |
| 01:12.37 | seanbright-home | i had a burrito the size of your head |
| 01:12.42 | Katty | nice |
| 01:12.43 | seanbright-home | and it was glorious |
| 01:13.58 | Katty | we're supposed to get some ice and snow tonight. |
| 01:14.08 | seanbright-home | where are you? |
| 01:14.11 | Katty | i'm actually kinda looking forward to it |
| 01:14.13 | Katty | southern missouri |
| 01:14.17 | seanbright-home | ah |
| 01:14.27 | seanbright-home | baltimore here |
| 01:14.39 | seanbright-home | haven't had any significant snowfall all season |
| 01:14.47 | seanbright-home | i blame the terrorists |
| 01:14.59 | *** join/#asterisk mkl1525 (n=some@212.204.47.147) |
| 01:15.03 | Katty | we've had a couple bad ice storms...took 15k homes out of power for about a day or so |
| 01:15.30 | Katty | ended up pup sitting a couple doggies so they didn't get cold (= |
| 01:16.00 | seanbright-home | awww |
| 01:16.20 | mkl1525 | Hi, (running a * 1.2) found that the i option on dial to ignore call forwardings came in 1.4. Is there any backport for this to 1.2? |
| 01:16.22 | Katty | [TK]D-Fender: you around? |
| 01:16.35 | Katty | mkl1525: you should go ahead and get 1.4 |
| 01:16.38 | Katty | mkl1525: well worth it. |
| 01:16.54 | seanbright-home | mkl1525: all the cool kids are using the 1.4 |
| 01:17.02 | Katty | mkl1525: i'm not sure if 1.2 has a work around. |
| 01:17.12 | Katty | mkl1525: lots of new features with 1.4 (= |
| 01:17.24 | Katty | mkl1525: and 1.6 will be out soon too i hope, in non beta form. |
| 01:20.13 | *** join/#asterisk BeeBuu (n=beebuu@59.38.97.193) |
| 01:20.17 | Katty | i can't wait till anthm gets the new chanspy stuff done |
| 01:21.53 | Katty | errr: http://angela.sleekgeek.org/2008/03/06/isymphony/ <- may you not beat your head on the wall. |
| 01:23.12 | seanbright-home | Katty: is he actually working on that? |
| 01:23.46 | Mercestes | Katty!!!!! |
| 01:24.14 | Katty | seanbright-home: i believe so. we were talking about it earlier. |
| 01:24.20 | seanbright-home | cool |
| 01:24.22 | Katty | seanbright-home: i promised to link my blog of what i did today |
| 01:24.57 | Katty | seanbright-home: i've got the software working...and mostly figured it out except for a couple things. it's kinda neat. i think i'd like to get a touch screen for an operator to drag things around. it'd be uber. |
| 01:25.03 | Katty | Mercestes: !!!! *hugskwish* |
| 01:25.14 | seanbright-home | Katty: you're using redirect? |
| 01:25.24 | Katty | seanbright-home: what? |
| 01:25.32 | Mercestes | Katty: *TACKLEPOUNCEmolestHUGS* |
| 01:25.37 | Katty | Mercestes: (= |
| 01:25.40 | Katty | Mercestes: what gives? |
| 01:25.49 | seanbright-home | Katty: you were talking about using redirect to get all the parties in a meetme yesterday |
| 01:25.58 | Katty | seanbright-home: ah, no... |
| 01:26.01 | Katty | seanbright-home: still working on that one |
| 01:26.07 | seanbright-home | ah |
| 01:26.15 | Katty | seanbright-home: it looks like an ExtraChannel api thing will work. |
| 01:26.24 | seanbright-home | so "the software" == "your entire app" |
| 01:26.27 | Mercestes | Finally got my work laptop today, so went with a Gentoo VDMK, but my netty wouldn't work so I went to go bug #gentoo about it...and decided to pop on here while I was there, and give some peeps bad advice. |
| 01:26.30 | Katty | seanbright-home: i've no experience with API, so that's gonna take some work |
| 01:26.32 | Mercestes | and market Polycom swag. |
| 01:26.43 | vap0rtranz | gentoo rules! |
| 01:26.50 | Katty | i'm swagged out. |
| 01:26.57 | Katty | but, just the same, enjoy the laptop (= |
| 01:27.02 | Mercestes | :) |
| 01:27.10 | Mercestes | it's a work laptop...nothin' special. |
| 01:27.10 | Katty | how's the wifey |
| 01:27.13 | Mercestes | IBM thinkpad ... |
| 01:27.16 | Mercestes | Wifey is good. |
| 01:27.39 | Katty | get to see her soon? |
| 01:27.40 | Mercestes | She *almost* gnawed through the ropes the other day, so I had to move to 6 gauge cable. |
| 01:27.53 | Mercestes | You want pics or a date? |
| 01:27.55 | Katty | seanbright-home: i plan on blogging it as soon as i have it figured out. |
| 01:28.04 | Katty | i.... think i'm good |
| 01:28.07 | Katty | heh |
| 01:28.13 | seanbright-home | Katty: what's the addr? |
| 01:28.20 | Mercestes | I can scan her pics and put them on my webpage if ya want. |
| 01:28.51 | Katty | seanbright-home: http://angela.sleekgeek.org/2008/02/29/simple-call-barge-on-asterisk/ |
| 01:28.55 | Mercestes | that remidns me, I gotta get with my host provider and see about fixing my direct links to pickys.... |
| 01:29.02 | Katty | Mercestes: i know what she looks like. i saw christmas photos (= |
| 01:29.16 | Katty | Mercestes: yeah, i need to get the gallery back up |
| 01:29.19 | Katty | Mercestes: just too lazy |
| 01:29.54 | Mercestes | Oh..you meant do *I* get to see her soon...? |
| 01:29.58 | Mercestes | I'm home now. |
| 01:30.04 | Katty | Mercestes: yes, i meant you |
| 01:30.07 | Katty | Mercestes: good (= |
| 01:30.10 | Mercestes | Yea, I was confused. =/ |
| 01:30.16 | seanbright-home | ugh... isymphony |
| 01:30.34 | Katty | seanbright-home: i can't seem to get HUD server installed. |
| 01:30.38 | *** join/#asterisk starlein (i=star@82.210.49.240.rev.worldbone.de) |
| 01:30.39 | Katty | seanbright-home: the setup guides are just to vague |
| 01:30.45 | Qwell | Katty: intentionally |
| 01:30.54 | fujin | that's cause it's shit |
| 01:30.54 | Qwell | they want you to pay for support |
| 01:31.01 | Katty | Qwell: figured as much. but i can't even find someone who blogged it |
| 01:31.07 | Katty | or a sample ANYTHING |
| 01:31.17 | Katty | Qwell: i'm going to figure it out. |
| 01:31.26 | Mercestes | Katty: Write a howto for me. ;) |
| 01:31.27 | Katty | Qwell: and i'm going to blog it to death ^_^ |
| 01:31.31 | seanbright-home | i spent about 20 minutes using isymphony until i was the one telling they were throwing null reference exceptions |
| 01:31.37 | Qwell | you're going to have to install an IRCD or something |
| 01:31.42 | Katty | i did. |
| 01:31.47 | Katty | hybrid. |
| 01:31.50 | seanbright-home | that was the worst typing ever, my apologies. |
| 01:31.59 | Katty | i think i'm stuck on the part about moving some stuff to /etc/asterisk |
| 01:32.08 | Mercestes | man mv |
| 01:32.08 | Katty | i can't find what stuff they're talking about. not even a locate can find it. |
| 01:32.14 | Mercestes | Oh. |
| 01:32.45 | Katty | http://angela.sleekgeek.org/2008/02/28/hudlite-installation/ <- that's pretty much what i did |
| 01:33.09 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 01:34.34 | Katty | maybe it's because i aliened them. |
| 01:34.41 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 01:35.11 | seanbright-home | there's only one command you need to know when it comes to isyphony... `rm -rf` |
| 01:35.16 | Mercestes | There is no good way to get an asterisk installation on a windows box, is there? |
| 01:35.20 | seanbright-home | gosh i'm clever |
| 01:35.29 | Qwell | Mercestes: format c:, insert debian CD |
| 01:35.38 | Katty | Mercestes: you could install it on a slimline machine. |
| 01:35.41 | seanbright-home | Mercestes: it should compile under cygwin. 1.6 at least. |
| 01:35.46 | Katty | Mercestes: then set it ontop of a windows box. |
| 01:36.08 | Mercestes | Katty: LOL. Nice. |
| 01:36.59 | Mercestes | Yea, I tried to get work to use VmWare OS on my laptop and then just install windows in a VM but they wouldn't do it. ;( |
| 01:37.28 | Katty | i need to find someone who knows gentoo to help me translate that HUD gentoo doc |
| 01:37.59 | Katty | i'm not doing anything tonight |
| 01:38.08 | Katty | except sitting here on the couch listening to nat king cole |
| 01:38.37 | Mercestes | Nat King Cole was a merry old soul, and a merry old soul was he.... |
| 01:38.43 | Mercestes | .....wait..that's "Old king cole" isn't it? |
| 01:39.52 | Katty | no idea. |
| 01:40.32 | Mercestes | Inspector gave house a major thumbs down. :( |
| 01:40.54 | Mercestes | link me to gentoo hud doc. |
| 01:42.06 | Katty | http://www.voipphreak.ca/2006/07/03/gentoo-asterisk-and-hudlite-installation-and-setup-howto/ |
| 01:43.51 | seanbright-home | good god |
| 01:44.14 | seanbright-home | thats... trivial |
| 01:47.21 | vap0rtranz | what's to know about gentoo?? |
| 01:48.28 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 01:49.24 | Katty | i'm sure it'd be very easy for some. |
| 01:49.45 | Katty | not for me tho (= |
| 01:50.49 | vap0rtranz | Katty: #gentoo is awesome. and yes, that's an insult to this community |
| 01:51.33 | Mercestes | vap0rtranz, nice to know. |
| 01:53.01 | vap0rtranz | Mercentes: well, the lease helpful ever is LFS. basically one might as well only speak in C over there |
| 01:53.43 | Katty | vap0rtranz: what's an insult? |
| 01:53.49 | Docfxit | Does anyone know of an IRC channel that I could get some help configuring a softphone? |
| 01:53.55 | Mercestes | I think he's trying to say we're not helpful. :( |
| 01:54.06 | vap0rtranz | Katty: hint hint. |
| 01:54.14 | Mercestes | Docfxit, #consultants |
| 01:54.26 | vap0rtranz | Mercestes: no, I'd imagine being just as annoyed my n00b's. |
| 01:54.28 | Docfxit | Mercestes » Not at all. Is it ok to ask about softphones here? |
| 01:54.30 | Mercestes | Which softphone?? |
| 01:54.37 | Qwell | vap0rtranz: try #pidgin sometime |
| 01:54.51 | vap0rtranz | Qwell: hehe |
| 01:54.51 | Qwell | vap0rtranz: mention gentoo too |
| 01:54.51 | Docfxit | Mercestes » Express Talk |
| 01:54.54 | Katty | yeah i'm completely lost. |
| 01:54.59 | Katty | but that's okay. i don't mind ;) |
| 01:55.11 | Mercestes | Or #freepbx... |
| 01:55.14 | Mercestes | or #plan9 |
| 01:55.20 | Mercestes | ...#openbsd |
| 01:55.31 | Qwell | the bsd channels are pretty bad too, from what I hear |
| 01:55.35 | Qwell | at least on efnet |
| 01:55.38 | vap0rtranz | Katty: it's been a long day. had to yank out sip on a box just to keep it up for processing over zap |
| 01:55.42 | Mercestes | they are a little brutal. |
| 01:56.15 | Mercestes | especially if your opinion of their product is, "is secure because no one knows how to use it." |
| 01:56.39 | vap0rtranz | Mercestes: "security by obscurity"? ... no that's not it ./.. |
| 01:56.51 | Mercestes | vap0rtranz, :D Exactly |
| 01:56.56 | Katty | <bkw> NEXT!!! |
| 01:57.16 | Docfxit | Mercestes » It's giving me an error saying "Did not respond to register" |
| 01:57.27 | Mercestes | Docfxit, I have no hands on experience with Express Talk. |
| 01:57.33 | Mercestes | Docfxit, Do you have an asterisk window open??? |
| 01:57.37 | Katty | Docfxit: that's because you didn't do a transaction. |
| 01:58.18 | Docfxit | Scheduling destruction of SIP dialog '1204773530-3932-GARYLT@223.1.1.128' in 32000 ms (Method: REGISTER) |
| 01:58.52 | Docfxit | That comes up in the asterisk debug screen. |
| 01:59.10 | Mercestes | That's not very helpful. |
| 01:59.18 | Mercestes | Turn debug off for a sec and turn on verbose 36 |
| 01:59.22 | Katty | maybe you should take it to the checkout. |
| 01:59.41 | Mercestes | then reboot your phone and see what * says about your registration. Look for "username/authname mismatch" or something silly like that. |
| 02:00.04 | Mercestes | Katty, one of those "self-serve" lines maybe... |
| 02:00.21 | Katty | Mercestes: i think everyone missed my silly register comment. |
| 02:00.29 | Katty | Mercestes: then again, i am tired ;) |
| 02:00.30 | Mercestes | except me, of course. |
| 02:00.36 | Mercestes | Of course, I'm the master of bad jokes. |
| 02:00.42 | Katty | Mercestes: i still don't get what vap0rtranz was talking about. |
| 02:00.53 | Mercestes | Katty, basically we suck. |
| 02:01.06 | Katty | Mercestes: fascinating. |
| 02:01.11 | vap0rtranz | Mercestes: no; i'm just frustrated. |
| 02:01.22 | Katty | i understand. |
| 02:01.28 | Katty | sucky people get confused easily ;) |
| 02:01.49 | vap0rtranz | Katty: or people who didn't sign up for voip |
| 02:01.50 | vap0rtranz | :P |
| 02:02.01 | Mercestes | vap0rtranz, Try h.323 sometime. |
| 02:02.13 | Katty | not all of start in medium to large businesses. |
| 02:02.41 | Mercestes | I didn't even sign up to answer a telephone and I've gone from POTS to Voip, to foip and now...h.323. :( |
| 02:02.47 | Katty | and anyone who has the patience to help me is NOT sucky. (= |
| 02:02.54 | Mercestes | and h.320 on occassion. |
| 02:02.57 | Katty | i wish more people would remember that. |
| 02:03.28 | Mercestes | :) |
| 02:03.33 | Katty | even a dog has more sense than that. |
| 02:03.34 | vap0rtranz | Katty: no peeps are sucky. some peeps are harsh. glob that everywhere |
| 02:03.44 | vap0rtranz | and that's cyclic reasoning of course |
| 02:03.47 | vap0rtranz | :) |
| 02:03.59 | Katty | glob does not parse. |
| 02:04.48 | Docfxit | Now that I increased verbose, I'm getting a line that says SIP/2.0 401 Unauthorized |
| 02:05.03 | Katty | heh, my boss just called me and left me a message to call him at his house. |
| 02:05.04 | Mercestes | Docfxit: During dial? |
| 02:05.16 | Katty | i think i'll pretend my phone is dead! |
| 02:05.17 | Docfxit | During registration. |
| 02:05.32 | Mercestes | Docfxit, Did you add it to sip.conf? |
| 02:05.33 | vap0rtranz | Katty: godz all bosses are the same |
| 02:05.44 | Docfxit | yes. |
| 02:05.47 | Katty | nice. he just IMed me |
| 02:06.00 | Katty | i'm going to go idle. back later (= |
| 02:06.13 | Mercestes | k |
| 02:08.56 | Docfxit | But I may have it wrong. I have the defaultIP set to a local IP 192.168.1.xxx I am currently remote. Is that correct? |
| 02:09.10 | eric2 | if _X. is for more than one digit, what is it for one digit or more? |
| 02:09.11 | Mercestes | Docfxit, try host=dynamic |
| 02:09.22 | Docfxit | Okay |
| 02:09.46 | Mercestes | eric2, It requires two lines, one for _X and one for _X. You don't want to do _. because it will cause crazy things. |
| 02:10.27 | *** join/#asterisk graphx (n=jeff@rrcs-67-79-144-150.se.biz.rr.com) |
| 02:10.29 | eric2 | hmm.... one of my customers wants single digit extensions for their pbx |
| 02:10.48 | eric2 | currently have it setup for extensions >= 2 numbers |
| 02:10.54 | eric2 | err.. digits |
| 02:11.04 | Mercestes | nice. |
| 02:11.14 | eric2 | hassle! |
| 02:11.15 | Mercestes | if you have like....10 phones, I guess. |
| 02:11.22 | eric2 | ya, only 5 phones |
| 02:11.33 | Mercestes | do a exten => [0-9] then for the extensions. |
| 02:11.36 | eric2 | but still... now I have to alter the 'master macro' |
| 02:11.58 | Mercestes | ...what master macro? |
| 02:12.09 | eric2 | I just mean the dial plan |
| 02:12.33 | eric2 | dial plan, macro... its all just coding |
| 02:13.24 | vap0rtranz | there needs to be a new release of Spencer's Handbook. It was so clear about 1.2. |
| 02:13.40 | vap0rtranz | In two weeks, the Digium docs will be 2 years old! |
| 02:14.21 | Qwell | ~book |
| 02:14.22 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 02:14.37 | Qwell | it has a foreword by Mark ;p |
| 02:14.42 | vap0rtranz | Qwell: obviously you missed my comment from earlier today |
| 02:15.08 | Docfxit | Mercestes » Tx for your help. |
| 02:15.18 | Mercestes | Docfxit: Np. |
| 02:15.42 | vap0rtranz | Qwell: the book doesn't discuss the changes from 1.2 -> 1.4 (like most revisions have done in the OReilly volume) |
| 02:15.52 | Qwell | UPGRADE.txt |
| 02:16.04 | graphx | eric2: what are the phones? |
| 02:16.28 | eric2 | snom 300 & 320's |
| 02:16.47 | graphx | you using voip to pots? |
| 02:17.18 | graphx | eric2: you using voip to pots? |
| 02:17.22 | eric2 | all voip, underlying carrier does termination |
| 02:17.32 | graphx | ic |
| 02:17.34 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 02:17.47 | *** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net) |
| 02:17.54 | eric2 | I just have to worry about the digital/software part of it |
| 02:18.22 | Mercestes | Oh, is that all? |
| 02:18.41 | vap0rtranz | Qwell: where's that? |
| 02:18.47 | Qwell | in the source dir |
| 02:20.15 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:22.32 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:22.47 | vap0rtranz | Qwell: thank you! an excellent read ... *reading* |
| 02:26.48 | *** join/#asterisk keith4 (n=keith@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 02:27.14 | keith4 | before I try to implment internal *69, is it already done somewhere, and I just can't find it using google? |
| 02:34.20 | *** join/#asterisk PepOSX (n=angeldav@190.72.144.198) |
| 02:35.50 | mosty | what's *69? |
| 02:36.18 | Mercestes | call return. |
| 02:36.22 | Mercestes | seriously. |
| 02:36.49 | *** join/#asterisk MmixX (i=mmixx@202.124.138.69) |
| 02:36.51 | mosty | it's a different code here |
| 02:37.03 | Mercestes | Ah. Someone here had a sense of humor. |
| 02:39.47 | vap0rtranz | hehe |
| 02:40.06 | vap0rtranz | return the favor? |
| 02:40.08 | vap0rtranz | lol |
| 02:40.14 | Mercestes | exactly. |
| 02:45.31 | Katty | hai |
| 02:46.22 | Mercestes | wb |
| 02:48.07 | vap0rtranz | Dilbert's boss go away? |
| 02:48.13 | Katty | no, i called him. |
| 02:48.41 | Katty | it wasn't exactly work related, really. |
| 02:48.41 | Katty | it was, but not in a Fix It sort of way |
| 02:49.31 | vap0rtranz | *sigh of relief* |
| 02:52.30 | *** join/#asterisk swiftkick (n=Miranda@mail.beanproducts.com) |
| 02:53.59 | Katty | i haz headache )= |
| 02:54.19 | swiftkick | hello |
| 02:54.32 | swiftkick | i have a question for the freenode #asterisk community at large... |
| 02:54.38 | *** join/#asterisk BeeBuu (n=beebuu@59.38.97.227) |
| 02:54.40 | vap0rtranz | *drum roll* |
| 02:54.46 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 02:55.03 | swiftkick | whats "the best" web based GUI for asterisk -- besides *now ? |
| 02:55.17 | JT | none of them? |
| 02:55.17 | swiftkick | for a single homed, multi-tenant installation |
| 02:55.41 | vap0rtranz | swiftkick: once you go gui, you'll never be able to troubleshoot |
| 02:55.49 | swiftkick | yeah im noticing this |
| 02:55.51 | apocn | Hello, Im making an AGI script (perl) and when I go to a for loop (to make 3 retries) the first iteration is made and everything starts working in the second iteration, why is this? |
| 02:55.56 | swiftkick | i have a multi tenant system running on Asterisk NOW |
| 02:56.00 | swiftkick | and its a bear to administer |
| 02:56.33 | swiftkick | not to mention asterisk now development moves very slow and is completely out of sync from the main asterisk development repository |
| 02:56.38 | swiftkick | s/is/seems to be |
| 02:57.11 | vap0rtranz | swiftkick: really? the latest *now is 1.4.17; the main is 1.4.18 |
| 02:57.25 | swiftkick | for *now ? oh that explains it |
| 02:57.42 | swiftkick | i hate the fact that *now development is completely isolated from * |
| 02:58.00 | BeeBuu | which sip/iax softphone can be installed to my cell phone? |
| 02:58.22 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 02:58.43 | swiftkick | vap0rtranz : yeah i was trying to use *now with the 1.6 betas |
| 02:58.49 | swiftkick | and watching it fail in numerous different ways |
| 02:59.51 | znoG | anyone know how to get Asterisk to compile app_realtime.c? |
| 03:00.27 | vap0rtranz | swiftkick: beta *shrug* i can't imagine *now being in a different Digium building. isn't user.conf the major difference? * wise i mean |
| 03:01.35 | *** join/#asterisk steliosk (n=Stelios@79.131.80.108) |
| 03:01.49 | swiftkick | probably. the problem with *now is, it is a whole linux distro/ISO. its tough to separate it out when youve got a hand-built asterisk installation |
| 03:02.44 | vap0rtranz | swiftkick: not for biz already staying with RH/CentOS and bosses who'll go along with the simplistic gui |
| 03:03.08 | swiftkick | well, whats this "conary" package manager? |
| 03:03.14 | swiftkick | thats not RH/centos is it? |
| 03:04.09 | swiftkick | note: apologies beforehand for bringing a *now discussion to the * channel... *now = ghost town channel, ghostly winds whistle as tumbleweeds roll bt |
| 03:04.13 | swiftkick | s/bt/by |
| 03:05.11 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 03:05.22 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 03:05.28 | vap0rtranz | swiftkick: no apologies. i take package managers like this: there's and ENDless supply. i haven't noticed any major things missing from the distribution servers. |
| 03:06.01 | swiftkick | vap0rtranz: since you seem in the know, do you know - is there any SIP based FAX solution for asterisk? |
| 03:06.19 | vap0rtranz | swiftkick: i know nothing ;) |
| 03:06.58 | BeeBuu | anyone know which sip/iax softphone can be installed to my cell phone? |
| 03:07.20 | plik | BeeBuu: probably depends which cell phone you have |
| 03:07.58 | vap0rtranz | swiftkick: i mean, if you really need a gui, most ppl here will tell you to go elsewhere. directly writing a dialplan is just simpler IMHO. i yes, we're talking more than one * box |
| 03:08.47 | swiftkick | vap0rtranz: yeah im almost to the point where i write a dialplan from scratch. i could certainly take the one that *now made, and modify it |
| 03:08.54 | swiftkick | however |
| 03:09.30 | BeeBuu | nokia n72 |
| 03:09.35 | swiftkick | i was hoping someone would have a commercial package to recommend, e.g. for multi-tenant situation, where we are letting at least one other business use our asterisk server |
| 03:09.43 | BeeBuu | plik: any idea? |
| 03:10.53 | plik | BeeBuu: none I'm afraid, but google or howardforums should be able to help you find a SIP client... not so sure about IAX though |
| 03:11.38 | znoG | how do I call a function in the dialplan? (not an application, but a function) |
| 03:12.08 | swiftkick | next question: I keep hearing about *telephone provisioning* being built into *now in the near future |
| 03:12.45 | ManxPower | swiftkick: As each phone is provisioned using whatever method the maker decides on, I think you should tell that person to lay off the booze |
| 03:13.05 | ManxPower | unless they know the specifics |
| 03:13.19 | swiftkick | ManxPower: word. i knew that. i heard something about cisco and polycom |
| 03:13.33 | ManxPower | but we all know that this channel is #AsteriskNotNOW, not AsteriskNOW. |
| 03:14.25 | swiftkick | hahahaha |
| 03:14.25 | keith4 | znoG: you mean a macro? |
| 03:14.30 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 03:14.39 | swiftkick | yeah no doubt |
| 03:14.51 | apocn | Hello, when I create a loop, the first iteration never runs my stuff, why is that? |
| 03:14.52 | ManxPower | znoG: look at extensions.conf.sample or the Wiki |
| 03:15.39 | apocn | for example I have: for($x=0; $x < 3; $x++){ $AGI->exec("NoOp", "Iteration No.: $x"); $AGI->get_data...;} |
| 03:16.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:16.30 | plik | swiftkick: provisioning Cisco & Polycoms probably means they're just going to add a few k's worth of tftp server to the nasty rPath linux distro they use |
| 03:16.35 | ManxPower | Or for those not using AEL: http://pastebin.ca/931365 |
| 03:16.51 | apocn | when I execute it I get: (NoOp) Options: (Iteration No.: 0) then (NoOp) Options: (Iteration No.: 1) then it executes the $AGI->get_data... and plays the file (the second iteration). |
| 03:16.53 | apocn | any help? |
| 03:17.16 | znoG | ManxPower: i had a look at the extensions.conf.sample and I see lots of lines with application calls (NoOp, Background, etc) but how do I call a *function* like REALTIME or ARRAY ? |
| 03:17.21 | ManxPower | apocn: very few of us use AEL. |
| 03:17.32 | *** join/#asterisk implicit (n=implicit@166.129.206.243) |
| 03:17.37 | ManxPower | znoG: Guess what I did when I need to know exactly the same thing. |
| 03:17.55 | ManxPower | I checked extensions.conf.sample AND the wiki |
| 03:17.56 | swiftkick | plik: yeah. or perhaps they are referring to the XML directory services etc too |
| 03:18.05 | swiftkick | plik: perhaps that falls under "provisioning" ? |
| 03:18.08 | znoG | ManxPower: :) i'm checking out the wiki now |
| 03:18.19 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:18.20 | swiftkick | SO - does anyone know of any way to get FAX functionality over a SIP circuit??? ??? ??? |
| 03:18.40 | keith4 | never going to happen |
| 03:18.46 | apocn | ManxPower, Im not using AEL, Im using AGI (with Perl) |
| 03:19.10 | keith4 | swiftkick: it's discussed in the wiki, i believe |
| 03:19.23 | swiftkick | keith: do you have a url to get me started? |
| 03:19.31 | swiftkick | im lame, i never consult the wiki |
| 03:19.37 | swiftkick | i find myself in the bugtracker far more often. :) |
| 03:19.53 | keith4 | swiftkick: http://www.voip-info.org/wiki-Asterisk+fax#SendingafaxtoaSIPdevice |
| 03:19.54 | mosty | swiftkick, it's a waste of time- unreliable |
| 03:20.21 | swiftkick | keith/mosty: we're finding its unreliable even using a SIPURA box pointing directly at the provider |
| 03:20.29 | swiftkick | -and yet- |
| 03:20.50 | keith4 | i pretty much gave up on faxing through asterisk |
| 03:21.11 | swiftkick | our T1 provider uses a Ciso 29something series box, it can provide 8 lines on its own. we dont manage it. but it sure provides flawless fax capability |
| 03:21.11 | mosty | swiftkick, fax over voice over ip is stupid anyway |
| 03:21.16 | keith4 | i spent a lot of time trying to get it to do fax detection, and direct the call to an actual fax machine on a zap channel... no luck |
| 03:21.34 | swiftkick | mosty: whats the alternative? |
| 03:21.54 | mchou | swiftkick: email |
| 03:22.00 | mchou | swiftkick: no joke :) |
| 03:22.02 | swiftkick | hahah |
| 03:22.08 | mosty | swiftkick, email-fax gateways, or a client-server setup like hylafax, or t.38 one day in the future maybe |
| 03:22.08 | swiftkick | we're a business, we need an actual fax |
| 03:22.23 | swiftkick | what puzzles me is |
| 03:23.16 | swiftkick | our telco provider gives us a Cisco IAD2431-8FXS, it has the PRI for our T1. gives us 8 potential phone lines and one ethernet connection. |
| 03:23.24 | vap0rtranz | i get so mad at *. i just registered 3 ATA's to our gateway today but this damn box won't budge! ARGH |
| 03:23.29 | swiftkick | fax works perfectly over those lines |
| 03:23.36 | swiftkick | therefore i am curious: |
| 03:23.44 | swiftkick | how is the provider doing it? |
| 03:23.56 | keith4 | sip != T1 |
| 03:23.59 | swiftkick | i understand |
| 03:24.02 | swiftkick | but i am wondering |
| 03:24.11 | swiftkick | what kind of telephony endpoint WOULD work under * |
| 03:24.22 | mchou | especially since it doesn't involve unsitabe speech codecs |
| 03:24.32 | mchou | unsuitable* |
| 03:24.33 | mosty | swiftkick, there is no good solution for fax with asterisk |
| 03:24.38 | vap0rtranz | swiftkick: ah, the fax question again. so how there needs to be a solution?! |
| 03:24.40 | swiftkick | is it just that there is no fax support under asterisk, or is there no fax support over SIP on asterisk |
| 03:24.49 | mchou | ~fax |
| 03:24.50 | jbot | Well, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically. |
| 03:24.57 | mosty | swiftkick, there is no reliable fax support in asterisk if it ever touches the network |
| 03:25.06 | swiftkick | ouch |
| 03:25.42 | swiftkick | what about FreePBX or trixbox? same deal? |
| 03:25.52 | mchou | swiftkick: same deal |
| 03:26.00 | mosty | swiftkick, that's still asterisk |
| 03:26.18 | znoG | ManxPower: am I right in saying that if I want to call the REALTIME function, i should be able to simply do: NoOp(REALTIME(arg1,arg2,arg3)); ? |
| 03:26.32 | swiftkick | mosty: nobody has a working fax-over-IP oslution for any telephony/PBX platform ? |
| 03:26.37 | ManxPower | znoG: I am NOT going to go to the Wiki and look it up for you! |
| 03:26.52 | znoG | ManxPower: no, i didn't ask you to do that, i was checking if i understood properly |
| 03:26.57 | ManxPower | swiftkick: I do, but it's not what you think. |
| 03:27.01 | mosty | swiftkick, google t.38, asterisk does not have a usable setup yet |
| 03:27.03 | znoG | ManxPower: if you don't know, then just say so and I will keep looking. |
| 03:27.03 | ManxPower | znoG: I would have to go look at my dialplans |
| 03:27.08 | swiftkick | i am just shocked that my proviers IAD2431/8FXS can provide 8 such flawless quality voice lines that support fax perfectly |
| 03:27.24 | mosty | swiftkick, is that a voip device? |
| 03:27.24 | swiftkick | maxpower: please do share this ? :) |
| 03:27.37 | swiftkick | mosty: yes it takes a T1 PRI and breaks it out into an ethernet jack and 8 phone lines |
| 03:28.13 | ManxPower | LEN is a function: |
| 03:28.14 | ManxPower | exten => _XXXX,1,GotoIf($[${LEN(${VOICE_MAILBOX})} != 0]?check-vm-status) |
| 03:28.29 | mosty | swiftkick, well i assume the fax machine connects to one of the phone lines, so it never touches voip |
| 03:28.34 | znoG | yes .. ok, thanks.. |
| 03:28.36 | ManxPower | swiftkick: your provider never uses voip |
| 03:28.56 | znoG | i think i was missing the ${} |
| 03:29.09 | mchou | swiftkick: dont confuse HW with SIP :) |
| 03:29.33 | mchou | swiftkick: or IAX for that matter |
| 03:29.34 | swiftkick | mchou: i realize that HW probably isnt using SIP. what I am wondering is what it is actually using |
| 03:29.45 | plik | swiftkick: PRI != VoIP... 8 lines = PRI ISDN the remainder (16 channels worth) is probably IP |
| 03:29.50 | swiftkick | and what kind of circuit etc. to the remote telco termination |
| 03:30.01 | swiftkick | SO |
| 03:30.11 | ManxPower | swiftkick: We use external hardware echo cancelers (the ones built into asterisk don't work well enough for us unless we screw with the audio gains and that messes up faxes), spanDSP and rxfax, then it is converted into a TIFF, turned into a PDF, wrapped up in an e-mail message and sent off. That fax got to the mailbox via IP. |
| 03:30.38 | vap0rtranz | # mv /etc/{asterisk,asterisk-sucks} && echo "start the fuck all over!" |
| 03:30.40 | mchou | ManxPower: that's CZRAZY!!! |
| 03:30.43 | ManxPower | swiftkick: That would be a T-1 or E-1 to from your location to their NOC |
| 03:30.52 | swiftkick | so -- if I used asterisk to terminate the T-1, does it then become possible??? |
| 03:31.07 | mchou | swiftkick: not thru asterisk :) |
| 03:31.11 | ManxPower | swiftkick: As long as you don't sent the call over the network, in theory |
| 03:31.21 | swiftkick | hahahah |
| 03:31.41 | ManxPower | swiftkick: a T-1/E-1 is not a "network" in this conversation |
| 03:31.46 | mchou | swiftkick: it's not as intractable as you think |
| 03:31.59 | ManxPower | T-1 PRI -> Asterisk -> Channel Bank -> Fax machine |
| 03:32.04 | swiftkick | ManxPower: understood. no doubt |
| 03:32.19 | ManxPower | That should work reasonably reliably as long as your audio levels are pretty good. |
| 03:32.28 | swiftkick | mchou: what do you recommend instead of asterisk to provide that function? |
| 03:32.50 | mchou | swiftkick: email fax gateways :) |
| 03:32.54 | plik | swiftkick: good luck, and be sure to write up a how-to for the wiki once you've managed it ;) |
| 03:33.37 | swiftkick | plik: i will write a how-to in the wiki about how I ran some spare cat-6 from the back of the Cisco IAD2431-8FXS to 4 more phone jacks in the office down the hall, one of which will be labelled "FAX" |
| 03:33.40 | swiftkick | :) |
| 03:34.06 | plik | :) g'nite |
| 03:36.19 | BeeBuu | how can i know what the commands can be used in current asterisk system? |
| 03:36.55 | ectospasm | BeeBuu: help |
| 03:37.11 | BeeBuu | type help in CLI? |
| 03:37.27 | ectospasm | BeeBuu: ("help" is the Asterisk CLI command to show you all commands) |
| 03:37.33 | ectospasm | try it, it won't hurt |
| 03:37.35 | znoG | does anyone know if the idea behind the REALTIME function is to use Cut() function to get whatever value I want from the comma delimited string? |
| 03:37.58 | BeeBuu | ectospasm: i did that,but no command page() |
| 03:38.03 | znoG | sorry, found it on the wiki |
| 03:38.05 | BeeBuu | show |
| 03:38.05 | znoG | ignore me |
| 03:38.16 | ectospasm | BeeBuu: try "help application page" |
| 03:38.50 | BeeBuu | ectospasm: is it work on your side? |
| 03:39.09 | ectospasm | BeeBuu: gimme a second |
| 03:39.57 | BeeBuu | thanks,buddy |
| 03:41.13 | ectospasm | BeeBuu: wait, it's not "help" it's "core show application <app>", although Page() is not registered with my asterisk system... |
| 03:41.46 | BeeBuu | ectospasm: OK,thanks,i got you. |
| 03:42.24 | BeeBuu | only with 1.4.10 or new,i think. |
| 03:43.38 | ManxPower | We use Page in 1.2 |
| 03:44.10 | ManxPower | what other applications are you missing? |
| 03:44.40 | BeeBuu | ManxPower: page can be use in 1.2? |
| 03:45.04 | ManxPower | BeeBuu: It is PART of 1.2 |
| 03:45.13 | ManxPower | "show application page" in 1.2 |
| 03:45.22 | BeeBuu | ManxPower: had you try? |
| 03:45.50 | BeeBuu | i got a problem with command page,could you help? |
| 03:46.09 | ManxPower | I cal tell you how we use page. |
| 03:46.44 | BeeBuu | i use like this: page(zap/g0/811&zap/g0/812|q) |
| 03:47.11 | BeeBuu | but only the extension 811 ring,what's wrong? |
| 03:47.20 | ManxPower | Page is for auto answering SIP devices |
| 03:47.54 | ManxPower | I don't know if Page even works with FXO ports |
| 03:48.27 | BeeBuu | only work for SIP protocol? |
| 03:48.41 | ManxPower | Do you know of any other protocol to make the phone auto answer? |
| 03:48.55 | BeeBuu | no |
| 03:48.58 | mosty | BeeBuu, do you have analogue phones which support auto answer? |
| 03:49.16 | BeeBuu | mosty: no |
| 03:49.25 | ManxPower | I don't KNOW if it will work or not. I just checked and we use chan_local for dialing the destinations. |
| 03:49.34 | mosty | BeeBuu, well Page is not going to work then is it? |
| 03:49.39 | ManxPower | BeeBuu: What exactly are you trying to do? |
| 03:51.18 | BeeBuu | ManxPower: i want to call some guys and put them to a meet auto. |
| 03:51.33 | BeeBuu | when they answer. |
| 03:52.27 | ManxPower | BeeBuu: you can try that with Page, it's not what it's designed for, however. I do not know if Page works dialing the way you want using channel groups. |
| 03:52.34 | BeeBuu | any application can do that? |
| 03:53.31 | ManxPower | BeeBuu: the alternative would be to use a .call file to spool the calls out, redirect them to a Meetme when they are answered |
| 03:53.44 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 03:53.58 | BeeBuu | ManxPower: i tried before,but still get problem |
| 03:54.14 | ManxPower | what problem did you have with using spool files? |
| 03:54.53 | BeeBuu | i made 2 call file,and move them to outgoing DIR,but only one call make at a same time! |
| 03:55.08 | JunK-Y | ManxPower: it doesnt work with really high volume? :) |
| 03:55.10 | BeeBuu | 2 call files,only one call out |
| 03:56.33 | ManxPower | BeeBuu: Is this a production system? |
| 03:56.47 | BeeBuu | what's your mean? |
| 03:56.48 | ManxPower | how many channels are in g0? |
| 03:57.02 | BeeBuu | ManxPower: only 2 for test |
| 03:57.04 | ManxPower | BeeBuu: is this a pbx that is in use everyday, or are you setting this up for the first time. |
| 03:57.19 | ManxPower | and have you successfully dialed out from two phones? |
| 03:57.24 | BeeBuu | it's my first time :-) |
| 03:57.36 | BeeBuu | yes,the lines is ok |
| 03:57.45 | ManxPower | then I suspect you are unable to make more than one call because of a config issue, not a dialing issue |
| 03:58.00 | BeeBuu | ManxPower: which config file? |
| 03:58.11 | BeeBuu | u mean the call file? |
| 03:58.23 | ManxPower | <PROTECTED> |
| 03:59.03 | BeeBuu | ManxPower: would you point out what is must be in that file? |
| 04:02.01 | ManxPower | What type of zap card do you have, how many ports, what kind of line coming into it. |
| 04:02.06 | BeeBuu | how can i fix that problem? |
| 04:02.21 | BeeBuu | 2 FXO & 2 FXS |
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| 04:02.28 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 04:02.39 | ManxPower | put a copy of your that file on pastebin.ca |
| 04:02.48 | ManxPower | WHAT MODEL of card do you have? |
| 04:03.04 | BeeBuu | pasting... |
| 04:03.14 | VJFROMGT | what does sip error 487 Request Terminated mean in asterisk? |
| 04:04.18 | trnzmeta | guys is there a device that converts isdn pri one side to pstn the other |
| 04:04.41 | BeeBuu | ManxPower: please help http://pastebin.ca/931405 |
| 04:05.24 | VJFROMGT | trn, asterisk should be able to do it |
| 04:05.29 | BeeBuu | thanks for your time |
| 04:05.50 | trnzmeta | VJFROMGT - having issues, trying to outrule the box |
| 04:07.24 | VJFROMGT | hmm, i believe i heard about something before |
| 04:07.52 | trnzmeta | I've seen it before, very small nifty device |
| 04:08.09 | BeeBuu | ManxPower: are you still there? |
| 04:08.43 | ManxPower | I was reading something else. I'll look at your paste now |
| 04:08.55 | trnzmeta | VJFROMGT - currently I have isdn --> asterisk --> pstn --> alarm receivers |
| 04:08.57 | BeeBuu | thanks |
| 04:09.40 | ManxPower | Can you do Dial(zap/g0/811&zap/g0/812) and will both of them ring? The file looks fine. |
| 04:09.56 | trnzmeta | the problem is something is happening that makes panels (dialers) call these alarm receivers multiple times |
| 04:09.56 | ManxPower | I am assuming that 1 ans 2 are the FXO ports. |
| 04:10.23 | BeeBuu | the command dial is call one by one,isn't it? |
| 04:10.49 | ManxPower | trnzmeta: how is information communicated between the two devices. |
| 04:11.07 | ManxPower | BeeBuu: Yes. Pick up one of your phones and dial an extension that runs the dial. |
| 04:11.19 | ManxPower | you have two fxs, two fxo, I am assuming you are dialing out the FXO |
| 04:11.33 | ManxPower | I've never heard of 3-digit phone numbers, but I won't question it. |
| 04:12.08 | BeeBuu | let me try. |
| 04:12.16 | mosty | BeeBuu, perhaps you should be using ZAP/1&ZAP/2 orZAP/3&ZAP/4 |
| 04:13.37 | BeeBuu | mosty: i think group is better |
| 04:13.52 | ManxPower | BeeBuu: Group is only correct for dialing via FXO ports |
| 04:14.07 | BeeBuu | ManxPower: just like now? |
| 04:14.19 | ManxPower | BeeBuu: are you dialing via FXO ports? |
| 04:14.24 | BeeBuu | yes |
| 04:14.40 | BeeBuu | as you seen in my zapata.conf |
| 04:14.57 | trnzmeta | ManxPower: proprietary alarm protocols --> ademco 685 (pretty much modem like) |
| 04:15.22 | BeeBuu | group=0 |
| 04:15.22 | BeeBuu | channel=1,2 |
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| 04:33.52 | ManxPower | trnzmeta: and you know what they say about modems and Asterisk, right? |
| 04:34.46 | trnzmeta | that's how they set it up initially, I'm left with fixing it |
| 04:35.17 | trnzmeta | so I'm trying to find a isdn to pstn device, so I can take on less out of the equation |
| 04:35.22 | ManxPower | They say "Don't expect to get more than 14.4k out of it, maybe as little as 9.6k |
| 04:36.05 | ManxPower | trnzmeta: my first thing to look at with fax machines (that also have a VERY modem-like protocol) is the in/out audio levels on the pstn interface. |
| 04:36.34 | ManxPower | Does the device REQUIRE an ISDN connection? |
| 04:38.09 | trnzmeta | ManxPower --> no, but we are taking 40 lines in |
| 04:38.38 | keith4 | crap |
| 04:38.45 | keith4 | i went from 1.2 to 1.4, and now asterisk won't start |
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| 04:39.54 | ManxPower | keith4: I'm surprised you are surprised. |
| 04:40.11 | ManxPower | That's a major upgrade. You read the upgrade.txt file right? |
| 04:40.14 | keith4 | i'm not surprised |
| 04:40.24 | VJFROMGT | keith what does log say? |
| 04:40.33 | keith4 | all kinds of stuff |
| 04:40.41 | keith4 | ManxPower: upgrade.txt? ;-) |
| 04:40.57 | ManxPower | keith4: run "asterisk -cvvv" paste the last half screen of output to pastebin.ca |
| 04:41.21 | keith4 | ah, colors! |
| 04:41.29 | ManxPower | keith4: yeah, the document that lists all the changes, removal, and updating of applications, features, and everyhing you need to know about before you upgrade. |
| 04:41.47 | ManxPower | go read it. it's in the asterisk doc/ directory |
| 04:41.48 | keith4 | well, i wasn't paying attention, and I did a debian dist-upgrade |
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| 04:43.39 | keith4 | ooh, errors |
| 04:43.53 | keith4 | ManxPower: the last half screen is not more useful than the first 4 screens |
| 04:44.53 | keith4 | these, however, are particularly worrisome: http://pastebin.ca/931424 |
| 04:51.04 | keith4 | and... upgrade.txt isn't included in the package. excellent |
| 04:55.34 | [TK]D-Fender | keith4, And did you upgrade libpri & zaptel at the same time? |
| 04:56.49 | ManxPower | keith4: now you know why we dislike packages so much. |
| 04:57.24 | ManxPower | keith4: the last line on the screen before it exits is usually exactly what we need to knwo |
| 04:58.14 | ManxPower | you have both a zap card and a VB card? |
| 04:58.21 | keith4 | I don't have a VB card |
| 04:58.39 | ManxPower | keith4: it's configured for one now. |
| 04:58.40 | keith4 | [TK]D-Fender sure! probably. I rebuilt the zaptel modules |
| 04:59.09 | ManxPower | keith4: you can use /etc/asterisk/modules.conf to noload => chan_vpb.so |
| 04:59.28 | ManxPower | I'm sure there are other ways of fixing that specific problem. |
| 05:00.07 | keith4 | maybe i should have used the packaged modules.conf.... |
| 05:00.16 | [TK]D-Fender | keith4, Go verify the versions of everything, and make sure you;re running current modules |
| 05:00.57 | keith4 | yessir |
| 05:01.30 | keith4 | yay, it starts now |
| 05:03.08 | keith4 | but my zap channels don't work |
| 05:03.09 | keith4 | hmmm |
| 05:03.38 | keith4 | but sip clients are registering |
| 05:13.49 | jameswf-home | hmmmmm |
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| 05:19.10 | Andy_WaveNet | my outgoing messages are very choppy, but music on hold is fine. anyone else have this problem? |
| 05:19.28 | mosty | all outgoing sound is choppy? |
| 05:19.47 | Andy_WaveNet | music on hold is fine, but announcements are choppy |
| 05:22.40 | mmlj4 | not enough oomph in the box? |
| 05:23.12 | Andy_WaveNet | i know there is, as it worked before. i just had to do a complete reinstall of the box. |
| 05:26.11 | SteveTotaro | ztdummy? |
| 05:26.17 | SteveTotaro | are you using iax? |
| 05:26.38 | SteveTotaro | convert your messages to slin |
| 05:26.50 | Andy_WaveNet | after the reinstall, i started having this problem |
| 05:27.30 | Andy_WaveNet | i'm using sip |
| 05:27.48 | Andy_WaveNet | my incoming did provider is freedigits. could the problem be there? |
| 05:28.37 | mosty | what do you mean by announcements? |
| 05:29.06 | Andy_WaveNet | IVR |
| 05:29.14 | Andy_WaveNet | phone system error messages, etc |
| 05:29.46 | mosty | what about when you talk? |
| 05:30.06 | JunK-Y | Andy_WaveNet: you mean ur digits doesnt work in ivr? |
| 05:30.09 | Andy_WaveNet | its fine |
| 05:30.11 | keith4 | yah, all fixed. thanks ManxPower (and [TK]D-Fender sortof ;-) |
| 05:30.23 | Andy_WaveNet | digits work fine in IVR |
| 05:30.26 | mosty | Andy_WaveNet, what codec are you sound files stored in, and what codec are you using on these calls? |
| 05:30.48 | Andy_WaveNet | i never thought to check that |
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| 06:14.11 | BeeBuu | anyone using command PAGE()? |
| 06:14.17 | fujin | yes, I do |
| 06:14.20 | fujin | not much though |
| 06:14.22 | fujin | it's not that useful |
| 06:15.36 | BeeBuu | fujin: would you like to help me fix problem? |
| 06:15.43 | fujin | BeeBuu: not particularly |
| 06:15.55 | mosty | BeeBuu, we mentioned before that Page is not what you want for your situation |
| 06:16.12 | mosty | and also Page will not work for FXO/FXS ports |
| 06:16.20 | fujin | page uses dummy channels doesn't it? |
| 06:16.31 | BeeBuu | mosty: are you sure? |
| 06:16.32 | fujin | I use page in a SIP only environment w/ ztdummy |
| 06:16.41 | fujin | with autoanswer on my linksys spa942's |
| 06:17.35 | BeeBuu | the book doesn't say only work for FXO/FXS prots |
| 06:17.48 | russellb | Page() will work fine for FXO and FXS ... if you have something on the other end that automatically answers the call |
| 06:18.11 | mosty | BeeBuu, said before that they do not have analogue phones that support auto-answer |
| 06:18.24 | BeeBuu | russellb: why must be auto answer? |
| 06:18.39 | mosty | BeeBuu, the purpose of Page is to call a phone and have it auto-answer |
| 06:18.39 | russellb | how else would you expect it to work? |
| 06:19.00 | russellb | Page just initiates a bunch of calls at once and dumps them into a conference ... |
| 06:19.07 | BeeBuu | russellb: i need call someones and put them to a meet room |
| 06:19.35 | fujin | so use meetme or app_conference like the rest of the world |
| 06:19.51 | BeeBuu | meetme can call out? |
| 06:20.05 | fujin | ugh |
| 06:21.09 | mosty | BeeBuu, initiate the call using call files or AMI or at the console |
| 06:22.01 | BeeBuu | mosty: as you known,i had try the call files,the problem is only one call made,even i get 2 call file. |
| 06:22.46 | BeeBuu | i made 2 call files,and mv them to outgoing DIR,but only one file work... |
| 06:22.48 | mosty | something is wrong with your 2nd call file then |
| 06:23.11 | BeeBuu | what's could be? |
| 06:23.41 | ManxPower | BeeBuu: Every single day one of my servers makes batches of 8 simul calls and all 8 happen at once. |
| 06:24.17 | BeeBuu | all same beside CHANNEL: zap/g0/810 and zap/g0/811 |
| 06:24.42 | mosty | BeeBuu, what do your logs say is happening when you try? |
| 06:24.52 | BeeBuu | logs.... |
| 06:25.45 | BeeBuu | let me check that... |
| 06:28.56 | BeeBuu | Mar 7 14:27:43] NOTICE[4701] pbx_spool.c: Call failed to go through, reason 8 |
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| 06:31.10 | ice_croft | hi all |
| 06:31.29 | ice_croft | how can i set max channels, those can pass through sip trunk? |
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| 06:31.45 | BeeBuu | <PROTECTED> |
| 06:32.01 | BeeBuu | mosty: any idea? |
| 06:32.19 | ice_croft | or it's unlimited? |
| 06:32.39 | mosty | BeeBuu, are all your g0 channels in use? |
| 06:32.51 | BeeBuu | no |
| 06:33.00 | BeeBuu | g0 get 2 channels! |
| 06:33.28 | mosty | can you explicitly call via each channel? |
| 06:33.40 | mosty | maybe there is something wrong with one of your lines |
| 06:34.56 | BeeBuu | i had tested. if i move all call file at a same time,only one call work;but if i move 2 call file one by one ,all work! |
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| 06:38.28 | mosty | one by one, but with the calls active simultaneously works? |
| 06:38.30 | BeeBuu | what's "reason 8"? |
| 06:38.50 | BeeBuu | yes, |
| 06:39.57 | BeeBuu | maybe not,cause i get the 2 phone ringing... |
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| 06:43.54 | mosty | i don't know what reason '8' is, try googling the error message |
| 06:52.58 | yang | I am trying to make SIP/66 ring if SIP/65 is being busy (in a call) with this setup - http://pastebin.com/m57558af6 |
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| 07:08.01 | ice_croft | somebody help me |
| 07:08.20 | ice_croft | i have planet vip 150 sip phone |
| 07:08.53 | ice_croft | and i can't dial from it to remote sip via trunk |
| 07:10.14 | ice_croft | http://pastebin.ca/931493 |
| 07:10.29 | ice_croft | it calls local sip clients well |
| 07:12.55 | yang | ice_croft: do you have extension like this Dial...(SIP/trunk/{exten}) ? |
| 07:13.30 | yang | you need to define trunk for outgoing calls |
| 07:15.10 | ice_croft | yang> i have a trunk, and my softphone xlite connects well via it |
| 07:15.35 | ice_croft | yang> where to dig? i'm dispaired |
| 07:15.53 | drmessano | Yoru trunk name is sr? |
| 07:15.59 | ice_croft | yang> xlite and vip150 have same config |
| 07:16.04 | drmessano | and you're dialing ext 912 @ sr? |
| 07:16.05 | ice_croft | yang> sip config |
| 07:16.13 | yang | you should set a parameter in extensions.conf ... but hey do you run your own asterisk ? |
| 07:16.15 | ice_croft | ice_croft> yes |
| 07:16.26 | ice_croft | yang> yes i am |
| 07:16.33 | ice_croft | drmessano> yes |
| 07:16.50 | yang | ice_croft: could you paste extensions.conf someqwhere |
| 07:16.51 | drmessano | sounds to me like far end isn't accepting the call |
| 07:17.11 | ice_croft | drmessano> no, logs r clear on remote * :( |
| 07:17.14 | drmessano | It's dialing it, but getting circuit busy |
| 07:17.15 | ice_croft | yang> min |
| 07:18.30 | ice_croft | drmessano> it has 407 error in the beginnin |
| 07:18.36 | ice_croft | drmessano> why? i don't understand |
| 07:18.43 | ice_croft | http://pastebin.ca/931498 |
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| 07:21.04 | ice_croft | drmessano> http://pastebin.ca/931493 line 34 |
| 07:21.10 | ice_croft | drmessano> what is it? |
| 07:23.14 | ice_croft | oh |
| 07:23.35 | ice_croft | softphone has same 407 error, but connects well! |
| 07:24.10 | ice_croft | so may be it's about max trunk channels? |
| 07:28.31 | ice_croft | it looks like trunk trouble for me |
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| 07:42.19 | greed | exit |
| 07:42.21 | greed | quit |
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| 08:19.31 | DiDjCodt | hello |
| 08:19.43 | DiDjCodt | anybody alive willing to help me with sip/dtmf ? |
| 08:20.20 | Stochastik | What's wrong? |
| 08:20.21 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:20.53 | Stochastik | Are you doing inband or rfc2833? |
| 08:23.45 | DiDjCodt | i have to cases |
| 08:24.02 | Stochastik | ? |
| 08:24.05 | DiDjCodt | i'm making a call from /var/spool/asterisk/outgouing |
| 08:24.42 | DiDjCodt | and at the end of the dialplan, i do a WaitExten |
| 08:25.06 | DiDjCodt | when calling my phone, the dtmf are in the rtp, so i guess they are inband |
| 08:25.16 | DiDjCodt | and asterisk is correctly receiving them |
| 08:25.40 | DiDjCodt | when calling another phone (sip phone, on another telecom company), i don't see the dtmf in the rtp |
| 08:25.53 | DiDjCodt | i don't know if it is rfc or info |
| 08:26.02 | DiDjCodt | btw, i need to be able to handle both cases |
| 08:26.22 | DiDjCodt | so i'm not sure adding a SIPDtmfMode() would do the trick :/ |
| 08:26.50 | DiDjCodt | am i missing an option in sip.conf or something like that ? |
| 08:28.42 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 08:29.50 | DiDjCodt | oh, the sip account i'm using to call from asterisk and the sip phone i'm calling which is not working are on the same telecom network |
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| 08:32.52 | DiDjCodt | Stochastik: any clue ? |
| 08:33.38 | Stochastik | Which dtmf mode is your sip phone set to? |
| 08:33.54 | DiDjCodt | i don't know :/ |
| 08:34.11 | DiDjCodt | how can i see the mode ? |
| 08:34.18 | Stochastik | Which phone is it? |
| 08:34.32 | DiDjCodt | both phones are thomson |
| 08:34.39 | DiDjCodt | same model |
| 08:35.28 | Stochastik | I would guess that you can see the setting in the web interface for the phones. |
| 08:35.58 | Lsodi | Hello, I cant figure ot whats causing the problem with SIP trunk, Sip trunk is between asterisk and cisco, calls from cisco to asterisk work fine, but calls from asterisk to cisco always fail, here is sip trun debug http://pastebin.com/d7d0849a6 |
| 08:36.32 | DiDjCodt | Stochastik: indeed, the phones are mgcp :/ |
| 08:36.41 | Stochastik | ahh |
| 08:36.45 | Lsodi | what does SIP/2.0 501 Not Implemented means? |
| 08:36.49 | DiDjCodt | sorry for the confusion, but one is working, not the other ;) |
| 08:37.15 | DiDjCodt | Stochastik: and asterisk is using a sip account |
| 08:37.54 | Stochastik | Have you watched the packets with tshark to see what is being sent? |
| 08:38.13 | Stochastik | You can see the rfc2833 if you tshark -R rtp |
| 08:38.25 | DiDjCodt | when i typed the dtmf, i don't have anything but the rtp with silence |
| 08:38.49 | Stochastik | And you can't get in to the phone web interface? |
| 08:40.30 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:40.55 | Stochastik | Lsodi: Maybe a codec mismatch? |
| 08:42.30 | Stochastik | Lsodi: a Cisco SIP gateway -> Asterisk? |
| 08:42.43 | Stochastik | AS5300/AS5400? |
| 08:44.17 | *** join/#asterisk qdk_ (n=qdk@85.235.253.139) |
| 08:44.51 | DiDjCodt | Stochastik: i don't see anything in the web interface |
| 08:46.22 | DiDjCodt | it's a ST2030 |
| 08:47.23 | DiDjCodt | and nothing about rfc2833 when tshark -R rtp |
| 08:48.24 | Stochastik | RTP EVENT Payload type=RTP Event, DTMF Five 5 |
| 08:48.40 | Stochastik | It would look something like that... so I guess it's inband eh? |
| 08:48.55 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) |
| 08:49.04 | Stochastik | You phone is connecting to Asterisk? |
| 08:49.06 | DiDjCodt | only lines like this : |
| 08:49.07 | DiDjCodt | 2965 34.640307 xxx.xx.xxx.xxx -> xx.xx.xxx.xxx RTP PT=ITU-T G.711 PCMA, SSRC=0x476731FF, Seq=1691, Time=3956838379 |
| 08:49.08 | Stochastik | your |
| 08:49.14 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 08:49.32 | DiDjCodt | i don't know the internal configuration between my asterisk and the phone |
| 08:49.55 | Stochastik | Can you put your sip.conf on pastebin? |
| 08:50.44 | DiDjCodt | 2s |
| 08:51.28 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 08:52.01 | DiDjCodt | Stochastik: http://djc.chewa.net/depot/sip_2008030701.conf |
| 08:52.16 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 08:52.42 | *** join/#asterisk obnauticus_ (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 08:53.08 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:53.15 | DiDjCodt | oh, i think the phone operator is using cirpack |
| 08:53.33 | obnauticus | Is it possible to load balance trunks on asterisk? |
| 08:54.07 | Stochastik | Have you tried dtmfmode=auto? |
| 08:54.07 | obnauticus | say you have 50 calls initiated, can you have asterisk automatically split 25 and 25 when they are initiated, so like 50% load on one trunk and 50% load on another |
| 08:54.27 | DiDjCodt | in sip.conf, in sip account section or general section ? |
| 08:54.53 | Stochastik | obnauticus: I'd use ser or openser |
| 08:55.13 | Stochastik | DiDjCodt: I put mine in the account section |
| 08:55.22 | DiDjCodt | ok |
| 08:55.36 | obnauticus | Stochastik openser :\ |
| 08:55.41 | Stochastik | obnauticus: Or you can ues DNS SRV |
| 08:56.19 | *** join/#asterisk adjohn (n=adjohn@p5155-ipad74marunouchi.tokyo.ocn.ne.jp) |
| 08:56.41 | DiDjCodt | Stochastik: nothing works now :) |
| 08:56.51 | Stochastik | lol |
| 08:56.54 | Stochastik | and you just added one line? |
| 08:58.08 | DiDjCodt | yup |
| 08:59.13 | DiDjCodt | changing to dtmfmode=inband and my phone is working again |
| 08:59.44 | Stochastik | You have that in general again? |
| 09:00.00 | DiDjCodt | indeed |
| 09:01.55 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 09:02.57 | Lsodi | Stochastik: Asterisk -> cisco isnt working. outgoing and incominc calls use g711ulaw, from cisco -> asterisk calls work. |
| 09:04.09 | obnauticus | Stochastik openser seems pretty kewl |
| 09:04.31 | Stochastik | obnauticus: That it is! |
| 09:05.50 | ThoMe | http://tm.muc.de/up/archiv/Unbenannt.PNG |
| 09:05.51 | ThoMe | ups |
| 09:05.53 | Stochastik | Lsodi: and you have ulaw allowed in your sip.conf, right? |
| 09:06.25 | Lsodi | Stochastik: yes, ulaw and alaw |
| 09:06.42 | DiDjCodt | Stochastik: well, when using another sip account for asterisk (one from another telecom company), the same phone is working... |
| 09:07.03 | DiDjCodt | Stochastik: i think b3g is not handling in the same way their dtmf when the call is withing their network or not |
| 09:07.38 | Stochastik | DiDjCodt: Maybe you need to set them to dtmfmode=rfc2833? |
| 09:07.55 | DiDjCodt | in sip.conf ? |
| 09:07.58 | Stochastik | yeah |
| 09:08.08 | Stochastik | worth a try anyway |
| 09:08.11 | DiDjCodt | well, trying in 15 minutes... |
| 09:09.35 | Lsodi | Stochastik: SIP trunk in asterisk http://pastebin.com/d7fa10a01 |
| 09:09.46 | Stochastik | Lsodi: Have you looked at debug ccsip? |
| 09:10.26 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:12.01 | tengulre | does the asterisk support FreeBSD? |
| 09:13.19 | Lsodi | Stochastik: no I havent, I'll do it (cisco is new to me, still googling and learning), is diap-peer 2002 correctly written? http://pastebin.com/de0b71f1 |
| 09:14.24 | yang | I am trying to make SIP/66 ring if SIP/65 is being busy (in a call) with this setup - http://pastebin.com/m57558af6 |
| 09:14.40 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 09:15.09 | yang | tengulre: i found an interesting asterisk project for freebsd - http://www.askozia.com/pbx/ |
| 09:16.34 | Stochastik | Lsodi: What is sip-server set to? |
| 09:16.38 | zeeesh | i can c at console my sip peer register for 3600 sec ... can i change it forever or 50000 or 100000 sec?????????? |
| 09:16.49 | Stochastik | nevermind |
| 09:16.51 | Stochastik | you have there there |
| 09:16.59 | Stochastik | looks good to me |
| 09:17.42 | Stochastik | s/there/it/ |
| 09:19.58 | Stochastik | Lsodi: http://pastebin.com/m1aed9a38 |
| 09:20.05 | Stochastik | Lsodi: one of my working ones |
| 09:21.10 | Stochastik | debug dialpeer |
| 09:21.12 | Stochastik | might help |
| 09:21.51 | Stochastik | along with 'terminal monitor' if you're in ssh or telnet |
| 09:21.51 | Nugget | telnet is eeeeeeevil! |
| 09:22.45 | Lsodi | Stochastik: thank you, I'll try |
| 09:45.13 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:49.12 | *** join/#asterisk stoffell (n=stoffell@24.138-136-217.adsl-dyn.isp.belgacom.be) |
| 09:50.36 | awk | hrm, bri asterisk repo is compltely messed.. |
| 09:50.45 | awk | anyone have the alatest bri stuff tar.gz files? |
| 09:50.58 | awk | I can't get thm off /www.asterisk-support.de |
| 09:51.03 | awk | downloads 10kb only |
| 09:57.38 | awk | its fine, thanks... found on other mirrors |
| 10:04.42 | stoffell | awk, get them from junghanns.net/downloads/ |
| 10:07.56 | awk | doesn't work.. |
| 10:08.03 | awk | downloads 10kb of the files onlyt |
| 10:08.10 | awk | tried from 3 different machines |
| 10:17.00 | *** join/#asterisk sbingner (n=john@pdpc/supporter/sustaining/sbingner) |
| 10:17.18 | zeeesh | i am dialing out with 00 ... can i dial with + as well .. can i mention in my context start with + .. is it possible ? |
| 10:19.13 | sbingner | zeeesh, does your phone have a + button? |
| 10:23.28 | *** join/#asterisk duckz (n=duckz@81.180.102.217) |
| 10:26.52 | *** join/#asterisk darviria (n=darviria@87-194-177-180.bethere.co.uk) |
| 10:28.11 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:29.28 | yang | awk: why arent you getting bristuff from debian, are you using debian? |
| 10:31.45 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 10:33.51 | *** join/#asterisk hank (n=hank@netwichtig.de) |
| 10:33.53 | hank | moin |
| 10:33.55 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 10:35.59 | hank | i am running asterisk 1.2 with some snom phones. in the snom-webinterface is a dialbox and it says 'enter telephone number or URL like info@snom.com'. i'd love to be able to just enter the initials of a colleague in there to dial his phone. where can i find more information on that feature? |
| 10:44.27 | zeeesh | <sbingner>: yeah .. my phone has button .. but asterisk is accepting calls just with 00 when i changed in extensions.conf 00 with only + .. it does't support???? |
| 10:44.41 | *** join/#asterisk grEvenX (n=even@pc105-230.ktv.no) |
| 10:46.47 | *** join/#asterisk af_ (n=getsmart@88-149-240-207.dynamic.ngi.it) |
| 10:50.20 | *** join/#asterisk agallo (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 10:57.09 | *** join/#asterisk Azam (n=azamzia@58-65-160-140.nayatel.pk) |
| 10:57.16 | Azam | Hello |
| 10:57.38 | Azam | i need to knw can i configure two asterisk servers with one realtime DB? |
| 10:57.57 | Azam | i am having some issue when i try to call from one server to the other |
| 11:01.06 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 11:03.37 | *** join/#asterisk FabiOne (n=FabiOne@host107-144-static.59-88-b.business.telecomitalia.it) |
| 11:09.10 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:20.56 | gego | hank: moin, as well. I'm not using snomes but grandstreams - do you mean a speed dial button? |
| 11:21.27 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 11:36.51 | flush | ahoy |
| 11:37.08 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-cfb53160638eea5e) |
| 11:45.19 | hank | gego: i just 'solved' it... snoms have a builtin http(s) server. on the 'website' is a box where you can enter a number to dial. but you can also enter a uri like 'info@snom.com'. |
| 11:45.31 | redax | hi |
| 11:45.52 | redax | Please help me, to choose the right E1 card for me, |
| 11:46.10 | redax | is the Digium TE121 PCIe card will work well for me? |
| 11:46.12 | ice_croft | i'm totally succeded with sip trunks! |
| 11:46.18 | ice_croft | horray to me! |
| 11:46.37 | redax | or should I use the old PCI ones instead of the PCI express one |
| 11:50.30 | flush | i have my fxo module set up and fxs too, i set up the echo thing it works, zttool shows "alarms: OK" and i have dial tone when i pick up the phone. however when a call comes in asterisk wont make the phone ring and when i hit a key it immediately gives me the fast busy tone, how can i place a call then |
| 11:50.38 | flush | what gives, any ideas? |
| 11:51.50 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 11:52.05 | kamisalami | can someone help me with this: |
| 11:52.06 | kamisalami | .[Mar 7 12:49:51] ERROR[8193]: chan_zap.c:9357 start_pri: Unable to open D-channel 55 (No such device or address) |
| 11:52.06 | kamisalami | [Mar 7 12:49:51] ERROR[8193]: chan_zap.c:11268 setup_zap: Unable to start D-channel on span 2 |
| 11:52.22 | kamisalami | i never defined an d-chan 55 |
| 11:52.44 | kamisalami | see http://pastebin.ca/931887 for configuration details |
| 11:52.54 | kamisalami | it's on ubuntu server 7.10 |
| 11:53.10 | awk | yang no not using debian, also debian repo's are outdated.. |
| 11:55.13 | flush | waht |
| 11:55.21 | flush | im using asterisk with ubuntu 7.10 too |
| 11:56.20 | kamisalami | i use the zaptel from hardy, because it provides me with vzaphfc |
| 11:56.47 | kamisalami | but can't get the hfc card configured in asterisk |
| 11:56.56 | kamisalami | in the kernel it seems to work |
| 11:57.41 | kamisalami | the other card from digium is working fine |
| 11:59.35 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 12:00.01 | mvanbaak | kamisalami: can you pastebin the output of: 'ztcfg -vvv' |
| 12:01.16 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:01.48 | kamisalami | ok |
| 12:02.21 | flush | yo seriously |
| 12:02.38 | flush | how the hell do you configure a single FXO module with a single FXS module only to be able to call your mom |
| 12:03.00 | flush | its like my fxo aint working |
| 12:04.00 | kamisalami | see http://pastebin.ca/931908 |
| 12:05.50 | kamisalami | flush: do you get anything on the asterisk console when you set debug and verbosity levels to something like 1000 ? |
| 12:06.52 | kamisalami | flush: is asterisk getting the call at all? or does your module create the dial tone? |
| 12:07.15 | kamisalami | mvanbaak: see http://pastebin.ca/931908 |
| 12:08.06 | hank | gego: muss weg... wenns dich interessiert, mach nen query bitte. |
| 12:08.08 | flush | my module creates the dial tone |
| 12:08.10 | hank | cu all |
| 12:08.10 | *** part/#asterisk hank (n=hank@netwichtig.de) |
| 12:08.14 | flush | i unplug the FXO from the wall, i still have dial tone |
| 12:08.29 | flush | kamisalami im a total newb sorry, how do i get debug and verbosity set to 1000 |
| 12:10.21 | flush | zttool says "no alarms" and "alarms: OK" even when i unplug the FXO from the wall jack |
| 12:10.59 | flush | :( |
| 12:11.03 | yang | I am trying to make SIP/66 ring if SIP/65 is being busy (in a call) with this setup - http://pastebin.com/m57558af6 |
| 12:13.31 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:14.42 | kamisalami | flush: core set debug level 10000 |
| 12:14.54 | kamisalami | flush core set verbose level 10000 |
| 12:15.07 | kamisalami | or 1000, will problably be enough :) |
| 12:15.58 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 12:16.08 | kamisalami | flush: probably it is not able to read the status at all, wich might be an issue with the driver, but i am not sure. I am not that experienced with it :) |
| 12:18.18 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:18.21 | kamisalami | flush wich fxo/fxs modules are yo using? |
| 12:18.27 | kamisalami | you |
| 12:18.29 | kamisalami | :) |
| 12:18.45 | flush | i dont know man |
| 12:18.48 | flush | i have tdm400p card |
| 12:19.20 | kamisalami | did you see http://www.voip-info.org/wiki/view/TDM400P ? |
| 12:19.45 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 12:19.53 | flush | negative |
| 12:20.19 | kamisalami | ok, have a look there first and see if you find anything usefull |
| 12:21.25 | flush | ok ill read thanks man hope i figure something out |
| 12:21.34 | flush | i gotta run i got courses have a good day, bbl |
| 12:22.31 | kamisalami | mvanbaak: you still with us? |
| 12:34.27 | kamisalami | someone still lurking here, or am i alone? |
| 12:34.59 | ice_croft | lurkin' |
| 12:35.02 | ice_croft | for sure |
| 12:37.25 | kamisalami | can you help me with my zaptel problem? |
| 12:37.34 | kamisalami | see http://pastebin.ca/931908 |
| 12:38.02 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:43.18 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 12:50.15 | *** join/#asterisk RoyK (n=roy@ip-202-5-149-91.dialup.ice.no) |
| 12:51.59 | nixguy | how do you guys use the i extension? |
| 12:52.24 | nixguy | one would thing that anything that doesent match goes to i but this is not the case |
| 12:52.36 | nixguy | are you guys maching with _. ? |
| 12:53.03 | nixguy | matching |
| 12:53.51 | *** join/#asterisk zenobic (n=zenobic@174-204-116-85.dsl.manitu.net) |
| 12:56.19 | *** join/#asterisk Dovid (n=Dovid@80.250.149.60) |
| 12:56.30 | Dovid | when asterisk does a core cump it goes in to /tmp ? |
| 12:57.38 | *** join/#asterisk jblack (n=jblack@pool-71-173-53-239.sctnpa.east.verizon.net) |
| 13:01.18 | redax | so, can you give me an advice what card E1 card should I choose for asterisk ? :) |
| 13:01.28 | mvanbaak | kamisalami: can you try it the other way around? use bchan 1-something,something-end for span 1 |
| 13:01.30 | nixguy | redax: depends on your need i guess |
| 13:01.34 | mvanbaak | ah well, you know what I mean |
| 13:01.46 | nixguy | redax: i myself use a te120p for my pstn connection |
| 13:01.46 | Dovid | redax: I like sangoma |
| 13:01.57 | Dovid | a lil hard to set up but works great |
| 13:01.58 | kamisalami | nixguy: did you put the i extension in the right context? |
| 13:02.00 | nixguy | 30chan primarey rate isnd |
| 13:02.06 | mvanbaak | hhmm |
| 13:02.12 | nixguy | kamisalami: yup its there alright |
| 13:02.13 | *** join/#asterisk nele (n=sanee@host-166-58.opticon.hu) |
| 13:02.15 | redax | nixguy: I've checked the Digium TE121 (PCIexpress one) |
| 13:02.21 | nixguy | somehow i just felt matching on everything feels wrong dont know |
| 13:02.27 | nixguy | i thought just i, |
| 13:02.28 | mvanbaak | hhmm |
| 13:02.29 | redax | is that the same good like the earlier TE120 ? |
| 13:02.31 | mvanbaak | your setup looks weird |
| 13:02.32 | nixguy | would sen all invalid ext's there |
| 13:02.38 | kamisalami | mvanbaak: mine? |
| 13:02.43 | mvanbaak | kamisalami: yes |
| 13:02.48 | redax | Dovid: which sangoma do you use? |
| 13:02.51 | kamisalami | why? |
| 13:02.58 | nixguy | redax: the standard modules for my distribution didnt have support for 120cards so i compuled modules from source, but after that it worked fine |
| 13:03.01 | mvanbaak | kamisalami: as you can see in ztcfg output |
| 13:03.10 | mvanbaak | you have channel 1-15 as b channels |
| 13:03.14 | mvanbaak | then a d channel |
| 13:03.35 | mvanbaak | so your zaptel.conf is wrong |
| 13:03.35 | nixguy | kamisalami: did you ask me? |
| 13:03.46 | mvanbaak | kamisalami: hang on, let me alter the pastebin |
| 13:03.47 | kamisalami | more tell you |
| 13:03.47 | redax | nixguy: zaptel-1.4.9.2 include the driver for your card, right? |
| 13:03.55 | *** join/#asterisk seanbright-home (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net) |
| 13:04.00 | kamisalami | mvanbaak: it is supposed to be that way |
| 13:04.07 | kamisalami | mvanbaak: i only use 15 lines |
| 13:04.17 | kamisalami | mvanbaak: it is from a working setup |
| 13:04.46 | nixguy | redax: hmm dont remember :) i just downloaded the source from digium |
| 13:04.47 | kamisalami | mvanbaak: it is the other card, lines 32,33 and dchan 34 that are not working |
| 13:04.54 | nixguy | and compiled them agains my linux-sources... |
| 13:05.29 | mvanbaak | http://pastebin.ca/index.php |
| 13:05.32 | mvanbaak | kamisalami: |
| 13:06.04 | redax | nixguy: checked zaptel, and it has support for TE120p TE121 and TE122 |
| 13:06.12 | nixguy | there ya go |
| 13:06.24 | nixguy | the card was like 600euro or something |
| 13:06.29 | redax | nixguy: so, you don't had any problem so far with your E1 card? |
| 13:06.31 | nixguy | not really that much money |
| 13:06.42 | nixguy | from a company perspective |
| 13:06.51 | nixguy | but i dont know might be cheaper alternatives that do the exact same thing.... |
| 13:06.59 | kamisalami | mvanbaak: where do i need the index for ? :) |
| 13:07.06 | redax | yep, we got E1 trunk from our Mobile provider, and that's why I need this card. |
| 13:07.15 | lirakis | N7n19rE |
| 13:09.06 | nixguy | redax: none what so ever actually |
| 13:09.15 | nixguy | redax: even if it was hell getting it to talk to pstn |
| 13:09.18 | nixguy | but thats a generic problem i think |
| 13:09.31 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 13:09.32 | nixguy | poorly documented ,magic config settings etc |
| 13:10.26 | redax | does those problems was linecode and crc related? |
| 13:10.38 | nixguy | no no purely configuration related |
| 13:10.42 | nixguy | my first asterisk ever |
| 13:10.46 | redax | ah. |
| 13:10.53 | nixguy | and with no prior experience in pstn or telephony |
| 13:11.02 | nixguy | the learning curve was quite steep |
| 13:11.11 | kamisalami | vanbaak: i'll try it |
| 13:11.46 | redax | uhh. that's good. I play a few years with asterisk, just this will be the first E1 card istall ;-) |
| 13:12.10 | jblack | Oh, I have a asterisk anecdote.... |
| 13:12.24 | nixguy | the module that came with debian was called t1xxp.ko i think |
| 13:12.36 | nixguy | it loaded but refused to work properly (or something like that) |
| 13:12.41 | nixguy | dont let the xx fool you |
| 13:12.50 | nixguy | using the sourcecompiled module t120.ko |
| 13:12.51 | nixguy | worked fine |
| 13:13.13 | jblack | A few weeks ago, I had a flood which took out my bedroom, so there's been a lot of workmen in there. I got tired of running upstairs to give them the phone every time the office called, so I dropped a line into the bedroom. :) |
| 13:13.34 | redax | now it's wcte12xp.ko for TE12x |
| 13:15.45 | kamisalami | mvanbaak: cat /proc/zaptel keeps insisting it is the other way around |
| 13:16.33 | redax | thanks nixguy for the informations, I think I take the PCIexpress WC TE121 |
| 13:17.01 | redax | found a shop where I get it for Eur 404 |
| 13:17.06 | nixguy | redax: you are welcome! |
| 13:17.24 | redax | that's sounds cheap ;-) |
| 13:17.53 | mvanbaak | kamisalami: sorry |
| 13:17.56 | mvanbaak | kamisalami: http://pastebin.ca/931978 |
| 13:18.00 | mvanbaak | <--- at work |
| 13:18.11 | mvanbaak | and ppl keep bugging me |
| 13:18.32 | kamisalami | me 2 , but we have some ppl who rotate what we call issue day |
| 13:18.40 | kamisalami | and i am not 1 of them luckily |
| 13:19.47 | hmmhesays | morning folks |
| 13:21.15 | *** part/#asterisk PepOSX (n=angeldav@190.72.144.198) |
| 13:21.29 | *** join/#asterisk PepOSX (n=angeldav@190.72.144.198) |
| 13:22.28 | *** join/#asterisk _Krieger_ (n=krieger@193.39.118.158) |
| 13:22.35 | kamisalami | mvanbaak: not sure what you changed? |
| 13:26.05 | yang | I am trying to make SIP/66 ring if SIP/65 is being busy (in a call) with this setup - http://pastebin.com/m57558af6 |
| 13:28.37 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:28.50 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 13:29.34 | kamisalami | mvanbaak: i removed configuration and modules for the digium card for now and loaded only the vhfczap module |
| 13:29.52 | kamisalami | mvanbaak: now i get: chan_zap.c:9357 start_pri: Unable to open D-channel 24 (No such device or address) |
| 13:30.21 | kamisalami | it *thinks* dchan is on channel 24, wich is nonexistent, should be on 3 |
| 13:31.02 | tzafrir_home | kamisalami, what version of Asterisk is it? |
| 13:31.24 | kamisalami | Asterisk 1.4.10 |
| 13:31.31 | tzafrir_home | Is it bristuffed? |
| 13:31.42 | kamisalami | yep |
| 13:32.03 | tzafrir_home | did you use signalling=bri_cpe_ptmp ? |
| 13:32.05 | kamisalami | standard asterisk from ubuntu 7.10 (deb) |
| 13:32.12 | tzafrir_home | or signalling=bri_cpe? |
| 13:32.26 | kamisalami | no |
| 13:32.40 | *** join/#asterisk DonAlex (n=DonAlex@mirafiori.demon.co.uk) |
| 13:32.44 | tzafrir_home | just use whatever genzaptelconf generates |
| 13:32.47 | kamisalami | does not seem to recognize those, think because i use the zaptel support for the hfc cards |
| 13:32.50 | tzafrir_home | It should work |
| 13:32.53 | DonAlex | Afternoon peeps :) |
| 13:33.17 | hmmhesays | heh wow I'm running into some really crazy stuff, using cmd read I lose all audio after it plays the sound file back I specified |
| 13:33.21 | tzafrir_home | hmmm..... |
| 13:33.26 | tzafrir_home | let's see... |
| 13:33.29 | hmmhesays | I'm still getting rtp back and forth |
| 13:33.44 | DonAlex | Need help from a guru methinks. Does anyone know how CUG work in asterisks? I mean authentication and strings and anything else? |
| 13:34.05 | DonAlex | That is PRI ISDN Closed User Groups etc.. |
| 13:34.15 | [TK]D-Fender | kamisalami: You're on BRI? |
| 13:34.24 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 13:34.35 | [TK]D-Fender | kamisalami: pastebin your zapata.conf and zaptel.conf |
| 13:35.37 | kamisalami | d-fender: i use support for hfc cards from zaptel, not sure if that is bri |
| 13:36.05 | tzafrir_home | kamisalami, what error do you get if you use cpe_bri_ptmp ? |
| 13:36.12 | kamisalami | <PROTECTED> |
| 13:36.17 | tzafrir_home | It is BRI |
| 13:36.22 | *** join/#asterisk FreezeS (n=nobody@82.208.157.125) |
| 13:36.31 | kamisalami | then i must be missing something |
| 13:36.55 | kamisalami | i was told that the asterisk modules in ubuntu are already bristuffed |
| 13:37.13 | kamisalami | maybe i have to load a module in asterisk? |
| 13:38.05 | tzafrir_home | bah, yet another half-baked ubuntu package :-( |
| 13:38.22 | tzafrir_home | And it still have our names on it, sadly |
| 13:38.37 | redax | I have a Digium B410P card (quad BRI) which is using mISDN drivers. time-to-time my pstn interfaces stops working, and have to restart asterisk to repair. |
| 13:38.51 | redax | is there any alternative to mISDN drivers, for the B410P ? |
| 13:39.01 | hmmhesays | this has to be some crazy nat problem |
| 13:39.06 | tzafrir_home | redax, how can you tell that pstn interface stops working? |
| 13:39.22 | DonAlex | No one at all.. any good with PRI ISDN and asterisk ? |
| 13:39.24 | tzafrir_home | BRI providers generally take the interface down when there are no calls |
| 13:39.27 | redax | you can dial out, nor dial in |
| 13:39.44 | tzafrir_home | ah, so that's not it |
| 13:39.45 | redax | call dropped immediatly, if I call from my mobile to the BRI numbers |
| 13:40.11 | [TK]D-Fender | redax: IIRC bristuff & PRI can't co-exist... |
| 13:40.11 | tzafrir_home | are you sure that this is not a dialplan issue? |
| 13:40.12 | kamisalami | tzafrir: i did use a recompiled package from ubuntu hardy, because the vzaphfc module was not compiled with the default package |
| 13:40.19 | tzafrir_home | did you try reloading asterisk? |
| 13:40.51 | redax | tzafrir: I _restart_ asterisk... should be enough a reload? |
| 13:41.00 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:41.22 | redax | [TK]D-Fender: this is an other pbx. where we have 4 BRI pstn connection... |
| 13:41.40 | DonAlex | Just we keep getting this hangup message that says code 90 which translates to invalid CUG |
| 13:41.50 | redax | tzafrir: why should be that dialplan issue, if I restart asterisk, everything works fine. |
| 13:42.12 | DonAlex | So either Asterisk is sending something and getting it wrong or it is not sending anything and getting it wrong |
| 13:42.51 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 13:43.03 | tzafrir_home | redax, well, you can easily emulate the same issue with a simple 'dialplan add' / 'dialplan delete' :-) |
| 13:43.16 | redax | tzafrir: and kernel messages going to be strange, after the bri interfaces stops working. |
| 13:43.37 | tzafrir_home | You restart , you also reset the channel driver. a reload doesn't |
| 13:44.16 | tzafrir_home | redax, ok. got me |
| 13:44.20 | redax | tzafrir: ;-) yah and the gate-keeper hacking asterisk ;-) |
| 13:46.02 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:46.28 | tzafrir_home | kamisalami, the packaging files for your package: http://svn.debian.org/wsvn/pkg-voip/asterisk/tags/1.4.10~dfsg-1/?rev=0&sc=0 |
| 13:46.39 | *** join/#asterisk susinths (n=susinths@sos3-1x-dhcp127.studby.uio.no) |
| 13:46.42 | tzafrir_home | now trying to figure out if bristuff is applied there |
| 13:48.30 | kamisalami | ok |
| 13:50.31 | tzafrir_home | that's funny |
| 13:50.52 | tzafrir_home | when I look at the patches in that tag, I don't find the bristuff patches |
| 13:52.13 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:52.13 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:53.16 | kamisalami | hmz... damn |
| 13:53.34 | kamisalami | i was misinformed ;) |
| 13:54.55 | kamisalami | there is nothing in the changelog about it being removed, but there are about it being included |
| 13:56.47 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 13:59.40 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 14:01.39 | x86 | well... I've got 4 of 6 T1's turned up now on my biggest Asterisk box |
| 14:02.07 | x86 | 48-port channel bank will make 6 :P |
| 14:02.25 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 14:02.26 | *** join/#asterisk dofear (n=arodef@202-91-197-146.intrapower.net.au) |
| 14:03.25 | dofear | Hi, my sip provider di dnot provide me with a username and password, they say it will accept calls based on my IP address ... how do I specify register in sip.conf for this provider? |
| 14:03.27 | kamisalami | tzafrir: seems like it is only included in hardy |
| 14:03.48 | kamisalami | tzafrir: package description does not include it for gutsy |
| 14:03.55 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 14:05.12 | dofear | if I try to send outbound call to this peer, I get this error "Got SIP response 503 "Service Unavailable" back from X.X.X.X" |
| 14:06.26 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 14:06.45 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:08.11 | x86 | dofear: you can't register without a username and password |
| 14:08.16 | x86 | well, at least a username anyway |
| 14:08.38 | dofear | can i send outbound call through a 'friend' without registering? |
| 14:09.16 | _ShrikE | dofear: If they did not provide you with credentials they don't want to you to register. Just create "friend" and send your calls that way. |
| 14:09.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:10.27 | dofear | i created the 'friend' in my sip.conf and added appropriate entries in extensions.conf ... but getting this 503 error |
| 14:10.30 | dofear | Got SIP response 503 "Service Unavailable" back from X.X.X.X |
| 14:10.55 | kamisalami | tzafrir: i do see bristuff patches in the patch file of gutsy |
| 14:11.12 | kamisalami | tzafrir: see http://packages.ubuntu.com/gutsy/asterisk |
| 14:11.44 | kamisalami | tzafrir: http://archive.ubuntu.com/ubuntu/pool/universe/a/asterisk/asterisk_1.4.11~dfsg-1.diff.gz |
| 14:12.05 | kamisalami | tzafrir: downloaded and extracted it, and it does contain bri patches as far as i can tell |
| 14:13.40 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 14:14.27 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:14.40 | x86 | dofear: you need a username and password for friends as well |
| 14:14.47 | x86 | dofear: you sure they're using SIP and not H323? |
| 14:14.59 | x86 | dofear: sounds like an H323 setup to me |
| 14:15.41 | _ShrikE | dofear: pastebin your sip.conf entry and sip debug for that peer |
| 14:15.51 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:16.15 | x86 | _ShrikE: he has no username and password for that SIP provider |
| 14:17.38 | _ShrikE | x86: Thats ok. You can have a sip peer without auth |
| 14:17.54 | x86 | ah... didn't think you could |
| 14:17.59 | x86 | at least, I could never get it to work :) |
| 14:18.26 | x86 | 32 active channels on our big Asterisk box right now... this is exciting :) |
| 14:18.59 | *** join/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 14:19.05 | x86 | we have an ancient toshiba KSU that had a T1 interface to the telco |
| 14:19.16 | dandre | Hello, |
| 14:19.48 | x86 | I put in a new telco T1 to a new provider directly to the asterisk box... then last night, I put the toshiba's T1 interface going into Asterisk also |
| 14:20.08 | x86 | that way both Asterisk and Toshiba can "share" the T1 channels heh |
| 14:20.14 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 14:20.16 | dandre | Is there any way to set de did number when a call is received on an analog zap channel if not provided by the operator? |
| 14:20.43 | x86 | dandre: what do you want the DID for? |
| 14:20.55 | x86 | dandre: each DID comes across a dedicated zap channel, correct? |
| 14:21.04 | dandre | yes |
| 14:21.15 | x86 | then you can put each channel in it's own context |
| 14:21.43 | ReDNeQ | how do i debug why the system isnt answering or calling out, yet system has been rebooted and zap is loading |
| 14:21.56 | dandre | and I want to share the dispatch logic in the dialplan |
| 14:22.08 | [TK]D-Fender | x86: No, a DID is not a zap channel. |
| 14:22.25 | x86 | [TK]D-Fender: no, but a DID can come in on a dedicated zap channel |
| 14:22.36 | x86 | [TK]D-Fender: aka, not a hunt group or otherwise |
| 14:22.38 | [TK]D-Fender | x86: if you arranged it as such with your telco. |
| 14:22.48 | x86 | [TK]D-Fender: right, that's what I was asking basically |
| 14:23.08 | x86 | [TK]D-Fender: if he knows the channel is always going to be the same for each DID, it's easy to put them in their own contexts |
| 14:23.45 | [TK]D-Fender | x86: You might do this is you have 1 PRI for a multi-tennent scenario where you want to basically "reserve" 12 B-chan's for CompanyA, and 11 B-Chans for CompanyB |
| 14:24.37 | [TK]D-Fender | x86: And you should not restrict your B-Chans for SINGLE DID's unless absolutely necessary . |
| 14:24.44 | *** join/#asterisk m160858 (n=m160858@mailserver2.kruger.com.ec) |
| 14:24.54 | m160858 | hi guys |
| 14:25.06 | dandre | I want to do something like: |
| 14:25.07 | dandre | if CID (from all ZAP channel) is some pattern, then dial extension 1 |
| 14:25.07 | dandre | if from channel 2 then dial 2 |
| 14:25.07 | dandre | else dial 3 |
| 14:25.39 | m160858 | i've a problem with zaptel 1.4.9, when a i try to activate the ECHO_CAN_MARK2 |
| 14:25.42 | hmmhesays | can read take multiple file names like playback ? |
| 14:26.06 | m160858 | *** Warning: "echo_can_init" [/usr/src/zaptel-1.4.9/kernel/zaptel.ko] undefined! |
| 14:26.10 | *** join/#asterisk ManxPower (n=manxpowe@97-81-241-251.dhcp.lds.al.charter.com) |
| 14:26.26 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 14:26.33 | [TK]D-Fender | dandre: Feel free to route based on weather, Patriots game last score, and anything else crazy you feel like. |
| 14:26.38 | jsmith | m160858: Why are you using the MARK2 echo can? Use MG2 instead -- it's much better (at least in my opinion) |
| 14:27.22 | x86 | [TK]D-Fender: Oh I agree, but if he's not dealing with a PRI it might be a different story |
| 14:27.31 | m160858 | ok, but that does not change my problem |
| 14:27.43 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 14:27.46 | coppice | use OSLEC. its a real echo canceller :-) |
| 14:27.51 | [TK]D-Fender | x86: If he's not dealing with PRI, then he typically doesn't have DID's, and this whole conversation becomes moot. |
| 14:28.00 | Katty | good morning, glories! |
| 14:28.07 | ManxPower | coppice saying Digium has a real EC? /me faints |
| 14:28.17 | [TK]D-Fender | coppice: I say cut the echo off at the source and shoot your callee ;) |
| 14:28.18 | Katty | it's a not so bright snowy day, but it doesn't matter! it's goign to be gggggggreat! |
| 14:28.29 | coppice | does Digium supply OSLEC? |
| 14:28.37 | ManxPower | coppice: is OSLEC really that good? |
| 14:28.49 | ManxPower | coppice: I assumed they did. They don't? |
| 14:29.22 | coppice | its GPL, so no they don't |
| 14:29.25 | ManxPower | Ah. |
| 14:29.37 | ManxPower | Did you write it? |
| 14:30.19 | coppice | David Rowe did the heavy lifting to make it robust |
| 14:30.23 | x86 | [TK]D-Fender: good point ;) |
| 14:30.29 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:30.40 | ManxPower | coppice: an endorsement from you IS pretty impressive. |
| 14:30.44 | BobLutz | Hello all! Information on a Asterisk-Dev kit please? |
| 14:30.45 | jsmith | coppice: No, we don't provice OSLEC... we provide the (commercial, I know) HPEC |
| 14:31.00 | ManxPower | I'll have to look at it when I'm running an Asterisk that it can work with. |
| 14:31.26 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 14:33.54 | x86 | 38 active channels :) |
| 14:34.37 | x86 | oh man this is awesome |
| 14:35.04 | dandre | yes [TK]D-Fender but my question was is there anyway to set the DID from an analog zap channel so that I could write something like |
| 14:35.04 | dandre | exten => s/123456,1,Dial(SIP/1) |
| 14:35.29 | [TK]D-Fender | dandre: that is not a DID. |
| 14:36.10 | Katty | exten => Katty'sNumber,1,Answer n,2,Dial(Zap/g1/misterfender)? |
| 14:36.22 | Katty | is that how a did works? |
| 14:36.31 | Katty | if i wanted to redirect it to you, minus the actual transfer. |
| 14:36.53 | [TK]D-Fender | Katty: DID's target an exten of the same number |
| 14:37.14 | Katty | so in sip.conf i'd have a [katty'sphonenumber] entry? |
| 14:37.19 | dandre | oups, sorry, read |
| 14:37.19 | dandre | exten => 123456,1,Dial(SIP/1) |
| 14:38.01 | ManxPower | There's nothing magical about DIDs and extensions. The part after the => and before the first , is the extension aka DID |
| 14:38.04 | Katty | [3334444] in sip.conf, then exten => 3334444,1,Dial(SIP/3334444)? |
| 14:38.13 | ManxPower | if that part is "s" then that is the "DID" |
| 14:40.35 | *** join/#asterisk simbol76ss (n=simbol@ip-212-26.sn1.eutelia.it) |
| 14:43.55 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:43.55 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:47.50 | x86 | ok, I've got an E&M wink T1 going from a Toshiba PBX to my Asterisk box... When calls come into Asterisk, the extension is set to "s", and the dnid is unknown... Is there a channel variable that will give me the number that the Toshiba is trying to call over the T1? |
| 14:47.56 | kamisalami | where do i need h323 for? |
| 14:48.16 | Katty | never used it. |
| 14:49.09 | kamisalami | http://www.voip-info.org/wiki/index.php?page=Asterisk+variables |
| 14:49.29 | kamisalami | look at the callerid ones |
| 14:49.32 | kamisalami | ? |
| 14:50.17 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:50.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:50.20 | ManxPower | no. Look at channelvariables.txt in the Asterisk source. The wiki is always out of date |
| 14:50.25 | Katty | anthm: good morning sunshine! |
| 14:50.31 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:50.32 | Katty | anthm: would you like some snow? we have plenty! |
| 14:50.42 | lmadsen | the wiki is good if you're like... running asterisk 1.0 |
| 14:50.45 | anthm | doh no thx |
| 14:50.47 | x86 | ManxPower: url me on that? |
| 14:50.57 | anthm | it's been feet high here for months |
| 14:50.57 | Katty | anthm: k, well, offer stands ;) |
| 14:50.58 | ManxPower | x86: asterisk is not GETTING the dialed numnber. If it did the call would not send the call to extension s |
| 14:51.01 | Katty | :< |
| 14:51.16 | Katty | anthm: we've just had ice storms here |
| 14:51.17 | x86 | ManxPower: sure it is, it's sending the call out the telco T1 just fine |
| 14:51.19 | ManxPower | x86: /path/to/src/asterisk/doc/channelvariables.txt |
| 14:51.31 | ManxPower | x86: Asterisk is not receiving the dialed digits. |
| 14:51.41 | anthm | it has to end soon right it's march! |
| 14:51.45 | x86 | not sure how it's knowing what number to dial out the telco T1 then |
| 14:51.50 | ManxPower | The DEFINITION of "s" is "asterisk did not receive dialed digits" |
| 14:52.01 | x86 | ManxPower: or call went to a macro? :) |
| 14:52.08 | lmadsen | 's' is not a 'catch-all' |
| 14:52.17 | ManxPower | x86: I meant when the call comes in. |
| 14:52.22 | lmadsen | it is exactly what ManxPower said |
| 14:52.31 | x86 | I realize this |
| 14:52.34 | BobLutz | Is it still possible to purchase an Asterisk-Dev kit? |
| 14:52.38 | ManxPower | If you are trying to troubleshoot without removing all your macros and other dialplan stuff then you don't deserve to get it fixed. |
| 14:52.53 | lmadsen | BobLutz: I think it's just the Asterisk Appliance now (it's out of beta / development) |
| 14:53.01 | BobLutz | :-/ |
| 14:53.09 | x86 | channel variables say extension is "s", but the call from the Toshiba PBX comes into Asterisk (Zap/g3) just fine, and is dialed out to the telco (Zap/g1) just fine |
| 14:53.33 | x86 | but if I run an AGI before placing the call to the telco interface, I can not find out what number is being called |
| 14:53.39 | ManxPower | x86: then something else is wrong. Simplify it, then paste the CLI outoput of a call to pastebin.ca |
| 14:54.10 | x86 | if it matters, I'm running E&M Wink on both interfaces |
| 14:54.41 | ManxPower | x86: If asterisk is not really getting the dialed digits then that might be a factor too. |
| 14:54.57 | x86 | ManxPower: how is it knowing what number to dial out the telco interface then? |
| 14:55.11 | ManxPower | x86: We will find out when I get that pastebin. |
| 14:55.25 | *** join/#asterisk angryuser (n=nononon@df01t2-213-44-80-96.d4.club-internet.fr) |
| 14:55.41 | ManxPower | x86: you've been here long enough to know there is a process to finding out what is happening and what is wrong. Let's try that process. |
| 14:56.14 | x86 | I never thought anything was "wrong" per se :) |
| 14:57.06 | x86 | in my AGI, if I dump all channel variables I don't see any at all |
| 14:57.19 | ManxPower | we are not talking about the AGI at this point. |
| 14:57.27 | ManxPower | we have much to find out before we get to that part |
| 14:58.34 | ManxPower | Is your question "why isn't asterisk getting the dialed digits" or is the question "why is my AGI not able to see the dialed digits"? |
| 14:59.06 | x86 | AGI not able to see them |
| 14:59.15 | x86 | not able to see any dialed digits though |
| 14:59.18 | ManxPower | Hell if I know. I would have to debug your AGI and I don't do that. |
| 14:59.27 | x86 | heh |
| 14:59.45 | ManxPower | But without a pastebin we are dead in the water anyway. Ask someone else to badger you until they get the info they need. That's too much work for me. |
| 15:00.06 | x86 | has something to do with this toshiba T1 interface... if I dial out of a local zap channel (phone) or a SIP channel connected to this asterisk box, the AGI works fine |
| 15:00.32 | ManxPower | pastebin.ca is harmless (it's Canadian after all!), there is no reason to be scared of it. |
| 15:01.08 | [TK]D-Fender | ManxPower: .com is faster, nicer looking, and jumps you to the linked paste immediately... it better now. |
| 15:01.36 | x86 | http://pastebin.ca/932134 |
| 15:01.36 | ManxPower | [TK]D-Fender: yeah, but you'll get branded a terrorist if you use it. |
| 15:01.54 | ManxPower | x86: I'm so proud of you! |
| 15:01.55 | x86 | pastebin.commie |
| 15:02.03 | x86 | ManxPower: spanx you!: ) |
| 15:02.08 | x86 | heh |
| 15:02.14 | BobLutz | Set(MY_VAR) <--- does this nullify? |
| 15:02.21 | ManxPower | -- Executing [s@macro-lddial:4] AGI("Zap/38-1", "callerid-test.agi") in new stack |
| 15:02.27 | x86 | right |
| 15:02.27 | Kobaz | nullification! |
| 15:02.30 | ManxPower | So you see what is wrong here, x86? |
| 15:02.35 | x86 | I'm jumping to a Macro :) |
| 15:02.38 | x86 | which I stated earlier |
| 15:02.39 | Katty | i'll nullify everyone here in a minute! |
| 15:02.44 | ManxPower | and the dialed number is a macro is....? |
| 15:02.46 | x86 | using the s exten |
| 15:02.48 | BobLutz | Whoa, easy there |
| 15:02.48 | ManxPower | "s" |
| 15:02.55 | ManxPower | so that is why you don't see the dialed numnber. |
| 15:02.55 | x86 | yep |
| 15:03.04 | ManxPower | x86: there's a pretty simple trick to handle that. |
| 15:03.08 | x86 | I want to know the original dialed number, which is stored in $ARG1 of that macro |
| 15:03.15 | x86 | ManxPower: cool! |
| 15:03.50 | ManxPower | exten => s,1,Goto(${MACRO_EXTEN,1} Then make the rest of your macro exten => _X.,1,whatever exten => _X.,n,whatever |
| 15:04.09 | ManxPower | your CDRs might look nicer as well. |
| 15:04.30 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 15:04.32 | x86 | hmm that's nifty |
| 15:04.34 | ManxPower | No! Based on that pastebin you are not passing ANY vars to the macro |
| 15:04.46 | ManxPower | ..er... sorry, let me go re-read the pastebin |
| 15:05.03 | ManxPower | so 9999 is the dialed number? |
| 15:05.05 | x86 | I can paste the macro if you like |
| 15:05.07 | *** join/#asterisk wmaulik (n=wmaulik@158.59.195.51) |
| 15:05.16 | x86 | 9999 is the account code, which is ARG2 in the macro |
| 15:05.22 | ManxPower | no, that is arg1 |
| 15:05.35 | ManxPower | Â -- Executing [17247442304@from-internal:3] Macro("Zap/38-1", "lddial|9999|7247442304") in new stack |
| 15:05.36 | *** join/#asterisk UVSoft (n=jnk467@80.254.48.215) |
| 15:05.36 | x86 | err you're right |
| 15:05.38 | UVSoft | hi there |
| 15:05.45 | x86 | i was mixxed up |
| 15:05.48 | ManxPower | see, the first arg is 999, the 2nd arg is 7247442304 |
| 15:05.53 | ManxPower | Is it mixed up in your code too? |
| 15:06.57 | x86 | nope |
| 15:06.58 | x86 | http://pastebin.ca/932143 |
| 15:07.00 | x86 | that's the macro |
| 15:07.05 | x86 | (with your modifications) |
| 15:07.25 | x86 | gah, you have typos ;) |
| 15:07.39 | BobLutz | Typo? |
| 15:07.44 | x86 | <PROTECTED> |
| 15:07.46 | x86 | fixed it |
| 15:07.50 | ManxPower | x86: *nod* |
| 15:08.01 | ManxPower | x86: it looks now like you don't really NEED to do that, but it doesn't hurt. |
| 15:08.41 | x86 | <PROTECTED> |
| 15:09.24 | x86 | yeah |
| 15:09.59 | *** join/#asterisk linxroute (n=linxrout@222.253.75.122) |
| 15:10.05 | linxroute | hi there |
| 15:10.28 | ManxPower | btw, I don't see ANYTHING wrong in that CLI pastebin, the call is dialing just fine |
| 15:10.30 | x86 | so why when I dial from a local zap channel (i.e. from a channel bank), or a SIP phone connected to this asterisk server, could I see the digits dialed in the DNID channel variable, but not when I call from the Toshiba T1 interface? |
| 15:10.39 | linxroute | i want to build a new * box |
| 15:10.40 | x86 | yeah, I know the call is dialing fine ;) |
| 15:10.44 | linxroute | from scratches |
| 15:10.48 | ManxPower | x86: Based on that pastebin you ARE seeing the dialed digits. |
| 15:10.49 | x86 | that was never the problem |
| 15:10.49 | linxroute | what OS |
| 15:10.55 | BobLutz | linxroute: NT |
| 15:10.55 | lmadsen | linxroute: what what do you know? |
| 15:10.56 | linxroute | do you guys recommend |
| 15:11.02 | ManxPower | Unless you didn't reproduce the problem in the pastebin. |
| 15:11.03 | x86 | ManxPower: right, never said i wasn't :) |
| 15:11.22 | linxroute | redhat ? |
| 15:11.24 | lmadsen | linxroute: as long as it's Linux, it doesn't matter what distro you use -- use the one you know the best |
| 15:11.40 | x86 | ManxPower: the problem is that I can't see the channel variables if I dial from the toshiba T1 interface, but I can if I dial from a local zap channel (phone) or a sip phone |
| 15:11.44 | linxroute | cost i'm only familiar with RH |
| 15:11.46 | BobLutz | linxroute: I use RH, its not my preferred distro *cough* Arch *cough* but it works |
| 15:11.52 | lmadsen | linxroute: then that's the one to use |
| 15:11.59 | ManxPower | linxroute: That's like asking what route to take to get to Los Angels, there are hundreds of routes, pick the one you know best. |
| 15:12.08 | lmadsen | linxroute: TFoT is built on CentOS 5, which is a RH style distro, so you could use that |
| 15:12.23 | lmadsen | linxroute: ^^^ what ManxPower said |
| 15:12.26 | ManxPower | x86: What that CLI pastebin a call coming from the Toshiba? |
| 15:12.26 | BobLutz | CentOS > RHES |
| 15:12.30 | linxroute | yes cos my friend wants to have some kind " GUI" |
| 15:12.42 | lmadsen | DO NOT RUN X WINDOWS ON AN ASTERISK BOX |
| 15:12.44 | x86 | ManxPower: yp |
| 15:12.45 | lmadsen | you're asking for trouble |
| 15:12.49 | linxroute | which i have not been using with asterisk for two years |
| 15:12.50 | x86 | ManxPower: yup |
| 15:12.57 | ManxPower | x86: I have no idea. |
| 15:13.05 | lmadsen | anyways, I need to do paid work now |
| 15:13.06 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:13.16 | ManxPower | but at least you understand extensions and call routing and macros a little better now. |
| 15:13.25 | linxroute | never mind, thanks for pointing out |
| 15:13.47 | x86 | ManxPower: had no problem with them before? :) |
| 15:13.53 | linxroute | by the way, anyone here used Oslec ? |
| 15:14.30 | nixguy | does exten => i,1,Playback(pbx-invalid) require pattern matching? |
| 15:15.05 | BobLutz | nixguy: no |
| 15:15.17 | [TK]D-Fender | nixguy: "i" is called because the user didn't MATCH another pattern. |
| 15:15.34 | BobLutz | nixguy: but if that is all that is in that extension "i" your call will probably hang up |
| 15:15.42 | BobLutz | after it plays back the sound |
| 15:15.53 | ManxPower | nixguy: it really only works from an IVR, not from a phone or line |
| 15:16.18 | linxroute | anyone here used Oslec ? |
| 15:16.24 | nixguy | [TK]D-Fender: ok cause i read on voip info that said the opposite of the manual |
| 15:16.28 | linxroute | i got a TDM400 |
| 15:16.32 | [TK]D-Fender | linxroute: Probably jsut as many people as the last time you asked. |
| 15:16.34 | nixguy | http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension |
| 15:16.36 | ManxPower | nixguy: the manual is correct. |
| 15:16.47 | nixguy | ok |
| 15:16.49 | linxroute | ? |
| 15:16.52 | linxroute | man |
| 15:16.52 | nixguy | i will dig some more then thnx. |
| 15:17.05 | linxroute | you always the same |
| 15:17.12 | ManxPower | nixguy: that is not at all what it says. |
| 15:17.19 | BobLutz | Does anyone know of a recently published book on Postgresql? |
| 15:17.21 | ManxPower | In fact it pretty much says exactly what I just said. |
| 15:18.03 | [TK]D-Fender | nixguy: I am correct in the context that it only for use with IVR's |
| 15:18.09 | nixguy | ahh ok |
| 15:18.11 | nixguy | i understand |
| 15:18.23 | [TK]D-Fender | nixguy: That will not catch any other kind of call. Those will "404". |
| 15:18.48 | nixguy | who do you guys do, do you match those calls? |
| 15:18.50 | [TK]D-Fender | nixguy: Which is why they show a wildcard match like "_." (EVIL!). Avoid unless absolutely necessary |
| 15:18.52 | x86 | ManxPower: got it figured out :) |
| 15:18.54 | *** join/#asterisk jbigbee (n=jbigbee@216.207.245.1) |
| 15:19.04 | linxroute | anyone here used Oslec ? |
| 15:19.10 | nixguy | [TK]D-Fender: thats why im asking i've read _. is bad for you so i wanna avoid it :| |
| 15:19.17 | BobLutz | linxroute: NOOOOOOOOOOOOOOOOO |
| 15:19.23 | [TK]D-Fender | nixguy: Waht are you trying to do? |
| 15:19.39 | [TK]D-Fender | linxroute: "unload chan_brokenrecord.so" |
| 15:19.40 | nixguy | just want invalid numbers to say invalid number basically when you call from our sip phones |
| 15:19.47 | linxroute | oh you dont |
| 15:19.54 | BobLutz | lol |
| 15:19.58 | linxroute | but some others might |
| 15:20.01 | nixguy | instead of a strange beeping sound that everyone will interpret as a faulty pbx here.. |
| 15:20.07 | linxroute | wtf |
| 15:20.27 | [TK]D-Fender | nixguy: Then that wildcard is what you'll need to do. make the first priority jump OUT into another context to actaully play the message. |
| 15:20.51 | [TK]D-Fender | linxroute: Are you seriously expecting like all 200 of us to waste our time to personally tell you "no"? That would flood the channel |
| 15:21.04 | ManxPower | nixguy: then you want _X. |
| 15:21.10 | [TK]D-Fender | linxroute: If someone has something to say, they will say it. Please don't badger. |
| 15:21.13 | ManxPower | exten i does not work unless you are in an IVR |
| 15:21.32 | [TK]D-Fender | nixguy: What you actually want to do is "_X!" |
| 15:21.49 | linxroute | i just looking for someone who might have the same problems like me |
| 15:21.56 | linxroute | if you dont bother |
| 15:21.57 | nixguy | [TK]D-Fender: ok |
| 15:22.06 | BobLutz | ~google |
| 15:22.06 | jbot | hmm... google is a search engine found at http://www.google.com/ |
| 15:22.15 | BobLutz | linxroute: ^ |
| 15:22.21 | linxroute | yeah i know that |
| 15:22.24 | nixguy | ManxPower: k thnx guys i will dig into it! |
| 15:22.36 | linxroute | but still got some errors |
| 15:22.44 | [TK]D-Fender | linxroute: You didn't even until just now even say you had tried it, let alone failed. And then compound things by not even describing anything useful about your problem. |
| 15:23.16 | linxroute | i need to know anyone used it first |
| 15:23.23 | linxroute | then i might ask |
| 15:23.26 | linxroute | right |
| 15:23.52 | coppice | repeating yourself gets annoying |
| 15:23.52 | *** join/#asterisk FabiOne (n=FabiOne@host107-144-static.59-88-b.business.telecomitalia.it) |
| 15:23.56 | FabiOne | hi all |
| 15:23.57 | coppice | repeating yourself gets annoying |
| 15:24.01 | *** join/#asterisk b11d|bbl (n=no@234-200-29-134.hcc.mnscu.edu) |
| 15:24.06 | coppice | repeating yourself gets annoying |
| 15:24.13 | coppice | repeating yourself gets annoying |
| 15:24.15 | FabiOne | my asterisk box tell me "What is wrong with you? You cannot use cause 34 number when in state 7!" |
| 15:24.15 | b11d|bbl | . |
| 15:24.20 | FabiOne | what's?? |
| 15:24.31 | hmmhesays | its freaking stupid read doesn't take multiple file names for playback |
| 15:25.52 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:26.16 | Qwell | hmmhesays: doesn't it? |
| 15:26.19 | Qwell | pretty sure it does |
| 15:26.38 | Zeeek | hay |
| 15:26.44 | Qwell | Read(variable[,filename[&filename2...]][,maxdigits][,option][,attempts][,timeout]) |
| 15:28.41 | *** join/#asterisk m160858 (n=m160858@mailserver2.kruger.com.ec) |
| 15:29.16 | Zeeek | so I hear Digium now delivers the g729 licenses by automatic email. Cool! |
| 15:29.29 | Zeeek | In my day, you had to wait for a human to answer ;) |
| 15:29.57 | Qwell | did they even have email in your day, Zeeek ? :) |
| 15:30.07 | BobLutz | Teehee |
| 15:30.08 | Zeeek | now that you mention it... I don't believe they did |
| 15:30.25 | Zeeek | didn't need asterisk, we used ham walkie-talkies |
| 15:30.56 | Zeeek | so here's my question: can you NOT hear the "thank you" file when the user hits # in voicemail app? |
| 15:31.03 | BobLutz | Zeeek: In my day we had to use smoke signals |
| 15:31.13 | Zeeek | We didn't have fire yet so that would have been hard |
| 15:31.23 | BobLutz | Zeeek: touchette! |
| 15:31.28 | [TK]D-Fender | My computer still gives me smoke signals... thats why I keep the extinguisher handy... |
| 15:31.33 | Qwell | Zeeek: carrier pterodactyls? |
| 15:32.04 | Zeeek | Would anyway care for a dark beer? Or perhaps a Bark Deer? |
| 15:32.54 | Zeeek | nice thing about living abroad, you ALWAYS drink imported beer |
| 15:32.55 | *** join/#asterisk ccvp (n=ax@66.0.46.210) |
| 15:33.55 | Zeeek | what about this "thank you" prompt in Voicemail() ? Can we get rid of it? |
| 15:34.53 | Zeeek | am I on that many ignore lists now? Sheesh :) |
| 15:35.31 | [TK]D-Fender | Zeeek: You've got the source... get busy... |
| 15:35.42 | Zeeek | It's not for me |
| 15:35.52 | ccvp | heh, when did Ubuntu take over "fedoras" roll as the preferred "masses" distro, meaning newbies, beginners, and some more advanced users? |
| 15:35.53 | Zeeek | But I wasn't sure it could be done without source? |
| 15:35.56 | ccvp | 1-2 years ago? or more like 2-3? |
| 15:35.59 | lnx | can you help me what does this string mean in CLI history? M\222\374\267\200x\375\267\240\233\362\267Y^D |
| 15:35.59 | ccvp | role |
| 15:36.03 | [TK]D-Fender | Zeeek: Message is the same, go feth the pteradactyl |
| 15:36.07 | [TK]D-Fender | fetch |
| 15:36.12 | Qwell | ccvp: after FC2 when Red Hat decided to change everything |
| 15:36.38 | [TK]D-Fender | ccvp: Maybe 2 years now |
| 15:36.51 | ccvp | heh |
| 15:36.54 | Zeeek | see I have a friend... who has installed asterisk... |
| 15:37.00 | ccvp | i barely remember feeling a sentiment |
| 15:37.05 | ccvp | that fedora was "where it's at" |
| 15:37.07 | Zeeek | and this... friend... wants to ditch the "thank you" |
| 15:37.09 | ccvp | so what's FC like these days? |
| 15:37.20 | [TK]D-Fender | Zeeek: The answer is not changing. |
| 15:37.29 | Zeeek | damn I thought it might |
| 15:37.42 | FabiOne | i've a problem with long call with * 1.4 |
| 15:37.51 | Zeeek | he's already intot he source, that's basically what I told him because I don't care about the "thank you" |
| 15:38.25 | FabiOne | suddenly the line mute afer about 25 mins. of call |
| 15:38.36 | FabiOne | in console i see |
| 15:38.37 | FabiOne | What is wrong with you? You cannot use cause 34 number when in state 7! |
| 15:38.44 | Zeeek | Incidentally, if anyone wants to download the funny answering machine files, you can find them on the end of this feed: |
| 15:38.46 | Zeeek | http://recordings.talkshoe.com/rss22622.xml |
| 15:38.48 | FabiOne | vith verbosity 999 |
| 15:38.54 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 15:39.13 | *** join/#asterisk fus10nx (n=fus10nx@140.239.27.40) |
| 15:39.50 | Zeeek | the best error message remains, "we should never get here" |
| 15:41.21 | apocn | Hello, Im making an AGI script (in Perl), when I use a for(...){ $AGI->get_data(...); } the get_data works after the first iteration. why is that? |
| 15:41.45 | *** join/#asterisk SteveTotaro (n=root@pool-71-166-126-209.bltmmd.east.verizon.net) |
| 15:43.48 | *** join/#asterisk jmacz (n=jmacz@190.90.35.212) |
| 15:44.58 | ccvp | Linux is nothing more then a fad. The same reason people goto starbucks, to look metrosexual in the morning before they goto work, so they can sip their preppy coffee cups, and have a "skinny dolce late"- - - Linux is of the same ilk. I use linux to look hip/metrosexual, because it's not Windows. aka: for attention, because it's free. The only legitimatecy of linux is: math/engineer students, and cheap companies who are poor. |
| 15:44.58 | ccvp | heh |
| 15:45.51 | Zeeek | I see he asked on the ML and geot a few answers |
| 15:46.39 | [TK]D-Fender | ccvp: Plenty of good reasons to use Linux. |
| 15:47.09 | apocn | [TK]D-Fender, can you help me? |
| 15:47.19 | [TK]D-Fender | apocn: Possibly. |
| 15:47.27 | Zeeek | Linux. In my day they didn't have linux |
| 15:47.35 | nixguy | ccvp: wrong |
| 15:47.38 | Zeeek | we used CPM and got on with our lives |
| 15:47.42 | nixguy | but this is not a discussion for this channel |
| 15:47.46 | [TK]D-Fender | apocn: nice big pastebin and we'll see. |
| 15:47.47 | jblack | Guys, he's trolling. |
| 15:47.48 | jsmith | Zeeek: All hail CP/M! |
| 15:47.56 | apocn | [TK]D-Fender, ok |
| 15:48.11 | jsmith | Zeeek: The "Z" in Zeeek stands for Z80, right? |
| 15:48.14 | Zeeek | I thought OS/2 was gonna rule |
| 15:48.19 | Zeeek | Of course |
| 15:48.24 | Zeeek | as in TRS-80 |
| 15:48.28 | ccvp | heh |
| 15:48.37 | Zeeek | ADD A,13 ; add 13 to A |
| 15:48.48 | Zeeek | comment champion |
| 15:48.49 | ccvp | jblack, its sarcasm, i use linux, but why don't we see it dominating |
| 15:48.50 | ccvp | the world |
| 15:49.07 | ccvp | the first step is china going non-MS in all gov't facilities, migrating to forms of slackware. |
| 15:49.10 | nixguy | simply because linux is not the singe answer to every problem |
| 15:49.19 | jblack | Inertia. |
| 15:49.19 | jsmith | Zeeek: Of course, I was an odd-ball -- I was running CP/M on a Commodore 128 |
| 15:49.19 | Zeeek | it might if MS gets kicked out of enough countries |
| 15:49.31 | jsmith | Zeeek: (my first dual-boot machine!) |
| 15:49.33 | Zeeek | yes, inertia |
| 15:49.44 | Zeeek | jsmith wow, that is advanced |
| 15:50.07 | jsmith | Zeeek: Well, that's my style -- use odd hardware, do cool things... |
| 15:50.10 | jblack | a 128? You spoiled brat! |
| 15:50.20 | Zeeek | I'm on a dual boot right now but I can only get X to work every two or three boots |
| 15:50.22 | jsmith | jblack: I know... I had to upgrade from the 64 |
| 15:50.36 | Zeeek | I had the "full" 48 K |
| 15:50.38 | ccvp | jblack, when we went to beijing last year to put call manager in 8 satelite offices, alot of the gov't reps we spoke to say that within 10 years |
| 15:50.42 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:50.42 | jsmith | Zeeek: I'm on a dual-boot right now (running Linux on a PowerBook with a PPC processor) |
| 15:50.46 | ccvp | none of their systems will be using Microsoft software. |
| 15:51.00 | jblack | ccvp: And? |
| 15:51.08 | Zeeek | I haven't bothered with dual on the MacBook. OS X is really good |
| 15:51.10 | *** join/#asterisk lftsy (n=lftsy@121.194.210.62.te-dns.org) |
| 15:51.15 | ccvp | it's a long term social plan, to ruin the american economy |
| 15:51.20 | ccvp | and make america less dominating |
| 15:51.23 | Zeeek | I wish I could run it on my other PC hardware |
| 15:51.37 | ccvp | and marginalize U.S companies who dominate the world economy |
| 15:51.40 | apocn | [TK]D-Fender, http://pastebin.com/m6dfc9553 |
| 15:51.42 | Qwell | Zeeek: people have been known to do that successfully |
| 15:51.50 | Zeeek | I think there's a serious chance that the MS inertia will die down worldwide |
| 15:52.13 | jblack | Don't take this wrong, but I'm not in the mood for your shit this morning. Can't you go off to a rest home and kick old people in the knees or something? |
| 15:52.18 | Zeeek | Qwell yeah, I've heard that. It reall is good IMO. When you want a prompt, the console is there |
| 15:52.27 | ccvp | jblack, just goto the pharmacy and take some zoloft/paxil |
| 15:52.30 | ccvp | and get some temporary seratonin :) |
| 15:52.33 | Qwell | Zeeek: I need to get a macbook some day |
| 15:52.43 | Zeeek | they rock, no shit |
| 15:52.43 | [TK]D-Fender | apocn: MORE please... |
| 15:52.45 | Qwell | I'd use osx over linux |
| 15:52.56 | apocn | [TK]D-Fender, this is the only thing Im executing now, nothing else after that |
| 15:52.59 | Zeeek | Qwell heh, rabble rouser |
| 15:53.11 | apocn | just the file header, including the module, using stricts, etc... |
| 15:53.18 | [TK]D-Fender | apocn: .... MORE CODE, and full CLI |
| 15:53.34 | ccvp | qwell........ |
| 15:53.43 | Qwell | ccvp......... |
| 15:53.46 | ccvp | don't even tell me your an apple fanboi, iPhone strapped to the side, instead of a blackberry |
| 15:53.49 | ccvp | macbook Air |
| 15:53.50 | Zeeek | OS X boots in like 40 seconds. It's faster than all my linux boxes without X |
| 15:53.52 | ccvp | etc :) |
| 15:54.07 | Qwell | ccvp: The last apple product I owned was 20 years ago. |
| 15:54.10 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 15:54.11 | apocn | [TK]D-Fender, http://pastebin.com/m455ba05b |
| 15:54.12 | drmessano | ccvp, need we remind you that are a Cisco fanboy? How can that not be the ultimate fail? |
| 15:54.13 | Qwell | I'm 25 now. |
| 15:54.24 | jblack | Booting? You have to boot your computer often enough that you know how long it takes? |
| 15:54.32 | Zeeek | 25. God. Just past Quarter life. |
| 15:54.39 | ccvp | qwell, just think |
| 15:54.41 | ccvp | in 5 years your 30 |
| 15:54.43 | ccvp | :) |
| 15:55.03 | Zeeek | One does boot a laptop daily rather than leave it on all the time |
| 15:55.18 | jblack | qwell: So, what are you going to do with the 18k days or so that you have left to live? |
| 15:55.23 | ccvp | so whose the oldest person in here? anyone over 50, 60? |
| 15:55.29 | Zeeek | I've got one foot in the grave compared to all you young'uns |
| 15:55.37 | Qwell | ccvp: Zeeek, no doubt |
| 15:55.47 | Zeeek | ccvp I am definitely the oldest person here |
| 15:55.52 | jblack | Zeeek: Bah. |
| 15:55.54 | ccvp | 70+ ? |
| 15:55.55 | Zeeek | and everywhere else |
| 15:56.03 | apocn | [TK]D-Fender, here it is: http://pastebin.com/m455ba05b |
| 15:56.05 | jblack | I've got you beat by a country mile. |
| 15:56.07 | Zeeek | I feel 90+ |
| 15:56.17 | *** join/#asterisk monkeytype (n=Bill@216.207.245.1) |
| 15:56.21 | ccvp | don't feel old because |
| 15:56.28 | ccvp | a 107 yr old wwI vet just visited bush |
| 15:56.30 | ccvp | in the white house :) |
| 15:56.46 | drmessano | Did Cheney shoot him? |
| 15:56.47 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:56.50 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:56.59 | Zeeek | I wish the Bush and White House were never in the same sentence |
| 15:57.00 | *** join/#asterisk bkw_ (n=brian@ppp-70-128-106-102.dsl.tulsok.swbell.net) |
| 15:57.07 | Zeeek | drmessano muhahaha |
| 15:57.11 | jameswf-home | ~bush |
| 15:57.12 | jbot | ooooh Bush me no likey bring on the tile baby... wanna borrow my razor.... |
| 15:57.22 | ccvp | well, if you people elect mccain |
| 15:57.26 | ccvp | it's just another 4 years of bush |
| 15:57.29 | ccvp | mccain = bush 2.0 |
| 15:57.30 | Zeeek | naw |
| 15:57.34 | Zeeek | no way |
| 15:57.36 | Zeeek | but we won't |
| 15:57.41 | jameswf-home | Paul/Nader 08 |
| 15:57.42 | Zeeek | seeing as Obama will win |
| 15:58.08 | Zeeek | Nader already made his contribution by helping elect the village idiot |
| 15:58.16 | Zeeek | now he should go away for good |
| 15:58.31 | ccvp | heh, nader |
| 15:58.31 | Zeeek | but we digress... |
| 15:58.34 | ccvp | he needs to fade away |
| 15:58.37 | drmessano | My predicition, the Digg militia will declare martial law and put ron paul in office |
| 15:58.41 | Zeeek | self-promo, nothing more |
| 15:58.59 | Zeeek | drmessano you are on today! Be sure to talk on the conference! |
| 15:59.06 | ccvp | I've generally come to the conclusion that most technology addicts, internet addicts, irc addicts etc |
| 15:59.07 | ManxPower | Cthulu 2008! Why settle for the lesser of two evils? |
| 15:59.10 | ccvp | tilt to the democratic side of things |
| 15:59.14 | drmessano | Nader needs to go back to checking seatbelts |
| 15:59.16 | [TK]D-Fender | ManxPower: LOL!! |
| 15:59.18 | ccvp | so I'm safe to assume that 95% of this room either will vote for |
| 15:59.22 | ccvp | obama or hillary :) |
| 15:59.27 | jameswf-home | no |
| 15:59.36 | [TK]D-Fender | ManxPower++ |
| 15:59.39 | Zeeek | hillary is jhust another pol in a skirt |
| 15:59.45 | drmessano | Ubuntu for President |
| 15:59.48 | jameswf-home | I will vote for Mickey Mouse |
| 15:59.51 | drmessano | Says Digg |
| 15:59.57 | Zeeek | you already have mickey mouse |
| 16:00.08 | ManxPower | My take on it is the either bama or hillary would make a good president. But bama would try more radical ideas like civil liberties, and stuff like that. |
| 16:00.16 | drmessano | I'm voting for Ubuntu... It can ran everything, including a country |
| 16:00.28 | [TK]D-Fender | apocn: FULL CLI OUTPUT with AGI debug, etc. |
| 16:00.29 | *** join/#asterisk sacitec (n=tobi@189.149.206.58) |
| 16:00.30 | drmessano | run |
| 16:00.42 | ManxPower | So the question is do you want the anti-christ (McCain), the idealist (bama) or the realist (hillary) |
| 16:00.50 | drmessano | If wonder if [TK]D-Fender uses that line on chicks |
| 16:00.59 | jameswf-home | Its 3am something has gone wrong in the world, the phone rings... Its an IVR and they want you to press 1 for spanish press 2 for english WTF |
| 16:01.13 | Zeeek | I want to feel good about the US in the world. The only choice is Obama for that |
| 16:01.23 | *** join/#asterisk _mre|666 (i=muffy@gateway/tor/x-3cbfd1a7cc09a4e1) |
| 16:01.24 | Zeeek | It sends a message |
| 16:01.32 | Zeeek | But we digress... |
| 16:01.33 | drmessano | SomeChick: You looking for some fun? [TK]D-Fender: FULL CLI OUTPUT with AGI debug, etc, FIRST |
| 16:01.42 | *** part/#asterisk _mre|666 (i=muffy@gateway/tor/x-3cbfd1a7cc09a4e1) |
| 16:01.45 | ManxPower | I thing racism is too rampant for 'bama to be elected. |
| 16:01.47 | ccvp | zeeek, obama has no foreign policy experience |
| 16:01.48 | ccvp | heh |
| 16:01.51 | jameswf-home | liberial facism |
| 16:01.53 | *** part/#asterisk BlunderBuss (n=BlunderB@65-123-225-213.dia.static.qwest.net) |
| 16:01.54 | ccvp | his only experience is "giving speeches" |
| 16:01.55 | ccvp | :) |
| 16:01.57 | jameswf-home | hmm thats what we want |
| 16:01.59 | drmessano | Racism won't matter |
| 16:02.00 | drmessano | Well |
| 16:02.06 | drmessano | It will, but not enough |
| 16:02.19 | ManxPower | drmessano: you don't live in the southern usa, do you? |
| 16:02.20 | Zeeek | ccvp I absolutely don't care. BUSH has never been anywhere and he and the gang of thugs have literally screwed up the whole world |
| 16:02.22 | Qwell | this is incredibly offtopic... |
| 16:02.29 | Zeeek | there's a topic? |
| 16:02.35 | Qwell | Zeeek: sharks |
| 16:02.39 | drmessano | ManxPower, I live more in the Southern USA than most.. so give me a break |
| 16:02.40 | Zeeek | oh |
| 16:02.44 | jameswf-home | ~topic |
| 16:02.47 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 16:03.02 | ManxPower | drmessano: I live near Birmingham, AL |
| 16:03.04 | [TK]D-Fender | Qwell: Yes, but its Friday and there really isn't anything else going on and it hasn't gotten too rowdy... |
| 16:03.10 | Zeeek | COme one come all: http://VoipUsersConference.org TODAY at 12 Noon EST |
| 16:03.14 | ManxPower | doesnt' get much more "south" than that from a cultural standpoint |
| 16:03.16 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 16:03.22 | drmessano | Georgia and Mississippi are the worst |
| 16:03.31 | Zeeek | You can hear all about some great new hardware |
| 16:03.39 | Zeeek | Asterisk related hardware |
| 16:03.50 | Zeeek | You likey asterisk, yes? |
| 16:03.58 | drmessano | Wuts Asterisk? |
| 16:04.06 | apocn | [TK]D-Fender, http://pastebin.com/m28d7412e -> Full AGI Output (with AGI debug) |
| 16:04.07 | BobLutz | Does anyone here use Asterisk? |
| 16:04.08 | drmessano | Im just here for the snack tray |
| 16:04.13 | Zeeek | It's that stuff that talks to X-Lite |
| 16:04.29 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:04.30 | agx | any idea how can i transform a 2-way call to a 3-way conference inviting another people to join it ? |
| 16:04.33 | drmessano | Oh, Asterisk is an ISP then.. I see.. |
| 16:04.41 | Zeeek | Snack tray? We have a full wet bar in the Asterisk^H^H^H VOIP Users Conference |
| 16:04.53 | drmessano | lol |
| 16:05.04 | Zeeek | TODAY only: free 6.5% dark beer |
| 16:05.17 | Zeeek | unless I finish the last one |
| 16:05.22 | drmessano | Sounds to me like I better get on the road.. I gotta run two errands and head to MY LAST DAY AT WORK YAY!!!! |
| 16:05.26 | dandre | I use the meetme application with I option to announce user join but the user oin is played twice. Why? |
| 16:05.33 | Zeeek | oh jameswf-home |
| 16:05.39 | Qwell | drmessano: giving up your practice? |
| 16:05.52 | [TK]D-Fender | apocn: Why don't those sound file name match? |
| 16:05.54 | drmessano | Giving up the practice of being paid shitty money |
| 16:05.56 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 16:06.01 | drmessano | LAST DAY! |
| 16:06.02 | *** join/#asterisk scoates (n=sean@iconoclast.caedmon.net) |
| 16:06.07 | scoates | hi |
| 16:06.12 | Zeeek | drmessano gonna live on the system? |
| 16:06.14 | apocn | [TK]D-Fender, I changed them now |
| 16:06.24 | drmessano | No, new job |
| 16:06.24 | apocn | (renamed them) |
| 16:06.33 | drmessano | Where thet PAY me REAL MONEY to do IT work |
| 16:06.35 | ccvp | ?msg #Politics Michelle Obama even said that if her husband is elected, she's going to walk the hallways of the Westwing in the whitehouse, and take down all of the old original oil paintings of the founding fathers, jefferson, washington etc, and put up paintings of african tribal leaders, MLK, kunta kinte, jesse jackson, etc.. and she says she isn't radical. She said she wants to re-heritage'ize the white house decorations/history. |
| 16:06.37 | ccvp | bleh |
| 16:06.38 | [TK]D-Fender | agx: You can't invite people to a 3-way call. You can transfer your current call to a MeetMe conference, join it yourself and wait for others to join however |
| 16:06.48 | drmessano | ccvp stop trolling |
| 16:06.51 | [TK]D-Fender | apocn: Show me something consistent. |
| 16:06.59 | scoates | I just installed ztdummy (because I need it for MeetMe), and that killed my *'s ability to play audio. It can still receive commands (DTMF), but none of the audio plays. Ideas, please? |
| 16:07.00 | drmessano | You keep having "Accidents" |
| 16:07.04 | apocn | [TK]D-Fender, this is consistent, just ignore the sound file names |
| 16:07.24 | apocn | what Im trying to figure out is why this is executing the first time and not waiting for user interaction |
| 16:07.35 | apocn | it works the second iteration but not the first, why is this? |
| 16:07.39 | apocn | that is my question |
| 16:07.45 | tzafrir_home | scoates, what is the output of 'zttest' |
| 16:07.52 | tzafrir_home | anything at all? |
| 16:08.15 | ManxPower | scoates: that is a common issue, look in the mailing list archives, there is something weird about your kernel |
| 16:08.20 | scoates | Opened pseudo zap interface, measuring accuracy... (then it stalls) |
| 16:08.29 | Zeeek | tzafrir_home hi |
| 16:08.33 | scoates | ManxPower: wonderful.. )-: any place specific to look? |
| 16:08.36 | tzafrir_home | Zeeek, hi |
| 16:08.54 | ManxPower | scoates: it has something to do with the kernel Hz |
| 16:09.04 | ManxPower | Supposed to be 1000 or 1024 and it's something really low |
| 16:09.13 | scoates | I meant URL(s) |
| 16:09.15 | ManxPower | I never use ztdummy, but I see the issue on the mailing list and here. |
| 16:09.20 | ManxPower | ~mailinglist |
| 16:09.21 | jbot | [~mailinglist] Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives |
| 16:09.28 | scoates | thanks |
| 16:09.30 | ManxPower | what crap is that. |
| 16:09.31 | Zeeek | ~Zeeek |
| 16:09.32 | jbot | somebody said zeeek was someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 16:09.41 | Zeeek | They should date those |
| 16:09.48 | ManxPower | search the mailing lists by prepending site:lists.digium.com to your google search. |
| 16:10.29 | apocn | [TK]D-Fender, there you go then: http://pastebin.com/m9eadf35 |
| 16:11.22 | scoates | found this, but no resolution: http://bugs.digium.com/view.php?id=11153 |
| 16:11.42 | Zeeek | Qiuck, name you favorite SIP or IAX provider |
| 16:11.46 | ManxPower | scoates: you saw my instructions on searching the mailing list? |
| 16:11.57 | ManxPower | Zeeek: Teliax and Vitelity |
| 16:12.00 | scoates | ManxPower: yep; that's what turned up the bug (-: |
| 16:12.05 | Zeeek | Quick, name you favorite hardware provider that works with asterisk |
| 16:12.17 | ManxPower | scoates: the problem is with your LINUX KERNEL |
| 16:12.33 | ManxPower | SIPuea and Polycom |
| 16:12.34 | scoates | yes, thanks. I understand that, but that doesn't tell me how to fix it |
| 16:12.38 | Zeeek | Teliax rocks, never tried the other but have heard ot it |
| 16:12.50 | ManxPower | scoates: you recompile your kernel |
| 16:12.58 | ManxPower | look for "kernel ticks" |
| 16:13.07 | hmmhesays | I love how malformed IF statments segfault asterisk every time |
| 16:13.13 | Zeeek | I used to like recompiling FreeBSD kernels. Never got into it with linux though |
| 16:13.16 | [TK]D-Fender | apocn: Novel though, NoOp your counter at the start of your loop |
| 16:13.34 | Zeeek | so should I eat the chocolate before or after the beer |
| 16:13.43 | apocn | [TK]D-Fender, ok, I will do it again and show you the output |
| 16:14.33 | SteveTotaro | what's up Mr. Messano? |
| 16:14.46 | *** join/#asterisk FciSoft (n=FabiOne@host107-144-static.59-88-b.business.telecomitalia.it) |
| 16:14.49 | Zeeek | Thats' Doctor |
| 16:15.31 | *** join/#asterisk juanjoc (n=juanjoc@host239.190-31-235.telecom.net.ar) |
| 16:15.37 | *** join/#asterisk FabiOne (n=FabiOne@host107-144-static.59-88-b.business.telecomitalia.it) |
| 16:15.44 | FabiOne | sorry |
| 16:15.47 | SteveTotaro | agx you can use app_bridge to bridge any channels to anywhere pretty much |
| 16:15.55 | [TK]D-Fender | ~drmessano |
| 16:15.56 | jbot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway |
| 16:16.10 | [TK]D-Fender | lol |
| 16:16.10 | apocn | [TK]D-Fender, http://pastebin.com/d3e1aeda3 |
| 16:18.03 | XnOSX | hey friends! anybody here can tell me about a good FXO gateway with 24 ports? |
| 16:18.18 | juanjoc | file: The fix for ticket #12148 worked partially. We're having audio drops that are longer in time but not as frequent. And we're seeing a new error message in the log:[Mar 7 11:15:05] WARNING[31326]: rtp.c:2047 ast_rtp_stop: Unable to cancel schedule ID 0. This is probably a bug (rtp.c: ast_rtp_stop, line 2047). |
| 16:18.18 | dandre | I don't understand why I get a user annouce twice when join a meetme root |
| 16:18.32 | scoates | ... is there a way to run meetme without ztdummy? |
| 16:18.44 | [TK]D-Fender | XnOSX: You know PRI is advised at that point I hope.... |
| 16:19.11 | [TK]D-Fender | scoates: Get a hardware timer |
| 16:19.20 | scoates | not an option |
| 16:19.39 | [TK]D-Fender | apocn: Ok, I'm at a loss, but you should now have enough info handy that someone who knows perl can help you out. |
| 16:19.49 | [TK]D-Fender | scoates: use app_conference instead |
| 16:20.06 | scoates | [TK]D-Fender: looking at that now, thanks. Does it work well? |
| 16:20.27 | apocn | [TK]D-Fender, well I dont think its a perl problem |
| 16:20.41 | [TK]D-Fender | scoates: Decently apparently |
| 16:20.50 | scoates | cool; thanks. downloading now. |
| 16:21.00 | apocn | maybe Asterisk::AGI module doesnt work too well then. |
| 16:21.06 | apocn | I'll use another language to program this |
| 16:21.19 | apocn | and see if that's the problem |
| 16:21.24 | *** join/#asterisk UserReg_CL (n=COB@95-41-246-201.adsl.terra.cl) |
| 16:21.39 | docelmo | Anyone in here from the UK? or can offer up a couple good providers in the UK for termination? |
| 16:21.42 | *** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 16:22.55 | Zeeek | voiptalk works well but is expensive |
| 16:23.04 | Zeeek | IMO |
| 16:23.19 | UserReg_CL | hi... need install QoS (have debian)... thank |
| 16:23.32 | Zeeek | docelmo high or low volume? |
| 16:23.42 | Zeeek | http://ideasip.com works very well |
| 16:23.43 | *** join/#asterisk mattx86 (n=matt@static2073.uctnwd.ken-tennwireless.com) |
| 16:23.49 | Zeeek | all over europe |
| 16:24.44 | mattx86 | any pointers for getting Asterisk+SIP+NAT to work? my only problem is there's no audio |
| 16:25.06 | Zeeek | yes, forward ports |
| 16:25.17 | Zeeek | 5060 and 10000-10020 |
| 16:25.24 | mattx86 | without that, if possible |
| 16:25.27 | Zeeek | you'll never look back |
| 16:25.34 | Zeeek | remove NAT |
| 16:25.41 | mattx86 | but what if I need another phone? |
| 16:25.47 | Zeeek | didn't I hear that 1.4 uses STUN? |
| 16:25.48 | TJNII | ~sipnat |
| 16:25.49 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:26.05 | docelmo | Did I actually see someone ask earlier if we all use asterisk in this channel? |
| 16:26.09 | docelmo | was he slapped? |
| 16:26.14 | Zeeek | no |
| 16:26.25 | BobLutz | docelmo: Its called sarcasm, have you heard of it? |
| 16:26.31 | JT | mattx86: are you just trying to access an ITSP with asterisk behind nat? |
| 16:26.32 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 16:27.11 | Zeeek | I'm behind NAT everywhere and I haven't had problems |
| 16:27.17 | mattx86 | JT: ITSP? I don't know what that is. |
| 16:27.24 | Zeeek | service provider |
| 16:27.33 | Zeeek | as opposed to phones registering |
| 16:27.35 | docelmo | BobLutz Nope.. Im more of a straight line guy.. Middle of the road.. |
| 16:27.38 | JT | ~itsp |
| 16:27.38 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 16:28.21 | mattx86 | JT: I'm trying to do: SIP phone -> (NAT/Router) -> our public Asterisk box |
| 16:28.42 | JT | so the sip phones are over the Internet? |
| 16:29.13 | mattx86 | JT: kind of; not necessarily the internet, because it's basically in the same building |
| 16:29.29 | mattx86 | JT: public IP's between the NAT/Router and Asterisk box though |
| 16:29.31 | Zeeek | shouldn't be a problem |
| 16:29.40 | SteveTotaro | some people use openpbx or freeswitch |
| 16:29.43 | Zeeek | use STUn on the phones themselves if need be |
| 16:29.46 | _ShrikE | Zeeek: PIKA will be on the conference today? |
| 16:29.50 | Zeeek | yes |
| 16:29.59 | Zeeek | http://voipusersconference.org/slides/ |
| 16:30.01 | _ShrikE | that should be interesting |
| 16:30.06 | Qwell | they have a presentation? |
| 16:30.08 | Zeeek | indeed it should |
| 16:30.15 | JT | mattx86: so is asterisk on a public ip? |
| 16:30.17 | Zeeek | see above URL |
| 16:30.48 | mattx86 | Zeeek: *shrug* already tried STUN. one way or another - I've gotten it to ring out no problem. but there's zero audio in either direction once the call is established |
| 16:31.01 | hmmhesays | 127 lines later I have an app that mimicks app_directory but is super customizable |
| 16:31.03 | Zeeek | those are the RTP ports then |
| 16:31.07 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:31.20 | Zeeek | open the high ports |
| 16:31.44 | JT | mattx86: is asterisk on a public ip? |
| 16:31.48 | mattx86 | JT: yes, and no. asterisk is behind a NAT too, but we have a public IP being forwarded (NAT'ed) completely in both directions |
| 16:31.49 | Zeeek | except usually the NAT problems are one-way audio, not NO audio |
| 16:32.07 | Zeeek | mattx86 in DMZ? Because that rarely works |
| 16:33.23 | mattx86 | Zeeek: DMZ - pretty much |
| 16:33.39 | Zeeek | DMZ has never worked for me |
| 16:33.40 | *** join/#asterisk jbigbee (n=jbigbee@216.207.245.1) |
| 16:33.44 | Zeeek | for anything |
| 16:33.46 | JT | dmz does not work |
| 16:33.48 | Zeeek | I don't know why |
| 16:33.53 | JT | give asterisk a public ip |
| 16:33.56 | Zeeek | Forward the ports and get over it :) |
| 16:34.02 | JT | or port forward udp 5060 and 10000 to 20000 |
| 16:34.03 | Zeeek | or public ip |
| 16:34.08 | Zeeek | "like minds" |
| 16:34.28 | Zeeek | you can get by with 10 to 20 RTP for a modest installation |
| 16:34.40 | JT | that's a bit small |
| 16:34.57 | Zeeek | We never had a problem: 3 people so no big deal |
| 16:35.04 | JT | and i usually see no reason to differ from the default |
| 16:35.13 | JT | try 5 people |
| 16:35.19 | Zeeek | there si a good reason: some routers only allow 20 FWD |
| 16:35.26 | JT | lol |
| 16:35.29 | JT | bin them |
| 16:35.49 | Zeeek | why? It all works perfectly with 10 RTP port range |
| 16:36.02 | jblack | You heard him zeeek. Throw your hardware away because JT thinks you' |
| 16:36.06 | jblack | you're not forwarding enough. |
| 16:36.16 | JT | it won't work with 5 or more calls |
| 16:36.24 | Zeeek | we can't have 5 or more. |
| 16:36.24 | JT | or so |
| 16:37.06 | Zeeek | 2 analog lines and a few SIP/IAX but they never need to be on all at once |
| 16:37.11 | JT | jblack: only a cheap piece of crap would limit you to 20 port forwards |
| 16:37.27 | Zeeek | this is practice, not theory |
| 16:37.29 | JT | most of even the cheapest router support forwarding ranges |
| 16:37.32 | jblack | and? |
| 16:38.06 | JT | clearly it's inferior garbage if it only does 20 port forwards |
| 16:38.27 | Zeeek | clearly it works perfectly so why would I mess with it? It's an 8 meg DSL modem. |
| 16:38.59 | JT | adsl? |
| 16:39.06 | Zeeek | Async |
| 16:39.10 | jblack | I imagine up is much lower than that. |
| 16:39.14 | Zeeek | DSL in the US I guess |
| 16:39.27 | Zeeek | up is 256k |
| 16:39.35 | jblack | ouch. |
| 16:39.43 | Zeeek | more than enough |
| 16:39.52 | Zeeek | for $22/mo |
| 16:39.54 | JT | i was thinking if it was symmetrical it probably would be an expensive modem which would be a good idea to keep |
| 16:39.58 | jblack | for 5 channels of calling? Yeah. |
| 16:40.09 | Zeeek | I don't need 5 |
| 16:40.16 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:40.16 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:40.29 | Zeeek | we mostly need 2 and 90% pf the time one |
| 16:43.05 | Zeeek | if anyone wants to join the conference in about 17 minutes: http://voipusersconference.org IRC #voip-users-conference |
| 16:46.46 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 16:47.38 | *** join/#asterisk datachomper (n=russ@static-68-236-214-95.nwrk.east.verizon.net) |
| 16:47.51 | BobLutz | My boss just told me to leave the office cause of the looming White Death! |
| 16:48.03 | datachomper | Anybody know where I may find more information on the specifics of what a signed linear audio frame looks like? |
| 16:48.16 | nixguy | Zeeek: so joining in would be as easy as: |
| 16:48.18 | nixguy | exten => 666,1,Dial(sip/123@66.212.134.192) |
| 16:48.28 | russellb | datachomper: what do you mean "looks like" ? |
| 16:48.49 | datachomper | I need to know how to manipulate the frame->data portion of a signed linear frame |
| 16:49.04 | Zeeek | nixguy yeah |
| 16:49.06 | datachomper | It's in 8k samples right? But how large are the samples? |
| 16:49.13 | russellb | 2 bytes / sample |
| 16:49.20 | russellb | 16 bit samples ... |
| 16:49.49 | nixguy | Zeeek: ok im still a littlebit noob i called in but didnt get any voice or anything, no sound until 15min?! |
| 16:49.52 | Kobaz | any idea why all my iax peersd would be unreachable according to asterisk, but yet i can ping and connect to the iax ports on all the peers i have listed |
| 16:49.54 | datachomper | signed samples? |
| 16:49.59 | Zeeek | we're talking now |
| 16:50.27 | nixguy | attempting native bridge according to asterisk |
| 16:50.36 | nixguy | <PROTECTED> |
| 16:50.36 | nixguy | <PROTECTED> |
| 16:50.37 | nixguy | <PROTECTED> |
| 16:50.44 | nixguy | :) sorry if im taking your conference time |
| 16:50.52 | mattx86 | woot. we have audio! |
| 16:51.28 | mattx86 | I had to configure .. externip, localnet, and nat in sip.conf. appears to work now |
| 16:51.38 | datachomper | So when I cast the void pointer to the data portion of a slinear frame. I can cast it as (signed int *) ? |
| 16:51.57 | ManxPower | datachomper: would not this be better on #asterisk-dev |
| 16:52.09 | nixguy | mattx86: arhg ive only used my pbx for interal sip calls and then pstn.. |
| 16:52.21 | Zeeek | Leaving now, rejoice! http://voipusersconference.org IRC #voip-users-conference |
| 16:52.24 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:52.36 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 16:53.00 | datachomper | Appologies |
| 16:53.53 | Yourname`` | Hello. I was wondering about cascading queues. Say there are 5 agents logged into a queue, the 6call that comes into the queue obviously has to hold till an agent is free. I was wondering if I can make asterisk detect that the 6th call won't have a place on the queue and so it "overflows" the call to another external DID (which is a queue, except setup on a different system than asterisk) Is... |
| 16:53.54 | Yourname`` | ...that possible? |
| 16:54.24 | docelmo | Yourname`` yes use timeouts |
| 16:54.38 | *** part/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 16:54.41 | scoates | anyone know how I might fix this when making app_conference? http://pastebin.ca/932263 |
| 16:54.43 | docelmo | they will sit there briefly then fall back into the dialplan and move on |
| 16:54.58 | the_5th_wheel | Hi. My isp seems to block sip ports, s i thought id try and change the bind port on the sip config in asterisk.but no matter what i do i cant get my phone to register. Im using a grandstream phone.Can anyone suggest a way i can test if the fault is on my phne, the ISP, or the server? Sipsak can connect to my server |
| 16:55.13 | jblack | scoates: Doesn't sound to me like you want queues at all. |
| 16:55.26 | scoates | jblack: ? |
| 16:55.28 | Kobaz | any idea why all my iax peers would be unreachable according to asterisk, but yet i can ping and connect to the iax ports on all the peers i have listed |
| 16:55.31 | Kobaz | the_5th_wheel: tcpdump |
| 16:55.32 | scoates | ah.. should be Yourname`` ^ |
| 16:55.38 | Yourname`` | docelmo: exten=> 100,n,Queue(testq|Tt|0), exten=> 100,n,Voicemail(100,u) |
| 16:55.47 | jblack | scoates: Try replacing the queue with Dial(first&second&third&fourth&fifth) |
| 16:55.48 | Yourname`` | docelmo: That's what it is right now. |
| 16:56.04 | jblack | and on the next priority Dial(thebackup) |
| 16:56.07 | docelmo | Yourname`` yep more or less that should do it |
| 16:56.11 | nixguy | argh i can call into the conference but i cant enter the callid!!! |
| 16:56.14 | nixguy | :) |
| 16:57.13 | nixguy | im behind nat, do i need to forward ports to my * |
| 16:57.21 | ManxPower | nixguy: of course you do. |
| 16:57.27 | nixguy | guessed so |
| 16:57.36 | ManxPower | much like if you were running a web server, e-mail server, etc |
| 16:57.37 | nixguy | one can hope it does nattraversal magically |
| 16:57.59 | *** part/#asterisk datachomper (n=russ@static-68-236-214-95.nwrk.east.verizon.net) |
| 16:58.01 | nixguy | but alas por Yorick , for i knew him well |
| 16:58.12 | Yourname`` | docelmo: Since it's Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI]): -> Shouldn't that arg be Queue(testq|tT||0|) -> ? |
| 16:58.17 | mattx86 | JT: jott this down: Asterisk/SIP behind DMZ - requires externip, localnet, and nat. apparently no port forwarding is needed on the sip phone side of it |
| 16:58.33 | mattx86 | it works :) |
| 16:58.36 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 16:58.37 | ManxPower | mattx86: JT has known that for YEARS |
| 16:58.51 | Yourname`` | lol |
| 16:59.00 | ManxPower | Most of us that have been here for more than a month know that. |
| 16:59.03 | Yourname`` | JT is one of the resident experts here. |
| 16:59.14 | Yourname`` | So is ManxPower. |
| 16:59.18 | Yourname`` | So is [TK]D-Fender |
| 16:59.23 | mattx86 | oh.. maybe I didn't explain myself well enough earlier |
| 16:59.24 | Yourname`` | And soon so is Yourname`` |
| 16:59.43 | mattx86 | but I got it working... thanks guys |
| 17:00.48 | ManxPower | mattx86: that is usually the case. |
| 17:04.30 | jsmith | jblack: http://www.vikingelectronics.com/ is your friend |
| 17:04.49 | jblack | Oh yeah, I found the right equipment. |
| 17:04.49 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 17:04.49 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 17:05.01 | jsmith | Cresl1n! |
| 17:05.03 | jblack | It's the effort of installation and cost |
| 17:05.12 | Cresl1n | jsmith!!!! |
| 17:08.13 | Yourname`` | Hey guys, am I doing something wrong here? Since it's Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI]): -> Shouldn't that arg be Queue(testq|tT||0|) -> ? |
| 17:08.40 | docelmo | I dont think 0 is valid.. same of it being not set. Try 1 |
| 17:10.16 | [TK]D-Fender | mattx86: ... |
| 17:10.18 | [TK]D-Fender | ~sipnat |
| 17:10.19 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:10.21 | [TK]D-Fender | ^^^ |
| 17:10.31 | Yourname`` | docelmo: I did that with [TK]D-Fender's help only because I want it to keep ringing in the queue if there is someone logged in. And it go to voicemail only if someone's not logged into the queue.. |
| 17:11.57 | TJNII | Huh... I pushed the volume up button on my phone and it rebooted. That's not good. |
| 17:12.25 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.117.5) |
| 17:14.32 | Kobaz | any idea why all my iax peers would be unreachable according to asterisk, but yet i can ping and connect to the iax ports on all the peers i have listed |
| 17:15.04 | ManxPower | TJNII: you have a grandstram? |
| 17:15.23 | ManxPower | Kobaz: port 4569 UDP? |
| 17:15.27 | Kobaz | yeah |
| 17:15.41 | ManxPower | remove the qualify= option |
| 17:16.09 | Kobaz | mmm |
| 17:16.09 | Kobaz | k |
| 17:19.25 | Kobaz | same thing |
| 17:19.32 | Yourname`` | [TK]D-Fender should be given awards or something |
| 17:19.40 | Kobaz | ManxPower: no workey :( |
| 17:19.54 | Yourname`` | Like, an Asterisk Oscar |
| 17:20.00 | [TK]D-Fender | Yourname``: Nothing says "I love you" like Paypal ;) |
| 17:20.13 | Yourname`` | I remember that! |
| 17:20.30 | Yourname`` | You should set that as your away message so people have your paypal without asking you ;) |
| 17:23.33 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 17:24.19 | *** join/#asterisk Guggemand (i=Guggeman@80.198.131.46) |
| 17:24.42 | nny_1 | hmm anyone ever find a work around for a centos ((Red Hat 3.4.6-8)) #1 SMP Mon Oct 22 17:57:16 EDT 2007, kernel version 2.6.18-8.1.14.3.cc ) udev error? Getting unhappiness when i try to do a make on zaptel here: |
| 17:24.58 | nny_1 | build_tools/genudevrules > /etc/udev/rules.d/zaptel.rules / build_tools/genudevrules: line 3: udevinfo: command not found |
| 17:25.08 | Kobaz | ManxPower: still all unreachable |
| 17:27.46 | x86 | man... this Toshiba PBX is retarded |
| 17:28.01 | TJNII | ManxPower: This was my new Snom |
| 17:28.06 | Juggie | nny_1, if you do a 'whereis udevinfo' does it find anything? |
| 17:28.08 | x86 | I've got a CAS T1 strung up between a Toshiba PBX and my Asterisk box (using the Asterisk box as basically a proxy to the PSTN) |
| 17:28.25 | x86 | they tell me the software on the PBX is capable of doing PRI, but the T1 card can not handle a PRI |
| 17:28.46 | x86 | (as I wanted to do PRI between the two boxes instead of CAS/E&M Wink |
| 17:28.47 | x86 | ) |
| 17:28.55 | x86 | how retarded is that? |
| 17:28.57 | ManxPower | x86: that is pretty typical for PBXs |
| 17:29.05 | x86 | T1 cards not supporting PRI? |
| 17:29.17 | ManxPower | We have a T-1 card that can do PRI in our nortel, but the key to unlock the PRI software is $5,000 |
| 17:29.20 | ManxPower | x86: correct. |
| 17:29.53 | x86 | that's retarded! |
| 17:29.54 | Juggie | nny_1? |
| 17:30.09 | *** part/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 17:30.16 | ManxPower | x86: that is "maximizing revenue" |
| 17:30.27 | x86 | lol |
| 17:30.52 | ManxPower | x86: you will find similar experiences with pretty much all traditional PBX vendors. |
| 17:30.59 | Juggie | tip: dont ask questions and then ignore our attempts to help you |
| 17:33.16 | x86 | ManxPower: wow... |
| 17:33.27 | x86 | ManxPower: I just dont understand how the hardware could be different... |
| 17:33.45 | x86 | ManxPower: I mean, a T1 is a T1 is a T1.... a T1 controller card should be able to handle a T1 interface |
| 17:34.00 | x86 | ManxPower: the fact that you're running PRI signalling should be software-dependant, imho |
| 17:34.06 | x86 | that just really burns me |
| 17:34.13 | ManxPower | x86: the hardware really isn't different. |
| 17:34.22 | x86 | thats what I thought! |
| 17:34.46 | *** join/#asterisk adjohn (n=adjohn@p5155-ipad74marunouchi.tokyo.ocn.ne.jp) |
| 17:35.12 | x86 | my toshiba vendor told me they would have to come install a PRI card into the cabinent.... they also said the version of software we already have is fully capable of running PRI |
| 17:35.25 | x86 | which sounds bass-ackwards to me |
| 17:36.15 | *** join/#asterisk RoyK (n=roy@ip-202-5-149-91.dialup.ice.no) |
| 17:39.01 | ManxPower | x86: the existing card is prolly designed to lock out PRI service |
| 17:39.05 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:39.16 | ManxPower | you can whine and complain all you want, but it's not going to get yo PRI service on the PBX for free |
| 17:40.19 | x86 | heh |
| 17:41.23 | ManxPower | when we needed a T-1 card for our Nortel, we managed to figure out the model number of the card and bought it on e-bay. new: $3,000, used: $300 |
| 17:41.42 | x86 | I've only had experience with little Avaya Partner systems (with POTS lines) and Inter-Tel systems (with PRI already), so I figured PRI capabilities in a system that already supports T1 would be a no brainer |
| 17:42.00 | x86 | never had to deal with much vendor fuckery |
| 17:42.08 | x86 | because that's exactly what that is... fuckery |
| 17:42.13 | x86 | (tm) |
| 17:43.15 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 17:43.32 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:47.47 | grandpapadot | lol, someone update jbot with 'fuckery' |
| 17:48.08 | grandpapadot | oops, sorry, fuckery(tm) |
| 17:48.11 | grandpapadot | lol |
| 17:54.36 | *** join/#asterisk CVirus (n=GoD@41.233.143.82) |
| 17:55.33 | tld | herlig... 1743 på fredag kveld svarer siemens med å si at kundeservicen er åpen 0800 til 1800 på ukedager. ;) |
| 17:55.42 | tld | (de svarte på spørsmålet også, så var forsåvidt greit nok) |
| 17:55.44 | anonymouz666 | what the fuck |
| 17:55.52 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
| 17:56.10 | tld | litt rotere forklart av meg... |
| 17:56.18 | tld | mailet dem ett spørsmål, og 1743 svarte de. |
| 17:56.32 | tld | var bare morsomt at de begynte mailen med å si jeg gjerne kunne ringe dem, men det måtte være 0800 til 1800 ukedager |
| 17:56.38 | anonymouz666 | tld: hur mar du? |
| 17:56.43 | tld | errr |
| 17:56.49 | tld | wrong channel |
| 17:56.54 | tld | wrong network even. ;) |
| 17:57.00 | tld | anonymouz666: "mar?" |
| 17:57.15 | anonymouz666 | I think it's swedish |
| 17:57.23 | anonymouz666 | :D |
| 17:57.26 | grandpapadot | tld: gotoiftime |
| 17:58.23 | tld | anonymouz666: Doesn't seem like swedish |
| 17:58.29 | tld | grandpapadot: gotoiftime? |
| 17:58.30 | tld | meh |
| 18:00.20 | znoG | anyone know why my 'i' extension is not working in a context? I'm using switches {} to call REALTIME and I was hoping that it would go to the 'i' extension if an invalid extension is dialed in that context. |
| 18:01.35 | ManxPower | znoG: "i" only works in IVRs |
| 18:02.01 | Greek-Boy | is it recommended to have the IAX2 jiterbuffer on? |
| 18:04.37 | Yourname`` | Is there a windows based mp3/wav -> ulaw converter anyone knows about? |
| 18:05.23 | tld | Yourname``: can probably do that with audacity? |
| 18:05.35 | Yourname`` | tld: Have you done it? |
| 18:05.50 | tld | Yourname``: I *think* so. Been a while though. |
| 18:05.57 | tld | Yourname``: Or maybe I did the other way around? |
| 18:05.59 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 18:06.04 | tld | Yourname``: Either way, it'll probably make you happy |
| 18:06.11 | Yourname`` | Yeah, no, lol. I want something that says "convert" source "mp3" target "ulaw", lol |
| 18:06.16 | tld | Yourname``: (it's free, and it'll take you just a few mins to test) |
| 18:06.42 | tld | Yourname``: Do you need it automated/scriptable? If you don't mind doing it through a GUI, it'll make you happy. |
| 18:06.48 | znoG | ManxPower: so what's the recommended way to do something like what I want to achieve? (look via realtime using "switch" and go to a Playback() call if not found) |
| 18:07.45 | znoG | s/look/lookup |
| 18:10.10 | *** join/#asterisk erstazi (n=erstazi@unaffiliated/erstazi) |
| 18:16.36 | Yourname`` | So hard to figure out how to remove disconnected numbers from the diallist :( |
| 18:19.24 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 18:23.21 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:35.31 | Greek-Boy | what is the strongest method of authentication for IAX2? is it RSA keys? or MD5/ |
| 18:37.46 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-be356ce0ee1a754e) |
| 18:39.18 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 18:40.28 | tzafrir_home | Greek-Boy, md5 hashing is always used (on the wire) |
| 18:40.58 | tzafrir_home | RSA keys means you don't send the secret on the wire, and thus it is more secure |
| 18:42.09 | Kobaz | ManxPower: any idea how to fix the underlying problem with iax peers being unreachable, other than turning off qualify? |
| 18:43.49 | Greek-Boy | thanks tzafrir |
| 18:43.53 | Greek-Boy | hi [TK]D-Fender |
| 18:44.00 | Greek-Boy | [TK]D-Fender how are you? |
| 18:44.10 | [TK]D-Fender | Greek-Boy: Doing OK |
| 18:45.10 | Greek-Boy | [TK]D-Fender: good to hear |
| 18:48.15 | ManxPower | Kobaz: did they become available after you turned off qualify? |
| 18:48.25 | *** join/#asterisk af_ (n=getsmart@88-149-240-207.dynamic.ngi.it) |
| 18:48.54 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 18:55.13 | *** part/#asterisk ManxPower (n=manxpowe@97-81-241-251.dhcp.lds.al.charter.com) |
| 18:55.19 | Kobaz | ManxPower: yeah i can dial out |
| 18:57.52 | *** join/#asterisk cr0n (n=d@dsl-240-27-108.telkomadsl.co.za) |
| 18:58.17 | *** join/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 18:58.44 | cr0n | hi, im having a problem with a tdm400p/asterisk not picking up a hangup in south africa, anybody? |
| 19:01.52 | Kobaz | cr0n: http://www.voip-info.org/wiki/index.php?page=Asterisk+Disconnect+Supervision |
| 19:02.53 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 19:04.50 | cr0n | Kobaz: thanks. |
| 19:04.54 | mattx86 | I don't have SIP working completely; when dialing from the SIP phone out, it works. when dialing in, the SIP phone rings, but there's no audio in either direction. asterisk says it can't find address for host extension 'xxx' |
| 19:06.22 | *** join/#asterisk msolomos (n=mw@85.72.134.101) |
| 19:09.23 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 19:15.32 | tld | Can I have Asterisk automatically accept all calls from one IP, not challenging it? |
| 19:16.47 | _ShrikE | tld: insecure=port,invite |
| 19:17.14 | tld | thanks. :) |
| 19:26.36 | *** part/#asterisk scoates (n=sean@iconoclast.caedmon.net) |
| 19:40.19 | TJNII | Has anyone here had trouble with Snom's rebooting? Mine has done it 3 times, but I haven't figured why. |
| 19:41.26 | Yourname`` | How can I reload musiconhold.conf? |
| 19:42.49 | jsmith | TJNII: Have you upgraded the firmware? |
| 19:42.59 | jsmith | Yourname``: "moh reload" from the CLI |
| 19:43.36 | Yourname`` | jsmith: Thanks! |
| 19:44.18 | Yourname`` | A quick question, let's say someone is waiting to talk to an agent and is in the queue. If that person hangs up, sometimes the call is still forwarded to an agent .. and obviously the agent notices dead air on his end. How can this be taken care of? |
| 19:45.08 | dmz | anyone know if it's possible to convert existing voicemail accounts into an imap based voicemail system? |
| 19:51.13 | *** join/#asterisk GMitre (n=nda@cavalao.nitnet.com.br) |
| 19:52.22 | GMitre | I'm configured a external line into my Asterisk, i can call out but when i try to receive a call i receive this error: WARNING[3789]: chan_sip.c:10011 check_auth: username mismatch, have ... Some one knows what´s going on ? |
| 19:57.55 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 20:00.25 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:01.14 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 20:04.56 | *** join/#asterisk gego (n=gego@dyndsl-091-096-098-208.ewe-ip-backbone.de) |
| 20:06.44 | *** part/#asterisk gego (n=gego@dyndsl-091-096-098-208.ewe-ip-backbone.de) |
| 20:07.03 | SteveTotaro | i have had moh reload crash systems |
| 20:07.17 | SteveTotaro | yourname, it depends on when they hangup |
| 20:07.38 | SteveTotaro | if it is just prior to the agent answering then there is no way around it |
| 20:07.59 | SteveTotaro | otherwise, what tech are you using to deliver the calls? |
| 20:11.52 | jameswf | I think I will try 276.5K calls per hour... |
| 20:12.13 | jameswf | then work down.... |
| 20:13.07 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 20:14.17 | *** join/#asterisk joshaidan (n=brianj@S0106001c1023e838.tb.shawcable.net) |
| 20:14.26 | mattx86 | dmz, you could have the voicemails e-mailed to an IMAP-capable mail server |
| 20:14.55 | *** join/#asterisk Pbx_help (n=Pbx_help@193.227.191.90) |
| 20:14.57 | Pbx_help | hello all |
| 20:15.05 | dmz | mattx86, ahh cool idea, thanks |
| 20:15.16 | Pbx_help | could anyone help me out with finding the best Linux distro to install asterisk on! |
| 20:15.33 | Pbx_help | am currently using redhat 4.X and im having all sort of errors and downtime |
| 20:15.36 | dmz | mattx86, i'll have to forge headers to get it to look right, but proably easier to just tell everyone to write down their messages and delete them :) |
| 20:16.36 | mattx86 | dmz: oh I see. I didn't realize there were imap settings for the voicemail. just an idea though ;) |
| 20:16.57 | *** join/#asterisk yenno (n=yenno@92.104.83.58) |
| 20:17.08 | yenno | hi, somebody knows a softphone that supports centralized conferencing? |
| 20:17.15 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:17.19 | dmz | well it should save the caller info & stuff (I'd assume voicemail in imap would have that) so i coudl have my cell phone(imap) or desk phone (cisco+*) "see" info about each msg |
| 20:17.40 | dmz | yenno why not use app_conference & call into an established conference on * (or meetme)? |
| 20:18.13 | Pbx_help | any1 knows of a possible solution for *white noise*! |
| 20:18.35 | yenno | dmz because i need to test centralized conference using INVITE conf@... ;-) |
| 20:18.55 | outtolunc | bigass comet? |
| 20:21.19 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 20:22.33 | GMitre | I'm configured a external line into my Asterisk, i can call out but when i try to receive a call i receive this error: WARNING[3789]: chan_sip.c:10011 check_auth: username mismatch, have ... Some one knows what´s going on ? |
| 20:28.49 | Katty | for one touch recording on zap lines, is it (Zap/g1/${EXTEN},,wW) or ${EXTEN),wW? |
| 20:30.59 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 20:32.21 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 20:33.31 | *** join/#asterisk galeras (n=Martin@190.144.130.142) |
| 20:34.25 | galeras | Dear Sirs, Any Idea Why a caller doesn't ear ringtone on an attended transfer? |
| 20:35.18 | *** join/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 20:36.15 | clyrrad | galeras: make sure you have Answer(), and also use the 'r' flag in Dial |
| 20:36.40 | asteriskmonkey | can you use goto to other contexts in realtime/ |
| 20:36.44 | clyrrad | you can also try Wait(1) |
| 20:37.00 | galeras | clyrrad: Cheking... |
| 20:37.56 | *** join/#asterisk CVirus (n=GoD@196.205.192.125) |
| 20:38.06 | *** join/#asterisk kharbooza (n=kharbooz@203.99.186.19) |
| 20:39.14 | galeras | clyrrad: Yep, i have answer and r flag in dial, weird thing is blind transfer generate ring tone while attended trasnfer doesn't |
| 20:39.42 | clyrrad | Does anyone know if its possible to use something like AppendCDRUserField() that will allow you to store other data with a CDR Record? Or is the only option to use 'userfiled' column, and use some sort of delimiter to seperate out the data? I would like to be able to set custom fields in the CDR table instead of having to use a comma seperated list in the 'userfield' column. Is this possible? |
| 20:41.10 | kharbooza | hi, for 4-5 days asterisk worked fine, today when i rebooted the computer for some reason asterisk daemon didnot start, i started it manually "service asterisk start". my problem is when i want to connect to asterisk via cli it says "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)". though that file exist. what should i do and yes tty 9 nothing there |
| 20:41.55 | clyrrad | kharbozza: try to start it like this asterisk -vvvvvvvvvvvgc, that will tell you why it dies |
| 20:42.06 | clyrrad | galeras: try to put a Wait(1) before the attended transfer |
| 20:44.26 | kharbooza | clyrrad: cli starts with this warning "WARNING[3307]: res_smdi.c:722 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener." rest of the stuff looks ok |
| 20:45.06 | clyrrad | kharbooza: if asterisk is not starting there must be some error, can you connect now with asterisk -r? |
| 20:45.30 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:45.43 | Katty | nice: sox: No such file or directory <- does that mean anything to anyone? |
| 20:45.58 | kharbooza | clyrrad: yes i can, but i want it to start as service |
| 20:46.03 | clyrrad | Katty: it means you dont have sox installed, or you didnt specify the full path to it |
| 20:46.20 | clyrrad | kharbooza: then something is wrong with your init script for Asterisk |
| 20:46.25 | Katty | clyrrad: what is 'sox' for? |
| 20:46.30 | Qwell | ~sox |
| 20:46.31 | jbot | sox is, like, Sound Processing Tool. URL: http://sox.sourceforge.net/ |
| 20:46.33 | Katty | clyrrad: besides something to do with recording audio |
| 20:46.49 | clyrrad | Katty: Qwell just linked you |
| 20:46.54 | Katty | yeah i saw it |
| 20:47.05 | clyrrad | Can anyone answer my question now? |
| 20:48.19 | kharbooza | clyrrad: when i got this problem first time i changed in asterisk.conf: astrundir => /var/run to astrundir => /var/run/asterisk. should i revert it |
| 20:49.13 | Katty | sox: invalid option -- m <- where do i go to look at what's giving options. |
| 20:49.23 | Qwell | it's soxmix now |
| 20:49.26 | Qwell | instead of sox -m |
| 20:49.42 | clyrrad | kharbooza: Give it a shot |
| 20:49.47 | Qwell | Katty: re-run ./configure |
| 20:50.00 | Katty | asterisk? or addons? |
| 20:50.15 | Qwell | asterisk |
| 20:50.18 | Qwell | ./configure && make install |
| 20:50.18 | clyrrad | Qwell: you know the answer to my CDR question by chance? |
| 20:50.37 | Qwell | clyrrad: delimit it |
| 20:50.49 | GMitre | I'm configured a external line into my Asterisk, i can call out but when i try to receive a call i receive this error: WARNING[3789]: chan_sip.c:10011 check_auth: username mismatch, have ... Some one knows what´s going on ? |
| 20:51.00 | clyrrad | Qwell: is it the only way? As in there is no way to append CDR Data to other columns? |
| 20:51.09 | Qwell | clyrrad: cdr_adaptive_odbc |
| 20:51.26 | kharbooza | clyrrad: i got this error: Starting asterisk: Unable to open pid file '/var/run/asterisk.pid': Permission denied |
| 20:51.40 | clyrrad | Qwell: hrm not famaliar with that, is it a config file? |
| 20:52.02 | clyrrad | kharbooza: ls -la /var/run/asterisk.pid, who owns it? |
| 20:52.20 | Qwell | nobody would own it - it wouldn't exist unless asterisk was running |
| 20:52.27 | clyrrad | he said before it exists |
| 20:52.44 | kharbooza | clyrrad: no such file or dir |
| 20:52.56 | Katty | Qwell: make and make install, too? |
| 20:53.00 | clyrrad | kharbooza: who are you logged in as? |
| 20:53.03 | Qwell | yeah |
| 20:53.33 | kharbooza | clyrrad: my personal account |
| 20:53.37 | clyrrad | Qwell: cdr_adaptive_odbc.c you mean would need to change the code correct? |
| 20:53.59 | clyrrad | kharbooza: did you build asterisk as root? |
| 20:54.16 | kharbooza | clyrrad: yes |
| 20:54.19 | clyrrad | thats why |
| 20:54.24 | clyrrad | su - root |
| 20:54.30 | clyrrad | and then service asterisk start |
| 20:54.31 | Katty | Qwell: any menuselect stuff i need to do? |
| 20:54.35 | Qwell | nope |
| 20:54.38 | Katty | kk |
| 20:54.45 | Qwell | you may need to make clean though if it doesn't work |
| 20:54.53 | Katty | aye aye |
| 20:54.56 | clyrrad | Qwell: did I understand you correctly? |
| 20:54.58 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 20:55.39 | kharbooza | i got the same error hey brb |
| 20:57.18 | clyrrad | Qwell: looks like the cdr_adaptive_odbc is for version 1.6 |
| 20:59.56 | clyrrad | Qwell: you there? |
| 21:01.04 | clyrrad | :/ |
| 21:01.13 | Katty | he probably went to the little boys room. |
| 21:01.20 | Katty | or to do something worthwhile :P |
| 21:01.30 | clyrrad | maybe.... |
| 21:01.43 | clyrrad | maybe he's ignorin me :P |
| 21:02.02 | Katty | jbot: festival? |
| 21:02.02 | jbot | festival is, like, a general multi-lingual speech synthesis system developed at CSTR. See http://www.cstr.ed.ac.uk/projects/festival/, or festival lite a much more compact festival http://www.speech.cs.cmu.edu/flite/index.html |
| 21:02.08 | Katty | ah, right. |
| 21:02.23 | clyrrad | Katty: should check out Cepstral too |
| 21:02.37 | Katty | i think not. |
| 21:02.46 | clyrrad | why not? |
| 21:02.52 | Katty | because i don't need it. |
| 21:03.05 | clyrrad | It does better job then festival |
| 21:03.46 | Corydon76-dig | and AT&T Natural Voices do a better job than Cepstral |
| 21:03.48 | Katty | yeah i don't need that either. |
| 21:04.02 | Corydon76-dig | The only difference is expense |
| 21:04.06 | Katty | everyone i know speaks english. |
| 21:04.07 | clyrrad | Corydon76-dig: yea |
| 21:04.11 | Katty | except JunK-Y |
| 21:04.16 | Katty | but he's pretty good at it now too |
| 21:05.14 | Katty | Katty vs sox, round two! |
| 21:05.21 | clyrrad | so looks like you can only append CDR fileds with Asterisk 1.6, no way to do it in Asterisk 1.4... |
| 21:05.30 | clyrrad | from what I can see |
| 21:06.07 | Katty | katty WINS |
| 21:06.08 | Corydon76-dig | clyrrad: eh? |
| 21:06.09 | Katty | Qwell: i love you. |
| 21:06.50 | clyrrad | Corydon76-dig: I am trying to append CDR fields to CDR records in PGSQL, it seems its not possible, that I would need to use the 'userfield' column and some sort of delimiter.... is that not correct? |
| 21:06.53 | Corydon76-dig | clyrrad: http://svncommunity.digium.com/view/tilghman/branches/1.4/ |
| 21:07.16 | *** join/#asterisk angryuser (n=nononon@df01t2-213-44-80-96.d4.club-internet.fr) |
| 21:07.24 | angryuser | hi all |
| 21:07.37 | clyrrad | Corydon76-dig: oh nice! I didnt know there was a backport for it |
| 21:07.48 | angryuser | i forgot the commant which let me see who is the sync master, which one was it ? |
| 21:07.53 | angryuser | command* |
| 21:08.08 | Corydon76-dig | clyrrad: also a backport for res_config_curl |
| 21:08.33 | Qwell | Katty: that's what they all say |
| 21:09.02 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 21:09.39 | angryuser | anyone ? |
| 21:09.41 | clyrrad | Corydon76-dig: now I just need to figure out how to use it, never installed a backkport before. Do I just download the files from the link you gave and compile? |
| 21:10.25 | Corydon76-dig | clyrrad: do an SVN checkout |
| 21:10.33 | Corydon76-dig | clyrrad: that's what the Makefile is there for |
| 21:10.47 | Corydon76-dig | ASTSRC=/path/to/asterisk-1.4 make |
| 21:11.24 | clyrrad | Corydon76-dig: gotcha, are these considered safe for production systems? |
| 21:11.38 | clyrrad | by safe i mean stable |
| 21:11.40 | angryuser | tzafrir you here ? |
| 21:11.42 | Corydon76-dig | clyrrad: all backports are at your own risk |
| 21:11.56 | Corydon76-dig | clyrrad: but then again, everything is already at your own risk |
| 21:12.19 | clyrrad | Corydon76-dig: yea, I hear ya.... if it was you would you use it or wait for 1.6? |
| 21:12.38 | Corydon76-dig | clyrrad: I'm the author. I've used it in 1.4 |
| 21:12.59 | clyrrad | Corydon76-dig: ah, so you have experience with it :D |
| 21:14.32 | Corydon76-dig | I had a need. So I wrote it |
| 21:14.55 | clyrrad | Yea I wanna try it out, just never used SVN before, I ususally just download the sources and compile |
| 21:14.59 | Corydon76-dig | svn co http://svncommunity.digium.com/svn/tilghman/branches/1.4 cdr_adaptive_odbc-1.4 |
| 21:15.24 | *** join/#asterisk champster (n=asterisk@69.71.17.7) |
| 21:15.29 | clyrrad | do I need to do that in my asterisk sources directory? Or I can be in a new directory? |
| 21:15.51 | angryuser | Fender awake ? |
| 21:16.13 | Corydon76-dig | clyrrad: new directory, but you point it back to the original source with ASTSRC |
| 21:16.22 | Corydon76-dig | See above |
| 21:16.34 | clyrrad | ok let me give that a shot |
| 21:16.41 | Corydon76-dig | It's a dependent Makefile, just like zaptel |
| 21:17.30 | clyrrad | gotcha |
| 21:17.50 | champster | How do I send an email after a call is hungup?? I want to send an email if someone calls 911 |
| 21:18.20 | Corydon76-dig | champster: see the 'h' extension |
| 21:18.30 | clyrrad | champster: you could use an AGI on the h exten |
| 21:19.34 | clyrrad | Corydon76-dig: do you need to recompile asterisk after this? |
| 21:19.36 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 21:19.56 | champster | Will 'h' catch evrytime? Is system riskier than agi? |
| 21:20.04 | Corydon76-dig | clyrrad: nope |
| 21:20.09 | clyrrad | k |
| 21:20.11 | Corydon76-dig | clyrrad: works just like the kernel |
| 21:20.16 | clyrrad | got it checked out :D |
| 21:20.28 | Corydon76-dig | clyrrad: as long as your sources match up to what you're running, you're fine |
| 21:20.52 | clyrrad | ok here goes nuttin :) |
| 21:21.28 | Corydon76-dig | clyrrad: just remember that you need the config file for it to do anything |
| 21:21.43 | clyrrad | cdr_adaptive_odbc.conf right? |
| 21:22.40 | Corydon76-dig | Correct |
| 21:22.48 | Corydon76-dig | There's a sample in that directory |
| 21:22.55 | clyrrad | yea i see some .so files now :D |
| 21:23.24 | clyrrad | just need to copy those into the modules directory i assume? |
| 21:24.33 | Corydon76-dig | 'make install' |
| 21:25.08 | clyrrad | Corydon76-dig: yea I tried that, but sais No rule to make target /usr/src/asterisk........ |
| 21:25.17 | clyrrad | Makefile must have that path hardcoded |
| 21:25.38 | Corydon76-dig | clyrrad: ASTSRC=/path/to/asterisk-1.4 make |
| 21:26.10 | clyrrad | Corydon76-dig: yea thats what I did |
| 21:26.21 | Corydon76-dig | Then same line and 'install' |
| 21:26.34 | Corydon76-dig | clyrrad: ASTSRC=/path/to/asterisk-1.4 make install |
| 21:26.41 | clyrrad | ah, thats my mistake |
| 21:26.42 | clyrrad | thanks |
| 21:26.55 | clyrrad | beauty that worked :D |
| 21:28.25 | *** join/#asterisk ether0 (n=afarring@apoc.digiport.com) |
| 21:29.10 | clyrrad | Corydon76-dig: reading your sample config file now |
| 21:29.43 | Corydon76-dig | clyrrad: btw, the category name is irrelevant. For your edification only. |
| 21:29.59 | twisted | yay |
| 21:30.02 | twisted | winter storm warning |
| 21:30.20 | Corydon76-dig | twisted: you headed out in a panic for milk and bread? |
| 21:30.27 | twisted | BWAHAHAHAHAHHAAHA |
| 21:30.27 | twisted | no |
| 21:30.36 | twisted | I actually KNOW how to get around and live a normal lief |
| 21:30.39 | twisted | *life |
| 21:30.55 | clyrrad | Corydon76-dig: I dont see 'category' name in the .conf file |
| 21:31.08 | Corydon76-dig | clyrrad: the name inside [here] |
| 21:31.32 | flush | mayday mayday |
| 21:31.41 | clyrrad | Corydon76-dig: ah, yea I was looking at that wondering how to name them |
| 21:31.42 | flush | now im totally lost |
| 21:32.18 | flush | how do you configure a TDM400P card to use with a single line, with a single analog phone |
| 21:32.21 | Corydon76-dig | clyrrad: I considered naming them [this] [doesn't] [matter] |
| 21:32.26 | clyrrad | Corydon76-dig: whats the reason for the multiple [contexts], is it so you can replicate the CDR info in multiple tables? |
| 21:32.31 | flush | i have configured my fxo port and fxs works, but it seems fxo is not working.. |
| 21:32.47 | flush | asterisk doesnt ring the phone when call comes in, and zttool shows "alarms: OK" even when the fxo port is not plugged |
| 21:32.47 | Corydon76-dig | clyrrad: not just multiple tables. Multiple databases, multiple connections... |
| 21:33.00 | clyrrad | Corydon76-dig: nice feature :D |
| 21:33.05 | Corydon76-dig | Think mysql, MS SQL Server, Sybase, Postgres, whatever |
| 21:33.21 | clyrrad | Yea I have PGSQL CDR's |
| 21:33.29 | Corydon76-dig | clyrrad: mvanbaak is using the filter feature, so that each customer's CDRs go into a separate table |
| 21:33.38 | flush | any ideas |
| 21:34.03 | Corydon76-dig | flush: fixed in 1.6 |
| 21:34.16 | flush | ? |
| 21:34.36 | clyrrad | Corydon76-dig: yea this is a nice feature |
| 21:34.41 | Corydon76-dig | 1.4 doesn't show red alarms on FXO ports |
| 21:34.44 | flush | i run Asterisk 1.4.10 |
| 21:34.52 | flush | god damnit |
| 21:35.05 | Corydon76-dig | unless you're using the old single span cards |
| 21:35.32 | Corydon76-dig | or a digital connection |
| 21:35.36 | clyrrad | Corydon76-dig: so you can have as many [contexts] as you like and your module will call each of them when a CDR is written is that how the logic works? |
| 21:35.44 | Corydon76-dig | clyrrad: correct |
| 21:35.48 | clyrrad | very nice |
| 21:36.32 | Corydon76-dig | btw, the native cdr_pgsql in 1.6 supports that as well, but without filters or aliases |
| 21:36.46 | Corydon76-dig | Also it doesn't support multiple connections |
| 21:36.57 | Corydon76-dig | but it will do additional fields, if they're named the same |
| 21:37.14 | Corydon76-dig | and it will also allow you to omit fields that you don't care for |
| 21:37.17 | flush | is there a up to date repository for asterisk on ubuntu |
| 21:38.02 | Corydon76-dig | clyrrad: are there fields you don't use? |
| 21:38.12 | Corydon76-dig | like amaflags? |
| 21:38.16 | clyrrad | yes |
| 21:38.21 | clyrrad | there are many fields I dont use |
| 21:38.30 | Corydon76-dig | clyrrad: exclude them from your table. It's adaptive. |
| 21:38.44 | clyrrad | wicked |
| 21:39.13 | clyrrad | connection=blah <- where is that pulling from? IE: what config? |
| 21:39.19 | Corydon76-dig | Dropping fields while in use doesn't work very well, but you can add fields and 'reload' the module |
| 21:39.37 | clyrrad | refering to your sample.conf file |
| 21:39.37 | Corydon76-dig | clyrrad: connection is pulling from res_odbc.conf |
| 21:39.41 | clyrrad | ok |
| 21:40.42 | clyrrad | your alias statments let you use CDR(ALIAS_NAME) in the dial plan correct? |
| 21:40.50 | Corydon76-dig | Also, cdr_addon_mysql in trunk is similarly adaptive |
| 21:40.59 | Corydon76-dig | clyrrad: correct |
| 21:41.07 | clyrrad | you thought of everything :) |
| 21:41.18 | Corydon76-dig | clyrrad: the main reason I put alias in, though, is to let you rename the standard cdr fields |
| 21:41.30 | Corydon76-dig | alias src => ani |
| 21:41.31 | clyrrad | yea its a good plan |
| 21:42.05 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 21:42.08 | Corydon76-dig | Not that there's a reason to, but you can do "alias src => dst" and "alias dst => src" |
| 21:42.20 | Corydon76-dig | That will flip them in the database |
| 21:42.27 | *** join/#asterisk cleone (i=cleo@41.251.91.103) |
| 21:42.42 | clyrrad | Corydon76-dig: thats some nice work bud! I just tested it works beautifuly :D |
| 21:42.50 | Corydon76-dig | Left side is the CDR name, right side is the column name |
| 21:42.55 | clyrrad | my new fields are populating in the DB :) :) |
| 21:43.06 | *** join/#asterisk Netgeeks (n=chris@gw0.office1.talkplus.com) |
| 21:43.09 | clyrrad | yea that makes sense |
| 21:43.16 | Katty | Netgeeks: hai. |
| 21:43.23 | clyrrad | Thanks for making this mod and telling me about it |
| 21:43.30 | Corydon76-dig | Think of the arrow as pointing out the data flow |
| 21:43.36 | Netgeeks | Katty: Hello! |
| 21:43.56 | clyrrad | Corydon76-dig: yep makes good sense |
| 21:44.09 | *** part/#asterisk asteriskmonkey (n=asterisk@69.77.169.14) |
| 21:44.46 | clyrrad | Corydon76-dig: this is so cool, thanks again, so much more efficient then using delimited string in the 'userfiled' |
| 21:45.07 | Corydon76-dig | clyrrad: correct, that was the point |
| 21:45.33 | Corydon76-dig | Someone came up with userfield2 and userfield3 because they couldn't cram everything into that one field |
| 21:45.47 | clyrrad | it took a fiar bit of C code to make it work from what i see |
| 21:45.48 | Corydon76-dig | That just made no sense to me |
| 21:45.59 | clyrrad | yea forsure.... your solution is much more efficent |
| 21:46.08 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 21:46.25 | Corydon76-dig | clyrrad: No, most of that code is just verification of the contents, to ensure that we don't fail on bad input |
| 21:47.03 | Corydon76-dig | clyrrad: also, if you put an integer into a field, it selects different things based on whether the field type is integer or char |
| 21:47.32 | Corydon76-dig | and ensures that strings fit within their appointed fields |
| 21:47.39 | clyrrad | I think my fileds will be all varchar's |
| 21:48.10 | Corydon76-dig | clyrrad: no datetime fields? |
| 21:48.29 | Corydon76-dig | Verifies the format on those, too |
| 21:48.33 | clyrrad | I think the ones that are there now are good |
| 21:48.41 | clyrrad | calldate field |
| 21:48.56 | Corydon76-dig | Make sure you "alias start => calldate" |
| 21:49.26 | clyrrad | start something special in the module? |
| 21:49.43 | Corydon76-dig | start is the name of the time the call started |
| 21:49.51 | Corydon76-dig | The CDR fieldname |
| 21:50.16 | Corydon76-dig | "start", "answer", "end", "duration", "billsec" |
| 21:50.23 | clyrrad | Yea i see in your code you have "start", "end" and "answer" |
| 21:50.31 | Corydon76-dig | First 3 are datetimes, last 2 are integers |
| 21:51.05 | clyrrad | any difference between your mods "billsec" and the regular cdr's "billsec"? |
| 21:51.23 | Corydon76-dig | clyrrad: nope, same thing |
| 21:51.56 | clyrrad | so really all I need to alias is calldate then |
| 21:52.04 | Corydon76-dig | Correct |
| 21:52.10 | clyrrad | very cool |
| 21:52.12 | clyrrad | i like this |
| 21:52.33 | Corydon76-dig | Most of my tools give people ideas |
| 21:53.31 | clyrrad | One day I would like to look though and figure out the structure of how asteirsk is coded so that I could make add ons like this. I know how to code in C, but havent delved into the layout of Asterisk, its a bit overwhelming :P |
| 21:54.19 | Corydon76-dig | My eventual goal is to make it so that most people will never have a need to code a module |
| 21:54.30 | clyrrad | How will you do this? |
| 21:54.32 | Corydon76-dig | or use AGI |
| 21:54.51 | clyrrad | I liked the PHP-AGI, just slow compared to C modules |
| 21:54.52 | Corydon76-dig | by making the existing modules so versatile that you won't need to |
| 21:55.24 | Corydon76-dig | func_odbc and cdr_adaptive_odbc are both tools of that goal |
| 21:55.25 | jameswf | sounds like web 3.0 |
| 21:55.50 | clyrrad | Corydon76-dig: func_odbc rocks I use it extensivly in 1.4 |
| 21:56.09 | clyrrad | can do so much with the option of using a database |
| 21:56.37 | mattx86 | is there anything I can tweak on asterisk or my SPA941, in order to avoid having to NAT inbound udp ports 10000-20000? everything works fine with the inbound NAT rule in place, but I'd rather not have to do that. |
| 21:56.56 | Corydon76-dig | Remember MacroExclusive in 1.2? In 1.6, you can use LOCK(), UNLOCK(), TRYLOCK(), and Gosub/Return in the dialplan |
| 21:57.29 | clyrrad | Yea 1.6 has a whole bunch of exciting new features, I was reading up on that last week |
| 21:57.29 | Corydon76-dig | and you cannot deadlock using LOCK() and UNLOCK() in the dialplan |
| 21:57.39 | Corydon76-dig | It has deadlock avoidance |
| 21:57.45 | clyrrad | nice |
| 21:58.18 | clyrrad | I just found in your code where you do the datatype checks that you were telling me about earlier |
| 21:58.44 | Katty | jbot: email |
| 21:58.45 | jbot | it has been said that email is the lifeblood of the pornternet |
| 21:58.52 | Katty | jbot: sendmail? |
| 21:58.53 | jbot | Powerful and flexible Mail Transport Agent. URL: http://www.sendmail.org/ |
| 21:59.08 | Katty | jbot: :< |
| 21:59.08 | jbot | methinks < is redirection of stdin to a program |
| 21:59.32 | Qwell | jbot: you're wrong |
| 21:59.33 | jbot | I reject your reality and substitute my own. |
| 21:59.34 | Katty | i wonder if i could use hula. |
| 21:59.47 | Katty | i always wanted to set that up |
| 21:59.49 | jameswf | ~nowwhat | Katty |
| 21:59.54 | Wayhigh | jbot: sendmail is also overly complicated |
| 21:59.55 | jbot | okay, Wayhigh |
| 22:00.12 | Corydon76-dig | clyrrad: if you find an obscure field that I haven't tested, you should get a warning, so please let me know if that happens |
| 22:00.24 | Wayhigh | jbot: sendmail? |
| 22:00.25 | jbot | Powerful and flexible Mail Transport Agent. URL: http://www.sendmail.org/, or overly complicated |
| 22:00.32 | clyrrad | Corydon76-dig: yes definatly will do |
| 22:00.38 | Wayhigh | that's much better :) |
| 22:00.51 | Corydon76-dig | clyrrad: but the worst case is that the field will simply remain blank. The module will still attempt to insert a record on the remaining fields |
| 22:01.14 | *** join/#asterisk CrashSys (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 22:01.17 | clyrrad | Corydon76-dig: yea and from what I see it plays real nice with varchars in PGSQL |
| 22:01.30 | *** join/#asterisk real0ne (n=up@41.251.91.103) |
| 22:01.53 | Katty | anyone have a link for getting an email server up and going with asterisk? |
| 22:01.59 | Katty | i've never done it before. |
| 22:02.00 | clyrrad | Corydon76-dig: is this a skeleton file you use then code your logic inside it? |
| 22:02.21 | Corydon76-dig | clyrrad: why? |
| 22:02.33 | Corydon76-dig | clyrrad: what'd I miss? |
| 22:02.40 | clyrrad | Corydon76-dig: cuz how can you know all those includes LOL |
| 22:02.57 | clyrrad | you didnt miss anyting that i found so far |
| 22:03.02 | Corydon76-dig | There aren't that many includes |
| 22:03.14 | clyrrad | well its more than stdio.h :P |
| 22:03.19 | Corydon76-dig | Heh |
| 22:03.23 | clyrrad | printf("Hellow World"); |
| 22:03.24 | clyrrad | haha |
| 22:03.38 | Katty | anthm: you have a doc on getting an email server working with *? |
| 22:04.01 | clyrrad | if there is a skeleton.c file for making modules would prolly take a lot of guess work out of what includes you need |
| 22:04.19 | Corydon76-dig | clyrrad: app_skel ? |
| 22:04.37 | Corydon76-dig | I don't use that one, though |
| 22:04.38 | clyrrad | yea, that'd be the one :) |
| 22:04.39 | anthm | Katty, not that i can recall |
| 22:04.49 | Corydon76-dig | other than for parsing |
| 22:05.04 | clyrrad | how long did it take you to code this? |
| 22:05.12 | Katty | anthm: which one you think would be the easiest to work with? sendmail? |
| 22:05.29 | Corydon76-dig | clyrrad: the main code or everything until now? |
| 22:05.33 | clyrrad | yea |
| 22:05.37 | anthm | what are you trying to do? |
| 22:05.39 | angryuser | please what was the file where it it written who is the sync master ? |
| 22:05.51 | Corydon76-dig | clyrrad: main code took a week or so |
| 22:06.12 | clyrrad | Corydon76-dig: yea looks like alot of work and testing |
| 22:06.34 | Corydon76-dig | All bugs and features, you're looking at 6-9 months |
| 22:06.49 | clyrrad | WOW |
| 22:06.51 | _ShrikE | Katty: get an email server working in what way? For voicemail? |
| 22:07.23 | Katty | _ShrikE: yeah, never done it before. |
| 22:07.28 | Katty | _ShrikE: i've got sendmail on this box tho |
| 22:07.48 | _ShrikE | Katty: with sendmail it has always "just worked" for me. |
| 22:07.50 | Corydon76-dig | clyrrad: I prefer my beer at room temperature, but thank you |
| 22:07.55 | clyrrad | ewwwwww |
| 22:08.05 | Katty | _ShrikE: i looked through ps aux and i don't see it running. |
| 22:08.08 | Qwell | good beer doesn't need to be cold |
| 22:08.19 | Corydon76-dig | clyrrad: the cold deadened the flavor |
| 22:08.25 | clyrrad | Never had a "good" warm beer |
| 22:08.39 | Corydon76-dig | clyrrad: not warm, just room temperature |
| 22:08.41 | _ShrikE | Katty: what distro? With centos its service sendmail start |
| 22:08.56 | clyrrad | Cold cold heineken mmmmmmm |
| 22:09.01 | Katty | _ShrikE: debian |
| 22:09.05 | Corydon76-dig | clyrrad: room temperature is still almost 30 degrees cooler than your mouth |
| 22:09.13 | clyrrad | this is true |
| 22:09.17 | Katty | _ShrikE: would it be listed somewhere else, maybe? |
| 22:09.40 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 22:09.49 | _ShrikE | Katty: any obvious activity in your maillog? |
| 22:10.32 | clyrrad | Corydon76-dig: im gonna have so much fun with this module, now I can track what outbound provider was used for a call, inbound provider so much stuff I couldnt track before (well not easily anyway) |
| 22:10.48 | Katty | empty |
| 22:11.02 | clyrrad | Does anyone actually use the 'amaflags' field? |
| 22:11.20 | Katty | _ShrikE: nothing in err, info.. |
| 22:11.32 | Greek-Boy | can someone please have a look at http://pastebin.com/d66b34d05 ? I am trying to manage emergency calls |
| 22:11.38 | bsdwarrior | Im trying to have a voicemail be emailed to more then one recipient. I have two addresses comma separated. is this even possible ? |
| 22:11.39 | Katty | _ShrikE: or warn. it's all empty. like it's never been turned on. |
| 22:11.54 | _ShrikE | Katty: I am really not familiar with debian, but it sounds like as you suspected, sendmail may not be running |
| 22:12.39 | Katty | i'll try figuring out how to start it (= |
| 22:12.41 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 22:13.11 | clyrrad | bsdwarrior: in voicemail.conf I dont belive thats possible if thats what your asking... |
| 22:13.12 | _ShrikE | once running, asterisk should use it and you should see mail transactions in the maillog. |
| 22:13.57 | Katty | it's up now. i'll give it a shot |
| 22:14.05 | Katty | i wonder if that means when i reboot the server, sendmail wont' come back up |
| 22:14.34 | bsdwarrior | i have the config in a database but still. |
| 22:15.30 | Greek-Boy | so now one can look at my dialplan? :( |
| 22:15.38 | znoG | Corydon76-dig: hi there. I'm the guy that reported a bug recently with res_config_ldap (i uploaded a proper backtrace, btw). I've a quick question regarding LDAP. I'm using a 'switches' statement in extensions.ael and I want some way to Playback a file if the extension entered does not match anything by res_config_ldap. Any ideas how to do that? |
| 22:15.46 | jameswf | ~pb |
| 22:15.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:15.54 | Katty | _ShrikE: anything in voicemail.conf i need to chagne? |
| 22:16.23 | Greek-Boy | http://pastebin.com/d66b34d0 |
| 22:16.27 | clyrrad | bsdwarrior: yea but the DB just replacing the config file |
| 22:16.40 | _ShrikE | Katty: just the email address AFAIK |
| 22:17.09 | *** join/#asterisk RoyK (n=roy@ip-183-20-149-91.dialup.ice.no) |
| 22:17.40 | bsdwarrior | you would think it would know how to email multiple recipients |
| 22:17.42 | Katty | hrmm |
| 22:17.44 | Corydon76-dig | znoG: yes, use an include |
| 22:17.47 | Katty | no emails, nothing in mail.log |
| 22:17.57 | Katty | anyone familiar enough with sendmail to tell me where to look? |
| 22:18.15 | Corydon76-dig | znoG: extensions are found in the current context as literals, switch, includes, in that order |
| 22:18.24 | anthm | Katty, dunno if this still works but i made this a few years ago |
| 22:18.27 | anthm | http://www.freeswitch.org/asterisk_stuff/fake_sendmail_pl |
| 22:18.49 | anthm | you can trick asterisk into using that and it will deliver it to some smtp gateway |
| 22:18.57 | znoG | Corydon76-dig: so basically include another context with s => { __X. => Playback(some-sound) } ? |
| 22:19.08 | Katty | anthm: nah. |
| 22:19.12 | Katty | anthm: i wanna get sendmail working (= |
| 22:19.17 | Corydon76-dig | znoG: correct |
| 22:19.26 | znoG | Corydon76-dig: interesting idea. I shall try it out :) |
| 22:19.30 | bsdwarrior | I can make an alias on the mail server but then the person knows who is getting copies of what |
| 22:19.31 | znoG | Corydon76-dig: thanks a lot |
| 22:19.47 | anthm | k |
| 22:20.53 | flush | so yo |
| 22:21.00 | flush | how can i know if my fxo module is working |
| 22:21.25 | Greek-Boy | how do I prevent some priorities from not being used in an extension unless I jump to them with Goto ? |
| 22:21.27 | Corydon76-dig | flush: call the mfg for support |
| 22:21.38 | real0ne | any one here from french |
| 22:22.49 | znoG | Corydon76-dig: one last question and I'll leave you alone :) i've managed to get res_config_ldap working, I can authenticate etc. I even managed to get switch working so it looks up the extension in LDAP. The bit I'm wondering about is how to DIAL a user. Say I'm registered using SIP with 'znog' as the username. Simply setting Application to Dial and Appdata to SIP/znog doesn't work (says no route to destination). Am I doing something wrong? |
| 22:24.27 | Corydon76-dig | znoG: if you're getting that error, I'd say the error does not lie within realtime |
| 22:24.46 | Corydon76-dig | or at least not within the realtime dialplan |
| 22:24.58 | Corydon76-dig | Might still be a problem with your sippeer |
| 22:25.02 | znoG | Corydon76-dig: so I'm setting the application and appdata correctly? (Dial(SIP/znog)) |
| 22:25.09 | Corydon76-dig | Correct |
| 22:25.38 | Corydon76-dig | Do you have an ipaddr= for peer znog? |
| 22:25.54 | znoG | in res_ldap.conf? dont think so |
| 22:26.08 | Corydon76-dig | for the sippeer, that's what you'd need |
| 22:27.06 | flush | how big is asterisknow distro |
| 22:27.10 | *** join/#asterisk VaNNi (n=VaNNi@216.70.165.200) |
| 22:27.20 | jameswf | 15 feet |
| 22:27.26 | znoG | Corydon76-dig: oh .. I see that I have ipaddr, storing it in "destinationIndicator" (as I didn't see a suitable field in the asterisk.schema) |
| 22:27.31 | flush | this is huge.. |
| 22:27.41 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 22:29.06 | *** mode/#asterisk [-o putnopvut] by russellb |
| 22:29.09 | *** kick/#asterisk [putnopvut!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (war!) |
| 22:29.40 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 22:29.40 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 22:29.43 | Corydon76-dig | ooooookay |
| 22:29.53 | jameswf | heh |
| 22:30.18 | russellb | ... he started it |
| 22:30.21 | Yourname`` | Is there anyway I can make the CLI show only stuff that I want to see? :S |
| 22:30.36 | flush | is asterisknow pretty cool |
| 22:30.49 | znoG | Yourname``: sure, by editing the source ;) |
| 22:30.53 | jameswf | flush try it |
| 22:31.04 | Yourname`` | lol |
| 22:31.09 | flush | thx for the tip |
| 22:31.44 | jameswf | ~fire |
| 22:31.45 | jbot | Bender : Light a fire for a man and he's warm for a night. Light a man on fire and he's warm for the rest of his life... |
| 22:32.13 | flush | ~stab |
| 22:32.14 | jbot | ACTION stabs flush |
| 22:34.40 | jameswf | ~orgy |
| 22:34.41 | jbot | ACTION straps on his jimmy hat to preform oral sex on all the channel. |
| 22:34.54 | jbigbee | wow |
| 22:35.43 | Corydon76-dig | Okay, now I'm curious as who this 'Jimmy' is and what his hat looks like |
| 22:35.43 | flush | how can i test my fxo module.. its configured in zaptel.conf but i cant figure out how to place a call and when a call comes in asterisk phone doesnt ring |
| 22:39.23 | Corydon76-dig | flush: this is not the forum for hardware support |
| 22:39.51 | Corydon76-dig | flush: please call the reseller or mfg of your device |
| 22:39.52 | flush | i just asked if someone had an idea, im aware its a irc channel |
| 22:40.42 | flush | arr |
| 22:40.45 | kamisalami | flush: join #asterisk |
| 22:40.51 | kamisalami | oops |
| 22:40.53 | kamisalami | hahaha |
| 22:41.03 | flush | lulz |
| 22:41.05 | kamisalami | lol, i have to get some sleep, thought i was in ubuntu |
| 22:44.33 | Yourname`` | How do I control what does into /var/log/messages again? |
| 22:44.56 | Nugget | edit /etc/syslog.conf then restart syslogd |
| 22:46.30 | *** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat) |
| 22:46.30 | Yourname`` | Err, sorry, I meant /var/log/asterisk/messages |
| 22:46.42 | dogmeat | can i use asterisk to build a front end voip client similar to what skype is? |
| 22:46.58 | jameswf | ~skype |
| 22:46.58 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 22:47.06 | dogmeat | e.g. with asterisk as the back end |
| 22:47.18 | Yourname`` | logger.conf |
| 22:48.06 | dogmeat | jameswf: i want to build a client similar to skype using asterisk as the back end, is that possible? |
| 22:48.10 | znoG | anyone know, with the Realtime() function, how to access ONE particular key/value pair out of the lot? |
| 22:48.47 | jameswf | I couldn't say. rephrased as cn it be done... yes, can you I dunno |
| 22:49.34 | dogmeat | jameswf, what do you mean 'rephrased as cn' |
| 22:49.46 | jameswf | s/cn/can/ |
| 22:49.56 | drmessano | lol |
| 22:50.07 | drmessano | As my good friend david used to say |
| 22:50.14 | drmessano | "Can denotes ability" |
| 22:50.22 | drmessano | Yes you CAN |
| 22:50.37 | dogmeat | im wondering if this would be a good implementation strategy |
| 22:50.41 | jameswf | ~nowwhat |
| 22:50.42 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E\ |
| 22:50.47 | jameswf | ^^^ drmessano look at link ^^^ |
| 22:50.47 | mvanbaak | why? because I can |
| 22:51.32 | drmessano | That needs to be fixed |
| 22:51.44 | jameswf | ? |
| 22:51.50 | drmessano | jbot: forget nowwhat |
| 22:51.51 | jbot | i forgot nowwhat, drmessano |
| 22:51.59 | jameswf | oh noz |
| 22:52.00 | drmessano | jbot: forget is <reply> So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 22:52.00 | jbot | i didn't have anything called 'is <reply> so you just installed asterisk now what? http://www.youtube.com/watch?v=ulgwbvj768e' to forget, drmessano |
| 22:52.07 | drmessano | crap |
| 22:52.14 | drmessano | jbot: nowwhat is <reply> So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 22:52.15 | jbot | drmessano: okay |
| 22:52.18 | drmessano | bad paste |
| 22:52.18 | drmessano | lol |
| 22:52.24 | drmessano | the / at the end |
| 22:52.28 | drmessano | err \ |
| 22:52.38 | jameswf | have you seen that video b4? |
| 22:52.39 | mvanbaak | ~nowwhat |
| 22:52.40 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 22:52.40 | drmessano | you hit \ and enter at the same time |
| 22:52.41 | drmessano | Nope |
| 22:52.58 | drmessano | Thats terrible |
| 22:53.11 | jameswf | :) |
| 22:53.13 | drmessano | It's just like.... |
| 22:53.17 | drmessano | A PBX |
| 22:53.28 | drmessano | Thats the new Trixbox commercial |
| 22:53.40 | drmessano | It's just like.... It's just like.... a PBX |
| 22:53.44 | jameswf | if you make it to the end I think ron paul kills a kitten |
| 22:53.55 | dogmeat | mvanbaak, that clip shows a furniture show room. |
| 22:54.08 | russellb | DrAk0: commercial for what? |
| 22:54.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:54.12 | drmessano | Turn the sound up |
| 22:54.15 | russellb | DrAk0: not you ... tab completion ... |
| 22:54.18 | russellb | drmessano: for what? |
| 22:54.28 | drmessano | ~nowwhat |
| 22:54.28 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 22:54.35 | drmessano | He needs to do the trixbox commercial |
| 22:54.39 | dogmeat | Screw Rick Roll - You Got Montgomeried! |
| 22:54.55 | drmessano | dogmeat: Turn the sound up and get the damn joke already |
| 22:55.21 | dogmeat | heh, no sound card on this PC |
| 22:55.30 | drmessano | Major Fail |
| 22:55.36 | *** join/#asterisk Miseryjdog (n=Miseryjd@216.207.245.1) |
| 22:55.47 | znoG | Corydon76-dig: what do you use for "ipaddr" ? |
| 22:56.04 | jameswf | 30 minutes after watching you will still be chanting.... the mini mall rap |
| 22:56.20 | drmessano | Time to take a failboat to failville and buy plate of failcakes |
| 22:56.45 | mvanbaak | 23:54 < jbot> So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 22:56.55 | mvanbaak | he has bigger tits then my wife |
| 22:57.44 | drmessano | He's just like.... he's just like... a mini mall |
| 22:58.03 | Corydon76-dig | znoG: the IP address of the phone, in dotted quad notation |
| 22:58.34 | drmessano | I want to know what year in the 80s he went back to with his time machine to get the drum machine they used for this spot |
| 22:58.36 | znoG | Corydon76-dig: i mean which attribute do you assign it to |
| 22:58.51 | Corydon76-dig | znoG: No idea |
| 22:58.53 | znoG | Corydon76-dig: there should be one for SIP and another for IAX |
| 22:59.19 | Corydon76-dig | znoG: I know next to nothing about ldap |
| 22:59.44 | znoG | Corydon76-dig: heh fair enough, I thought you did since you were involved with a big bug report about it, or maybe I'm wrong |
| 22:59.44 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 22:59.57 | znoG | i wish LDAP got more attention, it's way cool :) |
| 23:00.03 | Corydon76-dig | I was involved in getting it closed and committed |
| 23:01.22 | lunaphyte | in 1.2, i used the command iax2 debug - it seems to have gone away with 1.4? what's the equivalent command in 1.4? |
| 23:01.49 | Corydon76-dig | iax2 set debug on |
| 23:01.57 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:02.06 | lunaphyte | ah, thanks. |
| 23:02.33 | riddlebox | wohoo sold an asterisk system! |
| 23:03.37 | lunaphyte | hmm, the choices seem to be jb off trunk - i don't see on. |
| 23:04.08 | Corydon76-dig | Just omit the 'on' then... iax2 set debug |
| 23:05.18 | lunaphyte | oh, duh - thanks. |
| 23:07.20 | znoG | riddlebox: congrats ;) |
| 23:07.41 | *** join/#asterisk nirz (n=nir@89-138-73-246.bb.netvision.net.il) |
| 23:07.42 | riddlebox | yup the only issue I have with it is the guy insisted on grandstream gxp2000 |
| 23:07.44 | riddlebox | phones |
| 23:08.18 | jameswf | ~taze |
| 23:08.19 | jbot | DON'T TAZE ME BRO |
| 23:10.03 | Corydon76-dig | ~bzzzt |
| 23:10.04 | jbot | DON'T TAZE ME BRO |
| 23:12.43 | jameswf | ~bro |
| 23:12.49 | jameswf | bah |
| 23:12.55 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 23:19.21 | *** join/#asterisk dot36 (n=br0ken@CPE-65-26-242-251.wi.res.rr.com) |
| 23:19.55 | dot36 | howdy, i just installed asterisk and i'm having some trouble. i'm behind a router and i've got xlite going with a stun address, but i don't have any sound |
| 23:20.00 | dot36 | asterisk says it connected fine |
| 23:20.07 | dot36 | anyone have a tip? |
| 23:21.36 | Qwell | ~TANSTAAFL |
| 23:21.37 | jbot | extra, extra, read all about it, tanstaafl is "There Ain't No Such Thing As A Free Lunch", from Robert A. Heinlein's "The Moon is a Harsh Mistress" Heinlein, "The cat who walks through walls" |
| 23:26.27 | CrashSys | jameswf: Any samoa's? |
| 23:28.02 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 23:29.11 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 23:34.38 | *** join/#asterisk Zone-MR (n=no@router.jrdltd.co.uk) |
| 23:34.41 | Zone-MR | Hi |
| 23:36.28 | Zone-MR | Anyone know a UK sip/iax2 ddi provider with RDNIS support? |
| 23:38.47 | *** join/#asterisk seaq (n=seaq@correo.seaq.com.co) |
| 23:38.58 | seaq | hey all |
| 23:39.11 | seaq | any problems using asterisk with RHEL 64bits?? |
| 23:39.18 | *** join/#asterisk jamesrdorn (n=jamesrdo@adsl-99-135-235-94.dsl.rcsntx.sbcglobal.net) |
| 23:42.52 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 23:45.12 | jameswf | 15 minutes bahhhh |
| 23:48.04 | *** join/#asterisk markgreene (n=markgree@c-68-62-134-226.hsd1.al.comcast.net) |
| 23:48.25 | markgreene | Does anyone in here know the status of attempts to bring PacketCable into asterisk? |
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| 23:53.00 | *** join/#asterisk hfb (n=hfb@pool-72-67-142-193.lsanca.dsl-w.verizon.net) |
| 23:53.54 | *** join/#asterisk jamesrdorn (n=jamesrdo@adsl-99-135-235-94.dsl.rcsntx.sbcglobal.net) |
| 23:54.00 | [hC] | Question about the operator=yes behavior in Voicemail(). It is unclear which context is used to look up the 'o' extension when someone presses 0 during an unavailable greeting. Anyone shed some light? |
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| 23:55.25 | _ShrikE | [hC]: I believe it needs to be in the same context that launched the Voicemail application |
| 23:56.11 | [hC] | _ShrikE: the wiki is saying the opposite. it says its the context that the voicemail box we're looking for is in in the dial plan.... but-- that makes no sense. voice mail boxes arent in the dial plan. |
| 23:57.15 | _ShrikE | Iin my case those are usually the same. |
| 23:57.27 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 23:57.32 | [hC] | well what if it was extension 100, vm box "SALES" - there is no exten => SALES in the dial plan |
| 23:57.59 | [hC] | Its just i have a multi tenant install and it seems that the only place it could possibly be getitng its wires crossed is if its searching in the default context in sip.conf |
| 23:58.01 | [hC] | which is bizarre. |
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| 23:59.05 | _ShrikE | what context is sales under in voicemail.conf? |
| 23:59.32 | *** part/#asterisk Cresl1n (n=matt@216.207.245.1) |
