| 00:00.27 | syslod | Explains how to order a PRI from VZ. Or anything else for that matter. |
| 00:00.44 | Qwell | ahh |
| 00:00.51 | Qwell | Which link? |
| 00:01.00 | harryvv | Imagine being at the table like my stepdad was to watch craig sign in the first ever cell site in the world with the chicogo mayor :) |
| 00:01.21 | Sedorox | hehe |
| 00:01.43 | *** part/#asterisk fdp32 (fdp32@200.63.70.12) |
| 00:02.15 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 00:02.15 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 00:03.34 | harryvv | became top salemand for two years in a row for MCaw Cellular ;) how times have changes. MCaw cellular turnedinto Cellular one bought out by AT&T to become AT&T wireless now bought out by singular. |
| 00:03.50 | Qwell | syslod: I think I'm missing something big on this page |
| 00:03.58 | syslod | Qwell: Tariffs |
| 00:04.11 | syslod | Then your state. T |
| 00:04.17 | Qwell | heh, they're still using the gte domain |
| 00:04.28 | harryvv | Is that for charging customers? |
| 00:04.56 | Qwell | Where exactly did Verizon come from? |
| 00:05.02 | Qwell | Did GTE get bought out, or did they change their name? |
| 00:05.06 | syslod | GTE:CONTEL etc ect |
| 00:05.13 | Qwell | big ass merger? |
| 00:05.17 | harryvv | GTW was bought out by Verizon |
| 00:05.19 | syslod | They are alot of things |
| 00:05.23 | harryvv | GTE |
| 00:05.39 | Qwell | out here, SBC made a huge deal about their new name |
| 00:05.49 | Qwell | whereas Verizon was like, "y0, we're here...you pay us now." |
| 00:06.11 | kimosabe | how do i configure kphone to work with asterisk |
| 00:06.12 | kimosabe | ?? |
| 00:06.19 | Qwell | 5' outside my front door, theres still an old access grate with the GTE name on it |
| 00:06.30 | harryvv | Qwell, go and steel it |
| 00:06.31 | harryvv | :) |
| 00:06.34 | Qwell | its huge :p |
| 00:06.41 | Qwell | I could lay down on it... |
| 00:06.56 | harryvv | For historic sale ask if you can keep it :) |
| 00:07.00 | harryvv | sake |
| 00:07.02 | harryvv | :) |
| 00:07.04 | kimosabe | does any one use k phone with asterisk need a hand |
| 00:07.22 | harryvv | kim, that is for kde? |
| 00:07.36 | Qwell | syslod: I see "Pending Projects, Effective Tariffs, and Archived Tariffs" |
| 00:07.42 | kimosabe | k phone |
| 00:07.49 | harryvv | windows? |
| 00:07.52 | terrapen | <PROTECTED> |
| 00:08.01 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 00:08.23 | kimosabe | linux |
| 00:08.33 | syslod | Effective |
| 00:08.38 | harryvv | terrapen good at understanding why asterisk does not load because of some permission problems? |
| 00:08.45 | Qwell | This side is damn difficult to navigate, heh |
| 00:09.00 | harryvv | does not load at boot that is. Loads fine when done manually in root. |
| 00:09.19 | Qwell | site* |
| 00:10.09 | bkw_ | 1 gig just doesn't seem like enuf |
| 00:10.11 | bkw_ | :P |
| 00:10.13 | Qwell | bkw_: What kind? |
| 00:10.17 | Astrisk-boob | hello everyone! |
| 00:10.22 | bkw_ | I need a pair of 1gb 3200's |
| 00:10.30 | bkw_ | DDR baby |
| 00:10.34 | Qwell | yeah, can't help you there, heh |
| 00:10.46 | harryvv | ddr with ecc? |
| 00:10.58 | bkw_ | if it will work in an iMac sure |
| 00:11.01 | bkw_ | :P |
| 00:11.04 | Silik0n | hah |
| 00:11.07 | bkw_ | wants more |
| 00:11.08 | Astrisk-boob | at work here were on voip.. can i test from a pc connected to the same network? im super green to viop |
| 00:11.19 | bkw_ | so you can send your donations to brian@bkw.org via paypal |
| 00:11.21 | bkw_ | mmmmkay |
| 00:11.26 | syslod | Qwell: Thats the point. |
| 00:11.27 | terrapen | :P my dual g5 has 1.5Gb |
| 00:11.36 | Qwell | syslod: ahh |
| 00:11.43 | bkw_ | this imac is hella fast with 1gig |
| 00:11.46 | Qwell | syslod: so far the only number I've seenm is the one I would have called |
| 00:11.49 | harryvv | I have ecc. its needed for my "Bul*$%" Rma board that is at the mfg now :) A dual opteron. |
| 00:11.58 | bkw_ | it puts that poor 2.8ghz P4 to shame |
| 00:12.01 | charles___ | bkw_ Do you know which of the Calling Card app is still being maintained ? |
| 00:12.08 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
| 00:12.14 | charles___ | bkw_, the cvs in ast cc show july the last change |
| 00:12.21 | Silik0n | OS X ++ |
| 00:12.25 | charles___ | bkw_, areski looks very good |
| 00:12.25 | bkw_ | don't ask me that charles___ |
| 00:12.35 | charles___ | bkw_, don't you like billing stuff ? |
| 00:12.41 | bkw_ | fuck no |
| 00:13.01 | harryvv | what billing binaries are you using |
| 00:13.01 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
| 00:13.07 | syslod | billing stuff? |
| 00:13.18 | harryvv | hehe |
| 00:13.18 | ariel_ | bkw_, did you go the Philippines? |
| 00:13.35 | bkw_ | ariel_, not yet |
| 00:13.37 | bkw_ | I have been sick |
| 00:13.40 | bkw_ | 3 weeks of fucking crap |
| 00:13.43 | ariel_ | wow |
| 00:14.03 | bkw_ | ok now you can all paypal brian.west@mac.com |
| 00:14.03 | ariel_ | So that is why you have not been around here. |
| 00:14.04 | harryvv | Good oll PI |
| 00:14.04 | bkw_ | :P |
| 00:14.13 | bkw_ | ariel_, ya |
| 00:14.33 | harryvv | I could have gone to the PI when in the service |
| 00:14.34 | ariel_ | harryvv, yep it was great when I lived there. |
| 00:14.38 | jesster | i have a new asterisk setup and listening to mp3s sounds scratchy. I have the very same mp3 file on other asterisk so I know the file format plays ok. |
| 00:14.47 | harryvv | Would have gone to CAFB |
| 00:14.50 | bkw_ | your mpg123 version is wrong maybe? |
| 00:14.55 | jesster | bkw_: lemme check |
| 00:15.10 | kimosabe | does nay one use k phon e |
| 00:15.14 | file[laptop] | day after day, same mpg123 problems and fixes... I swear |
| 00:15.21 | jesster | bkw_: 0.59s-r9 |
| 00:15.27 | Qwell | /topic mpg123 0.59r only. NEXT! |
| 00:15.36 | file[laptop] | Qwell: that was in there once... |
| 00:15.39 | Qwell | heh |
| 00:15.42 | file[laptop] | still, people didn't understand |
| 00:15.43 | Qwell | Doesn't surprise me |
| 00:15.50 | Qwell | Also doesn't surprise me |
| 00:15.58 | bkw_ | you got it |
| 00:16.01 | ariel_ | you know asterisk has a make mpg123 built into it. use it. |
| 00:16.01 | bkw_ | 0.59r ONLY |
| 00:16.05 | jesster | heh, well google with site:lists.digium.com and various moh search strings yielded off topic stuff |
| 00:16.15 | bkw_ | who search for moh |
| 00:16.17 | bkw_ | damn boi |
| 00:16.23 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 00:16.49 | jesster | i did not use mpg123 in the search, i suspect that would have been key to finding it much quicker on my own |
| 00:16.49 | ariel_ | Well it's dinner time. I got to start cooking see you all later. |
| 00:16.50 | shmaltz | ~seen ManxPower |
| 00:16.52 | jbot | manxpower <~eric@dsl-209-205-172-111.i-55.com> was last seen on IRC in channel #asterisk, 20h 46m 56s ago, saying: 'Nugget, We should get tzanger's opinion!'. |
| 00:17.15 | bkw_ | http://lists.digium.com/pipermail/asterisk-users/2004-May/045735.html |
| 00:17.19 | afrosheen | wow my zaptel cards are acting crazy |
| 00:17.20 | harryvv | anyone ever see this when asterisk is being loaded by init.d eb 15 15:38:40 WARNING[1069]: Unable to open IAX timing interface: Permission denied |
| 00:17.20 | harryvv | Feb 15 15:38:41 WARNING[1069]: Unable to open '/dev/zap/channel': Permission denied |
| 00:17.20 | harryvv | Feb 15 15:38:41 ERROR[1069]: Unable to open channel 1: Permission denied |
| 00:17.20 | harryvv | here = 0, tmp->channel = 1, channel = 1 |
| 00:17.34 | eKo1 | harryvv: Check the permission on /dev/zap |
| 00:17.55 | *** part/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 00:18.05 | syslod | Anyone got any suggestions for configuring the new T1/E1 card to work with a 12FXO/12FXS CA Bank I ? |
| 00:18.07 | bkw_ | harryvv, you're not running as root are you |
| 00:18.08 | jesster | bkw_: ya, didn't see that since it doesn't match the content i thought would help for searches, ie moh, music on hold, sound quality, static, etc etc.. |
| 00:18.19 | harryvv | ek01 all the files are crw-r--r-- |
| 00:18.31 | bkw_ | harryvv, are you running as a NON ROOT USER? |
| 00:18.34 | eKo1 | harryvv: eh, who owns them? |
| 00:18.46 | harryvv | bkw no, this is only when debian is booting it gets these errors. |
| 00:18.55 | harryvv | its trying to load asterisk |
| 00:18.57 | bkw_ | doesn't debian muck i up to run as asterisk? |
| 00:19.04 | harryvv | but |
| 00:19.10 | bkw_ | no but |
| 00:19.13 | harryvv | When logged in as root it loads fine. |
| 00:19.16 | bkw_ | what UID is asterisk running as? |
| 00:19.18 | shmaltz | doesn anybody know how to use Local channel, I'm having problems with it |
| 00:19.22 | harryvv | manually that is |
| 00:19.31 | bkw_ | I bet you $$ the init script is fucked up trying to start it as a non root user |
| 00:19.35 | bkw_ | thats all it could be really |
| 00:19.48 | Mavvie | jbot: kewl start ? |
| 00:19.56 | afrosheen | I have 6 zap channels, and the only channel actually plugged in is 7. any reason why it's not answering even though asterisk thinks it is? |
| 00:20.12 | harryvv | bkw, possibly. bkw UID not sure. |
| 00:20.20 | bkw_ | no its all it could be |
| 00:20.46 | dsmouse | afrosheen: does it mention anything about the state of the line? |
| 00:20.55 | afrosheen | dsmouse: lemme look |
| 00:21.09 | bkw_ | some things never change :P |
| 00:21.22 | dsmouse | 'cause if it does you might want to disable busydetection and progress and see if then it works |
| 00:21.35 | harryvv | Well, I dont know Everything about debian or linux so need to know about the UID or user ID. It should be loading it as asterisk but by default /dev/zap is owned by root. |
| 00:21.35 | afrosheen | dsmouse: it thinks it's picking up the line and playing the ivr |
| 00:21.44 | kimosabe | does nay one know if i have to add a diffrent configuration to my sip.conf for k-phone instead of sipura |
| 00:21.44 | harryvv | chowned by root that is |
| 00:22.24 | dsmouse | oh, nm then |
| 00:22.26 | file[laptop] | bkw_: deep breaths |
| 00:22.33 | bkw_ | its a computer regardless of the OS on it you should know how to use it or make it do what you want |
| 00:22.46 | Zaw | i'm looking for suggestions on which OS to use for asterisk. i'd rather use FreeBSD if possible, as i'm most familiar with it. am i better off using a Linux distro, and if so which one is recommended (my personal preferences aside)? |
| 00:22.46 | afrosheen | dsmouse: it's weird, it really thinks it's doing it, but the phone calling in never hears it pick p |
| 00:23.00 | bkw_ | Zaw, then use gentoo |
| 00:23.06 | bkw_ | if you're a freebsd nut like me |
| 00:23.10 | ReVoK | oki, so i've install asterisk, and i want to use it as a server between 2 pc, to make VoIP call (using H323), but asterisk is install, but doesn't seems to have h323 module.? |
| 00:23.10 | bkw_ | gentoo will fit more like a glove |
| 00:23.12 | kimosabe | zaw xircom |
| 00:23.15 | afrosheen | Zaw: I've had success with mandrake |
| 00:23.21 | Zaw | bkw_: is that because freebsd does not have good support for asterisk as of yet? |
| 00:23.27 | afrosheen | lol |
| 00:23.28 | bkw_ | Zaw yep |
| 00:23.29 | bkw_ | and trust me |
| 00:23.31 | bkw_ | use gentoo |
| 00:23.35 | bkw_ | if you want asterisk and love freebsd |
| 00:23.40 | bkw_ | because you'll feel more at home with it |
| 00:23.43 | eKo1 | ReVoK: Why H.323? Use SIP. |
| 00:23.45 | Zaw | hmm |
| 00:23.48 | redder86 | bkw_: it can change, though. I used to think that about the HylaFAX discussion groups (mailing lists). Back in the 4.1beta days we got the same old questions over and over and over. It was horrible. Things have gotten better since then. The HOWTO cleared up a lot of that. Improvements with the documentation, both installed (manpages) and on-line (website and archives) are to account for most of that. Plus, it helped to do some coding in suc |
| 00:23.54 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.133) |
| 00:24.02 | harryvv | <PROTECTED> |
| 00:24.02 | harryvv | uid=105(asterisk) gid=105(asterisk) groups=105(asterisk),29(audio) |
| 00:24.06 | ReVoK | eKo1 : because it imposed by our teatcher :x |
| 00:24.07 | bkw_ | redder86, ya its getting there... |
| 00:24.09 | eKo1 | bkw_: gentoo isn't freebsd? huh? |
| 00:24.16 | bkw_ | redder86, we moved all our inbound faxing to rxfax on asterisk |
| 00:24.25 | bkw_ | and use Hylafax for outbound spooling only now |
| 00:24.29 | redder86 | bkw_: understandable |
| 00:24.31 | shmaltz | for those that are intersted: |
| 00:24.32 | shmaltz | http://story.news.yahoo.com/news?tmpl=story&ncid=1211&e=10&u=/afp/20050215/tc_afp/airtransporttelecoms&sid=96001017 |
| 00:24.35 | bkw_ | harryvv, thats why |
| 00:24.41 | eKo1 | ReVoK: Impose this on your teacher: ..!.. |
| 00:24.43 | redder86 | bkw_: especially since you were disabling V.17 anyway |
| 00:24.47 | bkw_ | redder86, yep |
| 00:24.53 | bkw_ | but you can enable V.17 in spandsp |
| 00:24.59 | bkw_ | its in there ifdef'ed out |
| 00:25.01 | redder86 | brw_: that is true |
| 00:25.07 | bkw_ | but I said fuck it |
| 00:25.07 | redder86 | bkw_: does it work well? |
| 00:25.08 | bkw_ | haha |
| 00:25.13 | bkw_ | no I didn't wanna chance it |
| 00:25.25 | bkw_ | ;) |
| 00:25.30 | redder86 | bkw_: I'd bet on a lot of quirks, as it hasn't been tested much, I'm sure. |
| 00:25.30 | file[laptop] | bkwwwwwwwwwwwww |
| 00:25.37 | bkw_ | fiiiiiiiiiiiiiiiiiiiiiiiile |
| 00:25.44 | file[laptop] | bkwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwww |
| 00:25.53 | bkw_ | fileeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee |
| 00:25.55 | bkw_ | now behave |
| 00:26.03 | file[laptop] | growing eh? |
| 00:26.04 | bkw_ | and acts like a 3yr old |
| 00:26.10 | Corydon-w | Welcome to the Mutual Admiration Society |
| 00:26.14 | file[laptop] | have you been looking at the special pics again? |
| 00:26.17 | ReVoK | eKo1 : actualy it part of a project and we have to use h323, so whate can i do to have the server suport h323? and how to use it after? |
| 00:26.19 | bkw_ | i'll always be like a kid |
| 00:26.21 | bkw_ | no reason to grow up |
| 00:26.38 | eKo1 | ReVoK: Don't use Asterisk, use OH323 or something. |
| 00:26.59 | bkw_ | outtolunc, I did come back.. but I might go away agay |
| 00:27.01 | bkw_ | er again |
| 00:27.03 | bkw_ | agay.. har har |
| 00:27.04 | afrosheen | freud |
| 00:27.04 | outtolunc | haha k |
| 00:27.09 | Corydon-w | Actually, both H323 channels suck |
| 00:27.16 | jesster | can I put a call on hold, and have the caller hear a choice of hold music |
| 00:27.25 | redder86 | bkw_: once Asterisk gets some adequate and concise documentation - stuff that will help newbies without generating questions - as well as perhaps improving some common stumblingblocks - at least in a way to assist the newbie in getting it fixed, then the monotony of stuff you see in the support forums will lessen, and the quality of the discussion materially generally will improve. |
| 00:27.29 | bkw_ | jesster, I find thats stupid |
| 00:27.31 | bkw_ | and a waste of time |
| 00:27.33 | bkw_ | don't bother |
| 00:27.35 | afrosheen | jesster: like 'for freedom rock, press 1' |
| 00:27.39 | bkw_ | if your people are on hold too much you're doing something wrong |
| 00:27.54 | bkw_ | redder86, when I get asterlinux fully done |
| 00:27.55 | bkw_ | it will |
| 00:28.04 | jesster | bkw_: i wasn't asking for opinions :) |
| 00:28.08 | Silik0n | redder86: you havent read too many OSS support mailing lists have you? |
| 00:28.14 | Zaw | bkw_: is your avoidance of asterisk on freebsd due to hardware compatibility issues or configuration problems? i noticed that there's a freebsd port for it, which is why i'm asking. |
| 00:28.18 | shmaltz | afrosheen: I actualy have such a menu (press 1 for this song....) |
| 00:28.24 | jesster | afrosheen: yea |
| 00:28.25 | bkw_ | Zaw, all of the above |
| 00:28.40 | afrosheen | jesster: shlameel is your man |
| 00:28.45 | redder86 | Silik0n: say what you want to say, don't say what you said, because I don't understand what you're trying to say. |
| 00:29.03 | bkw_ | Silik0n doesn't speak in complete sentences at times |
| 00:29.11 | afrosheen | redder86: isn't that an eminem song |
| 00:29.11 | bkw_ | its hard to follow him sometimes :P |
| 00:29.23 | redder86 | afrosheen: M&Ms ? |
| 00:29.24 | bkw_ | I mean that with love Silik0n |
| 00:29.31 | bkw_ | ok i'm getting my ibook and heading to the living room |
| 00:29.32 | bkw_ | bbl |
| 00:29.32 | Zaw | bkw_: my problem is this. we have suse linux and freebsd on all of our production systems. as such, our techs are only familiar with these OSs and their various quirks. i have reservations of throwing gentoo into production without staff that's supported it before. |
| 00:29.35 | afrosheen | redder86: a joke |
| 00:29.36 | Zaw | damn. |
| 00:29.50 | file[laptop] | VONNNNNNN, soon soon soon |
| 00:29.57 | dsmouse | anyone here use broadvoice? |
| 00:29.57 | afrosheen | Zaw: so put it on suse then |
| 00:30.06 | redder86 | Silik0n: if you were trying to say that many OSS support forums suck, then I think you're right. |
| 00:30.12 | jesster | afrosheen: what do you mean? shlameel? |
| 00:30.55 | afrosheen | jesster: woops, I mean shmaltz, he just said he has his MOH like that |
| 00:30.55 | Silik0n | redder86: i'm a long time OpenSOurce user... OpenBSD is one of the things I use quite often and it is very well documented and the answers are usually on the FAQ however the number of questions on the general mailing list on a daily basis that can be answered with RTFM or RTFF is insame |
| 00:30.55 | Zaw | afrosheen: have you run it on SuSe before, i'm guessing that it won't matter really so long as it's a Linux which distro you use |
| 00:30.55 | redder86 | Silik0n: I was trying to say that there is a way for the organizers and developers of those projects to elevate the level of discussion, though. |
| 00:30.58 | afrosheen | Zaw: as long as the kernel is decent you should be alright |
| 00:31.07 | Zaw | afrosheen: ok, thanks. |
| 00:31.20 | Silik0n | they can try all they want, but people will still ask the same questions that have been asked and answered 12T times |
| 00:31.20 | afrosheen | Zaw: I've had problems with tdm cards and kernel 2.6.x on mandrake but the 2.4.x series is perfect. |
| 00:31.26 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 00:31.26 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 00:31.30 | bkw_ | MOOSE PENIS |
| 00:31.37 | Silik0n | thats just the way it works |
| 00:31.58 | afrosheen | zaw: google for asterisk suse or look for suse on voip-info.org |
| 00:32.02 | Zaw | afrosheen: so i should stick to the 2.4 kernel rather than the 2.6? that's pretty old |
| 00:32.04 | afrosheen | I'm sure it's been done many times |
| 00:32.04 | redder86 | Silik0n: there will always be the dummies who don't read instructions and who don't pay attention, but that can be minimized |
| 00:32.09 | Zaw | afrosheen: ok, will do |
| 00:32.29 | afrosheen | Zaw: I'm just saying, different distros and different kernels have quirks, so ultimately the distro matters less than the kernel |
| 00:32.54 | Zaw | afrosheen: gotcha. thanks for the site, off to go do some reading. |
| 00:33.00 | afrosheen | zaw: :) |
| 00:33.24 | afrosheen | zaw: while you're at it, check out AMP at amp.voxbox.ca |
| 00:33.42 | Zaw | ok |
| 00:33.57 | Zaw | wow, nice. |
| 00:34.09 | afrosheen | zaw: we've been using it for awhile, it's effective |
| 00:34.20 | redder86 | Silik0n: but, I don't consider myself a dummy, yet I asked some real big dummy-like questions both here and on the Asterisk mailing lists simply because I couldn't find the answer anywhere where I was looking. As it turns out I simply looked in the wrong place, but what I'm saying is that over time the project can be improved to at least prevent people who are willing to do the learning from needing to ask dummy-like questions. Asterisk isn't an |
| 00:34.57 | dsmouse | wow that was long. you were cut off at "Asterisk isn't an" |
| 00:35.03 | outtolunc | ... |
| 00:35.21 | redder86 | dsmouse: anywhere near that. |
| 00:35.39 | Silik0n | redder86 I understand that I've done it myself, Documentation is someting thats VERY VERY nice, and I do understand that it needs to done... i would personally love to see better docs, but if I wrote them it would end up readling worse the chinese VCR instructions |
| 00:36.11 | fafnir | I've personally never had a problem with chinese VCR instructions |
| 00:36.13 | redder86 | Silik0n: yeah. And it really takes someone in a position of "authority" to write proper docs. |
| 00:36.22 | Silik0n | but even so in the real world even with excellent documentation, and things like mailing list archives that are google searchable etc, people will still ask the same questions over and over |
| 00:36.23 | *** join/#asterisk _PiGreco_ (~a@adsl-215-48.38-151.net24.it) |
| 00:36.44 | Silik0n | the great thing about OSS is you have the source you can write you're own docs... |
| 00:36.53 | _PiGreco_ | re ppl |
| 00:36.54 | Mavvie | redder86: I agree, but who wants to documentation on a project which is so chaotic? |
| 00:37.05 | Silik0n | that too... |
| 00:37.06 | bkw_ | asterisk changes so often |
| 00:37.10 | bkw_ | the docs would be fucked up in no time |
| 00:37.11 | bkw_ | haha |
| 00:37.15 | herag | is there a way I can change how often an sip registration is refreshed? |
| 00:37.15 | redder86 | Silik0n: that's true, but at least it would be nice to be able to limit those to those that just plain didn't read the instructions. |
| 00:37.24 | Qwell | Stupid question. Can I use vars in voicemail.conf? |
| 00:37.35 | Qwell | ie ${NORTHEXTEN} |
| 00:37.42 | Mavvie | redder86: digium should have a technical writer for it who makes the initial draft and keeps it up to date with the changes. |
| 00:37.52 | redder86 | Mavvie: doc changes need to happen with CVS commits, simultaneously. Doc changes should be part of the patch. |
| 00:37.53 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 00:38.27 | Mavvie | redder86: that's how it was done in my first job and that's was the best project documentation ever. No technical hacker, a technical writer :-) |
| 00:38.28 | shmaltz | ~seen ManxPower |
| 00:38.31 | jbot | manxpower <~eric@dsl-209-205-172-111.i-55.com> was last seen on IRC in channel #asterisk, 21h 8m 35s ago, saying: 'Nugget, We should get tzanger's opinion!'. |
| 00:38.31 | redder86 | Mavvie: I don't think that Digium has any commitment to documentation. I doubt that they'd hire anyone to do it. Nobody is going to pay them to. |
| 00:38.39 | Silik0n | someone really just needs to sit down and go thru the code and do some docbook patches but I dont see that as a high prority right now unless you just wanna get started on it |
| 00:39.28 | redder86 | I really wish that I had the kind of time to devote to Asterisk that I have been able to devote to HylaFAX, but I just don't. I'm all tapped out. Someone else will have to do it for Asterisk. |
| 00:40.20 | redder86 | It could be a technical writer - not the code writer - just as long as it happened at the same time. Just so long as all commits had to also pass through that technical writer's desk so that the documentation adjustments got made. |
| 00:41.07 | redder86 | Often someone will submit a patch to HylaFAX development without a doc update. We don't commit it without the doc updates, and so I usually will write those myself. |
| 00:41.48 | Mavvie | redder86: you're in the hylafax devteam? |
| 00:41.49 | redder86 | There's much, much more activity with Asterisk development than with HylaFAX. So, whoever is going to be involved there will need more time. |
| 00:41.56 | redder86 | Mavvie: yes |
| 00:42.16 | Mavvie | redder86: way cool. We're using your stuff for our deskfax server (~400 numbers) |
| 00:42.26 | redder86 | Mavvie: I'm glad to hear that. |
| 00:42.29 | shmaltz | redder86, souds great, but do c this happening anytime soon to* |
| 00:43.12 | redder86 | shmaltz: there are serious problems with how Asterisk is developed and maintained that prevent that kind of stuff from happening any time soon. |
| 00:43.42 | dsmouse | anyone here use teliax? |
| 00:44.01 | shmaltz | redder86, I am not sure if I agree on the part that there are serious problems, but it will not happen any time soon |
| 00:44.25 | Essobi | anyone looked at ICD? |
| 00:45.32 | Moc | redder86, I got a new Faxing Server on the way, as soon I finish this basic chan_unistim |
| 00:45.36 | redder86 | shmaltz: it *could* happen soon, but there are things that prevent it from happening, and those things have to do with the development structure, the way that the project is managed, that prioritize other things over things like documentation |
| 00:45.48 | Moc | Ive done the windows client for fax printing |
| 00:45.58 | Moc | look pretty much like rightfax ;) |
| 00:46.02 | redder86 | Moc: cool |
| 00:46.32 | shmaltz | redder86, at this point things are getting done without the documentation fairly well |
| 00:46.32 | Moc | here is a example GUI : http://pbx.moctel.com/SightFaxUtils.exe |
| 00:46.36 | Moc | it need .net thought |
| 00:46.38 | Essobi | We just use Email. |
| 00:46.53 | shmaltz | I mean I can get the new features with a bit of playing with it |
| 00:47.06 | Essobi | Everyone knows how to use email, and you can use it from anywhere. :) |
| 00:47.10 | Moc | I got the printing handeling to work, but Im currently looking into a client/server communication + How I will structure my database |
| 00:47.17 | shmaltz | I look at the bugnotes and after playing it usualy works, as they say RTFM |
| 00:47.23 | Essobi | Plus it fits nicely with Fax to Email gateways too. |
| 00:47.33 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.we.client2.attbi.com) |
| 00:47.55 | Moc | Essobi, alot of people still use fax, and will continue to use fax for legal reason |
| 00:48.06 | Moc | allthrought those reason are stupid... |
| 00:48.34 | redder86 | Ah, using a fax machine is way easier than using a scanner and attaching a file to an e-mail. |
| 00:48.36 | Mavvie | Moc: that hasn't stopped the telex from being eliminated :-) |
| 00:48.41 | *** join/#asterisk JunK-Y (~junky@modemcable056.110-81-70.mc.videotron.ca) |
| 00:48.48 | _PiGreco_ | redder86: is there any IRC channel for hylafax discussions afayk? |
| 00:49.28 | redder86 | _PiGreco_: I had thought about starting up a #hylafax here, and I actually have before, but I'm not sure that it would have much traffic. |
| 00:49.55 | terrapen | we must have the strangest music on hold of any company out there |
| 00:50.16 | redder86 | _PiGreco_: the mailing list these days is even rather quiet compared to, say, a year ago. |
| 00:50.36 | _PiGreco_ | yeah, i have noticed that |
| 00:51.16 | _PiGreco_ | mailing lists are something that stay there anyway, irc discussions vanish |
| 00:51.21 | redder86 | _PiGreco_: I think that there is less of a demand, quantitatively, for individuals and home users to have a fax system. |
| 00:51.24 | _PiGreco_ | but i prefer irc anyway :) |
| 00:51.39 | redder86 | _PiGreco_: interesting |
| 00:51.44 | Mavvie | _PiGreco_: you should post them to bash.org if you want them to stay alive :-) |
| 00:52.00 | terrapen | what do y'all use for your music on hold? |
| 00:52.04 | shmaltz | I like this one: |
| 00:52.04 | terrapen | anything interesting? |
| 00:52.05 | shmaltz | http://lists.digium.com/pipermail/asterisk-users/2005-February/089606.html |
| 00:52.10 | _PiGreco_ | redder86: well i had the occasion to set up a couple of fax server machine these days |
| 00:52.11 | Mavvie | terrapen: Blond with Hanging on the Telephone. |
| 00:52.25 | Essobi | Moc The stuff that MOST people are faxing these days, are in electronic format already. Word docs, PDFs, Forms, etc. |
| 00:52.29 | Mavvie | Blondie that is |
| 00:52.46 | _PiGreco_ | redder86: and its quite easy to make it work (apart a problem i got probably with a buggy/broken modem) |
| 00:53.02 | _PiGreco_ | Mavvie: irc logs are ok too :P |
| 00:53.03 | terrapen | what is that |
| 00:53.05 | redder86 | Essobi: not so, from what I see |
| 00:53.17 | Essobi | *SHRUG* Different places I guess. |
| 00:53.23 | redder86 | Essobi: most people are faxing hand-signed documents, completed forms, etc. |
| 00:53.50 | _PiGreco_ | and you have to scan them anyway..yeah, thats boring |
| 00:53.55 | nickv111 | Anyone here run asterisk on amd64? |
| 00:53.56 | Essobi | True enough, hand-signed docs do a lot. |
| 00:54.01 | *** join/#asterisk letherglov (~letherglo@8036aa5e.resnet.ucsd.edu) |
| 00:54.05 | *** join/#asterisk IsMe (~some@219.95.224.115) |
| 00:58.11 | *** join/#asterisk atmel (~vlad@wireless-am5.ucsd.edu) |
| 01:00.11 | file[laptop] | what a scary concept |
| 01:00.33 | JunK-Y | lo |
| 01:00.37 | JunK-Y | what a day! |
| 01:03.44 | *** part/#asterisk eKo1 (~bernd@207.42.191.66) |
| 01:07.10 | *** join/#asterisk hilkiah (~hilkiah@firewall.marpin.dm) |
| 01:08.03 | hilkiah | anybody home? |
| 01:09.51 | hilkiah | has anyone managed to get distinctive rings working? |
| 01:10.22 | *** join/#asterisk mxmasster (~maxc@rottie.media.net) |
| 01:10.25 | mxmasster | hi all |
| 01:10.29 | hilkiah | hi |
| 01:10.38 | hilkiah | r u an asterisk guru?? |
| 01:10.42 | mxmasster | i'm trying to configure an account with broadvoice |
| 01:10.51 | hilkiah | have u ever gotten distinctive rings working? |
| 01:10.56 | mxmasster | does anyone here have experiance with them? |
| 01:11.28 | hilkiah | we're either the only active ones here or something must be seriously wrong!!!! |
| 01:13.10 | hilkiah | anyone gotten distinctive rings working on internal lines??? |
| 01:13.18 | JerJer | zaptel? sure |
| 01:13.31 | file[laptop] | JerJer: So are you aborting teh VON trip? |
| 01:13.44 | hilkiah | good good |
| 01:13.48 | hilkiah | here's my problems |
| 01:13.51 | JerJer | file[laptop]: too expensive |
| 01:13.54 | hilkiah | i have 2 fxo + 2 fxs |
| 01:13.56 | JerJer | damn word travels fast |
| 01:13.57 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 01:14.04 | file[laptop] | JerJer: tight community :p |
| 01:14.09 | hilkiah | what i want is for a diff. ring when a call comes from a fxs line |
| 01:14.24 | hilkiah | thus i can know the diff. between an outside (fxo) call and an internal (fxs) call |
| 01:14.37 | hilkiah | how do i do this? |
| 01:15.01 | *** join/#asterisk wsmith (~wsmith@67.95.66.69) |
| 01:15.54 | wsmith | How can I add an interface to a Queue externally (i.e. through the Manager interface)? |
| 01:15.54 | afrosheen | hilkiah: I don't know |
| 01:16.13 | herag | is there a way I can change how often an sip registration is refreshed? |
| 01:16.29 | afrosheen | herag: yes |
| 01:17.09 | *** join/#asterisk kimosabe (~kimo@216.60.60.103) |
| 01:17.16 | hilkiah | i'm sure asterisk supports this scenario |
| 01:17.25 | afrosheen | herag: you can use qualify=yes for your extensions |
| 01:17.42 | afrosheen | herag: my phones check with * every 2 seconds |
| 01:17.45 | kimosabe | is there a way to make the sipura dial faster , ? becuase when i dial i dont get a ring for like 7 to 9 seconds |
| 01:19.04 | afrosheen | herag: we did it to get around the nat issue when we reset asterisk when making changes |
| 01:20.03 | ashus_ | hi. im trying to install asterisk-1.0.5+bristuff-0.2.0-RC5. when i try to start asterisk i get "== Parsing '/etc/asterisk/zapata.conf': Found; Unable to specify channel 1: No such device or address...chan_zap.so: load_module failed, returning -1". can anyone give a pointer whats goes wrong here? |
| 01:20.40 | Qwell | Are the zap modules loaded? Does ztcfg show that its ok? |
| 01:20.40 | afrosheen | ashus_: looks like ..uh... you didn't load the zaptel module? |
| 01:20.52 | moonwick | kimosabe: tweak your dialplan |
| 01:20.59 | afrosheen | ashus_: do a ztcfg -vvv and see what it says |
| 01:21.26 | kimosabe | moonwick what exaclly can i look for example |
| 01:21.52 | kimosabe | moonwick or a how or link thanks |
| 01:21.58 | ashus_ | afrosheen: 0 channels configured |
| 01:22.17 | ashus_ | afrosheen: but zaphfc os loaded successfully |
| 01:22.20 | ashus_ | is |
| 01:22.41 | afrosheen | ashus_: so something isn't configured right..what does lsmod | grep wx give you |
| 01:22.43 | moonwick | log in as advanced and as admin, then go to line 1 |
| 01:22.45 | *** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net) |
| 01:22.52 | moonwick | er, user 1 |
| 01:23.01 | ashus_ | afrosheen: nothing |
| 01:23.08 | wsmith | <PROTECTED> |
| 01:23.13 | Darwin35 | give me you weak your broken your none working PBX systems |
| 01:23.36 | Darwin35 | no |
| 01:23.49 | Darwin35 | it will not reload till after the channels are cleared |
| 01:24.01 | afrosheen | ashus_: what kind of card is it again |
| 01:24.17 | *** join/#asterisk MrEntropy (~entropy@ppp55-252.lns1.adl2.internode.on.net) |
| 01:24.19 | MrEntropy | yo |
| 01:24.21 | herag | afrosheen: but this isn't really for a context, it's the register line in my sip.conf |
| 01:24.23 | wsmith | Darwin35, So, as long as there are active channels, asterisk will wait to reload? |
| 01:24.45 | ashus_ | afrosheen: 1 hfc-s + 1 fritz(capi) problem seems to be hfc related |
| 01:24.49 | Darwin35 | it is suppost to unless you have a bug |
| 01:25.16 | Darwin35 | if it reloads before the channels are clear it should be reported |
| 01:25.20 | wsmith | I see. So, in a very active PBX, reloads can take awhile.... |
| 01:25.51 | Darwin35 | yes |
| 01:25.53 | ashus_ | afrosheen: do u have a bristuff enabled asterisk running? |
| 01:26.05 | afrosheen | ashus_: nope |
| 01:26.11 | Darwin35 | yeah was a nice day |
| 01:26.12 | hilkiah | guys....can one set each zap channel to ring in a diff. tone??? |
| 01:27.40 | wsmith | Darwin35, Does extensions reload work the same way? |
| 01:27.53 | Darwin35 | yes it is suppost to |
| 01:28.37 | wsmith | I'm trying to non-destructively change queue assignments from asterisk manager. Know of any cheap hacks? |
| 01:29.08 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 01:30.19 | ashus_ | afrosheen: might be my zapata.conf is wrong (noob here). http://rafb.net/paste/results/tf0kAr62.html do u see something obvious? |
| 01:30.32 | *** part/#asterisk _PiGreco_ (~a@adsl-215-48.38-151.net24.it) |
| 01:33.13 | sivana | on a fresh system, with no * software installed at all, only a T100P... should I see something when I do cat /proc/interrupts? |
| 01:33.42 | JunK-Y | i think yes. |
| 01:33.47 | sivana | or is it only when I load the zaptel drivers? |
| 01:33.59 | JunK-Y | never plugged a T100P w/out asterisk :) |
| 01:34.12 | sivana | I've only inserted it into the PCI slot :P |
| 01:34.21 | sivana | nothing plugged into it |
| 01:34.36 | sivana | fresh install... was wondering if it should show up in interrupts |
| 01:34.51 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 01:35.10 | Mneumonic | grrrrr |
| 01:35.18 | Mneumonic | anyone know why my music on hold wouldnt be working? |
| 01:35.26 | Mneumonic | mpg123 is installed and running |
| 01:35.35 | sivana | did you configure musiconhold.conf? |
| 01:35.35 | Mneumonic | but i place a call on hold and get NOTHING! |
| 01:35.38 | JunK-Y | sivana: try it, my guess is ya should see something. |
| 01:35.49 | Mneumonic | sivana - i used custom in musiconhold.conf |
| 01:35.54 | Mneumonic | anything specific in there? |
| 01:35.57 | sivana | JunK-Y: I don't see nothing in /interrupts :) |
| 01:36.11 | sivana | JunK-Y: I'll try it again when I download zaptel |
| 01:36.23 | sivana | Mneumonic: uncomment the first line |
| 01:36.44 | Essobi | Anyone used any good WiFi Phones? |
| 01:36.49 | Mneumonic | default => quietmp3:/var/lib/asterisk/mohmp3 |
| 01:36.51 | Mneumonic | that line? |
| 01:36.58 | sivana | ya, that's the default setting |
| 01:37.35 | Mneumonic | and that worked for you? you didnt have to enable anything in modules.conf or anywhere else? |
| 01:37.46 | sivana | no |
| 01:37.55 | JunK-Y | Mneumonic: i use default => mp3:/var/lib/asterisk/mohmp3 |
| 01:37.57 | sivana | compile,make,install mpg123 |
| 01:38.06 | sivana | modify musiconhold.conf.. that's it |
| 01:38.12 | JunK-Y | and reload |
| 01:38.15 | sivana | :) |
| 01:38.16 | afrosheen | yessir |
| 01:38.20 | Mneumonic | hmm.... hold on, lemme try |
| 01:39.09 | Mneumonic | err... im losing my mind.. .when i downloaded from CVS, where would the code for mpg123 go? |
| 01:39.26 | sivana | not sure.. go into /usr/src/asterisk |
| 01:39.32 | sivana | and do 'make mpg123' |
| 01:39.53 | sivana | then go into the mpg123 directory it creates and do 'make install' |
| 01:41.30 | Mneumonic | ok done, now all i need to do is reload and it should work after uncommenting the 1st line in musiconhold.conf? |
| 01:41.34 | JunK-Y | Mneumonic: if ya do a show modules like music, ya see res_musiconhold? |
| 01:41.39 | *** join/#asterisk vmlinuxz (~dc@wsip-68-15-253-140.dl.dl.cox.net) |
| 01:42.09 | Mneumonic | Junk - yes i do |
| 01:42.51 | JunK-Y | try it, ya should see some output in ur CLI |
| 01:43.05 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 01:43.10 | Mneumonic | got nothing |
| 01:43.20 | jalsot | did anybody try IAXy with fax? |
| 01:43.21 | Mneumonic | just called my cell from my X-Lite softphone |
| 01:43.26 | Mneumonic | placed the call on hold and got nothing |
| 01:43.36 | sivana | jalsot: fax over ip not reliable |
| 01:43.46 | sivana | unless private lan |
| 01:44.05 | jalsot | sivana: I know, it is in the LAN with ulaw, theoretically should work |
| 01:44.24 | sivana | jalsot: on a private lan, yes, it should be ok |
| 01:44.30 | sivana | not over the internet |
| 01:44.35 | jalsot | sivana: did you try? |
| 01:44.52 | sivana | jalsot: many times.. less than 10% success rate |
| 01:45.08 | jalsot | I tryed with asterisk-1_0 CVS and success rate is 0% [on LAN] |
| 01:45.21 | sivana | on LAN... I've used Sipura |
| 01:45.23 | *** join/#asterisk pbxman (~tmcarter@ip68-226-15-136.nc.hr.cox.net) |
| 01:45.41 | jalsot | I was curious how is it work CVS HEAD, and I'm surprised |
| 01:46.05 | jalsot | fax went through [even the box wasn't in the LAN this time] |
| 01:46.27 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
| 01:46.37 | *** join/#asterisk mrproper_ (~mrproper_@61.95.55.242) |
| 01:46.48 | mrproper_ | has anyone had any luck with SIP video to h323 video? |
| 01:46.49 | jalsot | I'm wondwering what is the difference between HEAD and 1_0 in the meaning of IAX2+IAXy co-operation |
| 01:47.05 | sivana | jalsot: not sure |
| 01:47.51 | *** join/#asterisk netsurfer (netsurfer@81-6-224-129.dyn.gotadsl.co.uk) |
| 01:48.22 | jalsot | sivana: do you think HEAD has better IAX2, or better zaptel support? |
| 01:49.01 | afrosheen | is something wrong with 1.0.x's iax2 and zap support? |
| 01:49.07 | sivana | jalsot: I have no idea. I use HEAD because of the new features, not sure if protocols are better |
| 01:49.12 | JerJer | head is far superior code |
| 01:49.21 | JunK-Y | afrosheen: i prefer head too. |
| 01:49.25 | file[laptop] | yes, give head a chance ... *G* |
| 01:49.29 | sivana | hehe |
| 01:50.07 | Darwin35 | head hell I want tails |
| 01:50.08 | jalsot | JerJer: thx, is it good to use HEAD in a production box? |
| 01:50.30 | JunK-Y | jalsot: i do all day long. |
| 01:50.58 | Darwin35 | whats the cvs command to get head now days |
| 01:51.07 | jalsot | JunK-Y: thx, just a good, working local copy of CVS is needed, right? |
| 01:51.24 | JunK-Y | jalsot: i dont understand ur questiuon. |
| 01:51.32 | JunK-Y | Darwin35: cvs checkout asterisk |
| 01:51.43 | Darwin35 | ok |
| 01:52.08 | jalsot | JunK-Y: while CVS HEAD is development version, it can easily have some untested codes, which are not good enough for production. |
| 01:52.20 | Darwin35 | 1.0.5 is nice I just wish maxsobo would egt the fbsd port uptodate |
| 01:52.26 | JunK-Y | jalsot: and stable doesnt have a lot of things too. |
| 01:52.37 | Darwin35 | Max has disapierd |
| 01:52.38 | JunK-Y | i run head and im satisfy with it. |
| 01:52.40 | afrosheen | Darwin35: tails are better any day |
| 01:52.46 | jalsot | JunK-Y: :) |
| 01:52.52 | *** join/#asterisk Caede (~chatzilla@h000f3d364f20.ne.client2.attbi.com) |
| 01:52.57 | netsurfer | I havent got head in over a week |
| 01:52.59 | Darwin35 | heads I win tails you loose |
| 01:53.10 | jalsot | JunK-Y: do you know about noticable improvements in the zaptel area? |
| 01:53.14 | afrosheen | Darwin35: I have a coin from a xxx theater with that on it :) |
| 01:53.18 | jalsot | JunK-Y: or in IAX2 area... |
| 01:53.39 | Darwin35 | I have a shirt with it on it |
| 01:54.01 | JunK-Y | jalsot: no, im still working on the same zaptel. |
| 01:54.27 | Darwin35 | shower time bbiab |
| 01:56.04 | jalsot | JunK-Y: do you use the zaptel from the stable version? |
| 01:56.33 | JunK-Y | jalsot: no cvs too. |
| 01:57.26 | netsurfer | anyone tried FastSMS ? |
| 02:02.20 | shido6 | ZzZz |
| 02:02.22 | *** join/#asterisk doughecka (~Doug@doughecka.user) |
| 02:02.38 | doughecka | So I have this 7940... without a power supply... |
| 02:02.47 | doughecka | what can I use to power this thing... =D |
| 02:03.05 | shido6 | 30 dollar power supply |
| 02:03.17 | doughecka | true |
| 02:03.21 | shido6 | or a poe injector I think |
| 02:03.27 | shido6 | want one |
| 02:03.32 | shido6 | I cna ship u a power supply |
| 02:03.35 | doughecka | sure |
| 02:03.59 | shido6 | pm me ur details |
| 02:04.27 | afrosheen | better be a special power supply |
| 02:04.33 | afrosheen | those ciscos are very weird |
| 02:07.46 | jcims | we're getting ready to drop coin on 5 new 7960's...are there any good reviews of them out there? |
| 02:08.24 | doughecka | ;) |
| 02:09.05 | *** join/#asterisk mrverizone (~mrverizon@pa-robinson1b-88.pit.adelphia.net) |
| 02:09.11 | mrverizone | Hello |
| 02:09.20 | mrverizone | any one home |
| 02:09.34 | doughecka | nope |
| 02:09.42 | mrverizone | ok, thanks |
| 02:09.45 | mrverizone | :O |
| 02:10.22 | doughecka | please hold |
| 02:10.43 | nickv111 | Can asterisk be used to call people using sip? |
| 02:10.51 | *** part/#asterisk jcims (~jcims@cpe-69-135-121-57.columbus.rr.com) |
| 02:11.13 | doughecka | sure |
| 02:11.18 | nickv111 | Awesome |
| 02:11.20 | doughecka | if you have a sip phone to talk through it |
| 02:11.25 | nickv111 | Oh |
| 02:11.35 | nickv111 | I need to find an open source sip phone |
| 02:11.38 | doughecka | ah |
| 02:11.51 | nickv111 | I was wondering if asterisk could serve as that |
| 02:11.53 | doughecka | voip-wiki.com |
| 02:11.56 | shido6 | i hate seeds in my orange |
| 02:12.02 | doughecka | well, never tried using asterisk as a phone |
| 02:12.21 | doughecka | lol |
| 02:13.01 | Darwin35 | ok back |
| 02:13.20 | Darwin35 | consol |
| 02:13.25 | Darwin35 | workd great |
| 02:13.32 | mrverizone | I am using sip |
| 02:13.35 | doughecka | Darwin35! :) |
| 02:13.41 | Darwin35 | Dough |
| 02:13.42 | mrverizone | I am using a zyxel phone with a sip server, |
| 02:13.49 | mrverizone | yeah, |
| 02:13.53 | mrverizone | I know darwin |
| 02:13.55 | doughecka | 48V |
| 02:13.56 | Luke-Jr | Anyone know a good IAX/SIP client for OpenZaurus? |
| 02:13.56 | Darwin35 | I have a xyxel and asterisk |
| 02:13.59 | mrverizone | he knnows some thing |
| 02:14.04 | doughecka | what the crap is it doing that it needs 48 volts |
| 02:14.05 | mrverizone | yeah, me to |
| 02:14.07 | Luke-Jr | KPhone transmits static :( |
| 02:14.17 | mrverizone | I belive you set it up for me darwin |
| 02:14.23 | mrverizone | mrverizone = jack earl |
| 02:14.27 | Darwin35 | ahh ok |
| 02:14.30 | mrverizone | :) |
| 02:14.30 | Darwin35 | I like it |
| 02:14.40 | mrverizone | yeah, how much did it cost you |
| 02:14.44 | Darwin35 | but I found anopther phone coming out soon |
| 02:15.03 | mrverizone | darwin how much did that zyxel phone cost you |
| 02:15.17 | mrverizone | :( |
| 02:15.17 | Darwin35 | I got it for free |
| 02:15.19 | nickv111 | lol, sflphone says it only costs one gnu general public license per user in your system... Wow, those are so expensive ;) |
| 02:15.31 | Darwin35 | beta tester |
| 02:15.33 | mrverizone | yeah, I paid for it, |
| 02:15.47 | mrverizone | darwin got it for free, at my cost, the cost of doing business |
| 02:15.51 | Darwin35 | but I am not getting the new one |
| 02:15.59 | mrverizone | you need to send if back |
| 02:16.00 | doughecka | ~seen bkw |
| 02:16.02 | jbot | bkw <~bkw@h55l114.delphi.afb.lu.se> was last seen on IRC in channel #freedesktop, 68d 11h 45m 5s ago, saying: 'when doing startx I get lots of Symbol _mesa_Uniform2fvARB from module /usr/X11R6/lib/modules/extensions/libGLcore.a is unresolved! lines'. |
| 02:16.03 | mrverizone | we need to get it replace |
| 02:16.06 | doughecka | ~seen bkw_ |
| 02:16.07 | jbot | bkw_ <~brian@bkw.developer.and.friend.of.asterisk> was last seen on IRC in channel #asterisk, 1h 38m 56s ago, saying: 'haha'. |
| 02:16.24 | mrverizone | me |
| 02:16.27 | mrverizone | na |
| 02:16.30 | mrverizone | do not know him |
| 02:16.30 | Darwin35 | mr verizon call me |
| 02:16.37 | mrverizone | call you what, |
| 02:16.42 | mrverizone | I am out of bad name |
| 02:16.44 | mrverizone | names |
| 02:16.46 | Darwin35 | on the landline |
| 02:16.53 | Darwin35 | punk |
| 02:17.02 | mrverizone | punker |
| 02:17.07 | Darwin35 | why you dis me like that |
| 02:17.08 | mrverizone | punk rocker |
| 02:24.54 | *** join/#asterisk QRPartner (~andy@ns1.accu-com.com) |
| 02:24.58 | QRPartner | hello |
| 02:25.42 | *** join/#asterisk kks (~kks@203.115.210.253) |
| 02:26.01 | *** join/#asterisk Mneumonic (Mnemonic@ool-18ba58b4.dyn.optonline.net) |
| 02:26.01 | QRPartner | When setting callerid the following can be used correct? |
| 02:26.01 | QRPartner | SetCallerID("John Doe" <123>) |
| 02:26.16 | JunK-Y | ya |
| 02:26.32 | QRPartner | Is there a limit to the charatcters you can use in the " "? |
| 02:26.39 | QRPartner | like , .- etc |
| 02:26.58 | JunK-Y | i dont think so. |
| 02:27.17 | JunK-Y | but the length has a limit for sure. |
| 02:27.33 | QRPartner | Where could I find that? |
| 02:27.46 | QRPartner | I searched SetCallerID, but didn't see it |
| 02:28.14 | JunK-Y | look in the code |
| 02:28.22 | JunK-Y | i use SETCIDNum instead. |
| 02:28.27 | JunK-Y | and SetCIDName |
| 02:28.53 | mrverizone | ok what happen to the call |
| 02:28.57 | mrverizone | darwin are you there |
| 02:29.02 | mrverizone | Hello |
| 02:29.07 | mrverizone | Darwin35 |
| 02:29.09 | doughecka | hi! |
| 02:29.21 | mrverizone | Hello doughecka |
| 02:29.44 | mrverizone | for the person that was asking, I am using the zyxel wireless sip phone |
| 02:29.59 | mrverizone | we have had a lot of luck with the asterisk box and this phone |
| 02:30.24 | mrverizone | we have set up extiontion, and are able to use the phone any where I can find an open access wifi point |
| 02:30.29 | Darwin35 | you hung up on me |
| 02:30.33 | Darwin35 | prick |
| 02:30.43 | mrverizone | if you have any any question, email me at jack@jackearl.com |
| 02:30.47 | mrverizone | :) |
| 02:30.55 | mrverizone | you will be okay, darwin |
| 02:31.04 | mrverizone | i think the t-mobile connection fail |
| 02:31.07 | Darwin35 | or bsdtech@runbox.com |
| 02:31.15 | mrverizone | should have been on my broad voice phone |
| 02:31.19 | mrverizone | with asterisk |
| 02:31.26 | mrverizone | asterisk kick ass |
| 02:31.34 | mrverizone | system |
| 02:31.34 | Darwin35 | call me on the zyxel |
| 02:31.37 | mrverizone | na |
| 02:31.41 | Darwin35 | lol |
| 02:31.42 | mrverizone | i am getting ready for sleepy |
| 02:31.51 | mrverizone | it is past my bed time |
| 02:32.00 | mrverizone | watching a movie, on the histroy channel |
| 02:32.03 | mrverizone | and going to sleep |
| 02:32.04 | mrverizone | see ya |
| 02:32.06 | Darwin35 | yep you old men have to go to bed early |
| 02:32.12 | tzanger | yeah |
| 02:32.12 | mrverizone | good night men |
| 02:32.21 | Darwin35 | night |
| 02:32.50 | doughecka | yo Darwin35! |
| 02:32.55 | doughecka | hows the mini asterisk box doin |
| 02:32.56 | Faithful | ls |
| 02:33.23 | Darwin35 | I have asterisk and bebian down to 200 megs |
| 02:33.24 | doughecka | passwd |
| 02:33.28 | doughecka | cool |
| 02:33.40 | Darwin35 | working on a case design |
| 02:33.52 | kks | is * support LDAP authentication (sip) yet? |
| 02:33.56 | Darwin35 | going to use a mini-itx |
| 02:34.15 | doughecka | yea, the dual proc mini-itx box? |
| 02:34.24 | Darwin35 | with a cf socket and a pcmcia socket |
| 02:34.40 | Darwin35 | this is a single cpu one the dual cpu is here also |
| 02:34.40 | doughecka | sweet |
| 02:35.00 | Darwin35 | the pcmcia is where I am putting a wireless card right now |
| 02:35.06 | doughecka | ah |
| 02:35.12 | Darwin35 | the board has dual nic onboard |
| 02:35.34 | Darwin35 | the cf drivesocket will have a ibm micro drive I think |
| 02:35.38 | doughecka | ah |
| 02:35.41 | doughecka | cool |
| 02:35.41 | Darwin35 | to statrt |
| 02:35.49 | Darwin35 | for vm storage |
| 02:36.11 | doughecka | neat |
| 02:36.18 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 02:36.20 | Darwin35 | then I can market it |
| 02:36.24 | vmlinuxz | Hey guys, I'm trying to get sjphone to connect to my asterisk server, but I keep getting errors saying registration failed messages. I have been over the configuration and it looks good, any ideas? |
| 02:36.33 | Darwin35 | as a mini pbx/soho |
| 02:36.52 | doughecka | how much |
| 02:37.00 | Darwin35 | 10 exten unit with 10 g720 licenses |
| 02:37.03 | Grooby | anyone here uses usb handsets w/ their softphones? |
| 02:37.19 | doughecka | ah |
| 02:37.56 | Darwin35 | well have to figure out a final cost |
| 02:38.00 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 02:38.16 | Darwin35 | it will depend on if we get this new wifi sip phone or not |
| 02:38.29 | Darwin35 | or if we use a basic sip to start |
| 02:38.37 | doughecka | :) |
| 02:38.55 | Darwin35 | I have to do some pricing |
| 02:39.20 | *** join/#asterisk d-tech (~dtc@node-423a1ebb.cle.onnet.us.uu.net) |
| 02:39.26 | Darwin35 | and I still have to get festival working right |
| 02:40.14 | d-tech | anyone aware of a cheap FXS PCI solution? |
| 02:40.40 | Darwin35 | the tdw400 |
| 02:40.43 | Darwin35 | tdm |
| 02:40.54 | Darwin35 | its only 125 |
| 02:41.06 | d-tech | bare maybe |
| 02:41.18 | Darwin35 | single port |
| 02:41.36 | Darwin35 | 305 for a full 4 port |
| 02:43.14 | Darwin35 | I want a 12 port card |
| 02:43.30 | d-tech | haha ... your funny |
| 02:43.36 | Darwin35 | there is one |
| 02:43.50 | Darwin35 | its made by another company |
| 02:43.59 | d-tech | voicetronix? |
| 02:44.06 | Darwin35 | think so |
| 02:44.14 | d-tech | 1350 |
| 02:44.57 | Darwin35 | the openswitch |
| 02:45.07 | doughecka | anyone here that can help me with the cisco firmware update to sip? |
| 02:45.16 | d-tech | probably cheap to TI and a channel bank at that point |
| 02:45.30 | d-tech | cheaper even |
| 02:45.46 | Darwin35 | I want it to for a small building for phone in each unit |
| 02:46.07 | Darwin35 | my frint runs a SRO and wants to put in a phone system |
| 02:46.33 | Darwin35 | 10 rooms + both front and back door for entry |
| 02:46.58 | mtqh | doughecka: What version are you running now? |
| 02:47.11 | Darwin35 | that would do the job |
| 02:47.19 | Darwin35 | but thats alot for a card |
| 02:47.26 | d-tech | yup! |
| 02:47.42 | doughecka | mtqh: none |
| 02:48.04 | doughecka | its running the callmanager version right now |
| 02:48.18 | Darwin35 | Computer Telephony Switch 6/12-port |
| 02:48.18 | Darwin35 | DNVT-V6PCIC |
| 02:48.25 | mtqh | doughecka: check what the version is |
| 02:48.51 | Darwin35 | add 555 for the 12 port |
| 02:48.57 | doughecka | mtqh: cant do that either... No power supply, I was just kinda getting everything ready for when I do get the firmware.. :) |
| 02:49.15 | mtqh | doughecka: It depends on what the currect version is on there |
| 02:49.18 | Darwin35 | 990 for the 6port |
| 02:49.20 | doughecka | well |
| 02:49.22 | doughecka | its brandnew |
| 02:49.28 | mtqh | doughecka: You have to upgrade them in order .... kinda.... |
| 02:49.28 | doughecka | so I assume its the latest and greatist |
| 02:49.43 | Sedorox | what would cause Asterisk to stop picking up on the zap interface? |
| 02:50.02 | d-tech | IRQ conflict |
| 02:50.17 | Sedorox | besides IRQ conflict... |
| 02:50.47 | Darwin35 | bad config |
| 02:50.55 | doughecka | solar flares |
| 02:50.55 | d-tech | bad card |
| 02:50.59 | Darwin35 | the drivers corrupt |
| 02:51.06 | Darwin35 | no power |
| 02:51.18 | doughecka | static cling from nylon underwear |
| 02:51.22 | Sedorox | lol |
| 02:51.23 | Darwin35 | the vibrator function cuts out |
| 02:51.32 | Sedorox | its a X100P clone... |
| 02:51.36 | doughecka | EWWWWWW |
| 02:51.40 | Darwin35 | lube on the circuts |
| 02:51.47 | Sedorox | ? |
| 02:51.49 | d-tech | ah ... a clone?! |
| 02:51.50 | *** join/#asterisk foxb_ (~chatzilla@hip2-247-93-148-209.dlup-mtl.progression.net) |
| 02:52.10 | tzanger | hmm |
| 02:52.14 | Darwin35 | clone me pls |
| 02:52.17 | doughecka | Sedorox: have you checked for corrosion on the 47.2 ohm resisters on the modem? |
| 02:52.17 | tzanger | asterisk -p just seems to give it -11 priority |
| 02:52.19 | Darwin35 | wich clone |
| 02:52.24 | Darwin35 | wich chipset |
| 02:52.48 | d-tech | probably the intel 537 |
| 02:52.55 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 02:52.55 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 02:52.58 | doughecka | its on sale |
| 02:53.02 | doughecka | hail bkw_ |
| 02:53.04 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 02:53.09 | d-tech | yeah ...$6.88 |
| 02:53.09 | Nukemizer | When getting a PRI to sysnc with a PBX, must you have an appropriate dial plan ? Meaning, would my b channels not come into service is i have not corect dial plan information in extensions.conf ?\ |
| 02:53.27 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 02:53.30 | Sedorox | ummm |
| 02:53.38 | Sedorox | hold on.. I know mine here is intel.. but not the same |
| 02:53.49 | d-tech | 3200? |
| 02:54.37 | Sedorox | [irq11: wcfxo0] |
| 02:54.40 | Sedorox | not IRQ conflict... |
| 02:55.06 | d-tech | are you sure? |
| 02:55.13 | Darwin35 | break time |
| 02:55.14 | Sedorox | how else do you check in FBSD? |
| 02:55.24 | foxb_ | Hi ! I've read some of documentation for * but there is no information about Processin power required. Can somebody help me? |
| 02:55.24 | Darwin35 | check what |
| 02:55.35 | Darwin35 | <== is on fbsd |
| 02:55.39 | Sedorox | what irq's are in use by what.. |
| 02:55.41 | Darwin35 | and linux |
| 02:55.42 | Sedorox | e.g. conflicts |
| 02:55.46 | Sedorox | I know in linux.. just not fbsd |
| 02:55.48 | Sedorox | wcfxo0@pci0:3:0: class=0x078000 card=0x00038086 chip=0x0001e159 rev=0x00 hdr=0x00 |
| 02:56.05 | Darwin35 | give me a min I have to remember the new command |
| 02:56.09 | Sedorox | kk |
| 02:56.13 | Qwell | foxb_: Depends on how many active calls you want at once. |
| 02:57.11 | nickv111 | Can anyone recommend one? |
| 02:57.12 | foxb_ | from 60-120 |
| 02:57.16 | nickv111 | Preferably a GUI |
| 02:57.46 | Sedorox | kphone I think |
| 02:57.57 | Sedorox | linkphone |
| 02:58.01 | Sedorox | linphone* |
| 02:58.03 | nickv111 | I tried kphone, but I got no sound |
| 02:58.24 | nickv111 | I couldn't get a gui on linphone, for some reason it needed gnome and not just gtk |
| 02:59.19 | chaoscon | Sedorox: IRQ's are vmstat -i |
| 02:59.22 | chaoscon | :P |
| 02:59.31 | doughecka | sieg hial kram |
| 02:59.38 | Qwell | foxb_: No private messages |
| 02:59.58 | foxb_ | ok |
| 02:59.59 | Sedorox | chaoscon: you told me before.. but just vmstat.. not the -i :-p so it never looked right |
| 03:00.06 | Sedorox | irq11: wcfxo0 271077080 795 |
| 03:00.10 | Sedorox | ok.. so it is on its on irq |
| 03:00.11 | doughecka | vmstat... |
| 03:00.16 | doughecka | that looks like a vmware command |
| 03:00.17 | doughecka | =D |
| 03:00.29 | chaoscon | Sedorox: I did tell you after figuring it out :P |
| 03:00.31 | Nukemizer | when guessing at the right configuration for my T1 card what proccess will give me consistant relfect my changes ? meaning in asterisk "stop now " "reload" or reboot ? which is the best to use in this case. |
| 03:00.41 | Sedorox | lol |
| 03:00.44 | doughecka | LOl |
| 03:00.51 | doughecka | Nukemizer: restart now |
| 03:00.53 | doughecka | =D |
| 03:00.57 | Sedorox | ok.. so what else could it be then... if no irq sharing... |
| 03:01.19 | chaoscon | like I said, my machine is tempremental :P |
| 03:01.28 | Nukemizer | thanks |
| 03:02.50 | foxb_ | Qwell: I plan to have 60 and more calls |
| 03:03.31 | Sedorox | funny.. I don't even see my fxo in my vmstat - |
| 03:03.32 | Sedorox | -i |
| 03:04.46 | chaoscon | did you load the drivers in? |
| 03:05.10 | foxb_ | Hi ! I've read some of documentation for * but there is no information about Processin power required. Can somebody help me? I plan 60 or more calls |
| 03:06.01 | *** join/#asterisk unixgeek (~unixgeek@216-220-234-197.exploremaine.com) |
| 03:07.00 | unixgeek | Good evening. Does anyone have any experience with D-Link's DVG-1402 VOIP router? |
| 03:10.59 | sivana | do I need "acpi" for anything? |
| 03:11.29 | doughecka | power management |
| 03:11.35 | doughecka | if its a server that stays on all the time... |
| 03:11.39 | PatrickDK | you don't need acpi, but acpi is nice |
| 03:11.48 | PatrickDK | acpi is still nice for servers |
| 03:11.50 | doughecka | yea |
| 03:11.59 | doughecka | for actully powering it off, correct? |
| 03:12.04 | PatrickDK | normally you can get tempaures and stuff from it |
| 03:12.16 | PatrickDK | no, I would never use it to power off anything but a laptop |
| 03:12.30 | PatrickDK | acpi has required stuff in it needed for smp |
| 03:12.33 | sivana | ok.. it's assigns IRQ too? |
| 03:12.36 | PatrickDK | so if you have a smp server ya need it |
| 03:12.44 | doughecka | ah |
| 03:12.57 | doughecka | smp/hyperthreading |
| 03:12.57 | Sedorox | thats weird... my FXO card doesn't have a IRQ... |
| 03:13.13 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4118784.sympatico.ca) |
| 03:13.15 | sivana | Feb 16 11:02:38 knox-1 kernel: ACPI: PCI interrupt 0000:00:08.0[A] -> GSI 11 (level, low) -> IRQ 11 |
| 03:13.28 | sivana | that's when I modprobe wct1xxp |
| 03:13.41 | sivana | and it has the same IRQ as eth0 |
| 03:14.12 | PatrickDK | sivana, na, apic does the irq thing |
| 03:14.23 | PatrickDK | acpi does the over 4g memory thing |
| 03:14.35 | PatrickDK | apic is for multicpu, heh, got it confused |
| 03:14.37 | sivana | PatrickDK: right |
| 03:14.42 | PatrickDK | acpi does the >4g memory PAE |
| 03:15.00 | PatrickDK | as far as servers go |
| 03:16.07 | sivana | maybe I need to explicitly set the IRQ on the slot |
| 03:16.42 | PatrickDK | sivana, that shouldn't work |
| 03:16.45 | Sedorox | is it even possible to run a FXO w/o a irq? |
| 03:17.00 | PatrickDK | sivana, changing the pci slot is normally the only possible solution |
| 03:17.11 | sivana | PatrickDK: there's only one in this box |
| 03:17.26 | PatrickDK | well, ya can try, it depends how they setup the pci bridge |
| 03:17.28 | sivana | in the BIOS I can set it explicit or leave it auto |
| 03:17.37 | PatrickDK | but normally eth0 and that pci slot will always use the same irq |
| 03:17.55 | PatrickDK | unless you can reroute the irq on the pci card to go to a different irq pin on the pci slot |
| 03:18.11 | sivana | I see.. so what's my options? :) |
| 03:18.29 | PatrickDK | I would try explicit, I give that a 10% chance of working |
| 03:18.48 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
| 03:18.57 | PatrickDK | otherwise, your stuck, different motherboard |
| 03:19.16 | PatrickDK | unless you really feel like retracing the pci ethernet card |
| 03:19.30 | PatrickDK | I mean the telelphone adaptor |
| 03:19.44 | sivana | shit.. this is suppose to be my * appliance :) |
| 03:20.08 | *** join/#asterisk Chuji (Chuji@pcp09929633pcs.tulipgrove.tn.nash.comcast.net) |
| 03:20.26 | NormAst | Hi all. |
| 03:20.26 | sivana | so the fact that they share the same IRQ will screw up operations? |
| 03:20.32 | syslod | sup |
| 03:20.37 | NormAst | sivana: yup |
| 03:20.46 | nickv111 | Linphone keeps crashing for me now and kphone has no sound and doesn't do anything when I call any number |
| 03:20.46 | PatrickDK | sivana, it shouldn't, by pci spec |
| 03:20.55 | nickv111 | Just sits there, and I can click disconnect |
| 03:20.57 | PatrickDK | in actuallty, your milage will vary |
| 03:21.12 | sivana | ya.. ok |
| 03:21.19 | PatrickDK | either it won't, or it will, about 50/50 |
| 03:21.27 | syslod | Anybody had experenice with AB I or II with FXO? |
| 03:21.28 | sivana | I don't have the physical box in front of me.. so I'll try the explicit setting tomorrow |
| 03:21.30 | dsmouse | nickv111: 'lsof | grep dsp' as root? |
| 03:21.33 | sivana | then pray :)P |
| 03:21.41 | NormAst | sivana: What system are you using for your * appliance. |
| 03:22.40 | PatrickDK | damn, my digium card isn't sharing an irq, amazing |
| 03:22.53 | sivana | ya |
| 03:22.56 | PatrickDK | I didn't even think to check that when I moved it around today |
| 03:23.15 | WilliamK | anyone seen coppice online lately? |
| 03:23.51 | syslod | WilliamK: I haven't |
| 03:23.55 | sivana | ~seen coppice |
| 03:23.57 | jbot | coppice <~chatzilla@245.195.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 1d 10h 30m 30s ago, saying: 'I wonder how many bounties actually get paid out'. |
| 03:24.02 | nickv111 | dsmouse: Sound works fine, it's just kphone that isn't working |
| 03:24.25 | nickv111 | I want linphone really, but when I click the "Call or Answer" button, it crashes |
| 03:24.44 | WilliamK | anyone know? |
| 03:24.46 | syslod | Argh!! * => channel bank suks. |
| 03:24.59 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 03:25.02 | *** join/#asterisk angler_ (~angler@suid.digium.com) |
| 03:25.04 | nickv111 | Heh, and gdb makes it just segfault right away |
| 03:25.20 | Chuji | syslod : adit 600 is good |
| 03:25.29 | Nukemizer | if I change dchan=9 in zaptel.conf "pri show span1" tells me that my D channel is still 24. Is ther another place to indicate my dchannel number ? |
| 03:25.52 | syslod | Chuji: I think I'm just having probs in general. I have tried a Adtran 750 and a CA AB I |
| 03:25.58 | syslod | Both do the same thing. |
| 03:26.23 | syslod | I can see signaling with a T-Berd but nothing is happening with * |
| 03:26.26 | Chuji | I've been real happy w/ the adit |
| 03:26.45 | syslod | Mind sharing your zapata and zaptel file? |
| 03:26.59 | Chuji | Yeah, they are pretty simple though |
| 03:27.44 | syslod | :) simple is good |
| 03:27.51 | Chuji | WHat is * complaining about? |
| 03:27.59 | Chuji | The framing? |
| 03:28.11 | syslod | Nothing. Everything seems to work but when I try to call nothing happens. |
| 03:28.23 | syslod | zttool says no alarm and * loads with no errors. |
| 03:28.36 | *** join/#asterisk BuckRogers (~none@ool-18bce89c.dyn.optonline.net) |
| 03:28.37 | Chuji | you got your fxo fxs declarations reversed right? |
| 03:28.39 | BuckRogers | hello |
| 03:28.42 | BuckRogers | all |
| 03:28.47 | Chuji | What is it fxo or fxs? |
| 03:29.07 | *** join/#asterisk clint_ (~clint@snap.helixsystems.com) |
| 03:29.14 | syslod | I've tried every combination and tried to map it out. FXO->FXS->FXO |
| 03:29.22 | BuckRogers | anyone here follow that ebay auction for the 500 sipura boxes that started at 100 dollars |
| 03:29.27 | syslod | I have a FXO and FXS. |
| 03:29.32 | *** part/#asterisk foxb_ (~chatzilla@hip2-247-93-148-209.dlup-mtl.progression.net) |
| 03:29.37 | syslod | The CA AB I has 12 of each. |
| 03:30.24 | Chuji | have you stuck your zaptel and zapata on pastebin? |
| 03:30.28 | Chuji | I'll take a look at them |
| 03:30.32 | syslod | I will just a sec |
| 03:31.02 | Chuji | Although I'm flying blind with that channel back |
| 03:31.04 | Chuji | bank |
| 03:31.15 | Chuji | Only point of reference I have is the adit |
| 03:32.46 | BuckRogers | do you think its better to have ata's do the start functions like *69 or have the asterisk server do it and keep the ata's dumb |
| 03:34.08 | syslod | http://pastebin.ca/5916 this is the adtran 750 connected currently |
| 03:35.35 | BuckRogers | any take'ers |
| 03:35.51 | Chuji | syslod : I thought it was 12 channels each |
| 03:36.19 | jetscreamer | !test |
| 03:36.29 | Chuji | !ticle |
| 03:36.38 | syslod | I have two channel banks. I can't get either one to work. The Adtran 750 is connected now it has 24 FXS ports. |
| 03:37.05 | Chuji | t100p or 410? |
| 03:37.30 | afrosheen | BuckRogers: keep the ata's dumb, centralize your configs |
| 03:37.55 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 03:37.58 | syslod | Its the new T1/E1 combined card. I'm not having any problems with the 4 port card and PRI. |
| 03:38.07 | syslod | I think is called the 110? |
| 03:38.08 | BuckRogers | yeah thats what i am talking about |
| 03:38.16 | BuckRogers | afrosheen |
| 03:38.42 | Defraz | Hello all, I am trying to get my asterisk box to talk to a NexTone iserver, I am having trouble figuring out the extention line to use for out going calls, I have it registering with the iserver |
| 03:38.45 | syslod | Anyone here familar with BAF, EMI, or AMA files? |
| 03:38.46 | Defraz | but I can't get a call to go out. |
| 03:39.12 | Chuji | syslod : and ztcfg, dmesg, and asterisk don't complain at all? |
| 03:39.35 | BuckRogers | afrosheen: what would be most featureless ata on the market |
| 03:39.44 | Defraz | I am expecting a _1NXXNXXXXXX and I want to pass to the ISERVER 2NXXNXXXXXX |
| 03:39.57 | syslod | Chuji: No |
| 03:40.07 | Defraz | Can I plaste my line to the chan and set you all take a look? |
| 03:40.18 | afrosheen | BuckRogers: no clue really |
| 03:40.24 | Chuji | Defraz : Just one line? |
| 03:40.30 | Defraz | yea |
| 03:40.35 | Chuji | go for it |
| 03:41.01 | Defraz | exten => _1NXXNXXXXXX,1,Dial(SIP/2085559760@65.101.69.113:5060/${EXTEN:2}) |
| 03:41.07 | Chuji | syslod : I'm stumped man. Doesn't look like a problem at all |
| 03:41.10 | *** join/#asterisk neuro_[rus] (~neuro_[ru@212.176.51.231) |
| 03:41.27 | BuckRogers | what i mean is that all the ata needs to do is creat the dtmf signals or send the voip equalilents to the asterisk server the asterisk server would do all the intellegence |
| 03:41.35 | BuckRogers | so if i hit star 69 |
| 03:41.37 | syslod | ??? Could it be something I need in extensions or something? Maybe the call is coming in but I'm not paying it any attention? |
| 03:41.55 | syslod | I don't seem to have any problems with PRI. |
| 03:42.00 | BuckRogers | the asterisk server would tell me what the last callers phone number was not the ata |
| 03:42.02 | syslod | Well PRI on the 4 port card. |
| 03:42.13 | Chuji | Defraz : That :2 is going to cut off the first two numbers |
| 03:42.19 | Chuji | Defraz : Is that what you want? |
| 03:42.35 | Defraz | okay I want to cut off the first number and replace it with 2 |
| 03:42.35 | syslod | Kinda got me stumped. I have like 20 of these running with no problems and I get the single port card and have all sorts of problems. |
| 03:42.55 | Chuji | syslod : You put logger.conf in debug? |
| 03:42.59 | Defraz | so I want to had the 1 off for example 12085482345 |
| 03:43.03 | Chuji | syslod : You will see the call come in if you do |
| 03:43.08 | syslod | Anybody intersted in EMI or BAF output from *? |
| 03:43.19 | syslod | ahh. I'll try that. |
| 03:44.17 | Defraz | I mean I want to hack off the 1 and replace it with a 2 |
| 03:44.28 | Chuji | Defraz : You want it to strip the 1 before passing it to the carrier? |
| 03:44.30 | BuckRogers | what there are no pear programers here |
| 03:44.33 | bjohnson | anyone have tips on fighting echo on a SPA 3000 when the fxo connects to the fxs on the same device through *? |
| 03:44.37 | BuckRogers | perl |
| 03:44.37 | syslod | pear??? |
| 03:44.49 | BuckRogers | my bust |
| 03:44.59 | *** join/#asterisk SafT (~bt@ip-202-37-230-5.internet.co.nz) |
| 03:45.04 | syslod | why use perl?? |
| 03:45.05 | Defraz | yea strip the 1 off and add a 2 there. |
| 03:45.22 | *** join/#asterisk xai (~pasta@user-0vvdb42.cable.mindspring.com) |
| 03:45.25 | SafT | woah, this channels grown a ton since i was here last |
| 03:45.39 | Chuji | exten => _1NXXNXXXXXX,1,Dial(SIP/2085559760@65.101.69.113:5060/${EXTEN:2}) |
| 03:45.48 | Chuji | oops |
| 03:45.49 | Chuji | shit |
| 03:45.50 | Chuji | sorry |
| 03:45.52 | xai | what does "dax" and "rbxdax" mean? |
| 03:46.11 | Chuji | exten => _1NXXNXXXXXX,1,Dial(SIP/2085559760@65.101.69.113:5060/2${EXTEN:1}) |
| 03:46.18 | Chuji | There you Defraz |
| 03:46.31 | BuckRogers | perl for least amount of overhead for most amount of effecintcy |
| 03:46.35 | Chuji | Don't know why you would do that, but that will do it |
| 03:46.37 | Defraz | okay let me give it a whirl. |
| 03:46.45 | syslod | asm |
| 03:46.47 | Chuji | /join #perl |
| 03:46.48 | SafT | anyone here played with cisco 12SP/30vip's? |
| 03:47.04 | Chuji | BuckRogers : it's an active channel |
| 03:47.08 | Grooby | hmmmmmm |
| 03:47.13 | Defraz | well, I am playing with my ISERVER and If I didn't want to hack off anything just leave 12085551212 i would just drop the :1 right. |
| 03:47.17 | Grooby | oishi desu ne? |
| 03:47.24 | file[laptop] | bad bad bad dial line, bad bad bad |
| 03:47.27 | Chuji | Defraz : Yeah |
| 03:47.28 | afrosheen | SafT: this is asterisk's year :) |
| 03:47.34 | nickv111 | Ughh |
| 03:47.37 | nickv111 | kphone is annoying me |
| 03:47.41 | afrosheen | kphone sucks ass |
| 03:47.54 | nickv111 | I call myself and I don't even get a busy message or anything |
| 03:47.57 | afrosheen | one of these days someone will make a decent linux client besides skype |
| 03:48.10 | sung | hey |
| 03:48.21 | sung | most of you assholes could certainly learn something from this: |
| 03:48.22 | sung | Hercules and the Wagoner |
| 03:48.22 | sung | A carter was driving a wagon along a country lane, when the wheels sank down deep into a rut. The rustic driver, stupefied and aghast, stood looking at the wagon, and did nothing but utter loud cries to Hercules to come and help him. Hercules, it is said, appeared and thus addressed him: "Put your shoulders to the wheels, my man. Goad on your bullocks, and never more pray to me for help, until you have done your best to help yourself, or depend upon it you |
| 03:48.22 | dsmouse | sjphone /works/ |
| 03:48.29 | sung | Self-help is the best help. |
| 03:48.35 | sung | /could/can/, anyways. |
| 03:48.43 | SafT | afrosheen - it seems so |
| 03:48.49 | Chuji | wtf you talking about sung? |
| 03:49.05 | afrosheen | what is a bullock and why would you goad on it |
| 03:49.17 | afrosheen | lol |
| 03:49.21 | nickv111 | I want to use linphone but it crashes whenever I try to make a call |
| 03:49.38 | afrosheen | nickv111: I've heard that xlite sometimes runs under wine |
| 03:49.40 | SafT | afrosheen - is goading bullocks legal these days? |
| 03:49.42 | afrosheen | lol |
| 03:49.54 | nickv111 | afrosheen: I want something open source |
| 03:50.07 | sung | afrosheen: that's not the important thing! |
| 03:50.12 | afrosheen | nickv111: sjphone is supposed to work, good luck compiling it, it crashes like mad for me |
| 03:50.21 | Sedorox | thats the ATA's that have a FXO and FXS? |
| 03:50.29 | Sedorox | for some reason I can't think of the name |
| 03:50.42 | afrosheen | sung: if you're gonna paste a condescending story, make sure it's in english |
| 03:51.05 | bjohnson | Sedorox: SPA 3000 |
| 03:51.07 | bjohnson | anyone have tips on fighting echo on a SPA 3000 when the fxo connects to the fxs on the same device through *? |
| 03:51.19 | Defraz | okay I put the register for the iserver in the sip.conf and it looks like I regged okay |
| 03:51.23 | Sedorox | no.. whats the manufactor (sp) name? |
| 03:51.25 | Sedorox | spuria? |
| 03:51.27 | bjohnson | Sipura |
| 03:51.31 | Sedorox | ah |
| 03:51.32 | Chuji | bjohnson : I have just a second or so of echo on mine |
| 03:51.32 | Sedorox | danka |
| 03:51.38 | Chuji | bjohnson : Goes away fast |
| 03:51.58 | bjohnson | about one in ten calls for me is a problem for my users |
| 03:51.58 | Defraz | so do I send the digits to that iserver by calling SIP/2089044795@65.101.69.113 |
| 03:52.02 | nickv111 | afrosheen: Is it open source? |
| 03:52.06 | Defraz | or how do I send to that register. |
| 03:52.09 | sung | afrosheen: it's not a condescending story |
| 03:52.11 | afrosheen | nickv111: yeah |
| 03:52.27 | sung | afrosheen: it's something to learn from |
| 03:52.28 | neuro_[rus] | Our company have several branches over the country, we need to consolidate our phones. needed feaches: conference, call recording. Where I can find typical detailed configuration(software + hardware)? May be somebody helps me? |
| 03:52.30 | afrosheen | nickv111: broken source :) but they say it's alpha, and it kinda works |
| 03:52.50 | Chuji | Defraz : I don't understand the 208* stuff, is that your sip account name on the "iserver" |
| 03:52.51 | nickv111 | afrosheen: Seems to be proprietary |
| 03:52.57 | nickv111 | According to voip-info.org |
| 03:53.03 | afrosheen | nickv111: oh no, you're one of those guys |
| 03:53.05 | Defraz | yea the sip extention |
| 03:53.13 | afrosheen | nickv111: I bet you don't have 3d for your nvidia card right |
| 03:53.18 | nickv111 | I do |
| 03:53.24 | afrosheen | but it's not open source! |
| 03:53.26 | nickv111 | I know |
| 03:53.31 | afrosheen | lol |
| 03:53.31 | sung | OMFG NOT OPEN SOURCE HOW EVIL |
| 03:53.40 | nickv111 | But I want tuxracer ;) |
| 03:53.53 | nickv111 | So I'd rather use a proprietary driver than not have tuxracer ;) |
| 03:54.00 | nickv111 | But still |
| 03:54.07 | afrosheen | nickv111: and you want to make phone calls..so Hercules said 'Goad your bullocks and sometimes you gotta use something which isn't open" |
| 03:54.22 | nickv111 | Well, linphonec works fine |
| 03:54.26 | nickv111 | Not linphone though |
| 03:54.42 | nickv111 | With SIP, can you send text? |
| 03:54.44 | sung | afrosheen: "poke your bull" |
| 03:54.45 | nickv111 | Just wondering |
| 03:54.51 | sung | goad your bullocks. |
| 03:54.51 | afrosheen | nickv111: do yourself a huge favor and get a polycom300, they're cheap and nice enough |
| 03:54.53 | sung | poke your bull |
| 03:55.06 | afrosheen | sung: what about goad your bollocks |
| 03:55.14 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 03:55.42 | afrosheen | and who's pushing a cart whilst poking bulls, you'd think you'd have your hands full with the cart :p |
| 03:55.50 | nickv111 | Wow, minisip sure looks nice |
| 03:55.58 | Chuji | Defraz : Are you sure it's not SIP/${EXTEN}@ ? |
| 03:56.13 | kks | i have problem communicating with Quintum gateway using h323, anyone experience it before? |
| 03:56.48 | Chuji | ~h323 |
| 03:56.49 | jbot | well, h323 is evil! Don't ask about h323 here. Ask JerJer if you need to bug someone. He says it works, but others don't. |
| 03:57.06 | Defraz | let me try it out |
| 03:57.35 | *** join/#asterisk Koshatul (~evangelio@202.9.38.223) |
| 03:57.42 | afrosheen | I hate our h323 polycoms, they accept a sip image but come with funky button caps |
| 03:57.54 | Chuji | Defraz : It all depends what your iserver is expecting. I'm not real sure what an Iserver is truthfully |
| 03:57.56 | afrosheen | we had to rearrange the buttons on 5 phones |
| 03:58.08 | afrosheen | Chuji: is it from apple? |
| 03:58.35 | *** join/#asterisk talkwebhosts (~admin@c-24-130-183-95.we.client2.attbi.com) |
| 03:58.36 | Chuji | Dunno, ask defraz |
| 03:58.43 | afrosheen | Chuji: it was a joke |
| 03:58.48 | unixgeek | I have a problem where much of the time I do not get any ring tones when calling another extension. Anyone have this problem before? |
| 03:58.51 | Chuji | ohh, "I" |
| 03:58.52 | afrosheen | apple puts i in front of everything.. |
| 03:59.16 | afrosheen | unixgeek: do you actually connect to the extension eventually? |
| 03:59.17 | Defraz | haha |
| 03:59.26 | Defraz | No it is a NextTone Iserver |
| 03:59.31 | Chuji | unixgeek : Yeah, the r flag in dial has that tendency |
| 03:59.51 | Chuji | unixgeek : Try answering the call first |
| 04:00.22 | unixgeek | chuji: yes, If the extension is picked up the two phones are connected. |
| 04:00.42 | Chuji | No, I mean in your dialpan |
| 04:00.48 | Chuji | s,1,Answer |
| 04:01.06 | Chuji | s,2,Dial (Zap/xxxxx,20,r) |
| 04:01.07 | Chuji | etc |
| 04:01.43 | unixgeek | Chuji: Not familar with the r flag. What does it do? |
| 04:01.55 | Chuji | unixgeek : Well, that is what I thought you were talking about |
| 04:01.59 | dsmouse | unixgeek: make it ring to the calling party |
| 04:02.01 | `Sauron | Grr. |
| 04:02.04 | Chuji | unixgeek : When it dials, the party hears ringing |
| 04:02.20 | unixgeek | OK. let me give it a try. |
| 04:02.20 | Chuji | I may have misunderstood what you were asking |
| 04:02.26 | afrosheen | he's talking about call progress tones I think |
| 04:03.33 | Chuji | I think the damn thing overheated |
| 04:03.34 | Chuji | heh |
| 04:03.59 | Chuji | Give me an excuse to buy something new |
| 04:04.23 | mishehu | while you're at it, buy me something new too |
| 04:04.53 | shido6 | lord |
| 04:04.56 | shido6 | call progress again |
| 04:05.05 | shido6 | Chuji u like mythtv? |
| 04:05.17 | Chuji | Yeah, it's great. |
| 04:05.17 | shido6 | I have an ati radeon 8500dv can I get mythtv to use that? |
| 04:05.23 | unixgeek | OK. that does not seem to do the trick. |
| 04:05.25 | mishehu | mythtv is good, when it doesn't crash for some unknown reason |
| 04:05.30 | unixgeek | Here is my configuration. |
| 04:05.38 | mishehu | shido6: for output, yes. |
| 04:05.40 | Chuji | shido6 Yeah, I think it can actually. |
| 04:05.46 | shido6 | not for recording? |
| 04:05.47 | shido6 | :( |
| 04:05.48 | Grooby | chuji sorry to hear that |
| 04:05.48 | Chuji | shido6 : you need a capture card |
| 04:05.54 | unixgeek | I have one Cisco 12sp+ calling a soft phone on my laptop. |
| 04:05.56 | Grooby | did you use 0.17? |
| 04:05.57 | shido6 | the 8500dv does do capture |
| 04:06.03 | afrosheen | recommended cap cards are the pvr250 or pvr350 from hauppage |
| 04:06.06 | shido6 | or does it fake it in software on windows |
| 04:06.08 | Grooby | i got 2 150 |
| 04:06.08 | mishehu | shido6: that's a rage theatre chip though? |
| 04:06.10 | Grooby | haven't put it in yet |
| 04:06.17 | Chuji | shido6 : I have a pvr 250 too |
| 04:06.19 | Grooby | + another HD3000 |
| 04:06.19 | Chuji | works great |
| 04:06.20 | shido6 | school me , please |
| 04:06.31 | unixgeek | When I dial from the 12sp+, I don't get any ringing coming through the handset. |
| 04:06.38 | mishehu | shido6: if it's rage theatre, then no, you can't do capture on it. and it's a software capture in that case. use a hauppage instead. |
| 04:06.50 | SafT | unixgeek - i have a quick question re: the 12sp |
| 04:06.51 | shido6 | the 8500 in the wifes box and the 9800 in mine both have theater rage chips I believe |
| 04:06.57 | mishehu | Grooby: I have yet to configure up my pchdtv3000 |
| 04:06.59 | Chuji | I don't have both the frontend and the backend on the same box though... I need to split them next time |
| 04:07.04 | shido6 | no worky for mythtvyee? |
| 04:07.09 | Chuji | unixgeek : 12sp? |
| 04:07.11 | mishehu | shido6: no worky for anything in linux |
| 04:07.15 | shido6 | sunuva |
| 04:07.26 | shido6 | I couldnt get it to work for the life of me in any flavor |
| 04:07.30 | shido6 | so I gave up |
| 04:07.34 | SafT | the gateway setting on it (the one set on the keypad, not the tftp) is that the network gateway, or the phoen gateway? |
| 04:07.40 | shido6 | then I bought a MSI tv card |
| 04:07.45 | shido6 | can I do anything with that? |
| 04:07.46 | Grooby | mishehu, i just don't got time....want to build a MBE and throw all these capture card in..just don't got a working MB/CPU yet |
| 04:07.52 | mishehu | shido6: I've tried gatos even, and you can only get either tv out OR capture, not both simultaneous, and thats if you can even get capture to work. |
| 04:08.04 | afrosheen | lesson for the day, avoid ati |
| 04:08.09 | shido6 | I used , no I tried to use gatos, I remember that |
| 04:08.14 | shido6 | I really gave up then |
| 04:08.20 | Chuji | Grooby : You seen the nano boards from via? |
| 04:08.21 | mishehu | afrosheen: ati's not bad overall. |
| 04:08.23 | mishehu | it's just a pain. |
| 04:08.26 | shido6 | I like ATI in XP |
| 04:08.26 | mishehu | at times. |
| 04:08.31 | shido6 | but on linux its a waste of money |
| 04:08.32 | unixgeek | SafT: I have the gateway as the network gateway. |
| 04:08.34 | afrosheen | pain, drivers, linux support = avoid me |
| 04:08.43 | *** join/#asterisk snewpy (~markl@203-217-67-238.dyn.iinet.net.au) |
| 04:08.55 | mishehu | afrosheen: if you have a pre-9100 radeon, it's not hard at all to get working |
| 04:09.12 | mishehu | it's the 9200's and later that require the proprietary drivers |
| 04:09.13 | SafT | unixgeek excellent, thanks |
| 04:09.16 | afrosheen | Chuji: those nano boards look hot |
| 04:09.48 | bjohnson | shido6: wintv (which I believe are hauppage 250) can be found occasionally on ebay for < $30 |
| 04:09.49 | Chuji | #mythtv-users is a pretty active channel if you guys are interested in myth |
| 04:10.15 | afrosheen | Chuji: I have a casual interest, waiting for it to mature, I'll probably build a box for it this summer |
| 04:10.27 | bjohnson | mature .. you're funny |
| 04:10.51 | Chuji | afrosheen : I find mine to be pretty stable |
| 04:11.01 | Chuji | afrosheen : It's got spousal approval |
| 04:11.25 | mishehu | as soon as I configure the lirc remotes for my mythbox, it will then have spousal approval |
| 04:11.50 | afrosheen | Chuji: that's critical for me :) |
| 04:12.14 | Grooby | so you guys using mythphone w/ your *? |
| 04:12.33 | mishehu | nah |
| 04:12.39 | Chuji | Yeah, between Asterisk, MythTv, and X10 home automation, my wife doesn't realize how much of our house is controlled by linux |
| 04:12.43 | mishehu | no time to set up mythphone. |
| 04:13.00 | Chuji | Yeah, I haven't fooled with it either |
| 04:13.02 | Grooby | nice |
| 04:13.11 | Chuji | I have callerID pumping to my mythbox |
| 04:13.13 | Chuji | that's it |
| 04:13.33 | Grooby | caller id humping your mythbox? hmmmf |
| 04:13.43 | Chuji | heh |
| 04:14.03 | *** join/#asterisk zoid_99 (~root@user-69-1-15-110.knology.net) |
| 04:14.10 | Chuji | heh |
| 04:14.25 | Chuji | zoid_99 : /join #scriptkiddies |
| 04:14.42 | Sedorox | lol |
| 04:14.44 | zoid_99 | and why would I do that? |
| 04:14.56 | Sedorox | never.. EVER... log onto IRC as root.... |
| 04:15.01 | Chuji | cuz they will appreciate you irc'ing as root |
| 04:15.12 | Grooby | lol |
| 04:15.13 | zoid_99 | ah.. heh.. my had |
| 04:15.19 | zoid_99 | quit |
| 04:15.23 | Sedorox | your right.. you had been had :-p |
| 04:16.43 | Chuji | bah, I'm going to sleep |
| 04:16.57 | *** join/#asterisk zoid_99 (~choward@user-69-1-15-110.knology.net) |
| 04:16.59 | afrosheen | funny because freenode probably already warned him |
| 04:17.09 | zoid_99 | it did |
| 04:17.19 | zoid_99 | i didn |
| 04:17.39 | afrosheen | oh |
| 04:17.40 | zoid_99 | i didn't realize that that was the term I was in |
| 04:19.41 | *** part/#asterisk zoid_99 (~choward@user-69-1-15-110.knology.net) |
| 04:23.47 | *** join/#asterisk zoid_99 (~choward@user-69-1-15-110.knology.net) |
| 04:25.16 | *** join/#asterisk brettnem (~brettnem@user-0ccsr2l.cable.mindspring.com) |
| 04:25.24 | brettnem | Hello everyone |
| 04:25.37 | file[laptop] | 'hello' is overrated, our new greeting is 'zerplatz' |
| 04:25.52 | brettnem | ah.. well zerplatz then |
| 04:25.57 | brettnem | zerplatz to all |
| 04:26.00 | file[laptop] | zerplatz brettnem! |
| 04:26.24 | mikegrb | http://crackmonkey.org/~nick/mail/zambozay-i-said <-- I laff [sic] every time I read it, so you should too |
| 04:26.33 | brettnem | you asterisk people always makin a simple thing complicated!! ;) |
| 04:26.45 | file[laptop] | complicated? no no |
| 04:27.21 | file[laptop] | oh dear me it's 12:30 |
| 04:27.34 | SafT | lies, its 5:30pm! |
| 04:27.46 | file[laptop] | LIE! |
| 04:28.03 | florz | 05:27 < SafT> lies, its 5:30pm! |
| 04:28.06 | brettnem | oh yeah? it's 10:16 here.. |
| 04:28.09 | florz | that's definitely wrong |
| 04:28.24 | mikegrb | it is 2228 you silly swines |
| 04:28.27 | brettnem | hmm.. how am I 15 minutes off |
| 04:28.39 | brettnem | hmm.. |
| 04:28.39 | SafT | 5:28pm them |
| 04:28.41 | SafT | fine!@ |
| 04:28.45 | brettnem | sun is not visable |
| 04:28.47 | SafT | i wanted to leave early |
| 04:28.48 | *** part/#asterisk cnj (~matt@host-24-225-150-239.patmedia.net) |
| 04:29.04 | brettnem | I bet one of you stinkers took it |
| 04:29.29 | file[laptop] | I moved it to /dev/null |
| 04:29.39 | brettnem | mikegrb: pretty funny.. :) |
| 04:31.24 | mikegrb | brettnem: :D |
| 04:31.31 | mikegrb | brettnem: nick is always good for many laffs |
| 04:31.46 | brettnem | he always write like this? |
| 04:32.05 | brettnem | hmm.. I have an unusual amount of whitespace in my speech today |
| 04:32.14 | brettnem | er.. writing.. doh |
| 04:32.15 | nickv111 | Hey mikegrb |
| 04:32.45 | florz | brettnem:thatsnotgoodindeed |
| 04:33.15 | brettnem | heh.. better than a broken spacebar |
| 04:33.25 | mikegrb | hello nickv111 |
| 04:34.05 | brettnem | so anyone got any clever alternatives to using the h extension for call cleanup? |
| 04:34.19 | brettnem | sorry to actually ask an asterisk related question... ;) |
| 04:35.12 | brettnem | mikegrb: is that "the" nick? |
| 04:35.44 | *** join/#asterisk mhnoyes (~mhnoyes@user-38ldslu.dialup.mindspring.com) |
| 04:35.58 | nickv111 | In the past, when one wanted to make compressed audio files, one had |
| 04:35.59 | nickv111 | to use the "rm" utility, which compresses all files to length 0. Then |
| 04:36.03 | nickv111 | Heh |
| 04:36.11 | nickv111 | -then |
| 04:36.13 | bjohnson | I think I'm going to go get Plasterisked |
| 04:36.33 | brettnem | oy |
| 04:36.41 | mishehu | and then engage in some asstricks? |
| 04:36.44 | bjohnson | that gets harder to say the more Plasterisked you get |
| 04:36.47 | mishehu | heh. |
| 04:37.28 | zoid_99 | don't get Plasterisked and irc as root..... you get called to the carpet |
| 04:39.27 | nickv111 | G2G guys |
| 04:39.32 | nickv111 | Bye |
| 04:39.34 | brettnem | so any asterisk stuff actually going on in here anymore?? ;) |
| 04:47.01 | mikegrb | brettnem: nah, nick moffit is the nick, the windows refund day organizer among other things |
| 04:47.06 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 04:47.34 | brettnem | windows refund day? |
| 04:47.45 | mikegrb | yes, several years ago |
| 04:48.05 | mikegrb | it's in the RevolutionsOS documentary as well as an interview with the nick |
| 04:48.31 | mikegrb | nickv111 is just evidince that I'm in way to many irc channels |
| 04:49.21 | zoid_99 | On Windows Refund Day (February 15, 1999), users of non-Microsoft operating systems will bring their Microsoft Windows original disks, manuals, and Certificates of Authenticity to the Microsoft office in Foster City, California to ask for a refund |
| 04:50.04 | zoid_99 | I remember people trying to organize it here as well |
| 04:50.39 | brettnem | so how did that go? |
| 04:51.19 | zoid_99 | not very.. Our local linux users group handed out free disc at best buy |
| 04:51.30 | zoid_99 | that was about it |
| 04:51.43 | brettnem | heh |
| 04:52.14 | *** part/#asterisk xai (~pasta@user-0vvdb42.cable.mindspring.com) |
| 04:52.48 | mikegrb | http://crackmonkey.org/~nick/mail/lets-get-it-on <-- awesome |
| 04:54.39 | mikegrb | Nugget: <3 |
| 04:55.19 | QRPartner | Anyone familiar with Aastra phones? |
| 04:56.23 | zoid_99 | the new Aastra phone? |
| 04:56.26 | brettnem | heh, pretty funny |
| 04:58.00 | QRPartner | zoid_99 -> The VoIP ones anyway |
| 05:01.22 | *** join/#asterisk pkwong (~stimmy@ool-44c087de.dyn.optonline.net) |
| 05:02.25 | pkwong | hi all.. quick question.. using an iax2 connection to nufone.. asterisk stops working after a few hours and i need to restart it to get it working again.. anyone experience this problem? and is there a way to solve it? |
| 05:02.57 | brettnem | what are the symptoms? can you be a little more descriptive? |
| 05:03.32 | PatrickDK | pkwong, restart as in just asterisk, or the box? |
| 05:03.58 | pkwong | well.. it works fine for a while.. i can make and receive calls.. then it doesn't work.. i try to make a call or call my did and nothing happens.. phone doesn't ring and it just sits there in silence.. |
| 05:04.07 | pkwong | console shows everything is fine.. |
| 05:04.20 | brettnem | what does the console actually show.. can you show us? |
| 05:04.21 | pkwong | i have to stop asterisk and restart it. |
| 05:04.25 | pkwong | sure. |
| 05:04.27 | brettnem | ie: do you see the call coming in? |
| 05:04.58 | PatrickDK | pkwong, what are you using? sip phone? fxo/fxs card? ata device? |
| 05:05.13 | pkwong | nope.. the call doesn't come in at all.. iax2 show peers shows nufone is connected.. |
| 05:05.18 | pkwong | it's a cisco 7960 |
| 05:05.37 | pkwong | when i dial out, it shows the call going to nufone.. (at least via console).. |
| 05:05.48 | brettnem | does it say the call was accepted? |
| 05:05.54 | brettnem | I wonder if the registration is timing out |
| 05:06.09 | pkwong | that's what i'm thinking.. |
| 05:06.17 | pkwong | but it shouldn't do that.. right? |
| 05:06.28 | `Sauron | set verbose 4 in * |
| 05:06.29 | pkwong | it definitely looks like registration timed out. |
| 05:06.32 | brettnem | not if it is set up correctl. ;) |
| 05:06.33 | `Sauron | And look at what it says |
| 05:06.38 | brettnem | +y |
| 05:07.10 | pkwong | heh.. just set it to 4.. now i have to wait a few hours before the problem resurrects itself. |
| 05:07.22 | brettnem | when it is "broken" what does a iax2 show registry say? |
| 05:07.32 | *** join/#asterisk jets (~brian@the.notsoblue.com) |
| 05:07.39 | jets | Anyone from Sacramento in here? |
| 05:07.42 | pkwong | iax2 show registry shows it's connected to nufone. |
| 05:07.52 | jets | or California in general? |
| 05:07.53 | brettnem | when it is broken it does? |
| 05:07.57 | pkwong | yes.. |
| 05:08.02 | brettnem | hm |
| 05:08.06 | pkwong | it's weird.. |
| 05:08.20 | pkwong | the only thing i can think of is that i'm connected via cable modem.. |
| 05:08.21 | Juggie | asterisk and my mitel5055 does that |
| 05:08.28 | pkwong | but still.. that shouldn't be the issue.. |
| 05:08.32 | Juggie | but the phone is dying |
| 05:08.35 | *** join/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
| 05:08.37 | pkwong | i am behind a linksys box. |
| 05:08.46 | brettnem | ah |
| 05:08.47 | Juggie | because the webserver on the phone dies |
| 05:08.58 | brettnem | I bet your nat xlate table is dying |
| 05:09.09 | pkwong | aha! |
| 05:09.13 | brettnem | hmm.. not sure if that will be a problem with iax.. I think it is tho |
| 05:09.26 | MrEntropy | what exactly is contained in the SIP 'headers'? |
| 05:09.29 | brettnem | do you have qualify set to yes? or something? |
| 05:09.38 | pkwong | let me see. |
| 05:09.38 | brettnem | MrEntropy: "stuff" |
| 05:09.55 | Qwell | brettnem: "data" |
| 05:10.02 | MrEntropy | brettnem: as off-shoot as that is, i need just a little more infomation =) |
| 05:10.08 | brettnem | MrEntropy: do a SIP debug and you'll see.. it's like the headers of an email... some of it is useful some isn't, some might be. |
| 05:10.26 | brettnem | MrEntropy: Sorry, feeling a little stressed out and tired tonight.. ;) |
| 05:10.29 | pkwong | i don't have a qualify statement in iax.conf at all. |
| 05:10.45 | MrEntropy | yeah, but when i do a sip debug, the whole sip message comes up...i want to know the difference between the header and body? |
| 05:11.15 | MrEntropy | or is body the payload? |
| 05:11.16 | brettnem | try setting qualify=yes or some number.. then an iax2 show peers should show OK and a response time instead of UNMONITORED |
| 05:11.57 | MrEntropy | ok, so what is a sip dialog? |
| 05:11.59 | brettnem | well most of the relevant data is in the headers.. some method types like NOTIFYs utilize the body to transmit data |
| 05:12.22 | brettnem | hmm.. I don't know.. sounds like a made up term maybe? can you use it in context? |
| 05:12.53 | `Sauron | Hum. |
| 05:13.01 | brettnem | oh.. well call setup data for INVITES is all in the body.. If I remember right.. SDP daa |
| 05:13.03 | brettnem | data |
| 05:13.03 | `Sauron | my iaxtel still shows unmonitored, instead of OK |
| 05:13.05 | `Sauron | bummer |
| 05:13.06 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 05:13.13 | brettnem | did you do a reload? |
| 05:13.27 | brettnem | I can't find a good way to do that.. but I think "reload" does it.. |
| 05:13.29 | `Sauron | Hum, maybe not. :) |
| 05:13.33 | brettnem | no iax2 reload command |
| 05:13.38 | `Sauron | I did a sip reload for a diff. phone, but not iax2 |
| 05:14.01 | `Sauron | iaxtel seems to be unavailable |
| 05:14.03 | pkwong | i just put in the qualify=yes in and it still shows unmonitored. |
| 05:14.18 | `Sauron | There we go |
| 05:14.27 | `Sauron | OK (569 ms) |
| 05:14.32 | `Sauron | sloow |
| 05:14.33 | Defraz | Has anyone got the asterisk box talking to a NexTone Iserver? |
| 05:14.40 | Defraz | I can't seem to get mine working quite right. |
| 05:14.42 | brettnem | pkwong: did you reload? |
| 05:14.47 | Defraz | I got it register from the entry in the sip.conf |
| 05:14.53 | Defraz | but I can't seem to push a call to it. |
| 05:15.08 | brettnem | you have a peer entry for it? |
| 05:15.17 | Defraz | peer entry? |
| 05:15.19 | Defraz | hmmmm |
| 05:15.21 | Defraz | I don't think so |
| 05:15.26 | Defraz | just a register entry |
| 05:15.29 | pkwong | ok.. i just put it in under the [nufone] context.. and it's fine now. |
| 05:15.36 | pkwong | it doesn't show unmonitored. |
| 05:15.40 | brettnem | you can't just register.. that only lets them know where you are.. doesn't tell asterisk how to use it |
| 05:15.44 | pkwong | will that solve the problem? |
| 05:15.52 | Defraz | oh I see |
| 05:16.02 | *** join/#asterisk jets (~brian@the.notsoblue.com) |
| 05:16.23 | brettnem | pkwong: well I know in the SIP channel qualifys are sent out every 60 seconds to good peers or 10 seconds to bad peers.. so this traffic typically is recommended to maintain NAT translations. |
| 05:17.09 | pkwong | ah ha! very very cool! heh.. hope it'll work.. i'll see.. thanks :) |
| 05:17.21 | brettnem | sure.. let us know how it goes. :) |
| 05:17.30 | pkwong | so qualify=yes keeps the connection alive? |
| 05:17.38 | pkwong | theoretically? |
| 05:17.41 | brettnem | well I don't know if I'd say that.. |
| 05:17.48 | brettnem | it just sends out packets periodically |
| 05:17.58 | pkwong | ok.. so it should then.. :) |
| 05:18.08 | brettnem | makes your linksys thingy happy.. if it doesn't see that nat translation being used for a while it'll kill it |
| 05:18.23 | pkwong | ok.. makes sense.. |
| 05:18.29 | brettnem | question for you pkwong, if you periodically used the phone (every hour or so) did you still have the problem? |
| 05:18.30 | pkwong | hopefully that solves the issue. |
| 05:18.34 | *** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net) |
| 05:18.37 | pkwong | nope.. |
| 05:18.47 | brettnem | or was it after long periods of complete inactivity (like overnight) |
| 05:18.49 | Defraz | okay I setup a peer deal in the sip.conf |
| 05:18.51 | pkwong | when i was using the phone during the day it doesn't happen. |
| 05:18.58 | pkwong | overnight.. it does. |
| 05:19.02 | brettnem | cool.. I bet that's it. |
| 05:19.08 | Defraz | now I have an extention line in my extension.conf |
| 05:19.35 | Defraz | can I paste that here to make sure it is right? |
| 05:19.37 | pkwong | i'm opening a store and hoping this solves it.. if it does, then i can offer * based pbx solutions. |
| 05:19.54 | pkwong | and i can lose verizon. |
| 05:20.00 | pkwong | and that's a GOOD thing. |
| 05:20.22 | Qwell | anytime "lose" and "Verizon" are in the same sentence, I get all giddy ;p |
| 05:20.27 | `Sauron | Ugh. |
| 05:20.30 | `Sauron | iaxtel blows |
| 05:20.36 | brettnem | heh |
| 05:20.48 | Defraz | exten => _1NXXNXXXXXX,1,Dial(SIP/2089044795@65.101.69.113:5060/2${EXTEN:1}) |
| 05:20.51 | brettnem | Defraz: sure if it's just a line.. more than that, please use pastebin |
| 05:20.58 | Defraz | yea just one line |
| 05:21.03 | Defraz | it isn't talking to the iserver |
| 05:21.06 | brettnem | Defraz: that's not how you dial to a registered phone |
| 05:21.08 | pkwong | heh.. i'm seeing lots of stuff coming in on the iaxtel link.. so i think that may have solved it.. :) |
| 05:21.19 | brettnem | btw, is the phone registered? |
| 05:21.29 | brettnem | pkwong: good! |
| 05:21.42 | *** join/#asterisk Silik0n (~krice@rso.suspicious.org) |
| 05:21.51 | pkwong | yeah.. showing iaxtel unreachable then 2 sec. later showing it's reachable again.. |
| 05:21.57 | Defraz | the NexTone said it regged with the asterisk |
| 05:22.00 | Defraz | then I setup a peer |
| 05:22.04 | pkwong | so hopefully it does that with nufone too. |
| 05:22.09 | pkwong | :) |
| 05:22.33 | pkwong | thanks :) i'm gonna go wait a few hours and see what happens! |
| 05:22.38 | brettnem | Defraz: ok so the peername is whatever is between the []'s in the sip.conf file |
| 05:22.40 | Defraz | yea and that is regged too it says in a sip show peers |
| 05:22.42 | pkwong | ttyl guys! :) |
| 05:22.43 | brettnem | sure thing |
| 05:22.47 | Defraz | yea iserver |
| 05:23.16 | Defraz | [iserver] |
| 05:23.26 | brettnem | so your dial line should be exten => _1NXXNXXXXXX,1,Dial(SIP/iserver) |
| 05:23.36 | brettnem | SIP/<PEERNAME> |
| 05:23.42 | Defraz | okay great |
| 05:23.45 | Defraz | lets see |
| 05:24.01 | brettnem | you'd do that other syntax for sending a call to a proxy or to a UA that does someting intellegent with DIDs.. |
| 05:24.03 | MrEntropy | does anyone know what the "sip dialog" is? |
| 05:24.07 | brettnem | a phone doesn't really |
| 05:24.14 | brettnem | MrEntropy: can you use it in context |
| 05:24.16 | brettnem | ? |
| 05:24.34 | MrEntropy | brettnem: no, it's a sip thing, nothing specifically to do with asterisk |
| 05:24.53 | MrEntropy | i'm sure asterisk uses it though |
| 05:24.58 | Juggie | finished phase 1 of my web enabled conf bridge today at work |
| 05:25.05 | brettnem | I hear ya, but If you can tell me what context you are reading about it, perhaps I can tell you what it means.. |
| 05:25.10 | Juggie | boss was impressed |
| 05:25.14 | brettnem | Juggie: you sharing with all of us?? ;) |
| 05:25.24 | *** part/#asterisk jets (~brian@the.notsoblue.com) |
| 05:25.32 | MrEntropy | it's in the SER dlg.h include file, i'm programming a SER module |
| 05:25.42 | Juggie | brettnem, working on getting approval from legal |
| 05:25.46 | Juggie | but dont count on it |
| 05:26.20 | `Sauron | silly legal people |
| 05:26.24 | brettnem | Juggie: you know GPL requires modifications to code to be contributed to the community or be commerically licenced from digium. Run that by legal. ;) |
| 05:26.24 | `Sauron | Grr. |
| 05:26.43 | Juggie | brettnem, not when the code is all php |
| 05:26.52 | `Sauron | brettnem: ONLY if you distribute it. Duh. |
| 05:27.04 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 05:27.05 | brettnem | oh.. well sure |
| 05:27.07 | `Sauron | Err, only if you distribute the product, do you have to supply the code. |
| 05:27.27 | brettnem | oh it's late |
| 05:27.32 | brettnem | ok |
| 05:27.32 | Juggie | i am getting a bunch of oss licenses approved right now |
| 05:27.33 | `Sauron | hehe |
| 05:27.47 | brettnem | so anyone have a good alternative to the "h" extension for call cleanup? |
| 05:27.54 | Juggie | but the conf brige is php/phpagi/code in context |
| 05:27.56 | Nugget | shutdown -r now |
| 05:27.56 | Juggie | no mods |
| 05:28.09 | Nugget | cleans up all the calls. |
| 05:28.19 | Defraz | brettnem: It is regged now when I do a sip show peers and so on but it doesn't seem to be connecting to the iserver when it sends the phoen number I dial. |
| 05:28.27 | brettnem | I'm trying to setup so that if a call is recorded and less than 5 seconds long it gets erased |
| 05:28.30 | Juggie | i did patch cdr_addon_mysql thogh |
| 05:28.41 | *** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net) |
| 05:28.53 | Juggie | but i havnt distroed it |
| 05:28.54 | brettnem | Nugget: Thanks... from the same wisdom that brought us "Alt-H" for higher access |
| 05:29.03 | Nugget | plus plus plus! |
| 05:29.20 | Juggie | i added another flag in the config file called servernumber |
| 05:29.33 | brettnem | Defraz: what kind of phone? |
| 05:29.37 | Juggie | and another field in the cdr table |
| 05:29.51 | Juggie | so you can dump records from many servers in |
| 05:30.04 | Juggie | and you know which server they are from |
| 05:30.04 | Defraz | well the phone is regged on the asterisk box and the phone is a WIP-5000 |
| 05:30.06 | brettnem | another field, another flag, another reason, for licencing your code... dooo be do be dooo |
| 05:30.21 | brettnem | WIP? |
| 05:30.23 | brettnem | hmm |
| 05:30.31 | Juggie | brettnem, thats just my private mysql cdr mod |
| 05:30.35 | Juggie | still testing |
| 05:30.45 | brettnem | heck.. I don't care really.. haha |
| 05:30.49 | Nugget | port it to postgresql. :) |
| 05:30.51 | Defraz | and I can make local calls out the analog line right off the asterisk box but when I do a 1NXXNXXXXXX I wanted it to go out to my LD provider who has a NexTone softswitch. |
| 05:30.53 | brettnem | EK |
| 05:31.02 | Juggie | Nugget, no point |
| 05:31.14 | Juggie | i could write a posgres module |
| 05:31.14 | Nugget | being able to avoid mysql is a worthwhile goal. :) |
| 05:31.16 | brettnem | heh.. well that dial line you added will make EVERY 1NXXNXXXXXX call goto the phone |
| 05:31.20 | Defraz | Wireless 802.11b VOIP phone; |
| 05:31.23 | brettnem | oh it's religon |
| 05:31.35 | Juggie | but the new database stuff is comming soon |
| 05:31.35 | brettnem | Nugget: how you been liking this weather we've been having? |
| 05:31.38 | Juggie | hopefully |
| 05:31.40 | *** part/#asterisk QRPartner (~andy@ns1.accu-com.com) |
| 05:31.49 | Nugget | I drove to dallas today, it was perfect convertible weather. |
| 05:31.58 | Defraz | okay well if I wanted to route every call any phone tha tis regged to go out threw the NexTone box. |
| 05:31.59 | brettnem | eekk WiFi sip phone.. this is your learning phone? ;) |
| 05:32.09 | Defraz | yea |
| 05:32.13 | brettnem | Can't beleive it was like 85 today.. crazy |
| 05:32.14 | Nugget | is there a wifi sip phone that doesn't suck? |
| 05:32.19 | brettnem | Um.. no |
| 05:32.25 | brettnem | but why not |
| 05:32.28 | brettnem | ACTUALLY |
| 05:32.32 | brettnem | I saw something cool |
| 05:32.34 | Nugget | I read a review of the senao that seemed to indicate it sucks just as badly as my zyxel. |
| 05:32.39 | brettnem | Nugget: you didn't goto astricon did you? |
| 05:32.42 | Nugget | nope |
| 05:32.54 | Defraz | I also have a SPA 2000 I am using |
| 05:33.06 | Nugget | I'll go to astricon europe if it's someplace fun. |
| 05:33.19 | `Sauron | brettnem: where you at? |
| 05:33.21 | brettnem | well Mark had a old fashioned BELL phone (with the pushbuttons) he had wired up an iaxy inside of it with a wireless card and had batteries in it.. that was pretty cool |
| 05:33.28 | Nugget | that's spiffy. |
| 05:33.31 | brettnem | I'm in Austin, Tx |
| 05:33.36 | `Sauron | Funny, that. |
| 05:33.41 | brettnem | heh it was pretty cool to hear that bell ring |
| 05:33.49 | brettnem | it was totally wireless... haha |
| 05:33.50 | `Sauron | It was indeed warm here today. :) |
| 05:33.56 | brettnem | where are you at? |
| 05:34.00 | `Sauron | same |
| 05:34.08 | brettnem | Ah.. cool.. where? |
| 05:34.18 | `Sauron | northwest, off 183 |
| 05:34.25 | Nugget | I was going to fly today, but it was foggy this morning. drove instead. |
| 05:34.28 | brettnem | oh, I think nugget it up that way |
| 05:34.41 | brettnem | spicewood and the like |
| 05:34.44 | Nugget | anderson mill and spicewood springs. |
| 05:34.44 | Juggie | pfft :) |
| 05:34.46 | `Sauron | 183/360 |
| 05:34.55 | brettnem | I work off 360/2244 |
| 05:35.00 | `Sauron | I'm not as northwest as y'all |
| 05:35.07 | brettnem | haha austin has so many numbered streets |
| 05:35.11 | Juggie | juggie = south of the 417 east of the 416 :) |
| 05:35.24 | Defraz | so on any phone that is regged with the asterisk should go out over the iserver phone when I dial a 1 first |
| 05:35.24 | brettnem | 360/2244 is south of 360/183 |
| 05:35.29 | `Sauron | I coulda said I'm off Research Blvd and Capital of Tx Highway |
| 05:35.30 | `Sauron | ;) |
| 05:35.34 | Nugget | heh |
| 05:35.45 | brettnem | or the arboretum |
| 05:35.50 | Nugget | who the fsck decided to call mopac a loop? |
| 05:35.51 | `Sauron | yup |
| 05:35.53 | brettnem | still pretty far south |
| 05:36.07 | brettnem | haha yeah, that's pretty silly |
| 05:36.19 | `Sauron | dunno, mopac does curve a bit, if you include 1325 |
| 05:36.27 | brettnem | it's more of a "C" |
| 05:36.28 | `Sauron | just like lamar is like, loop 276 or something |
| 05:36.34 | `Sauron | 279 |
| 05:36.36 | `Sauron | I can't remember |
| 05:36.42 | Defraz | or iserver sip connection |
| 05:37.00 | brettnem | well at least they'll all be toll roads soon.. I can't wait to start shoveling out some more change |
| 05:37.07 | Nugget | I never drive. |
| 05:37.24 | Nugget | today I bought my first tank of gas in 2005. |
| 05:37.26 | brettnem | ek |
| 05:37.29 | Nugget | (and the second tank, which sucked) |
| 05:37.30 | brettnem | ha! |
| 05:37.32 | brettnem | so you DO drive |
| 05:37.42 | `Sauron | Nugget: What you driving? |
| 05:37.44 | brettnem | you know.. you arn't supposed to drink that stuff |
| 05:37.47 | Nugget | a convertible. |
| 05:37.51 | `Sauron | ahha |
| 05:37.52 | Nugget | perfect for today. :) |
| 05:37.55 | `Sauron | no kidding |
| 05:38.06 | brettnem | hmm I should put the top down sometime.. |
| 05:38.06 | `Sauron | I just have a truck |
| 05:38.10 | terrapen | today is a great bike riding day |
| 05:38.23 | terrapen | can't wait till the thursday night ride |
| 05:38.25 | `Sauron | until I pimp out my van |
| 05:38.26 | Nugget | http://lnk.nu/slacker.com/lt |
| 05:38.27 | brettnem | `Sauron: you know... with the proper tools.. ANY car can become a convertible |
| 05:38.39 | `Sauron | brettnem: Granted. |
| 05:39.02 | brettnem | Nugget: you got BOFH? ha cool |
| 05:39.06 | Nugget | :) |
| 05:39.10 | `Sauron | nice |
| 05:39.16 | brettnem | wtf, you got that and don't drive.. what's the matter with you |
| 05:39.41 | brettnem | so how come you haven't suckered justin into asterisk yet? |
| 05:40.03 | *** join/#asterisk jets (~brian@the.notsoblue.com) |
| 05:40.11 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 05:40.19 | SexyKen | Anyone here do custom coding for Asterisk? |
| 05:40.25 | brettnem | HAHA |
| 05:40.28 | brettnem | oh sorry |
| 05:40.42 | brettnem | actually probably any of us will for the right price |
| 05:40.53 | SexyKen | Okay then give me a number to call you at. |
| 05:40.54 | Nugget | I'll drive again once it's warm all the time. |
| 05:40.58 | Nugget | not much fun to drive in the winter. |
| 05:41.10 | brettnem | woah kram! |
| 05:41.15 | kram | sup brett |
| 05:41.21 | SexyKen | brettnem You gonna give me a contact number or no |
| 05:41.25 | brettnem | good evening. :) |
| 05:41.36 | brettnem | SexyKen: sorry, I'm booked for now.. really.. |
| 05:41.39 | `Sauron | I think it's a matter of "the right price" |
| 05:41.54 | Qwell | kramage ;] |
| 05:42.20 | SexyKen | Oh no, there's no problem with money. It's the attitude that isn't helping some. |
| 05:42.24 | kram | sexyken: if you can't find anyone else, just give us a call at Digium, we'll be happy to loo at your dev work |
| 05:42.44 | brettnem | hmm |
| 05:43.07 | SexyKen | •kram• A: It's urgent. B: It's urgent. C: It's very urgent. So if you guys have time, I'll call you up right now and discuss what I need done and you give me an idea of cost and we'll start it up. |
| 05:43.47 | brettnem | SexyKen: have you tried making your actual request to the group? there is quite a bit of experience in here.. someone might be able to help |
| 05:43.49 | kram | call tomorrow morning and ask for Bill Hall |
| 05:43.57 | kram | he can help you out |
| 05:43.59 | SexyKen | •kram• Will do. |
| 05:44.07 | Nugget | SexyKen: your irc client is really annoying. |
| 05:44.12 | `Sauron | dum di dum |
| 05:44.25 | SexyKen | •Nugget• You can code me a new script for that if it bothers you enough. |
| 05:44.30 | SexyKen | ;-) |
| 05:44.36 | `Sauron | kram: Can you tell the sales drones to put some minimal amount of support pricing info on the digium site? :) |
| 05:44.46 | Nugget | script? you mean you're doing that on purpose? wow. |
| 05:44.51 | Faithful | Is call queuing that hard to set up? |
| 05:44.59 | `Sauron | Faithful: I don't think so. |
| 05:45.00 | brettnem | what's he doing? heh what am I missing? |
| 05:45.04 | brettnem | Faithful: not really |
| 05:45.06 | kram | sauron: talk to bill hall |
| 05:45.24 | `Sauron | kram: or I can email sales@digium, apparently. :) |
| 05:45.30 | kram | either way |
| 05:45.33 | Faithful | I guess if you can set up * as a newbie you can do anything :) |
| 05:45.33 | brettnem | I think that's bill's email |
| 05:45.46 | brettnem | except SS7 |
| 05:46.36 | `Sauron | sets |
| 05:46.36 | kram | sales is probably fine |
| 05:46.49 | brettnem | oh if I could only really do SS7 with my asterisk.. what a day that'd be.... I really want to toll out my switch |
| 05:46.57 | brettnem | er toss.. |
| 05:47.01 | brettnem | a freudian slip perhaps? |
| 05:47.10 | *** join/#asterisk habakuk (~chatzilla@24-116-201-143.cpe.cableone.net) |
| 05:47.48 | `Sauron | It's really spelled "Habbakuk" |
| 05:48.13 | `Sauron | Actually, that's wrong |
| 05:48.18 | `Sauron | Habakkuk |
| 05:48.23 | `Sauron | D'oh ;) |
| 05:49.10 | `Sauron | brett: Last I read, someone's (barely?) working on ss7 |
| 05:49.27 | brettnem | yeah, I've spoken to them.. I've very aware of their status.. |
| 05:49.41 | brettnem | and to put it plainly.. it's quite mediocre |
| 05:50.21 | brettnem | certainly is going the right direction, just not there yet.. more focused on international ss7 network architectures than US architectures |
| 05:51.03 | brettnem | in the community, most of that is being lead by steve underwood |
| 05:51.26 | brettnem | I'm just being impatient |
| 05:51.48 | `Sauron | hehe |
| 05:51.59 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 05:52.08 | shmaltz | ~seen ManxPower |
| 05:52.09 | jbot | manxpower <~eric@dsl-209-205-172-111.i-55.com> was last seen on IRC in channel #asterisk, 1d 2h 22m 13s ago, saying: 'Nugget, We should get tzanger's opinion!'. |
| 05:53.33 | `Sauron | Hehn. The fun of running HEAD with patches. |
| 05:53.36 | brettnem | why doesn't iax2 have a reload command? |
| 05:53.51 | `Sauron | Now I have to start from scratch, cuz latest cvs update clobbered some of the patched files |
| 05:53.58 | Juggie | ~seen juggie |
| 05:53.59 | jbot | juggie is currently on #asterisk (9h 46m 34s). Has said a total of 41 messages. Is idling for 1s |
| 05:54.04 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:54.04 | brettnem | fun |
| 05:54.09 | `Sauron | brettnem: Shouldn't be hard to add. Add it :) |
| 05:54.22 | brettnem | oh.. I'm just a user |
| 05:54.24 | brettnem | heh |
| 05:54.44 | brettnem | `Sauron: what do you do here in Austin? |
| 05:54.49 | `Sauron | network engineer |
| 05:54.52 | Juggie | `Sauron, how is cvs |
| 05:54.59 | Juggie | hows realtime looking? |
| 05:55.02 | brettnem | oh come on.. that's so generic.. ;) |
| 05:55.04 | `Sauron | Juggie: Well, it broke last night. :) |
| 05:55.19 | Juggie | i've been dying to get realtime in stable :) |
| 05:55.23 | `Sauron | brett: I largely do "other duties as assigned" for seton healthcare |
| 05:55.33 | brettnem | ah seton |
| 05:55.35 | habakuk | so what sort of interesting solutions have people come up with for fax support? I thought about getting an efax number and detecting a fax tone and force the call to my efax # |
| 05:55.37 | `Sauron | Juggie: realtime stomped on the ast_data patches. Grrrr. |
| 05:55.58 | brettnem | habakuk: there is some new fax detection stuff mentioned on the wiki.. go read |
| 05:56.13 | Juggie | `Sauron, realtime is a mix of ast_data and some other stuff isnt it? |
| 05:56.28 | brettnem | `Sauron: I'm the engineer for a CLEC based here in Austin |
| 05:56.31 | `Sauron | I dunno. Last time I looked at it (> week ago), it only did mysql.. barely |
| 05:56.38 | habakuk | brettnem, yes I'm aware of that.. unfortunately it's not reliable enough |
| 05:56.45 | `Sauron | _the_ engineer? :) |
| 05:56.50 | brettnem | :-D |
| 05:56.57 | brettnem | well.. you know how it is |
| 05:57.04 | Juggie | `Sauron, isnt there a odbc & mysql module for it? |
| 05:57.30 | brettnem | I'm the one who does switch interconection and VOIP |
| 05:57.34 | `Sauron | Juggie: ast_data also has pgsql ;) |
| 05:57.42 | habakuk | realtime is not realtime in my opinion. I use the ast_data patches |
| 05:58.06 | `Sauron | my only complaint with ast_data, is that there's difficulties with the dialplan lookups |
| 05:58.20 | brettnem | `Sauron: so is Seton using asterisk?? :) |
| 05:58.29 | habakuk | `Sauron, how so? |
| 05:58.31 | `Sauron | brett: We might, if I get it my way. |
| 05:58.35 | habakuk | works fine for me |
| 05:58.44 | brettnem | very cool |
| 05:58.54 | `Sauron | habakuk: in extensions.conf you can use include => blahblah to order the list of contexts |
| 05:58.57 | brettnem | I'll be having a baby at Seton on 38th in a month. :) |
| 05:59.08 | `Sauron | you can't do that when your dialplan is in sql |
| 05:59.13 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-80-254.sentechsa.net) |
| 05:59.18 | habakuk | yes you can |
| 05:59.21 | habakuk | you have both |
| 05:59.27 | habakuk | thats the default |
| 05:59.29 | `Sauron | I tried, it didn't work |
| 05:59.36 | habakuk | works fine for me |
| 05:59.42 | `Sauron | shrug |
| 05:59.47 | `Sauron | I'll have to re-patch |
| 06:00.23 | habakuk | I would also recommend the mwi patch as well that allows mwi while all your sip peers are in sql |
| 06:00.38 | habakuk | which is not available for realtime |
| 06:00.40 | `Sauron | hum |
| 06:02.41 | habakuk | brettnem, you were referring to spandsp earlier right? |
| 06:02.49 | brettnem | no |
| 06:02.56 | `Sauron | I need to try to hack spandsp into working with pppd/mgetty |
| 06:03.00 | brettnem | NVFaxDetect |
| 06:03.04 | brettnem | or something like that |
| 06:03.20 | brettnem | it's on the wiki |
| 06:03.26 | brettnem | ~google asterisk fax detection |
| 06:04.04 | brettnem | yeah, you'll find a link for it at that main wiki page at the bottom |
| 06:05.16 | `Sauron | habakuk: do you know if they've updated the ast_data patches recently (< 4 days) |
| 06:06.19 | habakuk | brettnem, yeah that works. Then I can send the call to my efax #, until I can afford a 5350 |
| 06:06.25 | Defraz | brettnem: when I dial 1NXXNXXXXXX it doesn't seem to try and talk to the ISERVER peer I setup. |
| 06:06.39 | habakuk | `Sauron, not sure, have you checked the site? |
| 06:06.44 | Defraz | I put a sniffer on the line and it looks like it dials itsself. |
| 06:06.45 | brettnem | Defraz: where are you dialing from and to? |
| 06:07.09 | `Sauron | I will in a few minutes |
| 06:07.35 | brettnem | Defraz: like I said before, with that line you have in extension.conf every dialied number will goto that iserver peer |
| 06:07.38 | Defraz | I am on a ip phone dialing for example 12082371212 my ip phone is regged on on the asterisk box and my asterisk box is regged to the iserver with is connected to our long distance provider. |
| 06:07.46 | *** join/#asterisk IsMe (~some@219.95.224.115) |
| 06:07.55 | brettnem | so if you are calling from the iserver peer and dial some random number it will goto the iserver peer (itself) |
| 06:08.01 | Defraz | hmm my phone says unavailable whenever I dail with a 1 |
| 06:08.21 | brettnem | that's a problem in your dialplan |
| 06:08.29 | jets | Defraz: are you a boise asterisk'r? |
| 06:08.30 | Defraz | but if I dial from a ip phone (friend) it should to out threw the peer. |
| 06:08.43 | Defraz | actually pocatello |
| 06:08.56 | brettnem | why? |
| 06:09.00 | jets | Sweet, I'm a Rupert/Twin Falls asterisk'r |
| 06:09.11 | brettnem | lets see your extension.conf.. use a pastebin |
| 06:09.11 | Defraz | neat |
| 06:09.11 | habakuk | hey I'm in boise |
| 06:09.21 | Defraz | what is the url for that? |
| 06:09.41 | Defraz | we will have to connect our boxes when I figure out what I am doing. |
| 06:09.43 | Qwell | ~pastebin |
| 06:09.44 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 06:10.05 | `Sauron | dum di dum |
| 06:10.19 | jets | Defraz: cool i work for a telco so i have some DIDs and stuff at my disposal |
| 06:11.32 | Defraz | so you work for PMT |
| 06:11.40 | Defraz | Yea I work for Direct Communications |
| 06:11.40 | jets | Yes, actually. |
| 06:11.43 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 06:11.46 | Defraz | I am getting some DIDs actually |
| 06:11.54 | habakuk | Defraz, can you get DID's anywhere in Idaho? |
| 06:12.57 | habakuk | err, jets / Defraz can any of you get me DID's in boise? |
| 06:13.09 | jets | habakuk: I have several DIDs blocks throught out idaho, basically anywhere syringanetworks has lit fiber. |
| 06:13.11 | brettnem | I'm only good for Texas |
| 06:13.21 | jets | habakuk: yes we can in boise |
| 06:13.32 | `Sauron | I'm only good for cat5 connectivity inside my apartment |
| 06:13.45 | `Sauron | and wireless connectivity slightly outside my apartment... |
| 06:13.46 | `Sauron | ;) |
| 06:14.04 | habakuk | jets, can you provide sip? |
| 06:14.16 | brettnem | I used to have a very large fiber network.. sigh... |
| 06:14.25 | brettnem | those were the days |
| 06:14.33 | `Sauron | Hum |
| 06:14.44 | `Sauron | who? ;) |
| 06:14.46 | jets | habakuk: yes we aren't really providing much sip termination yet |
| 06:15.07 | brettnem | `Sauron: do you know of EPGN, El Paso Global Network? They go by Alpheus now.. |
| 06:15.12 | habakuk | jets, ok what do you prefer? |
| 06:15.19 | `Sauron | I remember hearing about them years ago |
| 06:15.31 | brettnem | heh.. oh really? what did you hear?? :) |
| 06:15.47 | Defraz | yea I can get eastern idaho did access pretty easy |
| 06:15.53 | `Sauron | friend of mine was going to get gobs of connectivity from them, and resell |
| 06:16.00 | jets | habakuk: i guess I should have said "we don't really offer it to the public much." we can do sip or iax |
| 06:16.03 | `Sauron | then he bankrupted his company |
| 06:16.15 | brettnem | `Sauron: I think it's a great idea.. no one has really done it yet.. |
| 06:16.19 | brettnem | what company was it? |
| 06:16.26 | Astrisk-boob | how can i test my works voip network? i wanna poke around u know! |
| 06:16.38 | `Sauron | I don't know that I'm at liberty to tell |
| 06:16.40 | `Sauron | ;/ |
| 06:16.55 | Defraz | Brettnem I pasted that a few minutes ago, just had a phoen call inbetween. |
| 06:17.02 | brettnem | heh.. it's out of business, right?? heh.. ok well no pressure :) |
| 06:17.12 | `Sauron | he lives happily at by the Y at oak hill.. got wife and a dog. :) |
| 06:17.34 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 06:17.41 | brettnem | `Sauron: I was the original network architect for that network (Waller Creek Communications, Pontio, EPGN, Alpheus, whateve ryou want to call it) |
| 06:17.51 | `Sauron | It's one of those moments where a split second after you make a decision, you realise you've shot yourself in the foot for the next 7 years |
| 06:17.52 | brettnem | `Sauron: I live 1.52 minutes from the Y |
| 06:18.13 | jets | Defraz: are you one of the noc guys at dcdi? |
| 06:18.19 | Defraz | yea |
| 06:18.21 | bjohnson | Astrisk-boob: sign up for FWD |
| 06:18.22 | brettnem | ha |
| 06:18.25 | `Sauron | so what happened to y'alls fiber plant? |
| 06:18.33 | brettnem | `Sauron: EPGN? |
| 06:18.41 | `Sauron | ya |
| 06:18.52 | brettnem | It's still there.. of course, EPGN never had more than.. I'd say 6 miles of fiber |
| 06:18.57 | `Sauron | you said you used to be in charge of it.. implies youre either not there, or it's gone |
| 06:19.40 | brettnem | yeah El Paso came in, bought us (we were pontio communications).. the new company only had about 80-100 employees.. then they hired about 300 more people.. |
| 06:20.03 | jets | i'm excited another phone geek and syringa partner is working with asterisk... if you need to show your bosses any sucessful * deployments let me know. |
| 06:20.17 | brettnem | after a year, they realized that they didn't want to have anything to do with telecom.. so laid off 300 people.. Since I was in charge of new developments... and they wern't planning on expanding.. I got the axe.. blah |
| 06:20.20 | `Sauron | jets: I do, some day in the next 2 weeks or so |
| 06:20.49 | Defraz | Where have you used it? |
| 06:20.53 | brettnem | `Sauron: of course.. there was probably over 7,000 fiber miles in the network.. almost entirely SBC fiber |
| 06:21.09 | Defraz | I just have a few phones running with just some analog cards and IAX stuff going nothing to big. |
| 06:21.16 | brettnem | and 98% lit as OC-48/DWDM good stuff |
| 06:21.22 | jets | At a company in Salt Lake, and a deployment in our callcenter. Screen pops, reporting, etc. |
| 06:21.25 | Defraz | Jus twanted to tie it into a softswitch for the LD |
| 06:21.25 | `Sauron | brett: rofl |
| 06:21.36 | Defraz | nice that is great. |
| 06:21.59 | brettnem | `Sauron: I miss all that bandwidth sometimes.. |
| 06:22.04 | *** join/#asterisk outtolunc (~chatzilla@adsl-69-110-26-49.dsl.pltn13.pacbell.net) |
| 06:22.05 | brettnem | `Sauron: now I am a customer |
| 06:22.49 | jets | about 30 sip phones and a few pri's for other things we do with *. |
| 06:23.11 | Defraz | oh cool you mind helping me with goofing around with learning. |
| 06:23.24 | brettnem | `Sauron: they gave me a "visitor" badge.. a nice slap in the face for 5 years of slave labor installing over $60 million of telecom gear |
| 06:23.25 | Defraz | do you tie it into your Softswitch for your LD |
| 06:23.27 | brettnem | hehe |
| 06:24.24 | brettnem | actually, it was a lot of fun working for them.. I kinda miss it.. |
| 06:24.47 | SafT | <offtopic> i went thru el paso once </offtopic> :P |
| 06:25.04 | `Sauron | I went to el paso once. I'm never going back there. Ever. |
| 06:25.28 | SafT | i wasnt silly enough to stop |
| 06:25.41 | *** part/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 06:25.48 | SafT | it was the only place i saw mexico though :> |
| 06:26.21 | neuro_[rus] | asterisk-1.0.5 works on AMD64 |
| 06:27.13 | Qwell | neuro_[rus]: good to know |
| 06:28.13 | brettnem | heh.. I flew into Odessa once.. I was suprised to see that the runway was paved |
| 06:28.16 | brettnem | :) |
| 06:29.01 | brettnem | I want to goto sleep |
| 06:29.06 | `Sauron | I'm about to |
| 06:29.10 | `Sauron | alarm goes off at 6 |
| 06:29.14 | brettnem | I got to figure this CDR crap out |
| 06:29.30 | brettnem | yeah, mine too.. actually, my alarm is in the form of a 3 year old little girl. :) |
| 06:29.36 | `Sauron | rofl |
| 06:29.45 | SafT | thats a little young isnt it? ;o |
| 06:29.46 | `Sauron | much more effective than KHFI blaring out the clock radio |
| 06:29.54 | brettnem | pfft |
| 06:29.55 | `Sauron | saft: That's uncool |
| 06:30.13 | brettnem | no kidding |
| 06:30.41 | SafT | :-/ |
| 06:30.51 | SafT | jokes like that fly around our office :o |
| 06:31.08 | SafT | sorry if you were offended |
| 06:31.45 | mishehu | "was that the tornado siren or my brat?" |
| 06:37.11 | *** join/#asterisk wasim (~wasim@203.81.213.118) |
| 06:38.34 | *** join/#asterisk brettnem (~brettnem@user-0ccsr2l.cable.mindspring.com) |
| 06:38.38 | brettnem | doh |
| 06:39.04 | unixgeek | Can anyone give me more information on what a SIP error code 604 really means? |
| 06:39.52 | `Sauron | Hum |
| 06:39.58 | `Sauron | vonage's complaining up a storm |
| 06:39.59 | `Sauron | sheesh |
| 06:40.36 | outtolunc | 604 Does Not Exist Anywhere"Reorder""Not Found (604)" |
| 06:42.01 | outtolunc | http://www.voip-info.org/wiki-SIP+response+codes (for future refernce) |
| 06:42.03 | unixgeek | What does that really mean? I have just gotten the correct username/password from Lingo and it looks like the SIP registration suceeds. |
| 06:42.26 | unixgeek | When I try to place a call over the SIP connection, I get the 604 error. |
| 06:42.56 | outtolunc | are you sure the extens is valid? |
| 06:43.18 | unixgeek | You mean the number that was being dialed? |
| 06:43.27 | unixgeek | Yes, it was my cell phone. |
| 06:43.57 | outtolunc | meaning the extension dialed is VALID from the sip server |
| 06:44.13 | unixgeek | Yes, it should be. |
| 06:44.25 | outtolunc | should be is not an answer |
| 06:44.39 | unixgeek | If I do a sip show registry. It shows the connection to Lingo as registered. |
| 06:44.52 | three55ml | unixgeek: Registry has nothing to do with outgoing |
| 06:45.01 | three55ml | unixgeek: Registry just tells Lingo where to find you |
| 06:45.02 | outtolunc | once upon a time when asked if the earth was flat, there were alot of people saying 'should be' |
| 06:45.17 | unixgeek | :-) |
| 06:45.21 | three55ml | unixgeek: Is your outgoing context right? |
| 06:46.43 | unixgeek | I also have a SIP entry for outgoing connections which has the same basic information (username/password) and sets the context to default. Same as my other outgoing connections. |
| 06:48.29 | unixgeek | I have configured my dialplan to be like most business systems (i.e. 9 to get an outside line) and then pass the rest of the number off to Dial(SIP/kinetic_compute, ${EXTEN:${TRUNKMSD}}) |
| 06:48.30 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 06:48.31 | three55ml | Run with asterisk with -vvvgc and see what you get |
| 06:48.48 | `Sauron | hum |
| 06:48.50 | `Sauron | what's the g do? |
| 06:48.55 | unixgeek | I see the dial initiate. |
| 06:49.24 | `Sauron | ah |
| 06:50.33 | unixgeek | Is there somewhere I can clip the messages (SIP debug and verbose output) for other to see. Maybe there is something that I am not seeing. |
| 06:55.17 | outtolunc | unixgeek: what happens if you remove that ',' |
| 06:55.42 | unixgeek | I will give it a try. |
| 06:56.23 | `Sauron | Hum, well ast_data does apply to HEAD again |
| 06:56.33 | unixgeek | Actually here is the exact dial string that is executing: Dial("Skinny/11@tele1-2", "SIP/kinetic_compute/12073185646") |
| 06:58.53 | unixgeek | Does this look right? INVITE sip:12073185646@as.bw.iprimus.net SIP/2.0 |
| 07:05.03 | *** join/#asterisk outtolunc (~chatzilla@adsl-69-110-26-49.dsl.pltn13.pacbell.net) |
| 07:05.21 | outtolunc | grr |
| 07:05.29 | *** join/#asterisk Fanguin (~Fanguin@p50819084.dip0.t-ipconnect.de) |
| 07:06.04 | outtolunc | unixgeek: what happens if you replace the ',' with a '/' |
| 07:06.55 | outtolunc | or if kinetic_compute is a context, rearrange to Dial(SIP/${EXTEN:${TRUNKMSD}}@kinetic_compute |
| 07:07.02 | unixgeek | Actually the Dial string has a / not a , Here are the last two messages I sent that you might not have seen. |
| 07:07.04 | unixgeek | Actually here is the exact dial string that is executing: Dial("Skinny/11@tele1-2", "SIP/kinetic_compute/12073185646") |
| 07:07.06 | unixgeek | Does this look right? INVITE sip:12073185646@as.bw.iprimus.net SIP/2.0 |
| 07:07.33 | netsurfer | ur curly braces are all wrong, unixgeek |
| 07:07.49 | unixgeek | Well, kinetic_compute is the entry in sip.conf. |
| 07:07.50 | netsurfer | ${varname} is the format |
| 07:08.24 | netsurfer | Dial(SIP/kinetic_compute, ${EXTEN:}${TRUNKMSD}) |
| 07:09.12 | netsurfer | not sure why u have : in there either? |
| 07:09.14 | outtolunc | i'd probably just try ${EXTEN:1} first <G> |
| 07:09.17 | unixgeek | netsurfer: that seems counter intuative (sp! too late at night) |
| 07:10.10 | netsurfer | oops I see what ur doing now |
| 07:10.23 | netsurfer | (too early in the morning, not had coffee yet) |
| 07:10.30 | netsurfer | ur right, im wrong |
| 07:10.37 | unixgeek | Well, it seems to be doing the right thing. I dial 912073185646 and as you can see in the dial string from the verbose output, 12073185646 gets dialed. |
| 07:10.55 | unixgeek | Ok. |
| 07:11.41 | outtolunc | i'm a zap/iax guy <G> |
| 07:12.10 | unixgeek | When a call is being placed through a VOIP service like Vonage or Lingo, is it normal to show as sip:12073185646@as.bw.iprimus.net in the sip debug output? |
| 07:12.33 | outtolunc | looks fine |
| 07:12.57 | netsurfer | yes unixgeek thats ok |
| 07:13.35 | netsurfer | unixgeek - sip show registry |
| 07:13.47 | netsurfer | u registered ok? |
| 07:14.36 | unixgeek | OK. It just seems that Lingo sees the phone number in the SIP address and just comes back with the 604 code. |
| 07:14.37 | unixgeek | Yes. I seem to be. When I execute sip show registry, I do see a line for the connection and it is saying registered. |
| 07:15.00 | netsurfer | k |
| 07:15.24 | netsurfer | check their dial prefixes |
| 07:15.43 | netsurfer | does 12073185646 make sense to them? |
| 07:16.28 | unixgeek | I am not sure. I have the VOIP modem that they sent me and there was a quick reference in it. Let me see if they show anything. Be right back. |
| 07:17.15 | outtolunc | try 12076215923 |
| 07:17.43 | outtolunc | thats a dialup # in augusta (l3) |
| 07:17.59 | *** join/#asterisk pranav (~dawda_pra@202.63.174.250) |
| 07:18.14 | pranav | hello |
| 07:18.22 | pranav | hello everyone |
| 07:18.28 | *** join/#asterisk justinnn (~dsf@solid.mpa.net.au) |
| 07:18.30 | justinnn | hey ppl |
| 07:18.33 | justinnn | how do i convert a .gsm to .wav ? |
| 07:18.46 | moonwick | sox |
| 07:19.09 | unixgeek | outtolunc: I get the same thing. Are you in Maine? |
| 07:19.56 | pranav | i am able to make calls internally but how to make calls to a pstn |
| 07:20.26 | unixgeek | netsurfer: I don't see anything about a prefix. There docs seem to show just dialing the straight number as if it were PSTN. |
| 07:20.30 | pranav | what changes do i need to make in the dial plan(extensions.conf) |
| 07:21.48 | wasim | pranav: Zap/1/${EXTEN} |
| 07:22.41 | netsurfer | unixgeek - 12073185646 <-- would that number work on any pstn in oz ? |
| 07:23.10 | unixgeek | It should. Standard US number. |
| 07:23.53 | netsurfer | ah ur in the us? |
| 07:23.54 | Koshatul | hey, does anyone know if i have one iax2 connection setup for a provider and i want to buy a g729 license for it, do i need a license for every concurrent call ? or just a license for the single connection ? |
| 07:24.10 | unixgeek | netsurfer: yes. |
| 07:24.59 | netsurfer | ah, I thought iprimus were an aussie company |
| 07:27.42 | unixgeek | Yes, to my understanding Primus (parent of iprimus) is based in Australia. But they have definately been expanding. I have a client that had their webhosting company bought out by Primus. And they own Lingo too. |
| 07:28.13 | netsurfer | unixgeek try 0012073185646 |
| 07:29.09 | outtolunc | well guys, kernel update done, time for bed |
| 07:29.16 | unixgeek | I think my dialplan will squash that, but I will try it. |
| 07:29.23 | Sedorox | 4569 OK (1922 ms) |
| 07:29.24 | Sedorox | ouch |
| 07:29.40 | netsurfer | unixgeek yeah dont forget the 9 |
| 07:29.50 | unixgeek | Yep. congestion tones at the second 0. |
| 07:30.37 | netsurfer | unixgeek - try adding an extension to test it then |
| 07:31.59 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 07:39.16 | netsurfer | heh.. its only 7.40am |
| 07:39.20 | netsurfer | ;) g'nite |
| 07:47.22 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 07:50.25 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 07:53.11 | *** join/#asterisk Pulu (~chatzilla@64.200.224.158) |
| 08:03.36 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 08:03.48 | wasim | fu RoyK |
| 08:04.01 | wasim | s/fu/hi :) |
| 08:04.06 | RoyK | morning |
| 08:04.08 | RoyK | ? |
| 08:04.15 | *** join/#asterisk eivindtr (~Eivind@193.91.146.34) |
| 08:14.16 | *** join/#asterisk hesjar72 (~rajesh@196.11.146.97) |
| 08:14.28 | *** join/#asterisk Poincare (~jefffnode@dD5779B07.access.telenet.be) |
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| 08:24.34 | *** join/#asterisk finnjet (~finnjet@port-212-202-30-187.dynamic.qsc.de) |
| 08:26.24 | finnjet | Is there somebody arround who could tell me how to make asterisk to continue proceeding in a Dialplan after the callee has hung up? |
| 08:26.39 | zoa | Deadagi is made for that purpose |
| 08:26.47 | finnjet | The switch for the dial command alone isn´t doing the trick. |
| 08:27.00 | zoa | i dont think that switch works |
| 08:27.07 | zoa | the switch works only in one direction i think |
| 08:27.30 | finnjet | Thats what I had to find out as well! |
| 08:27.51 | finnjet | The deadagi script isn´t called either unfortunately! |
| 08:29.09 | finnjet | I´m setting up a callthrough gateway for me and the problem is that it is jumping back to the enter your number priority as it should but the Waitexten Command exits as soon as I enter any digit! |
| 08:29.34 | finnjet | Funny enough not when the call has been aborted for some error or me pressing the # key! |
| 08:30.04 | finnjet | is there any kind of command to reset every variable that can be set within a dialplan? |
| 08:32.11 | RoyK | ~docs |
| 08:32.13 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 08:33.00 | finnjet | Could you give me a hint what to search for? |
| 08:34.57 | brc_ | simple enough |
| 08:35.01 | brc_ | use the h extension |
| 08:35.05 | finnjet | The problem accours since I put the dial cmd in a macro. |
| 08:35.38 | finnjet | this is what I´m doing it says goto the extension where the whole callthrough starts! |
| 08:37.15 | finnjet | Exactly where the caller is send to! But for some reason after all these announcements the call will be disconnected when the user "came over the h goto". |
| 08:41.09 | *** join/#asterisk oej (~oej@54.Red-80-32-211.pooles.rima-tde.net) |
| 08:42.54 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 08:53.59 | Sedorox | question with queues if anyone is around at this time.... I can't get it to logout.. followed the AgentCallbackLogin Wiki.. but the logoff still isn't working |
| 09:05.30 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 09:07.43 | Sedorox | night all |
| 09:08.07 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 09:09.19 | *** join/#asterisk meppl (~mephisto@pD9E6935B.dip.t-dialin.net) |
| 09:09.50 | finnjet | Is there a difference in general whether a dial command is beeing excecuted in the context itself or in a macro? The point is that since i put the dial command into a macro i cant continue dialing after a successfull call! |
| 09:12.30 | *** join/#asterisk vs_ (vs@univac.spamcheck.net) |
| 09:12.35 | vs_ | howdy |
| 09:13.41 | vs_ | getting plenty of those: Bridge stops because we're zombie or need a soft hangup: c0=SIP/105-83df, c1=SIP/113-23a0, flags: No,Yes,No,No |
| 09:15.45 | finnjet | maybe the Hangupcause is the problem can this variable be reseted? |
| 09:16.04 | finnjet | I tryed setvar(Hangupcause=) not a real success |
| 09:17.11 | RoyK | finnjet: exten => t,2,SetVar(PRI_CAUSE=xx) |
| 09:17.14 | JerJer | that is a read-only variable |
| 09:17.14 | RoyK | perhaps... |
| 09:17.27 | RoyK | JerJer: PRI_CAUSE isn't |
| 09:17.42 | JerJer | read the README.variables |
| 09:18.31 | *** join/#asterisk hundra (hundra@xtc.df.lth.se) |
| 09:19.01 | hundra | howdy |
| 09:19.24 | RoyK | hej |
| 09:19.29 | TheEmperor | can anyone help, i am still having trouble putting together * to call out at a certain time |
| 09:19.56 | RoyK | JerJer: according to README.variables, HANGUPCAUSE is not ro |
| 09:20.17 | oej | Then we need to change that, because I think it is readonly |
| 09:20.32 | JerJer | RoyK: needs to cvs up |
| 09:20.39 | RoyK | JerJer: running stable |
| 09:20.43 | JerJer | that is your problem then |
| 09:20.58 | JerJer | i wish stable was named DON"T_RUN_ME |
| 09:21.05 | JerJer | it is what will BECOME stable |
| 09:21.15 | hundra | a question; i've setup asterisk (.deb from unstable) and configured extentions for meetme.. i dial in, everything works fine, and when i exit the call i see the BYE-packet from my GW, asterisk acks.. but the sip channel is still open i asterisk, how do i force it to close properly? |
| 09:21.28 | RoyK | hundra: don't use debian packages |
| 09:21.35 | RoyK | hundra: use the source, luke |
| 09:21.36 | JerJer | run cvs -head |
| 09:21.48 | JerJer | cvs co asterisk ; cd asterisk ; make install |
| 09:21.51 | RoyK | (if you can stand a few crashes) |
| 09:22.01 | RoyK | -stable really _is_ stable |
| 09:22.08 | JerJer | lol thats funny |
| 09:22.10 | RoyK | only it lacks a few things |
| 09:22.14 | JerJer | run about 20,000 calls thru stable |
| 09:22.22 | JerJer | and see how many deadlocks you get |
| 09:22.28 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
| 09:22.29 | hundra | ok, i'll try compiling from the sources instead |
| 09:22.30 | RoyK | haven't got any yet |
| 09:22.31 | hundra | thanks |
| 09:22.32 | JerJer | and memory loss |
| 09:23.12 | RoyK | I'd rather reboot the server once a week than risk segfaults |
| 09:23.30 | JerJer | lol |
| 09:23.53 | JerJer | so you would prefer to run known inferir software |
| 09:23.56 | RoyK | also, I guess the masterminds behind asterisk perhaps know a little more about the stability of -stable vs -head than JerJer does, even if he thinks he's the mastermind of the universe......... |
| 09:23.57 | JerJer | inferior |
| 09:24.13 | Poincare | Hmmm, alway nice if it works 'out of the box' |
| 09:24.18 | harryvv | jerjet what is is your * using |
| 09:24.24 | harryvv | JerJer I mean :) |
| 09:24.32 | harryvv | ID is it using |
| 09:24.32 | JerJer | cvs -head as of tuesday |
| 09:25.31 | JerJer | RoyK: ask any core developer what they run and why |
| 09:28.07 | oej | Well, core developers need to run head in order to stress test them selves and their server :-) They also have the knowledge to handle it. |
| 09:28.40 | oej | And yes, it's a better product, which is a good thing since it will soon be stable. |
| 09:29.16 | *** part/#asterisk oej (~oej@54.Red-80-32-211.pooles.rima-tde.net) |
| 09:30.09 | JerJer | RoyK: so now go back to your corner and cry |
| 09:30.29 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:30.32 | harryvv | :) |
| 09:30.47 | harryvv | JerJer do you know what id asterisk should have |
| 09:30.53 | JerJer | id ? |
| 09:31.01 | JerJer | like uid ? |
| 09:31.11 | harryvv | yes |
| 09:31.14 | harryvv | is asterisk |
| 09:31.19 | harryvv | id asterisk |
| 09:31.26 | Delvar | morning |
| 09:31.27 | *** join/#asterisk alakon (~carbon9@bluesocket-protected-gw.wireless.rochester.edu) |
| 09:31.36 | alakon | hey |
| 09:31.44 | JerJer | there is nothing requiring asterisk to run as root... so it can be any UID u want to run it as |
| 09:31.59 | JerJer | i make an 'asterisk' user and group and lock it down to that |
| 09:32.17 | JerJer | then if anyone happens to find a remote exploit all they get is a chroot jail |
| 09:32.32 | alakon | very nice |
| 09:32.41 | harryvv | well, there is some issues with init.d loading asterisk and its directories. It is hangining on /dev/zap with permission issues. |
| 09:32.51 | JerJer | change the permissions |
| 09:32.57 | harryvv | its the way it should be |
| 09:33.06 | harryvv | it was checked earler today. |
| 09:33.21 | JerJer | they are not right if it hangs |
| 09:33.47 | JerJer | i don't use the init.d crap |
| 09:33.53 | JerJer | i just fire off safe_asterisk |
| 09:33.55 | harryvv | all files in dev/zap are crw-r--r-- |
| 09:34.08 | JerJer | and owned by whom? |
| 09:34.15 | *** part/#asterisk alakon (~carbon9@bluesocket-protected-gw.wireless.rochester.edu) |
| 09:35.05 | harryvv | my use of chown is limited if thats what your asking. It was root:root |
| 09:36.06 | JerJer | that is a problem then |
| 09:36.28 | JerJer | cuz others can only read from zap |
| 09:36.39 | JerJer | which is not going to cut it |
| 09:36.54 | harryvv | bkw pointed out its a id issue. |
| 09:37.02 | JerJer | ? |
| 09:37.29 | JerJer | just chown it, problem solved |
| 09:38.33 | harryvv | anyway /var/log/asterisk/messages states that there is a issue with the path to /dev/zap permission denied. I did a id asterisk for him and he said thats the problem. I have chowned -R asterisk:asterisk then it creates a issue where the softphone stops communicating with asterisk and cannot log into it. |
| 09:38.49 | harryvv | but |
| 09:39.03 | harryvv | It did start asterisk when rebooting the server. |
| 09:39.34 | JerJer | i don't use the init.d so i have no clue about taht |
| 09:39.48 | harryvv | what distro are you using? |
| 09:39.51 | JerJer | mine |
| 09:40.10 | harryvv | debian/bsd,slack ? |
| 09:41.01 | JerJer | MINE |
| 09:41.13 | harryvv | you made your own os? |
| 09:41.20 | JerJer | doh |
| 09:41.28 | JerJer | stock linux kernel + busybox + tiny login |
| 09:41.32 | JerJer | find a clue |
| 09:41.34 | harryvv | I see |
| 09:42.04 | finnjet | On my asterisk PRI_CAUSE is writeable. |
| 09:42.16 | finnjet | Unfortunately this didn´t fix anything! |
| 09:42.50 | JerJer | RoyK has no clue what he is talking about |
| 09:43.11 | finnjet | Maybe it is because the Dial Command returns -1 when a call is finished normaly and not 0. Where is this return code read? |
| 09:43.19 | finnjet | can it be reset in any way? |
| 09:43.41 | JerJer | why does it need to be reset? |
| 09:44.37 | finnjet | Because I have no idea why my callthrough dialplan is exiting when I regulary completed a call and then goto the start extension and dial again! |
| 09:45.04 | JerJer | then your extension logic is not correct |
| 09:45.23 | RoyK | finnjet: what are you trying to do? |
| 09:45.28 | RoyK | PRI_CAUSE works for me |
| 09:45.35 | RoyK | on -stable |
| 09:45.42 | finnjet | The point is that its working when there is a error in the connection (like 404 with SIP) |
| 09:45.53 | RoyK | eh |
| 09:45.59 | RoyK | PRI_CAUSE only works on zap |
| 09:46.05 | finnjet | But PRI_CAUSE has already been reset. |
| 09:47.24 | finnjet | Its a dialplan that collects the digits with a variable and starts dialing when the user hits #. This calls another context and a macro within this context. |
| 09:48.55 | finnjet | When the dial command in the macro exits it jumps back to the h context where I SetVar(PRI_CAUSE=) and the same with HANGUPCAUSE and DIALSTATUS. But for some reason * hangs up after the first digit then! |
| 09:49.14 | JerJer | because your logic is evil |
| 09:49.32 | JerJer | go back and think through the problem again |
| 09:51.24 | *** join/#asterisk math_ (~math@mail.nlcom.nl) |
| 09:52.02 | math_ | i have a problem with receiving callerid's on my analogue phone... |
| 09:52.53 | math_ | im using a digium tdm400p |
| 09:53.34 | math_ | it doesnt send any number, i also dont see anything about sending a number in de debug log |
| 09:53.52 | JerJer | are you setting a number ? |
| 09:54.13 | math_ | no, when i dial from my sipphone to a zap channel |
| 09:54.27 | math_ | then i dont get any callerid on my phone |
| 09:54.53 | math_ | also some small note... im using dutch phone equipment |
| 09:55.21 | JerJer | have you set zaptel to dutch then? |
| 09:55.56 | math_ | you mean the language of the channels? in /etc/zaptel.conf ? |
| 09:56.30 | tzafrir | Q: What version is Asterisk? What quality? A: 1.0, beta. Source: http://freshmeat.net/projects/asterisk |
| 09:56.58 | math_ | ok double checked it... language is nl in zapata.conf and zaptel.conf |
| 09:57.06 | *** join/#asterisk ReVoK (ReVoK@82.224.60.46) |
| 09:58.01 | tzafrir | JerJer, what was the problem again? |
| 09:58.51 | tzafrir | I remember having strange problems in my old busybox system because things I took for granted ddi not exist. |
| 09:59.03 | math_ | tzafrir : i dont receive caller id's on my analogue phone |
| 09:59.20 | ReVoK | hi |
| 10:00.36 | JerJer | hoe |
| 10:05.09 | *** join/#asterisk abbas_ (nidobas@203.81.200.28) |
| 10:06.30 | abbas_ | <PROTECTED> |
| 10:06.52 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 10:07.10 | RoyK | ~h323? |
| 10:07.11 | jbot | hmm... h323 is evil! Don't ask about h323 here. Ask JerJer if you need to bug someone. He says it works, but others don't. |
| 10:07.29 | abbas_ | hhhahaha |
| 10:07.54 | abbas_ | Royk do u know any help channel for H323? |
| 10:08.06 | RoyK | perhaps #asterisk-h323 |
| 10:08.09 | abbas_ | actually carriers have 90% traffic on h323 |
| 10:08.17 | RoyK | create it and add it on the wiki :) |
| 10:08.18 | JerJer | if you say so |
| 10:08.38 | abbas_ | haha jer jer how r u |
| 10:09.11 | abbas_ | so no one ther ? |
| 10:10.17 | JerJer | nope |
| 10:13.24 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
| 10:15.31 | finnjet | so there is no chance to ignore or overwrite a return code of a macro to 0 isntead of -1 ? |
| 10:16.19 | hajekd | what is your experience with jitterbuffer? Looks like it is better to disable it...;) |
| 10:19.35 | talkwebhosts | anyone here run a hosting business? |
| 10:20.15 | math_ | JerJer : any clue about those callerid's ? |
| 10:20.23 | tzafrir | Anyway, Debian packages currently have h323 compiled in |
| 10:20.36 | tzafrir | I haven't tried it, actually |
| 10:21.09 | tzafrir | math_, do you have caller id inside your PBX? |
| 10:25.42 | JerJer | hajekd: yes disable it |
| 10:25.57 | JerJer | tzafrir: don't run packages... get the source and do it right |
| 10:26.04 | JerJer | math_: no |
| 10:26.58 | math_ | tzafrir : yes, im dialing from a sip phone |
| 10:27.01 | JerJer | the h.323 support on -stable (which is what was used for packages) is absolutely not supported |
| 10:27.10 | math_ | when i dial to another sip phone, it works fine |
| 10:44.30 | *** join/#asterisk djin (~marius@62.58.40.196) |
| 10:44.40 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 10:44.51 | abbas_ | jer jer: we tried to deploy * between Cisco AS5300 and a 4 port GSM gateway we just recieved call from AS5300 on * and forwarded at GSm SIP GW supporting 729 but there was no voice when we use any hardfone reg on * the cakll on same route workes well |
| 10:45.07 | JerJer | ok and? |
| 10:45.36 | abbas_ | the user behind AS5300 cant get voice neither he nor us |
| 10:45.49 | JerJer | this is supposed to help me figure your problem out how? |
| 10:46.13 | abbas_ | what u this where is the peroblem ? |
| 10:46.27 | abbas_ | is there anything in configuration of AS5300 and * |
| 10:46.39 | Mavvie | talk about nasty behaviour |
| 10:47.05 | JerJer | then change the signaling type |
| 10:47.06 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 10:47.15 | Mavvie | JerJer: change it to what? |
| 10:48.29 | abbas_ | jer jer: what u suggest for me ? |
| 10:48.48 | Mavvie | JerJer: oh sorry. thought you were replying to me. |
| 10:49.42 | abbas_ | changing signaling type was for me ???? |
| 10:50.12 | abbas_ | no sure not for me |
| 10:51.41 | JerJer | i'll let you two figure it out |
| 11:01.29 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
| 11:01.39 | bowman | hi. what does "unable to forward voice" in chan_sip.c mean? firewall trouble? |
| 11:04.09 | Poincare | cool, i got the same problem :-) |
| 11:07.40 | *** join/#asterisk TeLLuS (~johan@h187n2fls31o858.telia.com) |
| 11:10.17 | bowman | what does error 403 mean? |
| 11:10.39 | *** join/#asterisk qwerp (~abc@219.93.57.58) |
| 11:10.43 | qwerp | harlo.. |
| 11:10.58 | qwerp | anyone here uses TE110P ? |
| 11:13.28 | *** join/#asterisk soulz- (~Soulz-@host-137-132-45-213.imcb.nus.edu.sg) |
| 11:13.31 | soulz- | hi all |
| 11:13.49 | soulz- | i just got a quad card, tdm04b and tdm40b |
| 11:14.24 | soulz- | is anyone here? |
| 11:14.57 | qwerp | i have that too.. |
| 11:15.02 | qwerp | anything that i can help? |
| 11:15.20 | soulz- | thanks qwerp |
| 11:15.33 | soulz- | just want to find out, how to load the wctdm |
| 11:15.39 | soulz- | as i did modprobe wctdm |
| 11:15.54 | soulz- | on syslog it shows mod 1, auto fxo |
| 11:15.59 | soulz- | and so on until 3 |
| 11:16.08 | soulz- | and mod 0 auto fxs |
| 11:16.13 | soulz- | and so on until 3 |
| 11:16.31 | soulz- | but how do i reference if its a fxo and fxs? |
| 11:16.35 | qwerp | errmm.. |
| 11:16.45 | qwerp | guess u using the latest zaptel drivers. |
| 11:16.51 | qwerp | coz if u using v1-0, |
| 11:17.04 | soulz- | i just built it from cvs |
| 11:17.07 | qwerp | tdm is still called wcfxs,wcfxo |
| 11:17.09 | soulz- | so it should be latest |
| 11:17.17 | qwerp | i tried last time, |
| 11:17.25 | qwerp | it is still buggy.. |
| 11:17.52 | soulz- | it recognized the wctdm |
| 11:18.16 | qwerp | coz when i used that, the callerid on the fxs port won't work.. |
| 11:18.27 | soulz- | http://pastebin.ca/5929 |
| 11:18.46 | *** join/#asterisk pranav (sameer@202-149-48-200.broadband.isp.exatt.net) |
| 11:18.52 | pranav | hello |
| 11:18.57 | pranav | hello everyone |
| 11:19.15 | talkwebhosts | hello |
| 11:19.27 | soulz- | i am using fxs and fxo, on the tdm cards |
| 11:19.33 | qwerp | u modprobe wctdm ? |
| 11:19.38 | soulz- | yup |
| 11:19.41 | qwerp | humm.. |
| 11:19.41 | soulz- | when i do that |
| 11:19.42 | bowman | what does error 403 mean? |
| 11:19.42 | pranav | i am able to call internally but i dont know how to connect to pstn |
| 11:19.46 | soulz- | it picked up all 4 |
| 11:19.49 | soulz- | all 8 |
| 11:19.53 | soulz- | 4 for fxo |
| 11:19.58 | soulz- | and 4 for fxs |
| 11:20.12 | qwerp | i dun see any errors.. |
| 11:20.15 | soulz- | pranav: have u got ur 100xp? |
| 11:20.30 | soulz- | i know, i am not worried about that, i know its correct |
| 11:20.30 | pranav | soulz: yes |
| 11:20.40 | soulz- | but how do i reference it |
| 11:20.48 | soulz- | have u set up ur zaptel? |
| 11:20.54 | qwerp | yup.. |
| 11:21.02 | *** join/#asterisk beedauchon (foreigner@3ffe:bc0:8000:0:0:0:0:1d7) |
| 11:21.18 | soulz- | pranav: use ztcfg -vv |
| 11:21.24 | pranav | see i have a sipura device with 2 channels connecte to it and i can talk between them |
| 11:21.48 | pranav | ya it shows 1 channel configured |
| 11:22.01 | qwerp | brb |
| 11:22.07 | soulz- | then do this |
| 11:22.17 | soulz- | exten => _XXXXXX,1,Dial,Zap/g1/${EXTEN} ; |
| 11:22.20 | pranav | but on the cli prompt when i give zap show channels it says "unable to find the channel |
| 11:22.28 | soulz- | presuming u have set up in g1 |
| 11:22.44 | soulz- | maybe u should try with g1 for group 1 |
| 11:24.34 | pranav | ok tell me i have the two sipura phones which are internal, can i make a pstn call from that |
| 11:25.35 | soulz- | as long as ur in the same context |
| 11:25.37 | soulz- | sure |
| 11:25.42 | soulz- | gtg |
| 11:25.44 | *** join/#asterisk zeedo (~notroot@www.bsrf.org.uk) |
| 11:25.47 | *** part/#asterisk soulz- (~Soulz-@host-137-132-45-213.imcb.nus.edu.sg) |
| 11:26.37 | TeLLuS | Gaa, In 1.0.5 when using skinny with Cisco 7910 the phone is ringing while I'm in the middle of a call using it.. and then the call get disconnected. Ahh, now it feels better again.. |
| 11:27.17 | pranav | soulz: i have used the context default everywhere |
| 11:31.37 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011542342pcs.mainf01.in.comcast.net) |
| 11:36.43 | *** join/#asterisk hajekd (~hajekd@mail.idoox.com) |
| 11:36.59 | pranav | hello someone |
| 11:41.26 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 11:45.01 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 11:45.47 | pranav | hi zeeek |
| 11:46.58 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-180-108.dsl.scarlet.be) |
| 11:47.52 | wasim | re Zeeek |
| 11:51.06 | pranav | hi wasim |
| 11:52.47 | pranav | see when i give ztcfg -vv it shows 1 channel configured, but when on the cli prompt i give "zap show channel 01" it says unable to find channel 1 |
| 11:54.17 | *** join/#asterisk kks (~kks@203.115.210.253) |
| 11:54.30 | wasim | zap show channel 1 |
| 11:56.28 | pranav | it shows the same thing" unable to find the given command |
| 11:56.39 | qwerp | harlo |
| 11:56.51 | wasim | pranav: and you've configured it in zapata.conf? |
| 11:56.54 | qwerp | how can i configure a te110p card? |
| 11:57.05 | wasim | qwerp: give it a swift kick up the backside |
| 11:57.16 | qwerp | ???? |
| 11:57.30 | *** join/#asterisk Nix (~Nix@dsl81-214-9283.adsl.ttnet.net.tr) |
| 11:57.32 | qwerp | do i need to compile libpri also? |
| 11:57.37 | pranav | yes |
| 11:57.38 | wasim | qwerp: yep |
| 11:57.59 | qwerp | i compile it after i compiled zaptel and asterisk, is it ok? |
| 11:58.29 | tzafrir | asterisk need libpri |
| 11:58.46 | qwerp | so, meaning i have to recompile asterisk, am i right? |
| 11:58.57 | tzafrir | yup |
| 11:59.01 | qwerp | thanz... |
| 11:59.40 | tzafrir | according to the motto of that soundguy you could think Windows was destroyed in 1995 |
| 12:04.23 | *** join/#asterisk fuzza (~andrew@ppp171-147.lns1.per1.internode.on.net) |
| 12:04.26 | fuzza | hi all |
| 12:05.28 | fuzza | with a call queue (with only one member), is there a way to disable retrying the member on timeout, so it only actually calls once? (it's not overly important, I can just halve the intended timeout, just curious) |
| 12:09.48 | qwerp | tzafrir: can i ask u a question? |
| 12:09.57 | tzafrir | ask ahead |
| 12:10.33 | qwerp | why my te110p card, when i restarted my machine, i modprobe wcte11xp it cannot detected my card. |
| 12:10.55 | qwerp | everytime i have to shutdown, remove the card and insert it again, then it is able to detect. |
| 12:11.42 | finnjet | Is there a chance to ignore or overwrite a return code of a macro to 0 instead of -1 ? |
| 12:12.00 | tzafrir | qwerp, maybe the module was already loaded? |
| 12:12.11 | tzafrir | grep zaptel /proc/modules |
| 12:12.30 | qwerp | after i type modprobe wcte11xp |
| 12:12.38 | qwerp | it comes out error.. |
| 12:12.49 | qwerp | then i lsmod, it shows wcte11xp is loaded |
| 12:13.05 | qwerp | but when i zttool it, i dun see any card there.. |
| 12:13.27 | tzafrir | what I mean is: maybe its gets loaded somehow before your modprobe? Or maybe a different module? |
| 12:13.55 | tzafrir | what is the error? |
| 12:14.00 | qwerp | wait... |
| 12:14.39 | kks | can asterisk accept callerid with SPACE? |
| 12:16.43 | qwerp | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 12:17.07 | qwerp | FATAL: Error running install command for wcte11xp |
| 12:18.35 | tzafrir | where did I see this error? |
| 12:18.43 | jalsot | could anybody explain how does codec selection work on IAX2? e.g. pstn->*->iaxComm [which party decides which codec to use? |
| 12:19.09 | qwerp | the device is detected by the PC.. |
| 12:19.16 | qwerp | lspci, the device is there.. |
| 12:19.30 | qwerp | lsmod, module is loaded, |
| 12:19.38 | qwerp | but zttool, no device configured. |
| 12:19.53 | tzafrir | ztcfg -vv |
| 12:20.00 | qwerp | unless i shutdown it, remove the card and plug it in again.. |
| 12:20.41 | qwerp | a list of 31 channels and |
| 12:20.45 | qwerp | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 12:22.22 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 12:23.03 | qwerp | btw, show i configure all the digium cards using kudzu? |
| 12:27.40 | qwerp | tzafrir: so, any ideas? |
| 12:28.27 | tzafrir | qwerp, what version of zaptel do you use? |
| 12:28.42 | qwerp | v1-0 |
| 12:29.19 | qwerp | like i said, if i unplug and plug back in, it works fine. |
| 12:29.32 | qwerp | just when i restart the computer, the problem comes. |
| 12:33.09 | qwerp | somebody help me? |
| 12:37.16 | *** join/#asterisk foxb_ (~chatzilla@hip2-247-93-148-209.dlup-mtl.progression.net) |
| 12:38.01 | qwerp | tzafrir: how? |
| 12:38.17 | *** join/#asterisk GodThor (~ninja@212.110.95.139) |
| 12:39.25 | foxb_ | what are processor requirements per call? with 1GHz and 512 RAM how many simultinuosly call I can support? |
| 12:39.49 | *** part/#asterisk fuzza (~andrew@ppp171-147.lns1.per1.internode.on.net) |
| 12:40.03 | netsurfer | foxb_ - how long is a piece of string ? |
| 12:40.16 | Zeeek | this long |
| 12:40.23 | netsurfer | hey Zeeek |
| 12:40.44 | netsurfer | just got home to find my dev box has been playing around :( |
| 12:41.59 | tzafrir | qwerp, what version is the zaptel module? |
| 12:42.30 | tzafrir | anyway, this appears like bad hardware |
| 12:42.38 | qwerp | v1-0 |
| 12:42.39 | netsurfer | 0-order allocation failed <-- anyone seen that before ? |
| 12:42.44 | qwerp | checkout from CVS |
| 12:43.53 | foxb_ | netsurfer: A piece of string is long, but limited by realization. I want to know the processor requirements to maintain call. |
| 12:43.53 | tzafrir | any chance that the zaptel module and ztcfg are not from the same version? |
| 12:43.53 | qwerp | if that is the case, funny thing is when i unplug and plug it back again, it worked.. |
| 12:43.53 | qwerp | its the same... |
| 12:43.53 | netsurfer | foxb_ - depends if you are transcoding or just passing thru |
| 12:44.44 | GodThor | in db_connect.php says that PEAR must be install, what is pear? |
| 12:45.24 | foxb_ | netsurfer: passing thru mainly, but if you have the info for transcodin it will be usefull too... |
| 12:45.43 | GodThor | i think this is a problem because i cannot open asterisk manag. portal link , mysql error |
| 12:46.31 | tzafrir | PEAR is php-pear |
| 12:46.35 | tzafrir | php4-pear |
| 12:47.13 | GodThor | astersik started ok, fop started ok, db everything is ok(user,pass,host ,db) |
| 12:47.13 | netsurfer | kernel: __alloc_pages: 0-order allocation failed (gfp=0x1f0/0) damnit this looks bad |
| 12:47.13 | *** join/#asterisk pranav (sameer@202.149.48.200) |
| 12:47.25 | tzafrir | AMP uses mysql through PEAR's database abstracion layer |
| 12:47.25 | *** part/#asterisk qwerp (~abc@219.93.57.58) |
| 12:47.34 | foxb_ | netsurfer: it is about 90 calls and up. Can P4 3.0GHz handle it? |
| 12:47.44 | tzafrir | GodThor, what distro |
| 12:47.58 | netsurfer | foxb_ - for pass thru, no problem at all |
| 12:48.06 | GodThor | php-pear-4.3.9-3 i have installed already |
| 12:48.28 | GodThor | tzaf, what you mean? |
| 12:48.30 | pranav | i have pasted my zapata.conf in the pastebin.ca/5931 |
| 12:48.36 | netsurfer | foxb_ - I recall someone saying a p3 700 could handle over 300 calls on passthru |
| 12:49.30 | tzafrir | GodThor, and both it and mod_php itself are installed from distro packages? |
| 12:51.08 | foxb_ | netsurfer: will * gain if I use Opreron in 64 bit mode? Passthu - means from ISDN call to internet (using DSP on the digium card)? |
| 12:51.20 | GodThor | when i click on asterisk manag. portal link ,open blank page with: " mysql:// bla bla |
| 12:51.35 | pranav | can anyone tell me where is the mistake |
| 12:52.07 | foxb_ | netsurfer: or drivers are optimized for 32-bit mode? |
| 12:53.54 | finnjet | Is it possible that the g swith of the dial command causes almost nothing? |
| 12:57.03 | pranav | when it shows that it is configured in the root prompt then why it says that unable to find the channel in the cli prompt |
| 12:57.12 | *** join/#asterisk JunK-Y (~junky@modemcable056.110-81-70.mc.videotron.ca) |
| 12:57.18 | JunK-Y | in europe BRI is 30B+2D ? |
| 12:57.55 | RoyK | eh |
| 12:57.57 | RoyK | no |
| 12:58.10 | JunK-Y | its what exactly ? im just familiar with PRI |
| 12:58.12 | RoyK | PRI in europe is 30B+1D+1 channel for sync |
| 12:58.15 | JunK-Y | which is 23B+1D |
| 12:58.17 | RoyK | BRI is 2B+1D |
| 12:58.28 | GodThor | isdn |
| 12:59.32 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
| 13:00.02 | pranav | my asterisk is connected to a pstn via X100p card and not directly |
| 13:00.21 | pranav | so can i make calls to pstn via the internal phones |
| 13:00.54 | GodThor | tzaf, i found the solution :))) i missed to install php-mysql :)))))))))))) |
| 13:02.23 | hundra | there seem to be multiple zaprtc-sources available on the net, which one should i use? |
| 13:04.46 | soulz0 | hi all |
| 13:06.03 | *** join/#asterisk TheEmperor (TheEmperor@218.111.50.4) |
| 13:10.56 | *** join/#asterisk negativecreep (~yama@202.147.174.97) |
| 13:15.01 | *** join/#asterisk microlab (~chatzilla@203.88.33.179) |
| 13:15.48 | microlab | what is asterisk? how can I use it? |
| 13:16.50 | microlab | I know it is software for softswitch, but how to use it? |
| 13:16.58 | RoyK | ~docs |
| 13:16.59 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:18.52 | *** join/#asterisk benno2 (~benno2@host250-15.pool80182.interbusiness.it) |
| 13:19.16 | benno2 | hi, any good PDA that can be used as a voip SIP phone ? (without much echo issues) |
| 13:19.17 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 13:19.28 | microlab | thank you |
| 13:21.51 | *** part/#asterisk GodThor (~ninja@212.110.95.139) |
| 13:24.16 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk) |
| 13:25.29 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 13:25.32 | __Sparks_ | Hi! - I am having trouble dialling out with asterisk with a 100p card (Dialing from a SIP phone) - If I dial 9 then the number I want to call, i just get the PSTN dial tone |
| 13:26.32 | Delvar | it sounds like your TXGAIN is too low for the pstn ine to pickup diling sequance.. also is it set to tone or pulse dial? |
| 13:26.41 | __Sparks_ | (If i then redail the external number, it works) |
| 13:26.51 | Delvar | then forget everything i just said |
| 13:27.03 | __Sparks_ | okay! |
| 13:27.05 | _Brian | _Sparks_ what does your dialplan look like for external calls |
| 13:27.19 | _Brian | _Sparks_ could you show your diaplan and a debug in pastebin.ca? |
| 13:27.55 | _Brian | s/diaplan/dialplan |
| 13:28.03 | __Sparks_ | _brian, I am not sure :S - I have just installed Xorcom Rapid |
| 13:28.38 | _Brian | hmm...i never installed one of those prefangled Asterisk distributions |
| 13:28.42 | *** part/#asterisk foxb_ (~chatzilla@hip2-247-93-148-209.dlup-mtl.progression.net) |
| 13:28.44 | __Sparks_ | (I am very new to asterik! |
| 13:29.09 | _Brian | but, it should still be the same, dont you have a console connection to asterisk, so you can see what it is doing? |
| 13:29.09 | __Sparks_ | in my phones.conf... |
| 13:29.15 | Koshatul | __Sparks_: it sounds right, are you saying if you try to call back an external number it fails, but if you dial 9 for a external line you hear a dial tone ? |
| 13:29.43 | Koshatul | __Sparks_: what happens if you dial 9 then a valid number ... eg 90712345678 |
| 13:30.00 | _Brian | Koshatul: Hey!! why you giving out my phone number? |
| 13:30.02 | Koshatul | (of course that's not a valid number, and it's australian) |
| 13:30.04 | _Brian | :) |
| 13:30.24 | Koshatul | _Brian: your phone number? that's my luggage code ! |
| 13:30.29 | __Sparks_ | Koshatul, if i dial 9 followed by a valid PSTN number, i just get the PSTN dial tone, then if i dial the PSTN number again, it will connect |
| 13:30.41 | tzafrir | __Sparks_, I believe that there has been such a problem solved with the latest package from updates.xorcom.com |
| 13:30.46 | _Brian | Koshatul: rofl |
| 13:30.50 | *** join/#asterisk vmlinux (~dc@wsip-68-15-253-140.dl.dl.cox.net) |
| 13:31.11 | __Sparks_ | tzafrir, oh ok, i will have a look then! |
| 13:31.13 | Koshatul | __Sparks_: at a guess, the dial line is borked, the ${EXTEN:1} might be missing or mistyped, pastebin.ca all the way |
| 13:31.45 | tzafrir | __Sparks_, check in the menu (system information => package versions) what is the version of asterisk |
| 13:31.53 | tzafrir | Is it 1.0.5-1.8 ? |
| 13:32.09 | Koshatul | tzafrir: menu ? |
| 13:32.24 | __Sparks_ | it's 1.0.5 |
| 13:32.31 | *** part/#asterisk djin (~marius@62.58.40.196) |
| 13:32.53 | Koshatul | i had to go back to 1.0 because my iax supplier wouldn't seem to connect on the latest |
| 13:32.53 | tzafrir | Koshatul, of Rapid |
| 13:33.07 | tzafrir | __Sparks_, but is it -1.8 ? |
| 13:33.21 | Koshatul | tzafrir: ahhh, phone, i've only got two 7960's |
| 13:33.42 | Moc | it why I flushed IAX for sip now, doesnt have interoperabelity problems |
| 13:33.52 | *** join/#asterisk adnans (~adnans@noterik2.demon.nl) |
| 13:34.00 | __Sparks_ | sorry, I was in the wrong menu - o, it's 1:1.0.5-1.4 |
| 13:34.05 | ariel_ | morning all |
| 13:34.27 | Koshatul | moc:have you ever heard of jitter problems using hardware routers ? |
| 13:34.41 | *** join/#asterisk pointer (pointer@aj.catt.com) |
| 13:35.11 | Moc | nope, well yes if it has traffic and aint configure to give priority so audio packets |
| 13:35.13 | Koshatul | i bought a bigpong cable connection purely for the voip, and it was ok with a pc as a router, but i had issues with bgp, so i switched to static routes and a Netgear FR114P, now i get jitters .... i can't win |
| 13:35.37 | Koshatul | Moc: it has no other traffic, bar heartbeats and iax :) |
| 13:35.46 | Moc | try SIP... |
| 13:35.53 | Moc | that might fix your problem !! :( |
| 13:36.05 | Moc | I know it did for me |
| 13:36.10 | Koshatul | Moc: damn, that means contacting the supplier ... i hate doing that |
| 13:36.20 | Koshatul | i want a voip protocol that can survive a source change |
| 13:36.32 | Moc | Koshatul, they why there is alot of provider on the market with website |
| 13:36.38 | Koshatul | and i want bigpond cable to drop the routes when it isn't authed |
| 13:36.43 | pointer | who was the person that was going to start (or had) work on the application API? |
| 13:36.52 | Koshatul | Moc: ? |
| 13:37.15 | *** join/#asterisk microlab (~chatzilla@203.88.33.179) |
| 13:38.03 | Koshatul | the linux router would have supported tos defs |
| 13:38.07 | *** join/#asterisk simonides (simon@byte.unitycode.org) |
| 13:38.47 | Moc | there could be alot of factor for your problem.. |
| 13:39.03 | Moc | IAX is one of them, your provider link quality is another, your voip provider is a nother... |
| 13:39.34 | Koshatul | the problem i have with sip is (i believe) it has issues with quick change over of connection, iax just adds another agent (i believe) sip doesn't support that |
| 13:39.36 | Koshatul | scrap that |
| 13:39.47 | Koshatul | what provider do you use ? |
| 13:40.36 | Moc | currently, voiceconduits + a canadian provider for my DID and soon all my outgoing local |
| 13:40.45 | Koshatul | ahhh, i'm in aust |
| 13:41.33 | Koshatul | i'm using atp at the moment |
| 13:41.45 | Koshatul | i just hate contacting them, it's either no response or slow response |
| 13:42.09 | Koshatul | (the feeling i've got is that) if they think your question is stupid they don't reply. |
| 13:42.40 | Koshatul | when i was using asterisk latest, and iax was borked, i emailed asking if there were any issues .... i got no reply, still none |
| 13:43.34 | Moc | that a crappy provider |
| 13:44.16 | Koshatul | other than that it's been good |
| 13:44.32 | Koshatul | i've had some bad phone calls quality wise, but most of those were due to adsl load |
| 13:44.53 | Koshatul | i've got 1.5mb adsl here and "unlimited" cable, cable for voip, adsl for everything else |
| 13:45.15 | Koshatul | with a ibgp setup in the middle, so voip should fall over to the adsl |
| 13:45.30 | Koshatul | but the problem i have is that bigpond cable can be down, but you still have a route and connectivity |
| 13:45.54 | Koshatul | at least connectivity to the connected network, no furthur |
| 13:47.45 | Koshatul | and on the other side, pppd won't remove a default route and replace it, *or* add a default router with a lower metrix |
| 13:47.49 | Koshatul | metric even |
| 13:50.02 | __Sparks_ | Can someone tell me how to increase the gain on calls goung out of asterisk to the PSTN? |
| 13:50.09 | vmlinux | Does SJphone still work with asterisk? I'm new to all of this, but I have the demo up and going, and things seem to be set up right but i keep getting "Registration from 'sip:foo@bar.org' failed for '192.168.x.x' errors. Could toss a glance at my config and point out my failings? http://pastebin.ca/5934 |
| 13:50.20 | netsurfer | __Sparks_ - zapata.conf |
| 13:50.32 | vmlinux | could *someone" toss a glance |
| 13:52.30 | Koshatul | sjphone runs on a pda ? |
| 13:52.31 | *** join/#asterisk CpuID (~none@CPE-203-45-152-22.qld.bigpond.net.au) |
| 13:52.32 | Koshatul | sweet |
| 13:52.47 | CpuID | hey ppls, is there a cvsweb for asterisk/zaptel anywhere? cant seem to find one on asterisk.org |
| 13:54.08 | vmlinux | no |
| 13:54.16 | Koshatul | it sounds stupid, but i want a sip phone for my symbian phone :\ |
| 13:54.17 | vmlinux | it's on my laptop under xp at the moment |
| 13:54.25 | vmlinux | oh nm |
| 13:54.32 | vmlinux | thought you were replying to my question :P |
| 13:54.41 | Koshatul | :) |
| 13:54.44 | Koshatul | i'm downloading it now |
| 13:54.48 | vmlinux | That a good phone? |
| 13:54.51 | Koshatul | but i'm using asterisk 1.0 |
| 13:54.53 | vmlinux | I have a treo 600, I like it. |
| 13:54.58 | Koshatul | i've got a nokia 6600 |
| 13:55.12 | Koshatul | i don't quite like the idea of a ms phone, so i've avoided those so far |
| 13:55.20 | vmlinux | yea me too |
| 13:55.27 | Koshatul | but i can make my own apps for it :) |
| 13:56.02 | vmlinux | I kind of wish I knew how to write apps for palm os |
| 13:56.15 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 13:56.34 | vmlinux | I've only ever programmed anything decent in python. |
| 13:56.41 | Koshatul | i used to write some apps for palm, but they were pretty bad |
| 13:56.54 | vmlinux | It's really hard to go to other languages if you start out on something like python heh. |
| 13:57.06 | Koshatul | :) |
| 13:57.16 | ariel_ | <PROTECTED> |
| 13:57.20 | Koshatul | man, my friend in high school used to push me with assembler |
| 13:57.21 | ariel_ | <PROTECTED> |
| 13:57.25 | *** join/#asterisk lung (~lung@24-148-96-186.ip.mhcable.com) |
| 13:57.26 | ariel_ | <PROTECTED> |
| 13:57.28 | ariel_ | wow network just got me. |
| 13:57.35 | Koshatul | ? |
| 13:57.40 | vmlinux | heh, thanks for testing your lag on us kk thx :P |
| 13:57.52 | ariel_ | vmlinux, which is the setup for the sjphone on your pastebin |
| 13:58.10 | vmlinux | [linuxowns.org] |
| 13:58.18 | Koshatul | setting up sjphone now |
| 13:58.32 | ariel_ | vmlinux, actually it was my baby that hit a key on the keyboard. I did not see she had done it. |
| 13:58.38 | vmlinux | hehe |
| 13:58.51 | vmlinux | I've done worse, and I don't have a baby to blame it on :) |
| 13:58.59 | vmlinux | like accidentlaly pasting a binary |
| 13:59.09 | ariel_ | she is 23 month old and a hand full. |
| 13:59.10 | CpuID | hmm anyone here managed to get * running on ppc fine? :) |
| 13:59.13 | vmlinux | I'll bet |
| 13:59.21 | CpuID | just out of curiosity |
| 13:59.35 | vmlinux | I have enough trouble juggling my 2 left thumbs and a terminal :) |
| 13:59.43 | ariel_ | but back to your setup about the phone not working. |
| 13:59.58 | vmlinux | sorry I'm a newbie, I'm sure my config is awful |
| 14:00.03 | Koshatul | i'm still trying to setup sjphone |
| 14:00.07 | vmlinux | I've been reading the docs, just a lot to digest. |
| 14:00.11 | ariel_ | first lets add disallow=all and allow=ulaw for testing. |
| 14:00.21 | vmlinux | ok, to sip.conf right? |
| 14:00.29 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 14:00.29 | CpuID | oh actually ppls, who here has used snom handsets? any opinions? comparison of quality vs cisco handsets, just for basic featureset usage? |
| 14:00.39 | ariel_ | yes I would put them in the general section. |
| 14:00.51 | Koshatul | i'm going to sjphone first, since i have no idea how to set it up :) |
| 14:01.10 | ariel_ | you have dynamic 2 times and why fromuser? |
| 14:01.45 | ariel_ | Koshatul, have fun with it. I use xlite for most of the softphones here. |
| 14:01.46 | vmlinux | Well, I had fromuser commented out, but I uncommented it just trying to brute force it since I wasn't getting anywhere. |
| 14:02.16 | ariel_ | your on the same segment as the asterisk server? |
| 14:02.20 | vmlinux | I'm a victim of my own brute forcing a problem I don't understand :) |
| 14:02.24 | vmlinux | yes |
| 14:02.59 | vmlinux | the other system goes to linuxowns.org, which bounces off my nat and is directed to 192.168.0.5 |
| 14:03.15 | vmlinux | but both are on the same subnet |
| 14:03.16 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
| 14:03.21 | ariel_ | after you sip reload then turn off the softphone and back on see what it gives you the cli |
| 14:03.34 | vmlinux | ok |
| 14:04.20 | vmlinux | handle_request: Registration from '<sip:dcarroll@linuxowns.org>' failed for '192.168.0.1' |
| 14:04.41 | vmlinux | comment out fromuser? |
| 14:04.43 | elric | i am yet to find a decent *nix IAX2 softphone |
| 14:04.47 | vmlinux | I read that it isn't necessary usually |
| 14:05.09 | ariel_ | in the name remove the .org just use the linuxowns |
| 14:05.16 | vmlinux | ok |
| 14:06.23 | ariel_ | and your using user name on the sjphone as linuxowns correct. |
| 14:06.32 | vmlinux | just changed it |
| 14:06.38 | vmlinux | to linuxowns only |
| 14:07.06 | vmlinux | wait |
| 14:07.20 | vmlinux | no I was using linuxowns.org as the proxy name, dcarroll as the user name |
| 14:07.21 | ariel_ | iax2 in my view is great but for taking between asterisk servers. |
| 14:07.48 | ariel_ | ok your setup user name is linuxowns |
| 14:07.54 | vmlinux | omg that's it |
| 14:08.08 | vmlinux | thank you so much, I can't belive I beat my head against the wall on that all night |
| 14:08.14 | elric | i just like the idea that you dont have a seperate channel for signalling et al |
| 14:08.18 | ariel_ | so change the [linuxowns] to [dcarroll] |
| 14:08.26 | vmlinux | ok |
| 14:08.47 | vmlinux | woohoo, awesome :) this is going to be such a neat system |
| 14:09.05 | ariel_ | iax2 is great for getting you through firewalls but it has issues as well. |
| 14:09.11 | vmlinux | my gf only has a cell phone, and I'm wanting her to be able to call me from her computer so that she doens't hammer the minutes so bad. |
| 14:09.12 | *** join/#asterisk ReVoK (~ReVoK@82.224.60.46) |
| 14:09.21 | ReVoK | hello |
| 14:10.04 | ariel_ | vmlinux, if this is your first start at it. you should take a look at asterisk@home it's a great setup and fast to get it going. |
| 14:10.12 | vmlinux | ok |
| 14:10.14 | ReVoK | how many time to compile openh323 with p4 3.2 ?? :) |
| 14:10.18 | elric | yeah, my main concerns are firewalls right now though. i just need a *nix IAX2 client and theres only iaxComm which isnt too decent. |
| 14:10.44 | ariel_ | h323 is a always a problem and takes for ever to compile. |
| 14:10.55 | vmlinux | is that a distribution? |
| 14:11.09 | elric | ReVoK, wouldnt take too long on a 3.2ghz box provided it has decent amount of ram. |
| 14:11.09 | ReVoK | forever is more than 1 hour? |
| 14:11.10 | ariel_ | elric, for quick and dirty iax client I use diax. |
| 14:11.17 | elric | should be a lot better than my 450mhz system |
| 14:11.37 | vmlinux | bah, don't answer that I'll just check it out =) |
| 14:11.38 | elric | ariel_, diax it is. |
| 14:12.12 | vmlinux | if the about page will respond anyways. |
| 14:12.14 | elric | hrm no freebsd port for diax |
| 14:12.16 | elric | ah well |
| 14:12.32 | ariel_ | elric, it's windows |
| 14:12.41 | ReVoK | no frenchies? |
| 14:12.55 | elric | ah |
| 14:13.01 | elric | :| |
| 14:13.46 | elric | i compiled asterisk WITHOUT_H323 as well |
| 14:14.52 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-85-11.sentechsa.net) |
| 14:15.43 | slePP | http://pastebin.ca/news.php |
| 14:18.53 | ariel_ | slePP, nice hope you get some good sleep. |
| 14:19.02 | slePP | i'll definitely try :> |
| 14:19.33 | ariel_ | ok it's time to go and get ready for the day of work at customers sites. See you all later. |
| 14:20.45 | ReVoK | hum |
| 14:20.56 | ReVoK | compile done |
| 14:21.01 | ReVoK | chan_h323.so]Feb 16 15:20:15 WARNING[16549]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory |
| 14:21.02 | ReVoK | Feb 16 15:20:15 WARNING[16549]: loader.c:440 load_modules: Loading module chan_h323.so failed! |
| 14:21.03 | ReVoK | :x |
| 14:21.17 | *** join/#asterisk doughecka_ (~dheckaman@doughecka.user) |
| 14:21.34 | ReVoK | what's the best linux distrib for asterisk ? |
| 14:22.05 | Moc | ReVoK, read the doc... |
| 14:22.32 | ReVoK | arf :x |
| 14:22.38 | Moc | this libpt thing is becase you didnt define LD_... variable like it said in the README |
| 14:22.53 | riksta | i don't suppose anyone here that's in the UK, has an old 1U server they don't use i could buy off them cheap do they? |
| 14:23.29 | bjohnson | ReVoK: define best |
| 14:23.49 | ReVoK | compatibility, to make it work :) |
| 14:24.04 | bjohnson | ReVoK: atserisk@home likely for that |
| 14:24.13 | bjohnson | ReVoK: asterisk@home likely for that |
| 14:24.38 | hmmhesays | wow the new phpagi rocks |
| 14:25.35 | Moc | riksta, brandnew is about 539euro |
| 14:27.04 | riksta | Moc: i'm a student i can't afford that ;) |
| 14:27.06 | bjohnson | any tips on fighting echo with Sipura SPA 3000 units tied to Nortel CICS? |
| 14:27.34 | Moc | good luck then :( |
| 14:27.44 | mrempire | I have got my * working, I'm looking for a hardware phone, what's your advice? |
| 14:28.04 | bjohnson | riksta: maybe just nobody here right now .. try again later. Make sure they know you are doing code development (could make a difference) |
| 14:28.17 | riksta | bjohnson: yeah suppose so |
| 14:28.30 | riksta | bjohnson: how's things? |
| 14:28.31 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 14:28.48 | bjohnson | fine. Trying to figure out how to fix an echo problem |
| 14:28.58 | riksta | oh dear, where is your echo |
| 14:29.09 | riksta | oh...*reads up* |
| 14:29.24 | riksta | not familiar with that |
| 14:29.41 | bjohnson | I have 2 SPA 3000 connecting pstn (to fxo) and Nortel analog line in (fxs on the SPA) |
| 14:30.23 | ReVoK | moc t un vrai mito |
| 14:30.36 | bjohnson | I thought it was only a problem when the fxo and fxs on the same device were connected .. but as I try things I realize that I get echo even if fxo to fxs on the other device |
| 14:30.42 | *** join/#asterisk Banter (~glenrose@209.119.214.81) |
| 14:30.52 | Banter | Hello all |
| 14:30.52 | bjohnson | fxs -> * -> voip provider is good |
| 14:31.28 | *** join/#asterisk Luke-Jr (~luke-jr@207.192.219.246) |
| 14:31.42 | clive- | has anyone had any expereince with double nat and a sip service provider ? |
| 14:32.10 | bjohnson | found reference to similar problem on 'net but that group were playing with line impedence settings and were trying to get to work with UK settings .. so I couldn't really use it for reference |
| 14:32.35 | bjohnson | clive-: yes .. I couldn't get it to work |
| 14:32.52 | clive- | bjohnson, I was afriad someone would say that..:) |
| 14:32.59 | bjohnson | clive-: but not with a sip service provider since they don't normally have nat so it isn't "double" nat |
| 14:33.13 | bjohnson | and single nat is much different |
| 14:34.03 | clive- | well its like this ipphone---nat----nat----public_internet----sip_provider |
| 14:34.14 | bjohnson | clive-: sign up to fwd (sip) and follow their setup instructions .. worked for me when I follow their SPA 2000 STUN config example |
| 14:34.34 | clive- | this ipphone doesnt have stun..:( |
| 14:34.46 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 14:35.00 | bjohnson | clive-: that isn't what most people refer to as double nat .. but is exactly the system that I overcame by following the above procedure |
| 14:35.07 | tzafrir | does res_sqlite work with stable? |
| 14:35.24 | Cresl1n | ugh, sip+nat = yuck |
| 14:35.24 | Banter | can I replace my nortel switch with asterisk? |
| 14:35.26 | Cresl1n | lol |
| 14:35.29 | Cresl1n | Banter: sure |
| 14:35.37 | bjohnson | clive-: if a softphone .. change to an iax one |
| 14:35.37 | Cresl1n | why not? :-) |
| 14:36.00 | Banter | cool what about my existing phones |
| 14:36.12 | bjohnson | Banter: easy answer is replace them |
| 14:36.21 | Banter | hmmm lots of $$ |
| 14:36.22 | clive- | its not a hardophone, its a "planet sip phone" |
| 14:36.22 | bjohnson | Banter: there are more complicated options |
| 14:36.48 | Banter | I have lots of the 2616 display phones |
| 14:37.13 | bjohnson | Banter: send them to me .. I'll walk you through the * setup |
| 14:37.16 | clive- | I mean it is a hardphone,..,,not a soft phone |
| 14:37.28 | bjohnson | clive-: then you might be out of luck |
| 14:37.48 | clive- | cant one forward a lot of ports or somthing |
| 14:38.02 | bjohnson | clive-: try the tech support for the phone. You likely need a stun server or an outbound proxy. Did you try port forwarding? |
| 14:38.04 | pointer | who the guy was that was working on the "application API" (mentioned it on the dev conf last thursday)? |
| 14:38.27 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 14:39.16 | clive- | bjohnson, trying that |
| 14:39.43 | bjohnson | Banter: the more complicated answer is you can buy an adapter to convert those phones to sip phones. Or you can stick * out front of the nortel and have * feed lines to the nortel. or you can plug * into ATA ports into the nortel. or parts of all three. |
| 14:40.30 | bjohnson | Banter: you will likely encounter problems you have to overcome with all three of those options |
| 14:41.17 | Banter | Thanks bjohnson |
| 14:41.43 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 14:44.16 | *** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk) |
| 14:45.16 | jsandnes | Anyone here using the Micronet SP5100 SIP/h323 phone? |
| 14:46.50 | Darwin35 | wow coppice not see yo in awhile |
| 14:47.07 | *** join/#asterisk eKo1 (~bernd@207.42.191.66) |
| 14:47.23 | *** join/#asterisk nicolasg (~nicolasg@host-30.6.60.66-ta.adsl.netizen.com.ar) |
| 14:50.24 | bjohnson | Banter: btw .. Nortel's voip add-on basically sticks voip in front of the nortel. you end up with a hybrid |
| 14:51.51 | coppice | do you get bad echo from the hybrid? :-) |
| 14:52.13 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 14:52.26 | *** join/#asterisk bowman (~none@195.46.47.202.static.cablesurf.de) |
| 14:54.01 | bowman | Hi, I have problems with Zap channels. Asterisk 1.0.5, the box takes SIP connections and maps them to analog phones over the office PBX, which works fine for SIP calls from the internet. |
| 14:54.16 | *** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62) |
| 14:54.24 | bowman | As soon as an ENUM-mapped call comes in, Asterisk Dials Zap/g1/extension and instantly hangs up the channel again. why? |
| 14:55.40 | eKo1 | bowman: Stable? |
| 14:55.50 | eKo1 | or Head? |
| 14:55.58 | *** join/#asterisk _-Jon-_ (~jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 14:56.38 | bowman | eKo1: bristuff-0.2.0-RC5 package - it downloads Asterisk from digium's FTP - dunno what branch that is ;) |
| 14:57.22 | eKo1 | What does 'show version' say? |
| 14:57.35 | *** join/#asterisk multrix (~chatzilla@ALyon-252-1-62-80.w82-122.abo.wanadoo.fr) |
| 14:57.40 | bowman | Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 built by root@linux on a i686 running Linux |
| 14:57.42 | bowman | :) |
| 14:57.59 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 14:58.33 | bjohnson | coppice: I'm getting echo occasioanlly from mine that I'm trying to figure out |
| 14:58.42 | _-Jon-_ | Hey, has anyone else here been experiencing some major problems with VoicePulse? |
| 14:59.05 | `Sauron | Dum di dum. |
| 14:59.09 | eKo1 | bowman: Get the lastest Stable out of CVS. |
| 14:59.33 | bowman | eKo1: usually, that leads to huge problems with the Junghanns drivers - I tried that before |
| 14:59.56 | eKo1 | You're using a Junghanns BRI card? |
| 14:59.56 | `Sauron | Anyone here have * connected through lingo? |
| 15:00.02 | bowman | yep, quadBRI |
| 15:02.09 | eKo1 | bowman: Well, I can't really help much. I don't mess with BRI. |
| 15:02.42 | eKo1 | bowman: Is there a hangup() right after the dial(zap/...)? |
| 15:02.50 | bowman | eKo1: nope :) just a Dial().. |
| 15:03.04 | _-Jon-_ | Does anyone know of another service like VoicePulse Connect? I want a service that offers flat rate for incoming DIDs |
| 15:03.04 | shadebob | hi, can I do this <asterisk>-------T1-------<channel bank>-----ISDN phones? |
| 15:03.24 | `Sauron | broadvoice |
| 15:03.59 | `Sauron | I thought most voip providers offered flat rate for inbound calls |
| 15:04.15 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 15:04.53 | _-Jon-_ | Maybe I should be a little more specific.. I want to use IAX. I'm with BroadVoice but incomming calls don't always come through but I've never had that problem with AIX |
| 15:07.49 | riksta | any of you guys been using ADM? i'd appreciate some feedback...bugs etc |
| 15:08.27 | bjohnson | `Sauron: no |
| 15:08.45 | bjohnson | `Sauron: at least not for CDN DIDs |
| 15:09.11 | bjohnson | _-Jon-_: iax.cc has US ones (you pay more for monthly servcie) |
| 15:09.28 | bjohnson | like $10 instead of $2 |
| 15:10.40 | Grooby | welp..off to work |
| 15:10.41 | _-Jon-_ | Hmmm $10 vs $7.99 for something that works is worth it in my opinion |
| 15:10.43 | Grooby | talk to you guys later |
| 15:10.52 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 15:10.57 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 15:11.03 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivflc1.dialup.mindspring.com) |
| 15:12.05 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 15:12.06 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:12.06 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:12.09 | hmmhesays | heh interesting, you can originate a call with phpagi from a webpage |
| 15:12.15 | bjohnson | _-Jon-_: free incoming for $10 vs $2 for pay as you go depends how much you use it |
| 15:12.36 | bjohnson | hmmhesays: I think I saw an example on the tips and tricks page of the wiki |
| 15:14.16 | hmmhesays | I wasn't asking a question, was making a statement |
| 15:14.17 | bkw_ | _-Jon-_, its IAX not AIX |
| 15:14.38 | zoa | :) |
| 15:14.39 | zoa | heya brian |
| 15:14.43 | bkw_ | yo zoa |
| 15:17.39 | Essobi | Mmm. |
| 15:18.04 | Essobi | Ever thou it's set.. I can't get FOP to think I've entered a valid security number |
| 15:18.13 | *** join/#asterisk scrubb (~scrubb@OCI-19-41.OneCall.Net) |
| 15:19.09 | bjohnson | hmmhesays: I wasn't answering a question. I was volunteering assistance |
| 15:20.20 | benno2 | any idea how to set a password on the sipura ? |
| 15:20.54 | benno2 | (eg SPA 2000) it has only the "Password" field |
| 15:21.10 | *** part/#asterisk WildPikachu[BED] (~wildpikac@wildpikachu.user) |
| 15:21.15 | benno2 | but when I access the page it asks user and pass. what should I enter as username ? |
| 15:21.34 | PatrickDK | hmm, mine never asks for user/pass |
| 15:21.45 | PatrickDK | user is probably either user/admin |
| 15:21.54 | PatrickDK | since you can login under usermode or admin mode |
| 15:21.57 | benno2 | PatrickDK: yes because you did not set any. go to admin->System and set a password |
| 15:21.59 | Connor- | anyone had issues with SPA-2000's behind NAT? I've got it working, but, it takes a good 2-4 seconds for a call to start ringing when it's behind sip |
| 15:22.04 | benno2 | then go back and relogin |
| 15:22.21 | benno2 | it asks you login/pass (http authentication) but I don't know what enter as username |
| 15:22.35 | PatrickDK | try user or admin |
| 15:25.40 | bjohnson | Connor-: I didn't find any difference behind NAT than not once I got it working |
| 15:26.15 | bjohnson | Connor-: have you found a way to reduce the wait time after entering a number to dial (other than pressing #)? |
| 15:26.51 | *** join/#asterisk stickynomore (~jeff@nsc66.147.11-46.newsouth.net) |
| 15:28.00 | Essobi | It's been a while since I've used * heavily.. does -head still crash when you use the manager interface a lot? |
| 15:28.21 | ariel_ | Connor-, I have my spa 841 and 2100 behind my linksys router which it's nat and I dont have that issue. I press # key after everynumber I dial. |
| 15:29.23 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:30.29 | *** part/#asterisk TeLLuS (~johan@h187n2fls31o858.telia.com) |
| 15:30.41 | [Latre] | hi, i have a problem with iax ....i want configure one iAxy....the error is: No registration for peer , i follow the guide of digium!, can anyone helpme? |
| 15:31.16 | Connor- | ariel_ Other direction.. When CALLING the SPA |
| 15:31.52 | Connor- | I get something like this in asterisk.. Called spa2000/200 |
| 15:32.25 | Connor- | then 2-4 seconds later... I get the SIP/spa-1692 is making progress passing it to SIP/200 or whaterver |
| 15:32.32 | Connor- | at which point the SPA starts ringing |
| 15:32.47 | *** join/#asterisk klicTel (~Claude@207.107.208.137) |
| 15:32.52 | klicTel | morning all |
| 15:32.57 | *** join/#asterisk nazgool (~nazgoool@gatekeeper-e0.twc.de) |
| 15:33.00 | nazgool | hi |
| 15:33.03 | *** join/#asterisk brettnem (~brettnem@208.54.232.29) |
| 15:35.04 | nazgool | is it me or is there no AGI command to actually dial out (for the moment, my agi script just fills out a dialstring variable that it passes back, and the actual dialing is done back in extensions.conf, but i'd prefer to be able to call from the agi) ? |
| 15:36.07 | ariel_ | Connor-, just called my sipura 2100 from my sipura 841 and it worked right away. No delay. |
| 15:36.20 | Connor- | okay. |
| 15:36.38 | Connor- | You using STUN? |
| 15:36.51 | vaewyn | wow... the WIP5000 ROCKS! (so far... only played with it for 2 hrs now) |
| 15:37.27 | ariel_ | Connor-, no |
| 15:37.38 | *** join/#asterisk kiran (~kiran@203.212.254.27) |
| 15:37.40 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 15:37.59 | benno2 | vaewyn: cost in $ of the WIP5000 ? |
| 15:38.05 | Connor- | So your specifing your gateway IP in the SPA ? |
| 15:38.47 | vaewyn | benno2: 319.95$ from voipsupply.com |
| 15:39.01 | *** join/#asterisk km- (pgrace@67.105.178.133) |
| 15:39.07 | km- | Howdy! |
| 15:39.09 | benno2 | vaewyn: thanks. did you try to roam between access points on the same subnet. does it work or do you have big dropouts or call losses ? |
| 15:39.26 | *** join/#asterisk _PiGreco_ (~a@adsl-215-48.38-151.net24.it) |
| 15:39.32 | vaewyn | benno2: havn't had a chance to leave the building yet... but am going to try that soon |
| 15:39.42 | _PiGreco_ | re |
| 15:39.44 | vaewyn | (whole campus is wireless so... will get a good test) |
| 15:40.12 | benno2 | vaewyn: cool. what kind of wi-fi encryption does it support ? WPA-PSK and 802.1x too ? |
| 15:41.01 | CpuID | mmm wifi handsets :) |
| 15:41.36 | kiran | dsfr: hi can we chat on support for digium ? |
| 15:41.44 | _PiGreco_ | im getting mad, i set up an account for my iaxy device. it *seems* to register (actuallt i didnt understand where to get such info). i set up a simple context, default, for Answer and Playback a simple file for a test..just like the demo example, but when i pick up the phone and dial a number i get only busy tone..any hint? |
| 15:42.09 | bjohnson | Connor-: mine seems to work right away using STUN via FWD |
| 15:42.58 | km- | hey, do I need to fiddle with echo cancellation if I'm getting a TE405P? |
| 15:43.12 | bjohnson | km-: only if you get echo |
| 15:43.12 | km- | I remember how much fun it was to work that kink out with the x101 and tdm400p |
| 15:43.20 | benno2 | vaewyn: is the firmware of the hitachi VOIP handset stable or does it lock up sometimes ? |
| 15:43.24 | km- | I wasn't sure if it was possible to get echo on the t1 |
| 15:43.39 | _PiGreco_ | mh no hints uh.. :/ |
| 15:43.55 | vaewyn | benno2: 802.1x-md5 802.1x-tls WEP... etc.. |
| 15:44.05 | vaewyn | benno2: not sure... have only had it 2 hours :P |
| 15:44.13 | vaewyn | no lockups or glitches yet |
| 15:44.17 | nestAr | man, i'm getting some horrible echo problems with my SPA-3000 |
| 15:44.54 | benno2 | vaewyn: thanks. does it support WPA-PSK (pre shared key too) ? so you can get encrypted wi-fi in a SOHO enviroment without needing a RADIUS server |
| 15:45.14 | NormAst | km: Yes you need to play with echo can. on your TE405P |
| 15:45.28 | km- | what do you guys think, PII 266 enough juice for two t1 spans? |
| 15:45.43 | NormAst | Nope. |
| 15:45.44 | zoa | no |
| 15:45.47 | zoa | i wouldnt do that |
| 15:45.47 | km- | we've only got 16 channels on the t1 dedicated to voice |
| 15:45.48 | *** join/#asterisk jterrero (~jterrero@66.28.34.185) |
| 15:45.51 | km- | hmm |
| 15:45.57 | bjohnson | nestAr: I'm getting that too |
| 15:46.05 | vaewyn | benno2: havn't seen mention of it yet... but am still getting use to it |
| 15:46.58 | NormAst | km: The echo cancellation takes the power,.. |
| 15:47.05 | km- | gotcha |
| 15:47.18 | jterrero | Does anyone know where i can get decent documentation on how to implement MySQL as the cdr manager? i followed the instructions on voip-info.org but cant get it to work..... Downloaded the asterisk-addons, compiled and edited modules.conf to load cdr_addon_mysql.so |
| 15:47.20 | km- | I'm guessing it's got to be better than the last time I was fiddling with asterisk, about a year ago |
| 15:47.42 | km- | well, I'll just have to find some more cpu |
| 15:47.56 | nestAr | bjohnson: sad.. my X100P didn't have that problem.. but the caller id didn't work.. the caller id on the SPA works great... but now i got echo. |
| 15:47.56 | stickynomore | vaewyn: does that wip5000 show callerid correctly? |
| 15:48.26 | NormAst | km: You might be okay.. I can run 46 Channels of Echo can on a P3 600, but the box is only doin' voip. |
| 15:48.29 | nestAr | if i turn the gain down much more, people are going to complain that i'm too quiet.. |
| 15:48.36 | NormAst | km: no transcoding. |
| 15:48.39 | kiran | hi any one can help me on asterisk? |
| 15:48.50 | Koshatul | jterrero: did you put the load => into modules.conf ? |
| 15:49.02 | km- | NormAst: yeah, I wont know till I try, I guess. |
| 15:49.03 | vaewyn | stickynomore: so far yep... have been numeric only so far though... need someone on campus to call me for name |
| 15:49.08 | bjohnson | nestAr: my gain is just at factory default |
| 15:49.24 | km- | I'm going to be using asterisk as a hopping point, I'm essentially going to be pulling DID's at the border then shooting the rest of the T1 straight through to our existing NEC system |
| 15:49.35 | jterrero | Koshatul: yes load => cdr_addon_mysql.so |
| 15:49.53 | jterrero | Koshatul: am i suppose to put it in the globals area? i put it before the globals context |
| 15:50.00 | Koshatul | jterrero: in asterisk cli, can you run "cdr mysql status |
| 15:50.03 | Koshatul | " |
| 15:50.22 | km- | where's bkw when ya need him heh |
| 15:50.28 | Koshatul | jterrero: mine is in my [modules] section, not globals |
| 15:50.36 | NormAst | km: So you are using bridge mode... You don't need any echo cancellation. |
| 15:50.50 | jterrero | Koshatul: yeah thats where i put it. let me check on the cli |
| 15:51.02 | km- | NormAst: for the most part. New users will be coming off the system at the border and using sip phones for handsets |
| 15:51.11 | nestAr | bjohnson: i turned mine down a little as suggest in the FAQ on voxilla |
| 15:51.12 | km- | NormAst: so, the amount of juice needed will be mild |
| 15:51.15 | nestAr | to no avail. |
| 15:51.16 | jterrero | Koshatul: not currently connected to a mysql server |
| 15:51.25 | nestAr | all the echo can. on the SPA is turned on. |
| 15:51.27 | *** join/#asterisk robf (~robf@208.188.247.3) |
| 15:51.28 | Koshatul | jterrero: k, also try using the details in /etc/asterisk/cdr_mysql.conf work with a mysql client |
| 15:51.33 | NormAst | km: You can try it... I would personal go with atleast a P3. |
| 15:51.43 | NormAst | km: I sell pri cards if you are intersted. |
| 15:51.55 | km- | already ordered from digium, thanks for the offer though |
| 15:52.07 | robf | Quick question for someone... Didn't the Dial application at one time have an option to require answer confirmation by requiring the callee to press pound? |
| 15:52.18 | kiran | NormAst: what brand cards you sell |
| 15:52.18 | km- | Mark and I are old friends, so, I like to give him money directly :) |
| 15:52.37 | NormAst | km: if you start getting HDLC errors then you have either IRQ problems or overload CPU. |
| 15:53.10 | km- | I don't need HDLC -- the data side is already terminated at router on a seperate t1 |
| 15:53.14 | kiran | NormAst: can you specify more about cards |
| 15:53.22 | bjohnson | nestAr: mine's 0 on the pstn tab. What's the voxilla url to that tip? |
| 15:53.44 | NormAst | Kiran: What do you want to know? |
| 15:53.58 | Koshatul | jterrero: i just pm'd you my cdr_mysql.conf with user/pass removed |
| 15:54.09 | kiran | NormAst: do you sell digium or other brand pri cards |
| 15:54.20 | nestAr | bjohnson: http://voxilla.com/FAQ-index-myfaq-yes-id_cat-5.html#q42 |
| 15:54.34 | NormAst | both.. Depends on what the client needs.. |
| 15:54.40 | kiran | Normast : we are looking at e1 quad span cards |
| 15:55.07 | kiran | Normast : what are the other brand cards do you sell? |
| 15:55.12 | NormAst | kiran: I personal really like the sangoma cards... Done alot of testing.. |
| 15:55.39 | NormAst | Plus they support 5.0volts and 3.3.. |
| 15:55.51 | kiran | NormAst: can we chat in a private window? |
| 15:56.00 | NormAst | suure |
| 15:56.50 | ctooley | Is there any way to restrict VoicemailMain to a specific context without restricting it to a mailbox? |
| 15:56.52 | nestAr | :cyber: |
| 15:56.54 | *** join/#asterisk search_learn2005 (~Miranda@209.68.139.150) |
| 15:57.28 | robf | Quick question for someone... Didn't the Dial application at one time have an option to require answer confirmation by requiring the callee to press pound? |
| 15:58.25 | search_learn2005 | do I need an fxs module if I will only use public phone to the server, and server to the VOIP phones. Or do I only need fxo modules? |
| 15:58.47 | klicTel | Guys what happened to all the videos shot at Astricon? are they available somewhere? |
| 15:58.57 | ariel_ | search_learn2005, fxo |
| 15:59.31 | NormAst | kiran: Check for a private window. |
| 16:00.02 | search_learn2005 | how many fxo modules would I need to feed ~50 Voip phones, and what would be the server's requirements? |
| 16:00.36 | wasim | search_learn2005: 1 |
| 16:00.42 | Koshatul | search_learn2005: one billlion ! |
| 16:00.51 | Koshatul | search_learn2005: ok, just one |
| 16:01.00 | _PiGreco_ | does iaxy device really support adpcm? |
| 16:01.25 | search_learn2005 | but what if I have seven public phone lines coming into the building? |
| 16:02.13 | Koshatul | search_learn2005: pstn ? |
| 16:02.17 | ariel_ | search_learn2005, you will then need 7 FXO ports. |
| 16:03.02 | search_learn2005 | what kind of a server would I need? How much memory? For ~50 Voip phones with 7 fxos? |
| 16:03.04 | [Latre] | i have problems with iaxy, can anyone helpme......the iaxy can not log into server |
| 16:03.06 | ariel_ | What type of service is your Telco providing do you need more lines? |
| 16:03.47 | ariel_ | [Latre], first thing is to see if you have it on a switch that only allows 100mb the iaxy is 10 only. |
| 16:03.55 | bjohnson | nestAr: good info there |
| 16:04.10 | bjohnson | someone was looking for SPA REN info: The SPAs only have a REN of 3 |
| 16:04.10 | tclark | ariel_: do you have any adsi handsets these days ? |
| 16:04.38 | ariel_ | tclark, not any more. I am not working at the other place any more. |
| 16:04.39 | bjohnson | nestAr: is the gain only set on the pstn tab? |
| 16:04.54 | nestAr | as far as i can tell, yes. |
| 16:05.04 | brettnem | good morning all |
| 16:05.08 | nestAr | i'm sure there's a gain for the FXS port |
| 16:05.10 | nestAr | but i'm not using it.. |
| 16:05.12 | bjohnson | are you down to -6? |
| 16:05.24 | bjohnson | I didn't see a gain setting on the fxs |
| 16:05.29 | *** join/#asterisk bde (~bde@nanoworks-02.inc.rpi.edu) |
| 16:05.29 | nestAr | not yet.. i haven't had time to play with it.. |
| 16:05.37 | nestAr | i'm at like -2 |
| 16:05.37 | search_learn2005 | ariel_: what kind of a server would be good for 7 fxos and 50 voip phones? |
| 16:05.39 | ariel_ | tclark, long time no hear from you. How is the weather up in the NorthWest these days. |
| 16:05.47 | nestAr | but i can tell it's quiet already |
| 16:05.57 | bde | hi, can anyone tell me what kind of rates i can get for 1M minutes/month to the US/Canada? |
| 16:06.19 | bjohnson | nestAr: SPA To PSTN Gain or PSTN To SPA Gain? |
| 16:06.43 | nestAr | SPA to PSTN. I seem to be the only one that hears the echo. |
| 16:06.52 | ariel_ | search_learn2005, it depends on if your on the same network and your b/w. But any server running a p4 with at least 512mg and a good hdd will do fine with ulaw only. |
| 16:06.52 | nestAr | i hear myself |
| 16:07.06 | *** part/#asterisk pranav (sameer@202.149.48.200) |
| 16:07.13 | nestAr | but i think i set both of them to -2 |
| 16:07.41 | Zeeek | is anyone here familiar with "normal" pbxes? I have a simple question |
| 16:07.44 | ariel_ | search_learn2005, but it really depends on what you want to do as for internet access to the system and if you need to use meetme rooms and transcoding. |
| 16:07.57 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 16:08.00 | ariel_ | Zeeek, what is a normal pbx |
| 16:08.24 | Zeeek | when a call comes in to a pbx, if several phones ring and one is busy, then that person hangs up, will that phone begin to ring? |
| 16:08.32 | bjohnson | nestAr: I wonder if it should be the PSTN to SPA gain that gets lowered |
| 16:08.48 | Zeeek | IOW, is busy detection done before each ring? |
| 16:08.56 | Zeeek | (I doubt it but...) |
| 16:09.00 | nestAr | bjohnson: potentially. |
| 16:09.02 | search_learn2005 | ariel_: thank you. No, none of those. Only, automatic call forwarding, voicemail, and very simple services that we don't get with regular analog lines right now |
| 16:09.10 | bjohnson | Zeeek: I don't think so on our Nortel PBX |
| 16:09.15 | bjohnson | err Nortel CICS |
| 16:09.27 | *** join/#asterisk Brumle (~brumle@brumle.com) |
| 16:09.35 | Zeeek | so if someone hangs up when they hear another phone ring, their phone doesn't start ringing? |
| 16:09.37 | bjohnson | but you're only talking about a one or two second window |
| 16:09.50 | brettnem | Zeeek: I don't think soo. |
| 16:10.09 | brettnem | hmm. |
| 16:10.12 | brettnem | lets think.. |
| 16:10.14 | bjohnson | no I don't think so .. but even trying time the hangup to test that will be tricky |
| 16:10.15 | greg_work | WHY is there no press-0.gsm ?! |
| 16:10.19 | Zeeek | since this is the way asterisk behaves, I assumed so. In the case of a home with several phones, the opposite would be true |
| 16:10.28 | brettnem | I believe the Nortel SL1 protocol is based on BRI |
| 16:10.49 | Zeeek | but then come to think of it, home with one line, busy means BUSY! :) |
| 16:10.51 | bjohnson | Zeeek: I think depends on how it's configured |
| 16:10.56 | brettnem | which would mean that it missed the ALERTING from the switch |
| 16:11.12 | nestAr | i'm also wondering about the port impedance |
| 16:11.18 | Zeeek | well it all stems from the fact that my partenr haits call waiting |
| 16:11.24 | nestAr | voxilla's page says US uses 600 or 900 |
| 16:11.26 | Zeeek | but maybe she'll have to get used to it! |
| 16:11.26 | bjohnson | I haven't tested but .. I think if each phone is a separate handset, * will detect busy before first ring and not try that [hone anymore |
| 16:11.32 | brettnem | Zeeek: call pickup may be a good solution |
| 16:11.48 | brettnem | There is also a "retrydial" in the bugtracker |
| 16:11.48 | bjohnson | nestAr: I read some stuff about that too .. but what I found was all UK |
| 16:11.50 | Zeeek | brettnem yes, true |
| 16:12.01 | Zeeek | if she can remember what the call pickup code is |
| 16:12.09 | brettnem | make it a speed dial |
| 16:12.09 | search_learn2005 | What would be a cheap Voip phone which will be used for very simple purposes. Only listen to your voicemail, make a call. No conference calls or netmeetings and such. At least for now. |
| 16:12.10 | Zeeek | is it still hard coded as *8? |
| 16:12.27 | bjohnson | nestAr: I guess we just test options and coordinate our activities |
| 16:12.31 | nestAr | lol |
| 16:12.36 | brettnem | it's configurable in features.conf |
| 16:12.46 | nestAr | finding time to test this stuff has been a pain |
| 16:12.56 | km- | awesome, the call pickup stuff is finished? |
| 16:12.56 | nestAr | if i had a pots line here at the office, it'd be easier |
| 16:12.59 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 16:13.09 | Zeeek | from what version STABLE is it configurable? |
| 16:13.15 | brettnem | I don't know if it's finished, but it is in HEAD I believe. |
| 16:13.17 | brettnem | not sure |
| 16:13.19 | km- | oh |
| 16:13.20 | km- | it's in head |
| 16:13.47 | Zeeek | so for me it's still *8 which is fine |
| 16:13.59 | bjohnson | search_learn2005: one of the budgettones? maybe a fxs with an analogue phone? |
| 16:14.01 | Zeeek | what is *0 by the way? |
| 16:14.18 | brettnem | retrydial: http://bugs.digium.com/bug_view_page.php?bug_id=0003313 |
| 16:14.23 | brettnem | Zeeek: maybe flash? |
| 16:14.57 | Zeeek | if you are talking on zaptel, doesn't the flash key work? |
| 16:14.57 | bjohnson | nestAr: I have 2 POTS lines with 2 SPA 3ks. Problem is tracking which each one is doing separately |
| 16:15.13 | nestAr | lol |
| 16:15.15 | nestAr | i bet. |
| 16:15.17 | brettnem | yep |
| 16:15.43 | search_learn2005 | bjohnson: Thank you. I will check budgettones. We are thrying to use the existing 10/100 network at our school. I think VOIP phone would be better. Any drawbacks of these budgettones? |
| 16:16.04 | brettnem | the budgettone phones suck.. |
| 16:16.06 | eKo1 | Has anybody put a fax machine on a SPA-1001? |
| 16:16.08 | brettnem | really |
| 16:16.16 | brettnem | eKo1: I've put one on a 3000 |
| 16:16.30 | nestAr | but i won't be buying anymore. |
| 16:16.30 | eKo1 | On the FXS? |
| 16:16.34 | brettnem | yeah |
| 16:16.39 | eKo1 | And it works? |
| 16:16.46 | search_learn2005 | brettnem: Any ideas for a cheap VOIP phones? |
| 16:16.50 | brettnem | personally I think they suck. I won't buy another. my '#' key stopped working |
| 16:16.55 | nestAr | lol |
| 16:17.04 | brettnem | search_learn2005: you'll get more bang for your buck with sipura stuff |
| 16:17.16 | brettnem | and analog phones |
| 16:17.21 | eKo1 | Any special config., like turning off silence suppression? |
| 16:17.23 | brettnem | they are really quite nice |
| 16:17.24 | nestAr | a good ATA and a nice cordless phone = win. |
| 16:17.25 | [Latre] | ariel_: iaxy only allow 10Mb? |
| 16:17.38 | eKo1 | nestAr: That's what I have on my desk. |
| 16:17.38 | brettnem | eKo1: no, but it's dedicated LAN/WAN with QOS to PSTN gateway |
| 16:17.42 | ariel_ | search_learn2005, for cheap good phones I am now using the Sipura 841 |
| 16:17.55 | ariel_ | [Latre], yepper |
| 16:17.55 | brettnem | nestAr: I think that's te way to go |
| 16:18.18 | search_learn2005 | thanks everyone. |
| 16:18.23 | brettnem | eKo1: FoVOIPoInternet really has no guarentees |
| 16:18.30 | brettnem | search_learn2005: Look on the wiki for phone recommendations |
| 16:18.50 | brettnem | I use the Polycom IP500's.. they are around $140 or so and are really nice business class phones |
| 16:18.53 | ariel_ | search_learn2005, but in your case if you have that many phones it might be better to up up some adtrans and use analog phones. |
| 16:18.59 | brettnem | I hear the IP300s are nice too and are like 125 or so |
| 16:19.00 | search_learn2005 | is it voip wiki or asterisk wiki |
| 16:19.08 | [Latre] | ariel_: if i have connected to 100M doesnt work? but, in CLI * said No registration for peer 'user3' |
| 16:19.17 | brettnem | ariel_: great idea.. use a channel bank |
| 16:19.18 | km- | man, sip.conf has a crapload more goodies |
| 16:19.24 | bjohnson | Connor-: just remembered I had a connection delay issue that I think was caused by using jitterbuffer in my * conf file .. may be something for you to check |
| 16:19.32 | brettnem | search_learn2005: I'd look into ariel_ 's suggestion of using a channel bank.. |
| 16:19.44 | brettnem | it'll be more compatible with your existing infrastructure too I bet |
| 16:19.52 | brettnem | ~wiki |
| 16:20.11 | search_learn2005 | isn;t it better to move towards VOIP phone with this system? |
| 16:20.15 | *** join/#asterisk bill522 (~Bill@182-30.201-68.swfla.rr.com) |
| 16:20.15 | brettnem | hmm that didn't work as expected |
| 16:20.23 | brettnem | search_learn2005: why? |
| 16:20.26 | *** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl) |
| 16:20.38 | brettnem | I mean, they are nice.. but for a school it may be too expensive. |
| 16:20.49 | search_learn2005 | I checked the channel bank prices thay are quite expensive for 50 lines |
| 16:20.55 | brettnem | you probably won't notice a difference.. unless you are doing conference calling and transfering a whole lot |
| 16:21.13 | brettnem | search_learn2005: lok for the ADIT from CAC on ebay |
| 16:21.23 | ariel_ | search_learn2005, you can get great adtrans 750 on ebay for 400 to 500 each |
| 16:21.28 | brettnem | that plus a bunch of analog phones will be MUCH cheaper I think |
| 16:21.35 | brettnem | yeah that adtran is real nice too |
| 16:22.00 | ariel_ | put 3 adtrans one digium te410p card and your off using normal phones. and less echo and problems. |
| 16:22.17 | brettnem | search_learn2005: goto http://www.voip-info.org start there.. it's where ALL of the info is |
| 16:22.25 | ariel_ | Then your server can do a better job for voip service and meetme's. |
| 16:22.51 | brettnem | right.. you'll still be able to support voip.. like if you wanted to link campuses together or have remote extensions |
| 16:23.08 | brettnem | voicemail to email.. still get quite a bit of power |
| 16:23.25 | brettnem | I'd say you're not going to lose anything really |
| 16:23.32 | search_learn2005 | I thought there won't be any echo if I use VOIP phones |
| 16:23.47 | ariel_ | who gave you that Idea |
| 16:23.52 | brettnem | sometimes going PSTN to VOIP gives you echo if you don't have hardware echo cancelation |
| 16:24.01 | brettnem | heh |
| 16:24.23 | PatrickDK | heh, pstn always has echo, voip just makes it noticable |
| 16:24.25 | ariel_ | echo comes into play more over voip service then analog to pstn |
| 16:24.44 | *** part/#asterisk christo (~chris@office.enovi.com) |
| 16:24.45 | ariel_ | PatrickDK, your right |
| 16:24.52 | search_learn2005 | thanks |
| 16:25.01 | brettnem | ariel_: why would you say that.. voip is '4 wire' |
| 16:25.34 | Guy- | I thought there were these 'hybrid circuits' that cancelled echo? |
| 16:25.45 | Guy- | so that PSTN shouldn't have any, normally |
| 16:25.47 | PatrickDK | hybrid circuits make the echo |
| 16:25.50 | ariel_ | echo is due to many things. |
| 16:26.09 | PatrickDK | anytime you have analog you will have echo |
| 16:26.22 | brettnem | Guy-: what you are thinking about is a device called a hybrid that converts a 2 wire ciruit to a 4 wire circuit |
| 16:26.23 | ariel_ | echo is normal but it's more of a problem when you add the internet traffice into it. |
| 16:26.29 | PatrickDK | do to the signal crossing wires, not having correct impedance wiring |
| 16:26.31 | brettnem | it's an impedence matching problem |
| 16:26.40 | Guy- | brettnem: yes, that's what I seem to recall |
| 16:26.42 | search_learn2005 | doesn't the zaptel conf files cancel echo? |
| 16:26.52 | brettnem | it's software based.. can only be so good |
| 16:27.07 | brettnem | QOS |
| 16:27.22 | brettnem | anyone use the kentrox routers? |
| 16:27.26 | PatrickDK | I have found my sipura devices cancel echo really good, so do cisco |
| 16:27.37 | Guy- | what I don't quite understand is how I can have echo over pure VoIP links, where everything is supposed to be digital |
| 16:28.00 | coppice | Guy- easy :-) |
| 16:28.03 | PatrickDK | guy, then the echo is probably at the analon part, in the handset |
| 16:28.10 | ariel_ | Guy-, sound travels and sometimes the echo is really the stream not synced. |
| 16:28.10 | Guy- | that's what I thought |
| 16:28.18 | PatrickDK | I have that problem with my snom200 phones |
| 16:28.20 | coppice | The echo is between the earpiece and the mic |
| 16:28.29 | Guy- | I mean, I thought the handset on the other end is producing the echo |
| 16:28.33 | search_learn2005 | PatrickDK: what if I use VOIP handsets? |
| 16:28.39 | brettnem | heh.. coppice.. I had a feeling you'd pop into this one. :) |
| 16:28.43 | ariel_ | echo is always around. |
| 16:28.53 | PatrickDK | how the hell do you use a voip handset? just lession to the digital stream on a lcd? |
| 16:29.00 | coppice | that's why every cellphone and IP phone has an echo canceller built in |
| 16:29.00 | ariel_ | coppice, how is the far east. |
| 16:29.14 | coppice | dark at this moment |
| 16:29.54 | ariel_ | coppice, do you know anyone working on ss7 for digium or an asterisk system? |
| 16:29.59 | *** part/#asterisk search_learn2005 (~Miranda@209.68.139.150) |
| 16:30.08 | brettnem | I'm still waiting for A-Link capability with the SS7 solution |
| 16:30.10 | coppice | we have SS7 for *. Its commercial, not free |
| 16:30.21 | *** join/#asterisk Corydon76-home (two@pcp08665860pcs.500ash01.tn.comcast.net) |
| 16:30.26 | *** join/#asterisk search_learn2005 (~Miranda@209.68.139.150) |
| 16:30.32 | Zeeek | ~seen wasim |
| 16:30.39 | jbot | wasim is currently on #asterisk (9h 53m 28s). Has said a total of 9 messages. Is idling for 30m 3s |
| 16:30.39 | ariel_ | coppice, great I have a customer that is looking for it. Can you emal me some info on it? |
| 16:30.46 | search_learn2005 | Thank You Everyone |
| 16:30.50 | *** part/#asterisk search_learn2005 (~Miranda@209.68.139.150) |
| 16:30.56 | Zeeek | allright wasim come on out of there |
| 16:31.02 | coppice | brettnem: A-link is the SS7 on V.56 crap you get in the US, isn't it? |
| 16:31.03 | Zeeek | open the curtain |
| 16:31.12 | brettnem | coppice: I've been in touch with Hidar regarding the ss7 stuff.. Looks like it's coming along |
| 16:31.19 | brettnem | well X.25 |
| 16:31.40 | brettnem | coppice.. No one does f-links in the US.. and that's about all you guys are doing for now.. I think |
| 16:31.41 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 16:31.59 | coppice | yeah. we will get to that soon. working over a 64K slot works fine for the rest of the planet. |
| 16:32.07 | brettnem | coppice: it's NFAS.. right? so my signalling is way over here.. but my IMT is way over there (on a different platform) |
| 16:32.39 | brettnem | coppice: sure, I can understand that.. but us backward US folk just don't do it that way.. I can't get it like that if I begged. :) |
| 16:32.42 | *** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl) |
| 16:33.18 | brettnem | coppice: once you guys can support A-Link and TCAP stuff I'll be all over it. I'm dying to get rid of my softswtich |
| 16:33.22 | coppice | of course, your a-link comes in a 64 slot, but I understand the telco always breaks it out to a brain dead V.56 before you see it |
| 16:33.24 | __Sparks_ | Can anyone here help me configure Sipgate with Asterisk!? |
| 16:33.46 | MuppetMaster | Sure, I have a config working with a UK and DE number with Sipgate. |
| 16:33.55 | MuppetMaster | What do you need? |
| 16:34.01 | brettnem | coppice: well that 64k slot is either a DDS circuit or a DS0 of a T1.. and it comes from a different provider than where the IMT (ISUP trunk) is going |
| 16:34.15 | brettnem | coppice: and that's a pretty standard configuration for I'd say 99%+ of the US |
| 16:34.27 | coppice | of course SS7 is NFAS. Its always NFAS, even if the 64K slot it arrives on is in the same E1/T1 as some of the voice circuits it serves |
| 16:34.28 | *** join/#asterisk Wireless (~bad@220.233.77.87) |
| 16:34.40 | brettnem | I interconnect my ISUP trunks to SBC and get my SS7 ALink from Verisign |
| 16:34.55 | brettnem | coppice: heh, what do you call a 'facility' |
| 16:34.58 | Wireless | depmod: *** Unresolved symbols in /lib/modules/2.4.18-bf2.4/misc/ztdynamic.o .... etc when compiling zaptel - anyone? |
| 16:35.16 | brettnem | coppice: yeah, but in A-Links, the 64K slot is NOT EVER on the same T1 |
| 16:35.27 | bkw_ | I need someone to fax stuff to me for testing please |
| 16:35.36 | ariel_ | bkw_, number? |
| 16:35.39 | bkw_ | who is up for faxing ALOT of stuffs to me |
| 16:35.41 | bkw_ | 8882174459 |
| 16:35.43 | brettnem | coppice: and frequently goes to different providers |
| 16:35.47 | bkw_ | like 20 pages a whack please |
| 16:35.52 | sivana | bkw_: like garbage? |
| 16:35.57 | bkw_ | ya dont care what it is |
| 16:36.00 | ariel_ | 20 pages ok let me see. |
| 16:36.01 | brettnem | bkw_: arn't there fax porn autoresponders out there? ;) |
| 16:36.07 | bkw_ | brettnem, find me one |
| 16:36.10 | brettnem | haha |
| 16:36.41 | bkw_ | haha |
| 16:36.45 | bkw_ | that has your number on it doesn't it |
| 16:36.50 | sivana | hehe |
| 16:36.52 | bkw_ | I need more apple juarez |
| 16:37.03 | zoa | bkw, trying the rxfax ? |
| 16:37.15 | bkw_ | already tried it |
| 16:37.23 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
| 16:37.23 | zoa | good ? |
| 16:37.29 | zoa | do you get it to run stabkle ? |
| 16:37.34 | *** join/#asterisk ZX81 (matt@222-153-114-115.jetstream.xtra.co.nz) |
| 16:37.34 | bkw_ | it works great |
| 16:37.34 | bkw_ | so far |
| 16:37.36 | zoa | coppice |
| 16:37.40 | zoa | all faxes ? |
| 16:37.43 | MuppetMaster | Hello everyone |
| 16:37.54 | brettnem | coppice: I'd be interested in seeing how you guys are planning to supporting this a-link capability.. If you are.. that is.. |
| 16:37.59 | MuppetMaster | I am having a problem using the Monitor command with a local variable substitution. |
| 16:38.15 | coppice | brettnam: where's the problem? |
| 16:38.22 | brettnem | well.. |
| 16:38.36 | ZX81 | bkw_: did you end up having another copy of the recording? |
| 16:38.42 | bkw_ | ZX81, no |
| 16:38.43 | MuppetMaster | When I use Monitor(wav|my_recording|m) w/o variable substition it works fine |
| 16:38.45 | bkw_ | it was a bug :P |
| 16:38.49 | ZX81 | ah |
| 16:38.51 | brettnem | coppice: I need the capability of having the a-links (2 of them) in different cities.. obviously on differnt systems.. |
| 16:38.53 | ZX81 | all good |
| 16:38.56 | ZX81 | :) |
| 16:38.58 | ZX81 | heh |
| 16:38.59 | Juggie | when you use a .call file or sockets to have asterisk generate a call, do you have any control within asterisk to manipulate the number dialed... to eg select an interface? decide on local or internal call? |
| 16:39.30 | coppice | brettnem: no big problem |
| 16:39.34 | brettnem | coppice: then on further differnt systems, there will be ISUP trunks to providers with nothing but trunks on them, no signalling at all.. so that will require some kind of SS7 over IP.. and redundancy with the Alinks |
| 16:39.57 | brettnem | coppice: so we'll have to map TCICs and OPC DPC pairs to not only a platform, but T1 and DS0 |
| 16:39.59 | ZX81 | bkw_: how do you do scaling? |
| 16:40.03 | ZX81 | ser+asterisk? |
| 16:40.09 | MuppetMaster | But, when I use Montior(wav|${REC_FILENAME}|m) it does not like it, as I get a CLI message saying 'unknown format: wav|what was in the var. |
| 16:40.11 | brettnem | ZX81: loaded question eh? |
| 16:40.14 | ZX81 | :) |
| 16:40.21 | ZX81 | hehe |
| 16:40.26 | coppice | SS7 over IP is the most flexible way, but there are others we can use until we have the SIGTRANS stuff complete |
| 16:40.27 | MuppetMaster | So it appears to string the format (in this case wav) with the contents of the variable and then uses that as the recording type. |
| 16:40.31 | MuppetMaster | A bug? |
| 16:40.34 | zoa | coppice is ignoring me :p |
| 16:40.36 | zoa | evil coppice |
| 16:40.38 | bkw_ | ZX81, just keep throwing asterisk boxes at it.... |
| 16:40.40 | bkw_ | :P |
| 16:40.42 | ZX81 | :) |
| 16:40.53 | brettnem | coppice: so you guys will have an intern way until you can support sigtran? |
| 16:41.09 | brettnem | coppice: and will that allow a system without an A-Link to perform a TCAP query? |
| 16:41.23 | brettnem | coppice: like LNP, 1800,CNAM, LIDB, etc,etc,etc |
| 16:41.47 | brettnem | bkw_: does that work really? honestly.. |
| 16:41.52 | ZX81 | 48 Hours till I leave for Italy |
| 16:41.52 | Juggie | bkw, not a sip call or anything, that has a context... i do something like Action: Originate, Channel: Zap/g1/number |
| 16:41.54 | brettnem | bkw_: with redundancy |
| 16:41.54 | ZX81 | yaya |
| 16:42.23 | ZX81 | argh I lost my pipe |
| 16:42.27 | ZX81 | :( |
| 16:42.42 | ZX81 | wtf it was here a second ago... |
| 16:42.47 | brettnem | ew pipe |
| 16:42.54 | ZX81 | :) |
| 16:43.07 | ZX81 | ahaha it was on my lap |
| 16:43.08 | Juggie | should i just generate an event and do the dial in the context instead? |
| 16:43.53 | brettnem | hmm I think i burned him out on the SS7 stuff. ;) |
| 16:44.07 | ZX81 | is there a question involved? |
| 16:44.14 | ZX81 | :) |
| 16:44.28 | ZX81 | no, he's just running from the press |
| 16:44.29 | ZX81 | lol |
| 16:44.30 | ZX81 | :) |
| 16:44.32 | Juggie | ahhh.... |
| 16:44.37 | bkw_ | brettnem, just give us some time.. we have an idea on the board to solve this whole cluster issue in asterisk |
| 16:44.41 | Juggie | Extension |
| 16:44.45 | _PiGreco_ | im getting mad, i set up an account for my iaxy device. it *seems* to register (actuallt i didnt understand where to get such info). i set up a simple context, default, for Answer and Playback a simple file for a test..just like the demo example, but when i pick up the phone and dial a number i get only busy tone..any hint? |
| 16:44.50 | *** join/#asterisk brazil (~cleber@200.198.105.37) |
| 16:44.50 | ZX81 | Oooh |
| 16:44.54 | ZX81 | when is next conf? |
| 16:44.55 | brettnem | bkw: care to include me? :) |
| 16:44.58 | bkw_ | but it looks kick ass |
| 16:45.12 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 16:45.21 | brazil | good afternoon for all |
| 16:45.22 | ZX81 | dev conf == thursdays? |
| 16:45.39 | Tall-guy | Zx81: timex sinclair? |
| 16:46.08 | brettnem | the app_redirect stuff looked kinda promising.. |
| 16:46.13 | ZX81 | :) |
| 16:46.16 | brazil | I browse informations about QoS (HTB and SFQ) to user with asterisk.. Anybody can help me? |
| 16:46.27 | ZX81 | GOOOOOOOOO Sinclair ZX81!!!!!! |
| 16:46.28 | ZX81 | :) |
| 16:46.42 | Tall-guy | Zx81: man that takes me back |
| 16:46.49 | ZX81 | :) |
| 16:46.58 | bjohnson | here's a question that is probably too specific to my system for anyone to answer but .. I have SPA 3000s answering calls and hooked to my Nortel line in's. Ofetn when a call is hung up I get what I call a ghost ring on my Nortel. It seems the fxo port is still connected, senses the callerid, and re-rings into * as though it's a new call. Any ideas on this one? |
| 16:47.15 | Tall-guy | zx81: that big brick on the back full of ram...the cheezy little touch keys :) |
| 16:47.20 | ZX81 | :) |
| 16:47.26 | ZX81 | the keys rock!!!! |
| 16:47.34 | ZX81 | :) |
| 16:47.38 | Tall-guy | zx81: 16k? |
| 16:48.00 | ZX81 | I have an expansion pack to 64Tb |
| 16:48.01 | ZX81 | :) |
| 16:48.08 | Tall-guy | haha |
| 16:48.23 | ZX81 | and mesh computing using 512 ZX81's to run Asterisk |
| 16:48.23 | ZX81 | :) |
| 16:48.25 | ZX81 | hehe |
| 16:48.37 | ZX81 | jk |
| 16:48.41 | Tall-guy | zx81: case of beer for you if you can pull it off :) |
| 16:48.49 | ZX81 | lol |
| 16:49.03 | ZX81 | just imagine it |
| 16:49.06 | ZX81 | hehe |
| 16:49.12 | brazil | I have a simple script to classfied UDP packets but is not function... Where i can find more informations about HTB + SIP? |
| 16:49.19 | ZX81 | once you can do that, you can run asterisk on your calculator |
| 16:49.20 | Tall-guy | Anyone using IAXY's over internet.....I got latency issues.... |
| 16:49.31 | ZX81 | nah |
| 16:49.33 | ZX81 | not i |
| 16:49.47 | *** join/#asterisk jayden (~ircatjerr@65.170.43.34) |
| 16:50.03 | jayden | hello.... |
| 16:50.13 | Tall-guy | CoaxD: u alive? |
| 16:50.27 | ZX81 | :) |
| 16:50.41 | jayden | quick question, when using externip and localnet in sip.conf, what is the format for if you have multiple local networks? |
| 16:50.51 | tzafrir | anybody here uses destar? |
| 16:50.57 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 16:51.26 | ZX81 | nop |
| 16:51.29 | brazil | you can put 0.0.0.0 in the script sip.conf |
| 16:51.31 | ZX81 | what does it do again? |
| 16:51.52 | Zeeek | NoOp |
| 16:52.03 | [Latre] | someone configure a TDM04B with another PBX ? |
| 16:52.06 | ZX81 | :) |
| 16:52.07 | ZX81 | no |
| 16:52.12 | ZX81 | what does destar do |
| 16:52.13 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 16:52.15 | ZX81 | is it like AMP? |
| 16:52.24 | Zeeek | removes any remaining stars |
| 16:52.30 | ZX81 | hehehe |
| 16:52.38 | jayden | brazil: was that to me? |
| 16:52.57 | Zeeek | wasim |
| 16:53.01 | tzafrir | I'm trying to use it now. The approach seems healthier than AMP |
| 16:53.11 | tzafrir | even though I don't like python |
| 16:53.22 | tzafrir | (but I don't like php either) |
| 16:53.33 | brazil | yes |
| 16:53.42 | brazil | let get an example |
| 16:53.45 | brazil | just moment |
| 16:53.46 | Zeeek | php rocks |
| 16:53.53 | Tall-guy | poprocks! |
| 16:53.57 | Tall-guy | (and coke) |
| 16:54.00 | Zeeek | pop whose rocks? |
| 16:54.01 | ZX81 | guess what |
| 16:54.04 | ZX81 | SineAsterPanel is nearly finished |
| 16:54.06 | ZX81 | :) |
| 16:54.09 | ZX81 | I'll show you |
| 16:54.20 | Zeeek | mAsterPanel |
| 16:54.21 | jayden | internal Ip is 172.21.x.x, also hax other 172.2x.x.x and 192.168.x.x networks internally |
| 16:54.26 | tzafrir | What is SineAsterPanel? |
| 16:54.32 | brettnem | SineAsterPanel? |
| 16:54.43 | jayden | SineAsterPanel? |
| 16:54.43 | brazil | bindaddr=0.0.0.0 |
| 16:54.51 | brazil | externip=0.0.0.0 |
| 16:54.53 | ZX81 | :) |
| 16:54.57 | ZX81 | one sec |
| 16:55.00 | ZX81 | making a screenshot |
| 16:55.01 | ZX81 | :) |
| 16:55.01 | brazil | every address is listening |
| 16:55.06 | Zeeek | For a limited instruction set, try AsterRisc |
| 16:55.10 | brettnem | I'm really waiting for a web configuration GUI like AMP or Destar that simply supports contexts.. I can't figure out why no one has implemented that yet |
| 16:55.13 | bjohnson | hey ! There's new firmware for the SPA 2k and 3k !! |
| 16:55.27 | jayden | y, but how does it know what to send in the sip packet then, I am behind nat |
| 16:55.30 | ZX81 | ok |
| 16:55.33 | ZX81 | so, |
| 16:55.44 | ZX81 | http://zx81.sineapps.com/sine.jpg |
| 16:55.45 | Zeeek | baited breath |
| 16:55.48 | ZX81 | :) |
| 16:55.50 | tzafrir | brettnem, because it is not simple |
| 16:55.51 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 16:55.56 | brazil | yes, you are |
| 16:56.07 | ZX81 | with drag and drop contacts |
| 16:56.09 | brettnem | hehe.. yes it is.. :P |
| 16:56.12 | ZX81 | from outlook etc |
| 16:56.17 | ZX81 | and transfer etc |
| 16:56.22 | ZX81 | with no yucky # things |
| 16:56.26 | ZX81 | and call parking |
| 16:56.29 | ZX81 | and speed dials |
| 16:56.35 | ZX81 | oh and the buttons light up |
| 16:56.40 | Zeeek | OUTLOOK? di you say OUTLOOK? |
| 16:56.40 | ZX81 | if a person is on the phone |
| 16:56.42 | ZX81 | etc |
| 16:56.44 | ZX81 | heh |
| 16:56.45 | ZX81 | yeah |
| 16:56.46 | ZX81 | cos |
| 16:56.51 | Zeeek | same to you |
| 16:56.53 | ZX81 | Thunderbird doesn't send any data |
| 16:56.55 | ZX81 | lol |
| 16:57.03 | ZX81 | I can receive whatver |
| 16:57.12 | ZX81 | but the tunderbird drop contains nothing yet |
| 16:57.13 | ZX81 | lol |
| 16:57.17 | jayden | pretty |
| 16:57.31 | ZX81 | :) |
| 16:57.36 | brazil | i'm realy need a script to QoS (htb + sfq) to use together asterisk!! Anybody can help me |
| 16:57.37 | tzafrir | brettnem, what would you expect such a UI to do? |
| 16:57.40 | ZX81 | I can put in any other drop source |
| 16:57.50 | ZX81 | when you have debug=10 in the config |
| 16:57.51 | jayden | so, any answers for me :( |
| 16:58.00 | ZX81 | it tells you all data |
| 16:58.02 | ZX81 | to the log file |
| 16:58.04 | brazil | sorry jayden |
| 16:58.06 | ZX81 | :) |
| 16:58.10 | ZX81 | ah |
| 16:58.11 | jayden | hehe :) |
| 16:58.13 | brettnem | tzafrir: to be just like AMP, but you can define contexts and lots of DIDs.. think "multi-tenent" |
| 16:58.13 | ZX81 | what was the q? |
| 16:58.15 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 16:58.15 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:58.27 | Zeeek | 42 was the answer |
| 16:59.07 | brettnem | well 7*6 of course |
| 16:59.27 | bjohnson | how is "Change factory default <Time Zone>" considered a Feature Enhancement? |
| 16:59.36 | jayden | oky, well, gotta go to lunch.. bbiab |
| 16:59.36 | *** part/#asterisk hans (fugalh@falcon.fugal.net) |
| 16:59.51 | brazil | Jayden.. |
| 17:00.34 | MuppetMaster | Any comments on the issue with the Monitor cmd? |
| 17:00.52 | Tall-guy | zx81: where's the sinclair interface? :) |
| 17:01.16 | ZX81 | :) |
| 17:01.24 | ZX81 | skinnable too |
| 17:01.25 | ZX81 | :) |
| 17:01.43 | *** join/#asterisk rajo (~rajo@graphics.cs.uni-sb.de) |
| 17:02.11 | Tall-guy | zx81: is that an oppanel or a softphone? |
| 17:02.17 | ZX81 | oppanel |
| 17:02.19 | ZX81 | for FXS |
| 17:02.23 | ZX81 | but works with anything |
| 17:02.33 | ZX81 | I.E. sip/IAX/Zap tested |
| 17:02.34 | ZX81 | :) |
| 17:02.44 | ZX81 | makes your phone ring |
| 17:02.47 | ZX81 | you pick up |
| 17:02.50 | ZX81 | and the call starts |
| 17:02.51 | ZX81 | :) |
| 17:03.22 | ZX81 | internally it stores a whole lot of extra info |
| 17:03.25 | ZX81 | like cdr's etc |
| 17:04.01 | Tall-guy | zx: so it inself is not a phone, but similar to other op_panel's (sic) for call control |
| 17:04.06 | ZX81 | yep |
| 17:04.16 | ZX81 | you could run it with a softphone if you like |
| 17:04.17 | Tall-guy | (looks pretty!) |
| 17:04.18 | ZX81 | but the idea |
| 17:04.30 | ZX81 | is so that you can buy a shit phone but still have all functionality |
| 17:04.34 | ZX81 | with FXS devices |
| 17:04.39 | ZX81 | (I have 24 fxs) |
| 17:05.19 | Tall-guy | zx81: yeah, heck all you need is a pc....much cheaper than a phone! (yes, I'm being facetious!) |
| 17:05.45 | ZX81 | hehe |
| 17:05.54 | ZX81 | it's actually for a customer |
| 17:06.01 | ZX81 | but I'm gonna open source it |
| 17:06.03 | Tall-guy | zx: its nice, don't get me wrong. |
| 17:06.05 | ZX81 | once he's paid |
| 17:06.07 | ZX81 | :) |
| 17:06.10 | Tall-guy | zx: much nicer than a web based one. |
| 17:06.13 | ZX81 | He already agreed |
| 17:06.15 | ZX81 | :) |
| 17:06.19 | Tall-guy | zx: what did u write it in? |
| 17:07.19 | *** join/#asterisk Kresike (~crash@netmaster.tvnetwork.hu) |
| 17:07.22 | Kresike | hello all |
| 17:07.28 | brazil | hello |
| 17:07.39 | ZX81 | c# |
| 17:07.43 | ZX81 | using sharpdevelop |
| 17:07.45 | ZX81 | :) |
| 17:07.49 | mikegrb | c#! |
| 17:07.52 | mikegrb | ZX81: <3 |
| 17:08.07 | ZX81 | :) |
| 17:08.17 | ZX81 | are you the guy who told me about it? |
| 17:08.26 | mikegrb | doubful |
| 17:08.29 | ZX81 | oh |
| 17:08.30 | mikegrb | but I like it |
| 17:08.36 | mikegrb | unless it was in here |
| 17:08.37 | mikegrb | dunno |
| 17:08.41 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 17:08.41 | ZX81 | yah |
| 17:08.43 | ZX81 | was in here |
| 17:08.48 | __Sparks_ | I have a funny voicemail problem! - If i dial into my mailbox, then change the password, this works fine - but if I restart asterisk, the password reverts to the origanal one! - (Using Xorcom Rapid) |
| 17:08.49 | ZX81 | someone said: use c# |
| 17:08.51 | ZX81 | so I said ok |
| 17:08.54 | ZX81 | and did |
| 17:08.59 | ZX81 | and it's piss easy! |
| 17:09.00 | ZX81 | :)( |
| 17:09.04 | ZX81 | real nice |
| 17:09.04 | ZX81 | :) |
| 17:09.17 | ZX81 | __Sparks_: indeed that is a strange problem |
| 17:09.33 | tzafrir | yes, you have to change it from outside of asterisk |
| 17:09.51 | mikegrb | __Sparks_: make sure asterisk has write access to the voicemail.conf |
| 17:09.51 | tzafrir | either edit the file manually, or use the linux shell command: |
| 17:10.04 | mikegrb | tzafrir: nah you don't |
| 17:10.07 | tzafrir | ast-cmd -c NUM vm-pass |
| 17:10.19 | Kresike | __Sparks_ try using a database to store your passwords ... that way you can change passwords and keep them after restart too |
| 17:10.22 | mikegrb | tzafrir: asterisk will update the file if it has write permission |
| 17:10.28 | ZX81 | ast-cmd? |
| 17:10.37 | tzafrir | mikegrb, it is no in that file |
| 17:10.38 | ZX81 | what is ast-cmd?! |
| 17:10.44 | ZX81 | omg I never seen it? |
| 17:10.57 | bjohnson | damn .. need to find a use for this computer so I can get the 17" lcd ! http://click.linksynergy.com/fs-bin/click?id=CAqD7bLWUPI&offerid=85012.386269733&type=10&subid= |
| 17:11.07 | tzafrir | ast-cmd is a script we added in rapid for all the small scripts we needed |
| 17:11.13 | ZX81 | ah |
| 17:11.14 | ZX81 | ok |
| 17:11.15 | Beirdo | heya, mikegrb :) |
| 17:11.15 | ZX81 | :) |
| 17:11.21 | ZX81 | wow that was close |
| 17:11.23 | mikegrb | ZX81: is that a softphone or manager interface thingie |
| 17:11.29 | ZX81 | manager |
| 17:11.34 | ZX81 | dialer mostly |
| 17:11.36 | mikegrb | tzafrir: why does xorcom do it differently then, that seems silly |
| 17:11.38 | mikegrb | ZX81: spiff |
| 17:11.38 | ZX81 | +transfer bla bla |
| 17:11.41 | ZX81 | :) |
| 17:11.52 | mikegrb | ZX81: you get three gold stars on your file today |
| 17:11.53 | Beirdo | trying my hand at working today, still not 100% :( |
| 17:11.54 | ZX81 | will open source it in the next couple of weeks |
| 17:11.54 | mikegrb | Beirdo: howdy |
| 17:11.59 | ZX81 | yay!!!!! |
| 17:12.01 | ZX81 | :) |
| 17:12.11 | mikegrb | ZX81: I will hold my breath so you better make it quick, bish |
| 17:12.17 | tzafrir | mikegrb, It would also seem silly to update the entire config file for every small password change |
| 17:12.19 | ZX81 | hehe |
| 17:12.31 | ZX81 | it already works... |
| 17:12.39 | tzafrir | Also think about different permissions to the files that contain the passwords |
| 17:12.42 | ZX81 | I just have to wait for the customer to finish paying me |
| 17:12.47 | ZX81 | before I open source it |
| 17:12.48 | ZX81 | :) |
| 17:12.59 | mikegrb | tzafrir: so because you don't like the way asterisk changes things you change it in your distribution? |
| 17:13.06 | mikegrb | tzafrir: how about asterisk owns it |
| 17:13.08 | *** join/#asterisk jtodd (~jtodd@h-67-103-42-29.snfccasy.covad.net) |
| 17:13.11 | mikegrb | tzafrir: the default permissions work |
| 17:13.29 | mikegrb | tzafrir: all most all the config files have passwords of various sorts, do you take all of those out too? |
| 17:13.33 | Beirdo | I wish the voicemail SQL stuff was standard |
| 17:13.37 | tzafrir | But actually, I never realised that asterisk writes directly to its config file (and not to and #include-d file, which is a bug) |
| 17:14.41 | tzafrir | and it also seems that direct mysql access and h323 support are not the best of friends, license-wise |
| 17:15.18 | Beirdo | huh? |
| 17:15.38 | Beirdo | how do you figure? |
| 17:16.01 | tzafrir | The license of asterisk has exceptions from the GPL that allow it to link with some gpl-incompatible like libopenh323 (MPL) |
| 17:16.04 | *** join/#asterisk Buana (~thomasn@G721f.g.pppool.de) |
| 17:16.14 | Beirdo | so? |
| 17:16.21 | tzafrir | mysql's client libraries are GPL, with no such exceptions |
| 17:16.44 | mikegrb | h323 is stupid |
| 17:16.49 | Beirdo | I don't see a conflict there |
| 17:16.54 | tzafrir | people use and want h323 |
| 17:17.42 | tzafrir | either asterisk is standard GPL with no exceptions, so you can link to mysql, or there are some exceptions that allow you to link with libopenh323 |
| 17:18.31 | mikegrb | people shouldn't use asterisk if they want h323 |
| 17:18.31 | tzafrir | back to destar... |
| 17:24.00 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
| 17:25.28 | adjacent | irrelevant of complexity, can i build an asterisk box that has 6 incoming lines, puts the caller through a voice menu, then outputs the call to the selected phone. without a digium card? |
| 17:25.36 | *** part/#asterisk _PiGreco_ (~a@adsl-215-48.38-151.net24.it) |
| 17:26.22 | tessier_ | adjacent: How do you plan to get the calls into the box without a digium card? |
| 17:26.28 | Tall-guy | adjacent: u need SOME kinda card.... |
| 17:26.30 | *** join/#asterisk ZX81 (matt@222-153-114-115.jetstream.xtra.co.nz) |
| 17:26.38 | adjacent | tessier_: modem? i dunno, thats why im asking |
| 17:26.43 | tessier_ | adjacent: The only way without a digium card is to have someone send you the calls via SIP |
| 17:26.56 | adjacent | hmm. i have a VG with a PRI card. |
| 17:26.56 | tessier_ | adjacent: Why would you think you can do it with a modem? |
| 17:26.58 | JerJer | mikegrb: right on |
| 17:27.01 | Tall-guy | doesn't HAVE to be Digium card..... |
| 17:27.39 | JerJer | but it should be a Digium card |
| 17:27.51 | adjacent | i could do SIP on the internal network. thats ok. have calls go to the cisco router and be forwarded via SIP to the * |
| 17:28.12 | JerJer | sure |
| 17:28.40 | adjacent | except i have no PRI at the office location. and i cant buy a fractional PRI |
| 17:28.59 | ZX81 | what's libpri-matt? |
| 17:29.08 | ZX81 | why does matt have his own version? |
| 17:29.11 | JerJer | adjacent: then SIP it is |
| 17:29.37 | JerJer | cuz matt is special? |
| 17:29.41 | ZX81 | :) |
| 17:29.42 | JerJer | ed |
| 17:29.44 | ZX81 | cool |
| 17:29.48 | ZX81 | I'll go with that |
| 17:29.53 | ZX81 | :) |
| 17:30.05 | mikegrb | JerJer: :D |
| 17:30.59 | doughecka_ | my boss is a matt |
| 17:31.00 | doughecka_ | :P |
| 17:31.12 | tessier_ | I have a matt outside my front door. |
| 17:31.12 | ZX81 | cool, I'm your boss |
| 17:31.15 | doughecka_ | welcome matt |
| 17:31.17 | ZX81 | hehe |
| 17:31.19 | doughecka_ | dor matt |
| 17:31.21 | doughecka_ | door* |
| 17:32.27 | doughecka_ | something that would tunnel sip over iax to a box inside a nat |
| 17:32.27 | JerJer | wy would you want one? |
| 17:32.43 | *** join/#asterisk sangee (ravi@209.250.129.135) |
| 17:33.06 | doughecka_ | so I could plug in a sip phone, and have it talk to a server on der veb, and tunnel it inside a network that prevents sip from getting through |
| 17:33.09 | JerJer | so like why do you need anything special to to taht? |
| 17:33.24 | JerJer | this keyboard is borken this morning |
| 17:33.30 | doughecka_ | broken? :P |
| 17:33.31 | adjacent | JerJer: yeh. but how do i run 6 lines into my cisco? i have a voice card that accepts a smartjack |
| 17:33.47 | JerJer | 6 lines? |
| 17:33.52 | *** join/#asterisk RoyK (~roy@8.80-203-22.nextgentel.com) |
| 17:34.34 | adjacent | JerJer: yeh. i have 6 lines coming in here. 3 numbers |
| 17:34.40 | adjacent | two lines per number |
| 17:34.44 | JerJer | huh? |
| 17:34.51 | JerJer | is this a PRI? |
| 17:35.12 | adjacent | it is voip now. |
| 17:35.21 | JerJer | then you configure your VG |
| 17:35.28 | adjacent | but the call quality blows. not good enough for business use |
| 17:35.43 | adjacent | so i need to move the 3 numbers to copper and use voip for intrnal call management |
| 17:35.47 | JerJer | then fire your ISP and/or network admin |
| 17:36.02 | *** join/#asterisk Tili (~Tili@202-133-67-206-dialup.sat.net.pk) |
| 17:36.12 | adjacent | haha. i run an isp and im the net admin. =) |
| 17:36.19 | Tall-guy | ahahhha |
| 17:36.27 | doughecka_ | haaa |
| 17:36.36 | Tall-guy | ok, thats just damn funny |
| 17:36.40 | Tall-guy | sorry to laff at your expense adjacent |
| 17:36.48 | adjacent | JerJer: call quality sucks because i cant guarantee QoS once the call leaves my network |
| 17:36.58 | adjacent | yeh. its ok. i even laughed ;) |
| 17:37.02 | JerJer | then don't use VoIP |
| 17:37.18 | ZX81 | use satellite |
| 17:37.26 | ZX81 | I hear it's really good this time of year |
| 17:37.27 | ZX81 | :) |
| 17:37.29 | doughecka_ | use pigians |
| 17:37.31 | ZX81 | lol |
| 17:37.38 | doughecka_ | pigions |
| 17:37.39 | ZX81 | is that like pagans? |
| 17:37.40 | doughecka_ | whatever |
| 17:37.42 | ZX81 | hehe |
| 17:37.47 | doughecka_ | birds |
| 17:37.50 | ZX81 | lol |
| 17:37.54 | adjacent | JerJer: thats what i want to do. i need to figure out the logistics of moving the numbers from my PRI at my colo to the office |
| 17:37.56 | ZX81 | flying birds |
| 17:37.57 | ZX81 | :) |
| 17:38.00 | doughecka_ | antenallope |
| 17:38.03 | ZX81 | lol |
| 17:38.05 | ZX81 | kiwi's |
| 17:38.09 | ZX81 | oh no wait.... |
| 17:38.11 | ZX81 | :) |
| 17:38.14 | CpuID | hmm dang im gonna be tired tomorrow me thinks :) |
| 17:38.15 | JerJer | Auk's |
| 17:38.20 | doughecka_ | dodo |
| 17:38.22 | ZX81 | lol |
| 17:38.27 | CpuID | 3:40am so far |
| 17:38.29 | ZX81 | it's 6:39am here |
| 17:38.32 | ZX81 | it is tomorrow |
| 17:38.34 | CpuID | heh |
| 17:38.37 | ZX81 | the morning news is on |
| 17:38.37 | ZX81 | lol |
| 17:38.40 | JerJer | adjacent: port them |
| 17:38.44 | JerJer | adjacent: call your carrier |
| 17:38.53 | *** join/#asterisk schwagner (~andrew@68.143.92.248.nw.nuvox.net) |
| 17:38.56 | doughecka_ | its noon here |
| 17:38.59 | CpuID | lol |
| 17:39.08 | JerJer | nooner ! ! ! |
| 17:39.11 | ZX81 | :) |
| 17:39.58 | mikegrb | liar! |
| 17:40.09 | mikegrb | it isn't noon anywhere! |
| 17:40.13 | ZX81 | lol |
| 17:40.25 | adjacent | JerJer: yeh. but they are going to give me 6 twisted pair of telephone wire and let me do what i want. how do i bring these into the cisco? |
| 17:40.29 | ZX81 | yeah it is: GMT +x.1 |
| 17:40.30 | ZX81 | :) |
| 17:40.33 | doughecka_ | its actully 12:15 |
| 17:40.34 | doughecka_ | :P |
| 17:40.37 | ZX81 | heh |
| 17:40.41 | ZX81 | 15? |
| 17:40.41 | adjacent | im not a phone person, if that isnt obvous enough |
| 17:40.45 | junky[work] | 45 ? |
| 17:40.49 | ZX81 | you have a half timezone? |
| 17:41.05 | junky[work] | he lives between 2 tz :P |
| 17:41.05 | doughecka_ | yup |
| 17:41.08 | ZX81 | now I'm really confused...what's the time here |
| 17:41.10 | ZX81 | oh ok |
| 17:41.13 | ZX81 | phew!!! |
| 17:41.15 | ZX81 | :) |
| 17:41.15 | doughecka_ | ~date |
| 17:41.16 | jbot | Wed Feb 16 17:41:16 2005 |
| 17:41.16 | doughecka_ | ~time |
| 17:41.17 | jbot | rumour has it, time is 1 dimensional, or everlasting |
| 17:41.30 | ZX81 | ~me |
| 17:41.31 | jbot | [zx81] the creater of the Daily Asterisk News (see ~adn) |
| 17:41.32 | junky[work] | doughecka_ where are ya from? |
| 17:41.34 | ZX81 | wow |
| 17:41.36 | junky[work] | ~adn |
| 17:41.37 | jbot | somebody said adn was the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://www.sineapps.com/rssfeed.php for RSS |
| 17:41.37 | ZX81 | ok |
| 17:41.39 | ZX81 | :) |
| 17:41.42 | junky[work] | adn means dna in french :) |
| 17:41.47 | ZX81 | heh |
| 17:41.49 | mikegrb | reagan eats brains --> http://crackmonkey.org/~nick/mail/take-your-meds |
| 17:41.51 | doughecka_ | us |
| 17:41.53 | ZX81 | dyslexic? |
| 17:41.54 | ZX81 | hehe |
| 17:41.59 | doughecka_ | actully right across the river from louisville |
| 17:42.00 | *** join/#asterisk carbon60 (~adam@gw.techsupport.ca) |
| 17:42.32 | JerJer | cixelsyd |
| 17:42.53 | doughecka_ | ? |
| 17:42.56 | *** join/#asterisk eivindtr (~Eivind@062016241059.customer.alfanett.no) |
| 17:42.57 | *** part/#asterisk Buana (~thomasn@G721f.g.pppool.de) |
| 17:43.02 | JerJer | adjacent: you don't |
| 17:43.07 | JerJer | you buy the right piece of gear |
| 17:43.36 | JerJer | get a TA750 and a TE405P |
| 17:44.34 | RoyK | JerJer: do you read the -dev list? |
| 17:44.49 | RoyK | wtf is a ta750? |
| 17:44.51 | sangee | i want to send the userid and password when i dial out to some GW using oh323, how to do that |
| 17:45.17 | mikegrb | sangee: you don't use oh323 for starters |
| 17:45.20 | schwagner | anybody here using a T100P card? |
| 17:45.35 | tzafrir | mikegrb, you asked me why I put them in separate files? |
| 17:46.05 | tzafrir | I have just gone over the horrible code of app_voicemail where it tries to update voicemail.conf |
| 17:46.08 | mikegrb | I know why |
| 17:46.15 | tzafrir | It basically assumes that #include is not used |
| 17:46.16 | mikegrb | because you are a silly kanigit |
| 17:46.21 | carbon60 | Has anyone dealt with Polycom SoundPoint IP500s locking up hard? |
| 17:46.23 | sangee | is there anyway i can send userid and password? |
| 17:46.33 | tzafrir | And that code is a bunch of mess. |
| 17:46.41 | mikegrb | it is stupid to put every single mailbox in a different file |
| 17:46.44 | tzafrir | I wanted to be able to update things easily |
| 17:46.49 | mikegrb | you do it so you can feel important |
| 17:46.54 | mikegrb | ha ha |
| 17:46.59 | mikegrb | that is != easy |
| 17:47.14 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 17:47.33 | tzafrir | mikegrb, then please explain why it is stupid? |
| 17:47.56 | mikegrb | It is stupid, I have spoken, the end. |
| 17:48.29 | Nivex | such an enlightened answer |
| 17:48.43 | mikegrb | Nivex: well it matches his intellect |
| 17:50.07 | sangee | is it possible to send user/password with dial? |
| 17:50.28 | mikegrb | yes |
| 17:50.38 | sangee | how could do that |
| 17:50.51 | mikegrb | by not using oh323 |
| 17:51.28 | sangee | with sip, can i do it? |
| 17:52.17 | mikegrb | yes |
| 17:52.25 | sangee | how can i do that? |
| 17:52.26 | mikegrb | but you are better to put it in sip.conf |
| 17:52.43 | sangee | i alrady put it? |
| 17:53.51 | sangee | so the dial command will send the user and password to that GW? |
| 17:54.04 | sangee | if i have it in sip.conf? |
| 17:56.13 | sangee | \ |
| 17:56.56 | sangee | it automatically send userid and password to GW |
| 17:57.07 | sangee | take it from sip.conf? |
| 17:59.09 | doughecka_ | does a cisco phone really need 48 volts? |
| 17:59.12 | *** join/#asterisk CMike (~a_mike@c-304171d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 17:59.19 | Beirdo | mikegrb you're on a roll today :) |
| 18:00.25 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 18:02.11 | mikegrb | Beirdo: I so am |
| 18:02.19 | mikegrb | Beirdo: in more channels then you'd imagine |
| 18:02.23 | Beirdo | heh |
| 18:02.45 | mikegrb | doughecka_: all phones take 48 volts |
| 18:02.51 | mikegrb | sangee: yes |
| 18:03.23 | doughecka_ | yes, they take 48 volts |
| 18:03.37 | doughecka_ | but do they need it to operate... surely it just step it down inside |
| 18:03.56 | mikegrb | open it and see |
| 18:04.07 | doughecka_ | 48 volts is just for the poe standard |
| 18:04.20 | mikegrb | and the pots standard |
| 18:04.24 | doughecka_ | true |
| 18:06.21 | Beirdo | so I don't see the licensing issue with using openh323 with asterisk. as long as it's being used in a module that's dynamically linked... |
| 18:06.26 | [Latre] | i have a TDM04B i want configure this with my PBX Panasonic , in this moment when i dial an extension 114 (this is connected directly to port 2 of TDM) rang 2 times and then dial one extension sip but inmediatly sendme to voicemail..... |
| 18:06.50 | vaewyn | hmm... does the SIP presence stuff work under * ? |
| 18:06.55 | Beirdo | and there's a specific stated exception for openssl and openh323 anyways |
| 18:07.03 | mikegrb | Beirdo: but oh323 sucks anyway :p |
| 18:07.16 | Beirdo | well, that's another issue entirely :) |
| 18:07.30 | mikegrb | :D |
| 18:08.05 | Beirdo | as asterisk-oh323 wouldn't compile for me anyways, I gave up on it for now |
| 18:08.56 | Beirdo | so the other day, I signed up for voipjet.com, sent em $10 via PayPal |
| 18:08.59 | Beirdo | no DID though |
| 18:09.36 | Beirdo | will give me a good start |
| 18:10.23 | mikegrb | ja |
| 18:10.54 | Beirdo | why isn't mysql-vm-routines.h in asterisk instead of -addons? |
| 18:11.58 | mikegrb | silly license stuff |
| 18:12.07 | Beirdo | such as? |
| 18:12.12 | Beirdo | mysql is GPL |
| 18:12.17 | Beirdo | well it is *NOW* |
| 18:12.30 | Beirdo | I guess maybe it wasn't when that was written |
| 18:12.53 | vaewyn | They are moving away from database specific items anyways... going ODBC |
| 18:13.03 | vaewyn | which is IMO stupid but... I can see the points |
| 18:13.10 | Beirdo | ick |
| 18:13.21 | mikegrb | ja |
| 18:13.33 | `Sauron | vaewyn: Ick. |
| 18:13.36 | Beirdo | I guess it has some sense to it, people can use whatever backend they want |
| 18:13.41 | Beirdo | but ICK anyways |
| 18:13.49 | vaewyn | *nods* |
| 18:14.05 | `Sauron | What they should do, is slurp in perl or something |
| 18:14.15 | `Sauron | so you can use any DB that has a DBI:DBD interface |
| 18:14.40 | vaewyn | but then they still have to do server specific SQL... which is part of the allure of ODBC |
| 18:14.54 | `Sauron | Not neccesarily |
| 18:15.04 | `Sauron | You can write DB-portable SQL ;) |
| 18:15.08 | `Sauron | You just have to be careful |
| 18:15.09 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-252-cust.telepacific.net) |
| 18:15.19 | vaewyn | and can't use 95% of the features :P |
| 18:16.01 | `Sauron | Right. |
| 18:16.14 | `Sauron | Because few other DB's implement SQL92/99 properly |
| 18:16.31 | `Sauron | ask me again why I hate mycrapsql |
| 18:17.03 | vaewyn | all DBs are crap when it comes to sql standard conformance :} |
| 18:17.23 | `Sauron | shrug |
| 18:17.37 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 18:17.38 | `Sauron | doing the "farm interfaces out to someone else" method works wonderfully in openradius |
| 18:17.51 | Beirdo | blah blah blah :) |
| 18:17.58 | `Sauron | and, should I be braindead enough to want to interface with odbc, I can |
| 18:18.14 | `Sauron | Beirdo: do you actually do any development? |
| 18:18.30 | Beirdo | not on asterisk, but yes |
| 18:19.51 | brazil | hello all.. |
| 18:23.25 | *** join/#asterisk PTG123 (~PTG@ip67-153-155-242.z155-153-67.customer.algx.net) |
| 18:25.28 | bjohnson | here's a question that is probably too specific to my system for anyone to answer but .. I have SPA 3000s answering calls and hooked to my Nortel line in's. Ofetn when a call is hung up I get what I call a ghost ring on my Nortel. It seems the fxo port is still connected, senses the callerid, and re-rings into * as though it's a new call. Any ideas on this one? |
| 18:25.45 | ta[i]nted | Beirdo what do you do development on? |
| 18:26.09 | bjohnson | I develop my beer belly |
| 18:26.19 | bjohnson | it's in grand form |
| 18:26.23 | bjohnson | a complete success |
| 18:26.36 | bjohnson | next project -> a fat ass |
| 18:26.42 | bjohnson | and I've already started |
| 18:28.08 | bjohnson | it seems there isn't one way to interface to multiple DBs that stands out as the best way .. or there wouldn't be any discussion needed |
| 18:29.10 | bjohnson | I think serial increases Amps |
| 18:29.31 | *** part/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 18:29.54 | doughecka_ | increasing voltage |
| 18:30.00 | doughecka_ | like adding batteries |
| 18:30.22 | PTG123 | yah use 4 12v |
| 18:30.27 | vaewyn | Anyone wondering impressions on the Hitachi Cable WIP-5000 -> http://www.wwwrogue.com/voip/WIP5000.html only had the phone for 6 hours though so... :} more info to come |
| 18:30.28 | PTG123 | would work fine |
| 18:30.33 | PTG123 | amps would stay the same if its not in a series |
| 18:30.36 | PTG123 | voltage would increase |
| 18:30.43 | Nugget | vaewyn: cool, thanks. |
| 18:30.50 | doughecka_ | hmm |
| 18:30.54 | PTG123 | so 4 1am 12v power supplies, would make 48v 1amp PS |
| 18:30.59 | vaewyn | Nugget: np :} |
| 18:31.06 | doughecka_ | so 2 12V... |
| 18:31.19 | PTG123 | its smarter to use matched ones |
| 18:31.24 | doughecka_ | ah, true |
| 18:31.37 | *** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com) |
| 18:31.41 | PTG123 | don't want any fires :) |
| 18:31.44 | bde | can anyone tell me what kind of rates i can get for 1M minutes/month to the US/Canada? |
| 18:31.51 | PTG123 | aalso this is only true for DC power |
| 18:31.53 | PTG123 | not AC :) |
| 18:31.58 | PTG123 | is it DC? |
| 18:32.07 | bjohnson | bde: likely not much less than 1c USD/min |
| 18:32.09 | PTG123 | bde: depends on quality, etc |
| 18:32.22 | Zeeek | works for AC if there is exactly 180° phase diff |
| 18:32.26 | bde | ok, thanks |
| 18:32.29 | doughecka_ | yea DC |
| 18:32.48 | PTG123 | Zeeek: not without alot more circuity, since you can't guarantee the phase sync.. it could actually cancel each other out |
| 18:32.58 | PTG123 | its not a simple asp utting them all together then |
| 18:33.03 | Zeeek | and anyway I'm wrong - that would produce 0 |
| 18:33.07 | bjohnson | bde: http://www.livevoip.com/index.php?subject=1&content=usaCanadaRates |
| 18:33.08 | PTG123 | and actually you would want no phase difference |
| 18:33.15 | Zeeek | but it really isn't a good idea to put PS in series |
| 18:33.22 | Zeeek | DC or not |
| 18:33.33 | PTG123 | he isn't putting it in a series :) |
| 18:34.03 | doughecka_ | I am putting them in lpt |
| 18:34.06 | bde | bjohnson: have you used them in the past? |
| 18:34.07 | Zeeek | I'd wire up 4 cellphone batteris instead :) |
| 18:34.11 | doughecka_ | hah |
| 18:34.24 | Zeeek | wait better idea: |
| 18:34.40 | PTG123 | bde: you should talk to teliax tell him i sent you |
| 18:34.46 | Zeeek | run the serai port into a bit shifter until you have enough for 48v |
| 18:34.54 | doughecka_ | riiiight |
| 18:35.01 | bjohnson | bde: gee .. you want me to do your research and then you want me to research them? OK .. get me 1M minutes and I'll start right away |
| 18:35.01 | Zeeek | then it will be ok |
| 18:35.07 | doughecka_ | and like .00001 amps |
| 18:35.08 | doughecka_ | :P |
| 18:35.24 | Zeeek | not with a 1000uF cap |
| 18:35.30 | Zeeek | that'll stabilize her |
| 18:35.41 | bde | bjohnson: i was just asking if you've used them before |
| 18:35.45 | PTG123 | just build your own ps while yuo are at it :) |
| 18:35.52 | Zeeek | alternative #144: find an old TV set with CR tube. |
| 18:36.14 | bjohnson | alternative #145: a LOT of lemons !! |
| 18:36.19 | doughecka_ | PTG123: I am lucky enough to have a voltmeter |
| 18:36.21 | Zeeek | charge it up to 10KV, then find a 20gazillion ohm resistor and a rectifier bridge |
| 18:36.32 | Zeeek | (available at Radio Shack, $9.95) |
| 18:36.36 | Beirdo | ta[i]nted: sorry, had to run down to the machine room. Mainly on ancilliary tools for MythTV and a MUD |
| 18:36.37 | doughecka_ | rightg |
| 18:36.45 | PTG123 | doughecka_, well just remember 48v could be 49v, 46v, etc |
| 18:36.51 | Beirdo | well, not tools for the MUD, the MUD itself |
| 18:36.51 | PTG123 | when they say 48, they mean about 48 :) |
| 18:36.54 | Zeeek | yes, citirc acid and some copper and nickel will do it |
| 18:37.12 | doughecka_ | lol |
| 18:37.18 | doughecka_ | well, about is not 24V |
| 18:37.19 | doughecka_ | :( |
| 18:37.24 | PTG123 | true :) |
| 18:37.26 | bjohnson | need more lemons |
| 18:37.35 | doughecka_ | gonna freak my cow-orkers out |
| 18:37.41 | *** join/#asterisk [ro]nic3try (~iancu@81.181.199.39) |
| 18:37.47 | [ro]nic3try | re all :) |
| 18:37.53 | doughecka_ | yea, that whole powerstrip of powersuplpys are running a single phone |
| 18:37.54 | Zeeek | there is unfortunatley zeeeks law about the diminishing return of a large number of lemons |
| 18:38.19 | HitTop | anyone know ser here? |
| 18:38.22 | Zeeek | 1.41414 x $lemons |
| 18:38.59 | *** join/#asterisk c2bprojects (~c2bprojec@host213-122-27-227.in-addr.btopenworld.com) |
| 18:39.15 | bjohnson | I wonder how much solar panel he'd need for 48V |
| 18:39.29 | [ro]nic3try | i have a problem with mp3player form asterisk,.. when i try to put an mp3 it sounds like crap :( |
| 18:39.32 | PTG123 | one solar panel for 48v, but he wouldn't have the amps he needs :) |
| 18:39.37 | bjohnson | desk mounted wind turbine? |
| 18:39.51 | bjohnson | on a 50' pole? |
| 18:40.11 | PatrickDK | hmm, wind turbines produce alot of electricity |
| 18:40.11 | *** join/#asterisk Mneumonic (Mnemonic@ool-18ba58b4.dyn.optonline.net) |
| 18:40.20 | [ro]nic3try | pls ... |
| 18:40.23 | PatrickDK | a single one runs the whole 911 system here |
| 18:40.34 | PatrickDK | it hasn't touched power from the grid for 4 years |
| 18:40.37 | doughecka_ | sorry, cant help you, the wind died |
| 18:40.55 | bjohnson | check what verion of mpg123 you have |
| 18:40.55 | PatrickDK | doug, that is what one single 48v battery does :) |
| 18:41.12 | doughecka_ | haha |
| 18:41.17 | doughecka_ | ooh, 48V battery~ |
| 18:41.19 | doughecka_ | what I need |
| 18:41.20 | doughecka_ | :P |
| 18:41.23 | redder86 | I'd like to mount some solar panels on my house roof. |
| 18:41.28 | [ro]nic3try | its an oky.. only with the phone doesn't work |
| 18:41.40 | [ro]nic3try | xmms sounds ok |
| 18:41.59 | bjohnson | that's great! |
| 18:42.04 | bjohnson | now check what verion of mpg123 you have |
| 18:42.11 | bjohnson | version |
| 18:42.21 | [ro]nic3try | 0.59q |
| 18:42.44 | bjohnson | is that supposed to work? I thought just mpg123r |
| 18:42.48 | bjohnson | .59r |
| 18:43.10 | schwagner | [ro]nic3try: try disabling mmx, if you can; that worked for me |
| 18:43.18 | PatrickDK | heh, I have aot of 12v equipment, run them all off two nice deepcycle batteries, can keep the equipment up under full load for 15 days, then just run a basic charger to the batteries |
| 18:43.33 | [ro]nic3try | but when i log as root is 0.59r |
| 18:45.19 | schwagner | join #opencrx |
| 18:45.22 | schwagner | oops |
| 18:45.35 | bjohnson | vaewyn: glad you're happy with it |
| 18:45.45 | [ro]nic3try | schwagner: how should i do that ? |
| 18:45.49 | bjohnson | vaewyn: mind moving your notes to the wiki so they're easier to find? |
| 18:46.23 | schwagner | [ro]nic3try: happen to be running gentoo? |
| 18:46.23 | vaewyn | bjohnson: I would if the wiki you let me edit :} the moment I login it decides I can no longer edit any pages |
| 18:46.50 | bjohnson | strange .. works for me |
| 18:47.06 | bjohnson | vaewyn: galeon web browser here |
| 18:47.44 | vaewyn | bjohnson: yeah... not sure... firefox and mozilla here |
| 18:47.47 | *** join/#asterisk visik7 (~ciao@host11-39.pool80182.interbusiness.it) |
| 18:48.18 | [ro]nic3try | nope |
| 18:49.01 | redder86 | does dialing *81 pick up any ringing channels on the local system? |
| 18:49.04 | schwagner | [ro]nic3try: check the mpg123 lists then, sorry |
| 18:49.07 | *** join/#asterisk kimosabe (~natt@201.133.216.161) |
| 18:49.12 | *** join/#asterisk Jlau515 (~jackie@global-sf.keen.com) |
| 18:49.19 | [ro]nic3try | i have tryed to convert an mp3 to wav and then to gsm .. still doesnt work |
| 18:49.22 | Zeeek | join pr0ncenter |
| 18:49.26 | Zeeek | damn |
| 18:49.43 | Jlau515 | hi guys, how do u know which asterisk version you have, show version only tells me that i am running 1.0 |
| 18:49.57 | Jlau515 | is that the same as the tarbuild 1.0.5 |
| 18:50.05 | visik7 | Jlau515 form the cli |
| 18:50.18 | Jlau515 | yeh i rand show version from cli |
| 18:50.20 | Corydon-w | Probably not. It's probably 1.0.0 |
| 18:50.37 | Jlau515 | Asterisk CVS-v1-0-02/14/05-18:43:29 built by root@voip01-ntt.net.keen.com on a i686 running Linux |
| 18:50.52 | Jlau515 | i checkout it out of cvs |
| 18:51.05 | Corydon-w | Oh, then it's stable |
| 18:51.13 | Nugget | no, that's not stable. |
| 18:51.18 | Jlau515 | using this command cvs checkout -r v1-0 |
| 18:51.37 | *** join/#asterisk jjg (tink@216.253.86.223) |
| 18:51.38 | jjg | hi |
| 18:51.41 | *** join/#asterisk brazil (~cleber@200.198.105.37) |
| 18:52.00 | jjg | i heard that certain winmodems can be used as FXO cards. is there something similar for FXS cards? |
| 18:52.15 | *** join/#asterisk MrClean (~seabrook@store-fw.porchlight.ca) |
| 18:52.17 | Jlau515 | so is my asterisk 1.0 or 1.0.5? |
| 18:52.25 | kimosabe | jjg for fxs use sipura |
| 18:52.35 | Corydon-w | Neither. It's 1.0-2/14/05 |
| 18:52.37 | dsmouse | bah |
| 18:52.52 | Corydon-w | It's beyond 1.0.5 |
| 18:52.53 | jjg | kimosabe : oh yah, forgot about dongles and stuff |
| 18:53.20 | Jlau515 | ohhh okee, its the most stable release of 1.0 |
| 18:53.50 | Corydon-w | Well, no, it's the stable branch as of a certain date |
| 18:54.03 | kimosabe | jjg sipura u can interface directlly to a pbx in office for voip solution |
| 18:54.58 | Nugget | if you're tracking stable, remember to always use "make update" to update your local sources. If you just do the "cvs up" by hand, that date in the version string won't be updated. |
| 18:56.05 | dsmouse | ok, I've got a queue defined with a member Sip/5102... When a extention calls into the queue, 5102 rings, I hit answer but it doesn't connect me with the caller... |
| 18:56.12 | dsmouse | anyone have a pointer? |
| 18:57.46 | shido6 | uhh |
| 18:57.55 | shido6 | are you using "friend" in your sip.conf for this phone? |
| 18:58.00 | dsmouse | yes |
| 18:58.06 | shido6 | dont use friends |
| 18:58.10 | shido6 | less problems |
| 18:58.14 | shido6 | break it out to a user |
| 18:58.15 | shido6 | and a peer |
| 18:58.26 | dsmouse | mkay |
| 18:58.50 | nirs | hey all |
| 18:58.55 | nirs | how is everybody feeling tonight ? |
| 18:58.57 | kimosabe | does any one know how to config the x-lite to interact with asterisk |
| 18:59.09 | shido6 | yesh... |
| 18:59.10 | Zeeek | how kimosabe |
| 18:59.17 | shido6 | are you nat'd kimosabe? |
| 18:59.18 | Zeeek | turn ON transmit silence |
| 18:59.21 | nirs | kimo, you need to define the proper friend/peer section in SIP conf |
| 18:59.24 | kimosabe | no natt |
| 18:59.26 | nirs | and that's basically it |
| 18:59.28 | shido6 | good |
| 18:59.36 | Jlau515 | how do i know which version of chan_zap i have? |
| 18:59.38 | shido6 | leave the outbound proxy blank |
| 18:59.38 | *** join/#asterisk ^Fenris (~mazurbul@d3-31.rb.ot.centurytel.net) |
| 18:59.41 | shido6 | fill out sip proxy |
| 18:59.43 | shido6 | domain / realm |
| 18:59.45 | nirs | you can find a nice tutorial (well, more or less a tutorial) at http://www.voip-info.org |
| 18:59.46 | shido6 | dns server |
| 18:59.50 | kimosabe | shido6 ive tried but the counter doesnt start counting |
| 18:59.55 | Zeeek | turn ON transmit silence |
| 19:00.01 | Zeeek | audio settings |
| 19:00.01 | ^Fenris | where can I find an * voicemail setup tutorial? |
| 19:00.05 | shido6 | user, display, auth user as your username and ur done |
| 19:00.12 | Zeeek | turn ON transmit silence |
| 19:00.20 | kimosabe | oki let me try again |
| 19:00.27 | Zeeek | oh wait - I was transmitting silence |
| 19:00.32 | Zeeek | no one heard it |
| 19:00.43 | tzafrir | ^Fenris, basic voicemail is pretty simple. Use that attiude and the basic samples |
| 19:01.17 | junky[work] | any good link related to span timing? |
| 19:02.09 | nirs | say, has any encountered a situation where asterisk would perform a full shutdown at random times ? |
| 19:02.14 | nirs | I'm talking about the stable version |
| 19:02.16 | MrClean | Question: Is it possible to trigger additional steps in a dialplan after a call that has been in a queue is disconnected? I know the Dial() command has the "g" option that tells it to continue stepping through the plan on hangup, but in my dialplan the Dial() command is used to call an agent, bridge them to the call in the queue, then hangup. I want to check if the call that was bridged hangs up, not the call out to the Agent. Anyone know ho |
| 19:02.27 | nirs | without any indication of an error in the log files or anything like that ? |
| 19:02.48 | nirs | yes MrClean, it's called DeadAGI, and it's run once a channel dies |
| 19:03.04 | MrClean | Thanks, I'll look into that. |
| 19:04.53 | nirs | any if the Digium wizard around ? |
| 19:05.32 | vaewyn | Ok... Zyxel/Wisip can bite me... even at 300+$ this WIP5000 kicks their @#$@#$ |
| 19:05.32 | kimosabe | shido6 how do i save the config on x lite |
| 19:05.51 | CpuID | hehe vaewyn, im guessing you like that phone :) |
| 19:05.59 | vaewyn | CpuID: hell yeah! |
| 19:06.04 | *** join/#asterisk roamer323 (~sing@HSE-MTL-ppp72937.qc.sympatico.ca) |
| 19:06.06 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 19:06.06 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 19:06.09 | CpuID | did you find out if it could do WPA-PSK by any chance? |
| 19:06.13 | CpuID | :) |
| 19:06.17 | shido6 | back |
| 19:06.25 | vaewyn | Am still surfing the menus |
| 19:06.28 | CpuID | ah yep np |
| 19:06.34 | kimosabe | oki |
| 19:06.39 | CpuID | so have you tried roaming between ap's now? |
| 19:07.05 | shido6 | kimo |
| 19:07.05 | dsmouse | shido6: :( still isn't working |
| 19:07.10 | shido6 | it saves it for ya |
| 19:07.11 | vaewyn | Yeah... 2-4 seconds of dead air while our switches recognize where the MAC is now but it never drops the call |
| 19:07.17 | shido6 | dsmouse what isnt working, specifics, please :) |
| 19:07.23 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 19:07.26 | vaewyn | need to see if I can speed the switches up on that |
| 19:08.20 | dsmouse | OK, I'm using SJphone as a SIP clint at extention SIP/5102 and is a member of queue default. I have a call comming in on zap/1, goes into the queue, |
| 19:08.24 | CpuID | ah fair enough |
| 19:08.25 | vaewyn | CpuID: http://wwwrogue.com/voip/WIP5000.html |
| 19:08.32 | kimosabe | shido i put all the correct parameters in menu then i go to dial the number i take line 1 dial number but the counter doesnt start counting |
| 19:08.49 | CpuID | nice one :) |
| 19:08.50 | shido6 | kimosabe does it place the call? |
| 19:08.52 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 19:08.57 | dsmouse | 5102's like rings, and I hit answer but the call remains in the queue. In fact, 5102 will ring again in a few seconds |
| 19:09.07 | vaewyn | need to get my camera and take some pics tonight |
| 19:09.09 | shido6 | dsmouse show me your sip.conf |
| 19:09.12 | shido6 | pastebin.ca your sip.conf |
| 19:09.17 | kimosabe | shido all i see is where i enter the number how can i see if it placed the call |
| 19:09.24 | shido6 | and your dialplan and your queues conf |
| 19:09.25 | CpuID | yea was gonna say :) |
| 19:09.42 | dsmouse | shido6: all of it or just the parts for 5102 |
| 19:09.43 | shido6 | kimosabe you will know it placed the call when ur talking to someone on the other end... |
| 19:09.44 | vaewyn | pretty blue leds..... mmm... |
| 19:09.58 | shido6 | dsmouse use pastebin.ca to paste your sip.conf I need to see it |
| 19:10.12 | Essobi | Anyone familiar with queues and agent penalties? |
| 19:10.12 | shido6 | along with the dialplan referring to that extension |
| 19:10.25 | shido6 | and yoru queue conf file |
| 19:10.27 | dsmouse | ok |
| 19:10.44 | dsmouse | if it helps, that extension can make other calls normaly |
| 19:12.19 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 19:12.42 | *** join/#asterisk mzungu (~tony@mlango.nbi.dtske.com) |
| 19:13.23 | kimosabe | shido the xlite says awaiting proxy login information |
| 19:13.46 | kimosabe | shido but all is in the menu filed all the corect info |
| 19:16.00 | shido6 | dsmouse u have two [general] stanza's in sip.conf |
| 19:16.28 | nirs | any one has an idea why would asterisk stable shut itself down at random times ? |
| 19:17.04 | kimosabe | nirs run safe_asterisk |
| 19:17.06 | dsmouse | I... don't know.... |
| 19:17.25 | nirs | kimo, I do run with safe_asterisk |
| 19:17.30 | nirs | it restarts it |
| 19:17.40 | dsmouse | shido6: I'll comment out the latter for now |
| 19:17.46 | nirs | but every 4-6 minutes, asterisk would restart, killing all the calls on the box |
| 19:17.58 | nirs | not very good for a calling card system, is it ?> |
| 19:18.28 | RoyK | ~seen wasim |
| 19:18.29 | jbot | wasim is currently on #asterisk (12h 41m 18s). Has said a total of 9 messages. Is idling for 3h 17m 53s |
| 19:18.38 | nirs | hey roy |
| 19:18.39 | RoyK | fecking paki. |
| 19:18.43 | RoyK | hi |
| 19:18.55 | nirs | roy, any idea about my question? |
| 19:19.06 | RoyK | what was that? |
| 19:19.07 | nirs | you're one of the more older users here |
| 19:19.14 | nirs | any one has an idea why would asterisk stable shut itself down at random times ? |
| 19:19.20 | nirs | every 4-6 minutes, asterisk would restart, killing all the calls on the box |
| 19:19.24 | Essobi | I'm damn aggravated at the lack of documentation reguerding call penalties. |
| 19:19.27 | Essobi | and queues |
| 19:19.30 | nirs | in the log it shows |
| 19:19.40 | nirs | Feb 16 19:00:27 ivr01 asterisk: asterisk shutdown succeeded |
| 19:19.41 | nirs | Feb 16 19:00:31 ivr01 asterisk: asterisk startup succeeded |
| 19:19.41 | nirs | Feb 16 19:01:00 ivr01 CROND[14662]: (root) CMD (nice -n 19 run-parts /etc/cron.hourly) |
| 19:19.41 | nirs | Feb 16 19:01:00 ivr01 CROND[14663]: (root) CMD (/etc/init.d/asterisk check >/dev/null 2>&1) |
| 19:19.41 | nirs | Feb 16 19:02:01 ivr01 CROND[14685]: (root) CMD (/etc/init.d/asterisk check >/dev/null 2>&1) |
| 19:19.41 | nirs | Feb 16 19:03:00 ivr01 CROND[14742]: (root) CMD (/etc/init.d/asterisk check >/dev/null 2>&1) |
| 19:19.43 | nirs | Feb 16 19:03:21 ivr01 asterisk: asterisk shutdown succeeded |
| 19:19.45 | nirs | Feb 16 19:03:24 ivr01 asterisk: asterisk startup succeeded |
| 19:19.48 | nirs | Feb 16 19:04:00 ivr01 CROND[14854]: (root) CMD (/etc/init.d/asterisk check >/dev/null 2>&1) |
| 19:19.49 | nirs | Feb 16 19:05:00 ivr01 CROND[14895]: (root) CMD (/etc/init.d/asterisk check >/dev/null 2>&1) |
| 19:19.51 | mikegrb | nirs: YOU ARE EVIL |
| 19:19.54 | mikegrb | nirs: YOU ARE EVIL |
| 19:19.55 | nirs | sorry |
| 19:19.57 | mikegrb | nirs: YOU ARE EVIL |
| 19:19.59 | RoyK | ~pastebin |
| 19:20.00 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 19:20.18 | nirs | I keep forgeting that pastebin thingy |
| 19:20.32 | RoyK | nirs: try running asterisk in a screen. run it as asterisk -gvvvvvvvvvvvvvc in a known directory |
| 19:20.41 | nirs | already did that |
| 19:20.44 | nirs | it doesn't core |
| 19:21.11 | RoyK | you said you ran safe_asterisk |
| 19:21.14 | RoyK | asterisk -gvvvvvvc |
| 19:21.24 | RoyK | and do a while loop around it |
| 19:21.30 | RoyK | inna screen |
| 19:21.31 | nirs | well, I also tried running asterisk in the forground with -vvvvvcgp, and it didn't core |
| 19:21.42 | nirs | it just dropped |
| 19:22.19 | RoyK | -g should always dump a core if it gets SIGSEGVd\ |
| 19:22.38 | nirs | well, there is no core |
| 19:22.38 | junky[work] | RoyK: u familiar with span timings? |
| 19:22.39 | nirs | nothing |
| 19:22.42 | nirs | nada |
| 19:22.49 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 19:22.52 | junky[work] | ya make valgrind? |
| 19:23.16 | RoyK | junky[work]: not more than what's said in the docs |
| 19:23.42 | nirs | well roy, any idea ? |
| 19:23.48 | *** join/#asterisk DevilFish (~me@staff211.qtm.net) |
| 19:24.12 | DevilFish | anyone have any experiane mapping PolyCom hard keys to speeddial ? |
| 19:24.36 | DevilFish | not getting it to work and I'm sure I have a syntax problem in my ipmid.cfg |
| 19:26.06 | Essobi | ~mailing list |
| 19:26.13 | Essobi | ~mail |
| 19:26.15 | jbot | well, mail is on it's way :-) |
| 19:26.26 | Essobi | ~list |
| 19:26.26 | jbot | one warez list being sent |
| 19:26.29 | Essobi | :| |
| 19:26.37 | Essobi | FFS Jbot. |
| 19:26.40 | mikegrb | Essobi: asterisk.org has all the lists |
| 19:27.16 | tzafrir | nirs, use a wrapper script and add plenty of vvvs to the command-line to get a trace. |
| 19:27.30 | tzafrir | Or maybe run it under strace (and the hell with performance) |
| 19:27.56 | nirs | tzafrir, already did that |
| 19:28.04 | nirs | nothing out of the ordinary came up |
| 19:28.22 | nirs | it just looks as if asterisk simply shuts down, as if it received a -x flag |
| 19:29.03 | Essobi | Can you not search current mailing lists anymore? |
| 19:29.56 | vaewyn | man... These polycoms are fine to set up once you have the base XML stuff... but helping a friend set them up from scratch over the phone sucks |
| 19:32.56 | Sedorox | having a problem with it too? lol |
| 19:34.56 | Essobi | Anyone know how to make it ring the next member in the list if the first doesn't answer? round robin just sends them all the the first member. |
| 19:35.00 | Essobi | That's retarded. |
| 19:37.34 | dsmouse | Mr. Jones, |
| 19:37.34 | dsmouse | We can provide VoIP services starting at $24.95 per month. With a one time |
| 19:37.34 | dsmouse | activation fee of $34.99 which can be waived with a one ear commitment. |
| 19:37.43 | dsmouse | One ear seems very expencive. |
| 19:38.06 | mikegrb | well that still leaves you an ear to listen when talking on the phone |
| 19:38.49 | Essobi | haha |
| 19:39.54 | *** join/#asterisk mac_7 (~karsten@c167138.adsl.hansenet.de) |
| 19:40.11 | _-Jon-_ | lol |
| 19:40.45 | doughecka_ | lol |
| 19:41.30 | _-Jon-_ | Is anyone here using LiveVoip? |
| 19:42.28 | _-Jon-_ | Their 1.8c/min rate seems good for a US/Canada toll-free number but I'm wondering what the quality is like |
| 19:42.53 | mac_7 | chan_vpb.c: In function `void get_callerid(vpb_pvt*)': |
| 19:42.53 | mac_7 | chan_vpb.c:553: error: `vpb_cid_decode2' undeclared (first use this function) |
| 19:42.53 | mac_7 | chan_vpb.c:553: error: (Each undeclared identifier is reported only once for each function it appears in.) |
| 19:43.10 | mac_7 | need some help with this compile problem cvs |
| 19:43.59 | bjohnson | new bash srcipt available for providing a quick sum of usage from csv based CDR records: http://www.voip-info.org/tiki-index.php?page=Asterisk+CDR+csv+handling2 |
| 19:44.37 | bjohnson | Essobi: superdial macro on the wiki |
| 19:45.03 | bjohnson | Essobi: it's not a queue thing though .. just a sequential dialer with extra options |
| 19:46.24 | bjohnson | _-Jon-_: 1.8? I thought they were down to 1.2c |
| 19:46.49 | bjohnson | "Call USA & Canada Only 1.2 Cents a Minute" |
| 19:46.57 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 19:46.59 | bjohnson | ahh toll free |
| 19:47.06 | _-Jon-_ | bjohnson, yeah incoming toll-free |
| 19:47.49 | _-Jon-_ | Hmm maybe it is 1.2 |
| 19:48.02 | eKo1 | Hmm...for some reason, Asterisk isn't loading cdr_odbc.so. |
| 19:49.10 | bjohnson | _-Jon-_: where do you find info about the tolls working in Canada? |
| 19:49.36 | eKo1 | OK, it's not even in there. How can I make cdr_odbc.c? |
| 19:49.42 | Essobi | bjohnson Naw, I need queues. |
| 19:49.45 | bjohnson | bunch of us are looking for CDN toll free for < 5c CDN / minute (easy to get that rate from telcos) |
| 19:49.56 | eKo1 | Bear in mind, it is installed in a non-standard location. |
| 19:50.03 | Essobi | But it seems app_queue won't escelate up the priorities |
| 19:51.02 | _-Jon-_ | bjohnson, I'll let you in aon a little secret.. When I signed up with LiveVoip, I tested it and it works from Canada :P |
| 19:51.16 | eKo1 | Guess it's time to play with the Makefile.... |
| 19:52.06 | _-Jon-_ | And what's better is I didn't recognize the charge on my credit card by LiveVoip so I did a charge-back and they don't seem to interested in fixing it |
| 19:52.06 | bjohnson | _-Jon-_: $0.50 cents each - per Month? |
| 19:52.24 | _-Jon-_ | bjohnson, yeah the 50c/month # |
| 19:52.54 | Essobi | bjohnson I have a set of queue users, that the queue nmeeds to ring in a certain order. they also need to be able to dynamically login and log out. |
| 19:53.06 | Essobi | I can't seem to achieve both features at the same time. |
| 19:54.46 | Essobi | Which is rather silly since that's a basic ACD scheme. |
| 19:55.00 | bjohnson | Essobi: sorry, I don't use queues .. I just ring all phones |
| 19:56.13 | Sedorox | I'm using queues.. but to ring all |
| 19:56.58 | Essobi | I'm useing a ring all and a "Screw the J-man" priority as he puts it. Heh. There's an ordered preference of dialing extensions. |
| 19:58.28 | RoyK | øszjc mg89p |
| 19:58.37 | Essobi | I'm adding two members to the queue dynamically.. one with a penalty of 0 and one with a penalty of 1.. I use round robin, call in.. penalty 0 rings.. times out, penalty 0 rings, times out, penalty 0 ring, and etc.. never reaching panalty one. |
| 19:58.50 | HitTop | is there any SER user arround here? |
| 19:59.14 | mac_7 | I'm unable to find a declaration for vpb_cid_decode2 any hint |
| 19:59.14 | mac_7 | chan_vpb.c:553: error: `vpb_cid_decode2' undeclared (first use this function) |
| 19:59.55 | moonwick | so, ah... are the two status LEDs on the back of the T100P supposed to say anything at all about the status of the link? |
| 20:00.05 | moonwick | I've got a green LED, and there's nothing even plugged into it. |
| 20:02.43 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 20:07.46 | Godsey | are there any devices like the SPA-2000 that also support iax? |
| 20:08.16 | *** join/#asterisk clive- (~pirch@rrba-146-94-228.telkomadsl.co.za) |
| 20:08.37 | modulus_ | iaxy box yes |
| 20:08.41 | eKo1 | Godsey: Hah! That'll be the day. |
| 20:08.54 | dsmouse | Godsey: IAXy |
| 20:08.57 | eKo1 | The IAXy is only one-port. |
| 20:09.03 | dsmouse | oh |
| 20:09.08 | eKo1 | SPA-2000 is two-port. |
| 20:09.28 | modulus_ | isn't there one with 2 ports? |
| 20:09.44 | clive- | eko any advice on getting a SPA-2000 to work on sip through nat? |
| 20:09.45 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 20:11.35 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 20:11.43 | bjohnson | Godsey: the iaxy does not have a web interface |
| 20:11.51 | eKo1 | clive-: NAT on the same network or NAT on different networks? |
| 20:12.03 | bjohnson | Godsey: there are iax devices on ebay that are supposed to support iax and have a web interface |
| 20:12.36 | bjohnson | Godsey: no reports here yet about how good they are. Do you feel like testing and reporting? They are going for about $40 USD each |
| 20:12.40 | clive- | eko its like internet---nat----spa2000 |
| 20:12.48 | Godsey | sure |
| 20:12.55 | clive- | really $40 ? |
| 20:13.12 | eKo1 | clive-: you mean asterisk----internet----nat-----spa? |
| 20:13.26 | eKo1 | or asterisk----nat-----internet-----nat-----spa? |
| 20:13.38 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
| 20:13.44 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 20:14.22 | clive- | eko, im not using asterisk for this one...its jser |
| 20:18.28 | clive- | bjohnson, what are these iax units called? |
| 20:20.41 | bjohnson | searching ebay now .. don't see any right now |
| 20:20.51 | bjohnson | an you search for ended auctions? |
| 20:22.23 | Sedorox | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61839&item=5752456419&rd=1 |
| 20:22.26 | Sedorox | that what your looking for? |
| 20:23.02 | djin | http://www.eezeephone.com/ |
| 20:23.02 | Ayano | How easy are polycom to use with asterisk? |
| 20:23.11 | Sedorox | was looking at getting one of those... but haven't had the money.. |
| 20:24.10 | eKo1 | AIX2?! |
| 20:24.18 | Sedorox | mistype |
| 20:24.19 | Sedorox | dunno |
| 20:24.20 | Sedorox | lol |
| 20:24.24 | bjohnson | Sedorox: yeah that's it |
| 20:24.29 | eKo1 | Yeah right. |
| 20:24.32 | *** part/#asterisk pointer (pointer@aj.catt.com) |
| 20:24.38 | eKo1 | That shit doesn't support IAX. |
| 20:24.57 | Sedorox | lol |
| 20:25.04 | Sedorox | dunno |
| 20:25.10 | Sedorox | when I get the money I'll probably pick one up |
| 20:25.15 | Sedorox | and we'll see |
| 20:25.30 | Beirdo | wonder what shack in Taiwan that's made in? |
| 20:25.31 | bjohnson | I think it's the one here: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61839&item=5752454353&rd=1 |
| 20:25.33 | Ayano | eKo1: what doesn't? |
| 20:25.34 | eKo1 | That eezeephone looks interesting. Anybody have one. |
| 20:25.45 | clive- | eko I do |
| 20:25.47 | bjohnson | http://www.atcom.com.cn/engweb/product.html |
| 20:25.49 | *** join/#asterisk NoOS (~askme@cust.8.241.adsl.cistron.nl) |
| 20:25.54 | bjohnson | Beirdo: that one ^^ I think |
| 20:25.56 | clive- | its the pa186 phone |
| 20:25.57 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 20:26.02 | eKo1 | clive-: Do you have it running IAX? |
| 20:26.04 | NoOS | hi |
| 20:26.51 | clive- | eko, I have the sip running, but some versions of the firmware supposedly support iax |
| 20:27.22 | eKo1 | Well, if you get it working with IAX, please make a note of it somewhere on the wiki. |
| 20:27.36 | Sedorox | I don't think they support IAX2... |
| 20:27.40 | Godsey | dang got pulled away |
| 20:27.41 | Sedorox | they being the link I put |
| 20:28.01 | NoOS | Can any1 advise me on the hardware I need for asterisk and an isdn line? I need callwaiting.. |
| 20:28.05 | eKo1 | Dang it, * won't compile because of /usr/bin/ld: cannot find -lodbc |
| 20:28.09 | bjohnson | $89.99 includes shipping to USA and Canada !! for the eezeephone |
| 20:28.34 | bjohnson | Sedorox: the one I posted spelt IAX2 correctly |
| 20:28.36 | *** join/#asterisk basv (~bas@datarack.xs4all.nl) |
| 20:28.38 | Godsey | I was hoping for 2 fxs ports but that's ok :) |
| 20:28.41 | Godsey | and http configs |
| 20:28.52 | basv | hello everybody |
| 20:29.00 | bjohnson | if it's the one at atcom .. it lists http config |
| 20:29.04 | Sedorox | yea.. butg when you look on the site you sent.. the atcom.com.cn |
| 20:29.07 | Sedorox | Support H323 V4 ,MGCP,SIP |
| 20:29.17 | Sedorox | I was talking about the IAX, not http... |
| 20:29.20 | Ayano | How easy are polycom to use with asterisk? |
| 20:29.26 | *** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 20:29.31 | bjohnson | Sedorox: I'm just guessing that it's the same one .. looks the same |
| 20:29.35 | basv | have a small question about using 2 cisco 7960's with asterisk |
| 20:29.47 | Sedorox | yea.. |
| 20:29.49 | basv | phones are converted to sip and they register with asterisk |
| 20:29.58 | Ayano | okay |
| 20:29.59 | basv | dialtone on both phones |
| 20:30.00 | bjohnson | Beirdo: pm |
| 20:30.07 | Godsey | I need devices w/ 2 fxs ports |
| 20:30.18 | bjohnson | Godsey: buy 2 and report back |
| 20:30.24 | Godsey | ha :) |
| 20:30.24 | Sedorox | LOL |
| 20:30.27 | Sedorox | -caps |
| 20:30.33 | basv | I'd like to know how I can call the other phone, where's the best information on dialing plans? |
| 20:30.38 | bjohnson | Godsey: you're going to pay $35 per port for 2 port anyway |
| 20:30.38 | NoOS | Can any1 advise me on the hardware I need for asterisk and an isdn line? I need callwaiting.. |
| 20:30.45 | Godsey | it doesn't matter |
| 20:30.49 | bjohnson | Godsey: pay $40 per port and get 2 of them |
| 20:30.54 | Godsey | business requirment is 2 port per unit |
| 20:31.00 | Sedorox | hmm |
| 20:31.04 | Sedorox | 2line phone? |
| 20:31.48 | Godsey | probably not |
| 20:32.01 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk) |
| 20:32.25 | eKo1 | bjohnson: Eh, doesn't the Sipura phone have 2 FXS ports? |
| 20:32.40 | Sedorox | Godsey: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61839&item=5750225388&rd=1 |
| 20:32.43 | bjohnson | eKo1: the SPA 841? |
| 20:32.53 | eKo1 | Yeah. |
| 20:33.13 | bjohnson | I thought it just had one CAT 5 port |
| 20:33.19 | eKo1 | Never mind. |
| 20:33.25 | Sedorox | bjohnson: what do you think of the above link? |
| 20:33.58 | eKo1 | Well, it doesn't make sense to have an FXS port on a phone. |
| 20:33.58 | *** join/#asterisk TokyoJimu (~jimmy@shasta.nccom.com) |
| 20:34.21 | bjohnson | eKo1: it makes as much sense as having 2 fxs ports in a ATA |
| 20:34.31 | clive- | godsey there is a website called www.iaxtalk.com with iax devices |
| 20:35.24 | eKo1 | bjohnson: No, 2 FXS on an ATA is equivalent to 2 ethernet ports on a phone. |
| 20:35.31 | _-Jon-_ | Does anyone know of a simple way for me to use my phone to record greetings? Like dial an extention and it records a nice gsm file for me use? |
| 20:35.34 | TokyoJimu | I'm trying to dial out on a specific span & channel but when I use the syntax specified I get an error "unknown option '-' |
| 20:35.36 | TokyoJimu | Unable to create channel of type 'Zap' |
| 20:35.36 | bjohnson | Sedorox: don't know anything about that one .. and I love Sipura .. that looks similar to SPA 2001 which is $90 I belive |
| 20:35.46 | TokyoJimu | exten => 19164930010,1,Dial(Zap/7-1/18189950699) |
| 20:35.51 | Godsey | stupid |
| 20:35.52 | Sedorox | with two fxs's? |
| 20:35.54 | bjohnson | _-Jon-_: yes .. record() |
| 20:35.55 | TokyoJimu | Ignore that Zap error |
| 20:35.56 | Beirdo | 2 FXS on an ATA makes perfect sense. |
| 20:36.01 | Godsey | they only ship to ebay confirmed addresses |
| 20:36.03 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || chan_zap in 1.0.4 had a bug, so 1.0.5 has been released || Dev Conf 2PM CST FEB 17th -> IAX2/guest@66.250.68.194/996 |
| 20:36.15 | Sedorox | Godsey: most times if you email them.. its alright |
| 20:36.19 | _-Jon-_ | bjohnson, ah that works :P |
| 20:36.34 | Godsey | sedorox: I can't finish the buy now page :) |
| 20:36.42 | TokyoJimu | Let's try again: I use this: |
| 20:36.42 | TokyoJimu | exten => 19164930010,1,Dial(Zap/7-1/18189950699) |
| 20:36.56 | TokyoJimu | and get error "unkown option '-'" |
| 20:37.10 | bjohnson | what is 7-1? |
| 20:37.15 | TokyoJimu | span-channel |
| 20:37.28 | Sedorox | hmmm :-/ |
| 20:37.30 | bjohnson | the rest looks ok |
| 20:37.33 | TokyoJimu | according to http://www.voip-info.org/tiki-index.php?page=Asterisk+ZAP+channels |
| 20:37.37 | Godsey | I'll have someone at work buy it |
| 20:37.59 | Godsey | I will never tell paypal about my checking account :) |
| 20:38.01 | TokyoJimu | Oh, maybe that's a newer option and this is an older release... |
| 20:38.24 | Sedorox | Godsey: I opened a second checking account just for paypal... keep everyting seperate |
| 20:38.31 | hardwire | grr |
| 20:38.38 | hardwire | linux based SIP cluient that works |
| 20:38.40 | hardwire | and is not kphone |
| 20:38.42 | hardwire | anybody? |
| 20:38.43 | hardwire | err |
| 20:38.48 | hardwire | not Asterisk either :) |
| 20:38.50 | Mw3 | linphone |
| 20:38.57 | hardwire | apt-get install linphone |
| 20:38.57 | hardwire | yay |
| 20:38.59 | hardwire | no codecs |
| 20:38.59 | FuRR_ | does asterisk support Distinctive ring, and how would i setup distinctive ring to ring certain extensions |
| 20:39.55 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 20:40.09 | bjohnson | Beirdo: pm? |
| 20:40.26 | Beirdo | yeah, it's 3:40PM :) |
| 20:40.33 | Beirdo | go ahead... |
| 20:42.41 | slePP | http://pastebin.ca/news.php |
| 20:42.44 | slePP | anyone have any suggestions? |
| 20:44.03 | __Sparks_ | I have a problem with Asterisk and Sipgate - If I call my sipgate PSTN number, i get a call on my sip phone, but audio drom the PSTN to Astersk is not working? |
| 20:47.47 | netsurfer | __Sparks_ - sipgate.de or .co.uk ? |
| 20:47.56 | *** join/#asterisk jaiger (~jaiger@fire.innovationsw.com) |
| 20:48.15 | *** join/#asterisk Weezey (Weezey@lan6.LO.iasl.com) |
| 20:49.14 | *** join/#asterisk clint_ (~clint@snap.helixsystems.com) |
| 20:49.42 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 20:50.24 | __Sparks_ | co.uk |
| 20:50.35 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 20:52.17 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 20:52.19 | ManxPower | ~docs |
| 20:52.20 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:52.37 | ManxPower | OK guys, I have an unusual issue. |
| 20:52.59 | *** join/#asterisk darby_t (~tom@dmx42.neoplus.adsl.tpnet.pl) |
| 20:53.14 | ManxPower | I have the following setup (don't complain that it sucks, just help) CLEC -> T-1 -> Asterisk -> Channel Bank -> (analog) PBX |
| 20:53.25 | TokyoJimu | gg:q |
| 20:53.45 | brettnem | ok? |
| 20:53.49 | moonwick | heh, ouch |
| 20:53.59 | ManxPower | The Channel Bank is an Adtram. The CLEC manages the channel bank over the T-1. How can I make sure they can still manage the channel bank when we put Asteirsk between the CLEC and their channel bank. |
| 20:54.18 | brettnem | are they using FDL? |
| 20:54.31 | *** join/#asterisk FryGuy- (fryguy@c-67-174-57-164.client.comcast.net) |
| 20:54.34 | ManxPower | BTW, we are using the Zaptel DACS to cross connect the channels between the T-1 and the Channel bank that Asterisk doesn't care about. |
| 20:54.45 | ManxPower | brettnem, no clue. What is FDL? |
| 20:54.48 | brettnem | I don't think you can.. |
| 20:55.14 | brettnem | Facilities Data Link.. it's a management channel in the overhead of the T1 |
| 20:55.47 | brettnem | http://groups-beta.google.com/group/comp.dcom.telecom.tech/browse_thread/thread/79fbe448fe23a77a/394b435a4617e1a3?q=channel+bank+fdl&_done=%2Fgroups%3Fq%3Dchannel+bank+fdl%26hl%3Den%26lr%3D%26client%3Dfirefox-a%26rls%3Dorg.mozilla:en-US:official%26sa%3DN%26tab%3Dwg%26&_doneTitle=Back+to+Search&&d#394b435a4617e1a3 |
| 20:55.49 | brettnem | woah |
| 20:56.17 | clint_ | In the following setup: PSTN -(PRI)> Asterisk -> -(T1)> channel bank -> analog phone, when a number on the PSTN is busy, I receive a 'reorder' tone as opposed to a busy tone (fast busy.) Is this normal behavior whenever asterisk encounters a busy line? Is there any differentiation between busy destination and busy facilities in between (which is when a reorder should be presented?) |
| 20:56.34 | brettnem | <PROTECTED> |
| 20:56.38 | `Sauron | lo, brett |
| 20:56.39 | brettnem | oops |
| 20:57.08 | brettnem | hey there `Sauron |
| 20:57.15 | netsurfer | __Sparks_ - iv noticed some strange problems with them over the last few weeks - for instance 3 nights ago all their numbers were ringing out 'engaged' |
| 20:57.20 | *** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au) |
| 20:57.22 | brettnem | ManxPower: from that link " If you are muxing the T1 up to the SONET level or any other medium, the FDL must be carried through. The exception to this rule is if the T1 terminates into a switch that will break it down into its component DS0's. The FDL in this case would terminate at the switch." |
| 20:57.55 | *** join/#asterisk clive-- (~pirch@rrba-146-94-228.telkomadsl.co.za) |
| 20:58.07 | `Sauron | brett: Why would you have a network that wasn't 100% ip? ;) |
| 20:58.26 | brettnem | what do you mean? are you refering to FDL? |
| 20:58.32 | `Sauron | pointone |
| 20:58.39 | Nugget | `Sauron uses NetBEUI ;) |
| 20:58.39 | brettnem | heh |
| 20:58.39 | ManxPower | brettnem, Yes, but WHERE in the T-1 is the FDL data carried? |
| 20:58.47 | `Sauron | "Out network" says "100% IP NETWORK" |
| 20:58.53 | `Sauron | I'm like, no really?!?!!??? |
| 20:58.54 | `Sauron | :) |
| 20:59.00 | Essobi | Pssh. |
| 20:59.09 | brettnem | ManxPower: I'm not sure.. try to find T1.403 |
| 20:59.15 | Essobi | Queue penalties look broke as all hell. :\ |
| 20:59.25 | ManxPower | brettnem, If I know where it's carried I may be able to transport it. |
| 20:59.40 | brettnem | `Sauron: I'm not pointone.. but we are family |
| 20:59.46 | `Sauron | Hum, ah. |
| 20:59.56 | `Sauron | Just noticed that'd who your IP belongs to. |
| 21:01.14 | brettnem | ManxPower: right.. and FDL tunneling would be an interesting addition |
| 21:01.14 | `Sauron | dum di dum |
| 21:01.15 | brettnem | `Sauron: yep.. :) |
| 21:01.15 | `Sauron | waiting for this ebay auction to end |
| 21:01.15 | brettnem | `Sauron: indeed, I do have a pointone IP address |
| 21:01.15 | brettnem | But to answer your question. their point is that it's a 100% IP network that is 100% voice traffic. |
| 21:01.16 | brettnem | heh, except for this IRC session aparently.. ;) |
| 21:01.21 | `Sauron | Eh, I figured it was all marketing speak anyway |
| 21:01.32 | `Sauron | I just had to laugh at it |
| 21:01.38 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 21:01.43 | ManxPower | damn internet |
| 21:01.48 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 21:01.48 | brettnem | `Sauron: actually, I think it's a good point for a voice network.. |
| 21:02.26 | vaewyn | Anyone wondering about the WIP-5000s abilities can check out: http://wwwrogue.com/voip/WIP5000.html end of the webpage has a full map of the menu system in it (which answers most of the capabilities questions) |
| 21:02.35 | Nugget | yay vaewyn. |
| 21:02.47 | Nugget | trade you a zyxel for it. :) |
| 21:02.58 | vaewyn | nope |
| 21:03.00 | `Sauron | Nugget: Don't like the zyxel? |
| 21:03.01 | vaewyn | :} |
| 21:03.08 | `Sauron | I saw you had pics on it on slacker |
| 21:03.12 | Nugget | the zyxel is possibly the worst piece of equipment I've ever owned. |
| 21:03.21 | Nugget | it's tremendously awful. |
| 21:03.41 | brettnem | so the WIP5000 is actually decent?? |
| 21:03.46 | moonwick | I think the only reason nugget doesn't get rid of his is it makes everything else he owns look good by comparison. :) |
| 21:03.54 | Nugget | haha. yes, exactly. |
| 21:04.13 | Sedorox | what is it? |
| 21:04.20 | clint_ | ManxPower: According to my pet telephone switch tech, FDL robs a few bits per second from the last ds0 for its purposes. |
| 21:04.25 | vaewyn | brettnem: so far it kicks ass |
| 21:04.39 | `Sauron | vaewyn: I want pics. :) |
| 21:04.50 | FuRR_ | does * support modem detection? |
| 21:04.54 | vaewyn | `Sauron: will be taking them tonight... cameras at home :} |
| 21:04.58 | FuRR_ | i know it does fax detection, but can it do modem detection |
| 21:05.06 | `Sauron | hum |
| 21:05.07 | `Sauron | I got paid |
| 21:05.08 | Nugget | modems don't send announce tones like faxes. how would you detect one? |
| 21:05.12 | `Sauron | I should spend some money |
| 21:05.14 | `Sauron | dum di dum |
| 21:05.14 | vaewyn | It does have pretty blue LEDs though :P |
| 21:05.21 | FuRR_ | Nugget: uhm, yes they do |
| 21:05.33 | `Sauron | www.gumstix.com |
| 21:05.51 | `Sauron | Gunna get their connex board, with the CF daughterboard |
| 21:06.05 | Nugget | FuRR_: if you think so, more power to you. Good luck with that. |
| 21:06.11 | brettnem | woah.. robing bits for FDL?? I don't think so.. that doesn't sound right |
| 21:06.33 | *** join/#asterisk BoRiS (~boris@S0106006097b94339.wp.shawcable.net) |
| 21:06.37 | Sedorox | Godsey: also... |
| 21:06.39 | Sedorox | http://www.voipsupply.com/product_info.php?products_id=252 |
| 21:06.41 | Sedorox | two port FXS... |
| 21:06.46 | Sedorox | if your still around |
| 21:06.48 | FuRR_ | Nugget: call yourself with a modem, as soon as the channel picks up, the calling party starts talking...i think |
| 21:06.56 | Nugget | you think. I see. |
| 21:07.29 | Nugget | am I supposed to say "uhm" here to indicate that I think you're a moron? |
| 21:07.32 | Nugget | I think that's the protocol. |
| 21:07.51 | `Sauron | * can detect modems |
| 21:08.04 | `Sauron | but I think it does it by quirk, rahter than by any smart way... |
| 21:08.07 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 21:08.13 | `Sauron | s/modems/fax |
| 21:08.25 | Nugget | yes, you can detect a fax because calling faxes send announce tones. |
| 21:08.28 | Nugget | modems do not. |
| 21:08.32 | `Sauron | nigget++ |
| 21:08.35 | clint_ | Anyone have any ideas on that busy tone cadence thing? |
| 21:08.39 | `Sauron | s/i/u |
| 21:08.44 | `Sauron | Nigget :) |
| 21:08.48 | PBXtech | is the outbound system not recognizing file dates in the future to schedule a call? |
| 21:09.23 | *** join/#asterisk ctooley (~chatzilla@206-81-243-139.client.cypresscom.net) |
| 21:09.27 | *** join/#asterisk BoRiS (~boris@S0106006097b94339.wp.shawcable.net) |
| 21:09.36 | ctooley | how do I strip the 1 off the beginning of ${EXTEN}? |
| 21:09.51 | PBXtech | ${EXTEN:1} |
| 21:09.54 | ctooley | I thought I knew how ot do that, but apparently I'm wrong. |
| 21:09.57 | ctooley | Oh crud |
| 21:10.45 | netsurfer | anyone recognise this: NOTICE[3850]: app_dial.c:911 dial_exec_full: Unable to create channel of type 'Zap' (cause 0) |
| 21:10.48 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 21:11.32 | ManxPower | netsurfer, Yes, it means Asterisk doesn't see any zaptel cards/drivers/configs. |
| 21:11.54 | netsurfer | oh.. not again :( |
| 21:12.09 | ManxPower | Or you are doing something like Dial(Zap/999/1234) when there is no Zap channel 999 |
| 21:12.17 | ManxPower | netsurfer, forgot to tell the server to load zaptel on start? |
| 21:12.59 | moonwick | hm, I'm tempted to get an SPA-841 |
| 21:13.10 | ManxPower | moonwick, I have two of them |
| 21:13.12 | moonwick | I wonder if that'd satisfy my urge to buy a cisco phone. :P |
| 21:13.15 | moonwick | how are they? |
| 21:13.28 | brettnem | the cisco's are nice, but not really anything special |
| 21:13.33 | brettnem | it's a good phone |
| 21:13.46 | ManxPower | moonwick, Not bad. There's an issue with microphone/speakerphone gain that I hope will be addressed in fiture firmware version. |
| 21:14.05 | moonwick | I've been happy as a clam with my SPA-2000, so if the 841 has the same level of quality, I suspect I'd be just as satisfied |
| 21:14.10 | ManxPower | The phone and handset are a little on the light side, the display is not backlit |
| 21:14.18 | moonwick | ah |
| 21:14.19 | ManxPower | But it's the best IP phone on the market for under $100 |
| 21:14.34 | netsurfer | ManxPower - im modprobing wcfxo then ztcfg then starting * |
| 21:15.00 | ManxPower | netsurfer, wcfxo? Make sure you have no IRQ conflicts and try moving the card to another slot. |
| 21:15.22 | ManxPower | Also, I've seen that if you stop and start asterisk without unload/load of zaptel I have seen issues. |
| 21:15.46 | *** join/#asterisk pryk (~tmalkut@fire2.orasoft.net.pl) |
| 21:16.21 | netsurfer | ManxPower - its using its own IRQ - worked fine earlier today |
| 21:16.48 | ManxPower | netsurfer, I've seen the card sometimes not recognized on a warmboot, but is fine after a coldboot. |
| 21:16.56 | netsurfer | ManxPower - when I try unloading zaptel I get device/resource is busy |
| 21:17.02 | netsurfer | ok |
| 21:17.56 | ManxPower | netsurfer, then you have ztmonitor or asterisk or zttool still running. |
| 21:22.18 | netsurfer | no, asterisk isnt running.. and I cant find any trace of zttool |
| 21:22.32 | hardwire | have you tried installing it? |
| 21:22.40 | *** join/#asterisk Connor- (~billy@198-144-174-5.knx.tn.nxs.net) |
| 21:22.56 | Connor- | Damnit.. Having problems upgrading a 7960G from SCCP v5 to Sip 7.2 |
| 21:23.10 | Connor- | Damn thing isn't requesting the SEPDefault.cnf file.. |
| 21:25.43 | ManxPower | Have you tried upgrading to older SIP loads? The SIP firmware README talks about those issues. |
| 21:25.45 | PBXtech | is the outbound system not recognizing file dates in the future to schedule a call? |
| 21:25.53 | *** join/#asterisk jterrero (~jterrero@mcse-irc.isys-networks.com) |
| 21:26.31 | ManxPower | PBXtech, It always did for me. |
| 21:26.45 | Connor- | No. But, I'm whatching it via tcpdump, it's only requesting the OS79XX.TXT file and a .cnf.xml file.. |
| 21:26.59 | Connor- | normally they request the SEPDefault.cnf file.. this one isn't |
| 21:27.10 | PBXtech | like this right -> touch -d 200502172000 test.call |
| 21:27.25 | schwagner | Connor-: isn't it SIPDefault.cnf? |
| 21:27.31 | Connor- | only with SIP |
| 21:27.35 | jterrero | anyone know anything about sendmail trying to relay to itself? its trying to send all email to 127.0.0.1 |
| 21:27.59 | PBXtech | join #sendmail and ask :) |
| 21:28.18 | Beirdo | that's likley the submit program punting to the daemon |
| 21:28.23 | ManxPower | PBXtech, http://pastebin.ca/5963 |
| 21:28.52 | PBXtech | thank ManxPower |
| 21:28.53 | bkw_ | Please someone I need about 100 people to fax me a few pages to 8666799920 |
| 21:29.00 | bkw_ | please please please |
| 21:29.05 | bkw_ | :P |
| 21:29.24 | Beirdo | who pissed you off this time? |
| 21:29.46 | bjohnson | the FBI? |
| 21:29.47 | NoOS | Do I need a wildcard TE110P with my isdn line? |
| 21:30.09 | schwagner | Connor-: this a brand new phone? |
| 21:30.10 | ManxPower | NoOS, Is it an ISDN PRI or and ISDN BRI? |
| 21:30.55 | Beirdo | nice answer |
| 21:31.06 | moonwick | one of you guys answer the phone :) |
| 21:31.59 | *** join/#asterisk Xander77 (~alex@exten-halls-243.soton.ac.uk) |
| 21:32.27 | *** join/#asterisk NoOS (~askme@cust.8.241.adsl.cistron.nl) |
| 21:32.35 | NoOS | Do I need a wildcard TE110P with my isdn line? |
| 21:32.47 | ManxPower | NoOS, Is it an ISDN PRI or and ISDN BRI? |
| 21:32.48 | *** part/#asterisk mac_7 (~karsten@c167138.adsl.hansenet.de) |
| 21:33.00 | ManxPower | I'm not going to ask a 3rd time. |
| 21:34.03 | wasim | NoOS: you need 24 of them actually |
| 21:34.09 | bjohnson | next time we all fax to noos |
| 21:37.14 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 21:37.14 | *** topic/#asterisk is Asterisk: The Open Source PBX || chan_zap in 1.0.4 had a bug, so 1.0.5 has been released || Dev Conf 2PM CST FEB 17th -> IAX2/guest@66.250.68.194/996 |
| 21:37.16 | NoOS | probably I have BRI (cheapest solution) |
| 21:37.32 | ManxPower | NoOS, Then Digium does not make a card compatable with your ISDN |
| 21:37.33 | wasim | NoOS: you can use an CAPI card, or the quad-bri |
| 21:38.16 | greendisease | bkw_: can i talk to you in msg |
| 21:39.06 | [Latre] | ManxPower: the iaxy is register....and works.....but what happend if my user go out, what happend with the diaplan because in the dial plan i put the IP of this iaxy.... |
| 21:39.18 | terrapen | GREEN SNAKES ON THE CEILING |
| 21:39.29 | ManxPower | [Latre], then why are you dialing by IP address? |
| 21:39.38 | greendisease | bkw_: well we need a conference room for the whole day |
| 21:39.44 | ManxPower | Dial(IAX2/iaxconfentryforiaxy) |
| 21:39.57 | greendisease | some developers from europe want to be involved in devel discussion |
| 21:40.03 | terrapen | how well does the IAXy work? |
| 21:40.09 | *** join/#asterisk r0d3nt|m (RatMan@65.60.102.18) |
| 21:40.15 | terrapen | i can't decide between one or a cheap cisco phone |
| 21:40.22 | greendisease | we have a * server but idg failed to get us an outisde ip |
| 21:40.32 | [Latre] | ManxPower: nop....i dialing for extension....but my dialplan is: exten => 600,1,Dial(IAX2/userx@192.168.1.151/s) where 192.168.1.151 is a ip of my iaxy |
| 21:41.04 | ManxPower | [Latre], You cannot do that if your IAXy changes IP addresses. |
| 21:41.18 | ManxPower | [Latre], either listen to my advice or not, I don't especially care. |
| 21:41.23 | *** join/#asterisk r0d3nt|m (nthwlx@perverseengineering.org) |
| 21:42.02 | [Latre] | :~| |
| 21:42.27 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:42.59 | *** part/#asterisk darby_t (~tom@dmx42.neoplus.adsl.tpnet.pl) |
| 21:43.06 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 21:43.22 | Xander77 | Ive got an asterisk box behind nat and its trying to register with an internet sip provider. It does seem to register but doesnt actually work. Im noticing the 'Contact' header asterisk is sending is the local ip, which is useless to the provider. Ive set nat=yes in sip.conf. any ideas? |
| 21:43.55 | RaYmAn-Bx | externalip and localnet settings in sip.conf (check the wiki) |
| 21:44.04 | Xander77 | yeh they're set too |
| 21:44.35 | RaYmAn-Bx | sorry, externip of course |
| 21:44.56 | [Latre] | ManxPower: if a quit the ip of iaxy in my dialplan, not ring....inmediatly sendme to voicemail |
| 21:45.10 | RaYmAn-Bx | then I have no idea..with those lettings my * sends the correct ip |
| 21:45.21 | tessier_ | RaYmAn-Bx: What version of * ? |
| 21:45.28 | tessier_ | Xander77: What version of * ? |
| 21:45.58 | Xander77 | Asterisk 1.0.3, |
| 21:46.09 | Sedorox | [Latre]: use Dial(IAX2/user@iaxytel/s) |
| 21:46.19 | Sedorox | where iaxytel is the userid you have in iax.conf |
| 21:46.40 | Sedorox | this way it changes when the iaxy registers with the box.. send it to whatever is currently registered |
| 21:46.46 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Dev Conf 2PM CST FEB 17th -> IAX2/guest@66.250.68.194/996 |
| 21:48.27 | *** join/#asterisk wankel (nobody@ohno.mrbill.net) |
| 21:48.45 | Xander77 | oh god how embarresing. 2 typos. Its annoying how id been looking at it for ages and its not still some says check this you notice |
| 21:48.51 | [Latre] | Sedorox: exten => 203,1,Dial(IAX2/usuario3@iaxtel/s) this one? |
| 21:48.55 | Xander77 | till |
| 21:49.08 | [Latre] | Sedorox: if i do that inmediatly send me to voicemail |
| 21:49.15 | Sedorox | is you have [iaxtel] in your iax.conf |
| 21:49.17 | Sedorox | if* |
| 21:49.39 | Sedorox | you don't need the /s on it.. I don't think.. |
| 21:50.31 | [Latre] | a ok...works |
| 21:50.39 | [Latre] | :) |
| 21:51.00 | Sedorox | :) |
| 21:52.31 | [Latre] | Sedorox: you has configured TDM04B |
| 21:53.05 | Sedorox | nope |
| 21:56.13 | `Sauron | buying stuff on ebay is always fun |
| 21:56.26 | `Sauron | gotta bid just at the right time |
| 21:56.28 | [Latre] | i was configure TDM04B and i can comunicate with Panasonic TD1232.....only have a problem with dialplan......if i make a call to extension X(this is connected to TDM port 2)giveme 3 seconds fot dial other extension in asterisk and i can talk with any extension in asterks from my local analog phones! |
| 21:57.04 | [Latre] | but, viceversa i cant do it |
| 21:57.13 | Sedorox | `Sauron: ebay is addicting.... |
| 21:57.30 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 21:57.46 | Sedorox | I would have money to spend on voip stuff if I didn't find ebay 3 years ago... |
| 21:57.57 | ManxPower | [Latre], Then your IAXy is NOT registering |
| 21:58.23 | Sedorox | ManxPower: little late :-p |
| 21:58.43 | ManxPower | [Latre], "iax2 show peers" should show the current IP address of the IAXy |
| 21:59.12 | *** join/#asterisk tzafrir_home (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 21:59.23 | Sedorox | hmmmm |
| 21:59.48 | Essobi | ManxPower You familiar with app_queue? |
| 21:59.54 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 22:00.05 | ManxPower | Essobi, no! |
| 22:00.09 | Essobi | Damn. |
| 22:00.13 | shmaltz | hi ManxPower |
| 22:00.29 | Essobi | It's dynamic member functionality is retarded. |
| 22:00.47 | shmaltz | ManxPower, do you still have the weather script? |
| 22:00.48 | Sedorox | agent call back? |
| 22:00.54 | Essobi | Excuse me.. Dynamic member with penalties. |
| 22:00.57 | Essobi | Sedorox Yes. |
| 22:00.58 | Sedorox | heh |
| 22:01.07 | Sedorox | I have callback.. but I don't use the penalties.. |
| 22:01.15 | Essobi | Yea? |
| 22:01.26 | Essobi | You have a predefined order or just use ring all? |
| 22:01.37 | *** join/#asterisk afrosheen (afrosheen@txprotoa8.august.net) |
| 22:01.38 | Sedorox | just ringall... don't have a need for any order yet... |
| 22:01.43 | Essobi | I do. |
| 22:01.57 | Sedorox | yea.. I saw stuff in the queues.conf for it.. but didn't really read it fully |
| 22:02.00 | Essobi | and penalties don't kick in unless an agent is logged out, which is kinda dumb. |
| 22:02.01 | afrosheen | hey, has anyone seen or used one of the new Soyo VOIP phones? |
| 22:02.19 | [Latre] | ManxPower: i put userx@iaxtel and works |
| 22:02.42 | ManxPower | I guess [iaxtel] = iaxconfentryfortheiaxy |
| 22:03.08 | jesster | Having problems outbound faxing from cisco ata 186 through PSTN, asterisk says unknown codec 100 received, got any ideas? |
| 22:03.38 | ManxPower | jesster, most people have problems faxing over voice over ip. |
| 22:03.42 | [Latre] | ManxPower: yes...i dont see before |
| 22:03.45 | johnnyb | What does it mean when "n" is used as a priority in extensions.conf? |
| 22:03.47 | jaiger | jesster, I hear bad things about faxing over VOIP |
| 22:03.57 | jesster | ManxPower: i thought as long as a Zap interface was used, it'd work? |
| 22:03.57 | ManxPower | ~google "unknown codec 100" |
| 22:04.07 | jaiger | all my fax machines are on a T1 interface through my channel bank |
| 22:04.08 | ManxPower | ugh! |
| 22:04.16 | ManxPower | ~google site:lists.digium.com "unknown codec 100" |
| 22:04.32 | ManxPower | ~google site:lists.digium.com unknown codec 100 |
| 22:04.32 | dsmouse | it is at? |
| 22:04.48 | jesster | 2003..how recent |
| 22:04.58 | Essobi | My choises are either hack app_queue to working with dynamically added agents and penalities, or get AgentLogin to allow redirecting to nonlocal extensions. :| |
| 22:05.07 | *** join/#asterisk Nethab (~Anon5416@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 22:05.11 | Essobi | Either way.. that's going to suck. |
| 22:05.18 | ManxPower | Essobi, redirect to Local/blah extensions |
| 22:05.23 | Essobi | I tried that. |
| 22:05.36 | ManxPower | Essobi, Now you know why I don't currently use queues. |
| 22:05.59 | afrosheen | hey, has anyone seen or used one of the new Soyo VOIP phones? |
| 22:06.10 | afrosheen | walmart.com is selling them cheap |
| 22:06.14 | Essobi | AgentCallbackLogin attempts to verify that an extension is a registered phone. |
| 22:06.22 | dsmouse | Essobi: non-local extentions isn't hard |
| 22:06.22 | ManxPower | Soyo? Is that like Vegan VoIP? |
| 22:06.28 | jesster | So those that need outbound faxing are using Channel banks -> T1 -> Digium card? |
| 22:06.37 | terrapen | walmart is selling voip phones? |
| 22:06.38 | terrapen | wtf? |
| 22:06.40 | tzanger | jesster: TDM430P is working ofr me |
| 22:06.45 | dsmouse | Essobi: I defined a extention to call a non-local number and can log in with it |
| 22:06.47 | tzanger | just not for receiving faxes to a real fax machien through it |
| 22:06.48 | Essobi | dsmouse With AgentCallbackLogin? |
| 22:06.53 | dsmouse | Essobi: yea |
| 22:06.56 | Essobi | How? |
| 22:07.02 | ManxPower | jesster, All Astreisk related faxing we do is ALWAYS TDM, no VoIP. |
| 22:07.18 | afrosheen | terrapen: yep :) they're $100 or so |
| 22:07.20 | Essobi | Tehe. |
| 22:07.22 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 22:07.37 | Essobi | dsmouse Using agents.conf? |
| 22:07.44 | *** join/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net) |
| 22:07.48 | terrapen | really? i might just have to pick one up |
| 22:07.49 | Sedorox | I got non-local working too.. but anyway... |
| 22:07.50 | jaiger | jesster, it's probably not a requirement but it works well for me |
| 22:07.52 | jesster | ManxPower: is this by design? |
| 22:07.56 | kuj | howdy |
| 22:08.18 | dsmouse | essobi I have agent => 6000,abcd,Phillip as a line in agent.conf |
| 22:08.24 | ManxPower | jesster, Yes. |
| 22:08.40 | ManxPower | jesster, Fax over Voice over IP is just not stable enough for us. |
| 22:08.45 | ManxPower | And it really never will be. |
| 22:08.46 | Essobi | dsmouse Okay. |
| 22:09.03 | jesster | ManxPower: ok. wierd. I've seen other outfits saying they have VOIP fax - i thought it'd be more common/stable.. |
| 22:09.14 | Sedorox | bbl |
| 22:09.22 | terrapen | goddammit! |
| 22:09.32 | dsmouse | Essobi: and when I log in it askes for an extention, I use 4555, which I have included in [default] |
| 22:09.33 | terrapen | i spilled sticky starbucks coffee all over my nice new apple keyboard! |
| 22:09.34 | ManxPower | jesster, Yes, some people use it. I dom't know what they sacraficed to The Gods, but I'll stick to TDM fax, thankyouverymuch. |
| 22:09.34 | terrapen | fuck! |
| 22:10.01 | Nethab | working? on *? |
| 22:10.28 | Essobi | dsmouse And what.. you have 4555 routed off *? |
| 22:10.43 | terrapen | afrosheen, you know why buying the sayo phone is a sure bet? |
| 22:10.50 | afrosheen | terrapen: no, why |
| 22:10.54 | terrapen | because you can take anything back at wal-mart |
| 22:10.57 | Essobi | haha |
| 22:10.59 | Essobi | he's right |
| 22:10.59 | dsmouse | I have that routed as 4001,1,Macro(dialout,91234567) |
| 22:11.00 | Essobi | you can |
| 22:11.05 | afrosheen | terrapen: this is walmart online ONLY |
| 22:11.07 | Essobi | Ahhh. |
| 22:11.07 | [Latre] | i was looking for a dialplan in digium and voip for make call from xlite to zap channels of TDM04B, some knows where i found that? |
| 22:11.14 | _-Jon-_ | hey why would this not work: Record(/tmp/asterisk-record/:gsm)? |
| 22:11.22 | terrapen | "you can take diapers back three years later and go, 'Hey, these diapers already got shit in em!'" |
| 22:11.24 | afrosheen | terrapen: but soyo is selling these all over the place |
| 22:11.33 | *** join/#asterisk SuperMMan (~graphic@d209-89-191-155.abhsia.telus.net) |
| 22:11.38 | terrapen | "We're real sorry about that, sir. Run back and get ya another pack!" |
| 22:11.41 | dsmouse | Essobi: which basicly just figures out what the Dail() whould be |
| 22:12.02 | netsurfer | ManxPower - that error we spoke about.. is appearing on SIP channels also |
| 22:12.09 | afrosheen | http://64.233.187.104/search?q=cache:K0RZElVmghkJ:www.bizrate.com/buy/products__cat_id--11510903,keyword--Ip%2520Phone.html+polycom+soundpoint+500+power+supply&hl=en |
| 22:12.16 | afrosheen | ack what |
| 22:12.26 | afrosheen | hang on, I'm retarded today |
| 22:12.43 | afrosheen | http://www.walmart.com/catalog/product.gsp?dest=9999999997&product_id=3429311&sourceid=1500000000000002028790 |
| 22:12.45 | afrosheen | that's it |
| 22:13.34 | dsmouse | Essobi: the only down side is logging out is a bit odd; basicly, you have to enable "autologoff" |
| 22:13.44 | dsmouse | I wish there was a AgentLogout() |
| 22:15.31 | *** join/#asterisk darkskiez (~mhb@host-84-9-91-127.bulldogdsl.com) |
| 22:16.12 | *** join/#asterisk jarrod (jarrod@dipole.informationwave.net) |
| 22:16.34 | jarrod | hey what is the most popular prepaid calling card software to use with asterisk |
| 22:18.34 | terrapen | jarrod, write your own |
| 22:19.01 | jarrod | rather than re-inventing the wheel, i was wondering if there was already an application available i could build upon |
| 22:19.14 | jarrod | i would definitely update to meet all of my needs |
| 22:19.57 | bkw_ | _-Jon-_, its filename.gsm not :gsm |
| 22:20.00 | bkw_ | the dot now |
| 22:20.03 | doughecka_ | bkw_: hail |
| 22:20.03 | bkw_ | with CVS head |
| 22:20.12 | bkw_ | doughecka_, my man.. what did ya need? |
| 22:20.15 | file[laptop] | BEAUTIFUL CHILD! |
| 22:20.19 | doughecka_ | I found it :) |
| 22:20.23 | doughecka_ | but gots a question |
| 22:20.32 | bkw_ | ok ask |
| 22:20.35 | redder86 | do Grandstreams not support attended transfer? |
| 22:20.49 | doughecka_ | when a 7940 fires up, is it usally blank with a green light on the front? |
| 22:20.59 | doughecka_ | for a few seconds? |
| 22:21.09 | file[laptop] | brrrrrr cold |
| 22:21.11 | jaiger | redder86, I don't think so |
| 22:21.25 | redder86 | jaiger: that's what I seem to see |
| 22:21.30 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
| 22:21.37 | redder86 | jaiger: looks like I'll need to set up parking |
| 22:23.07 | redder86 | jaiger: ah, but... http://lists.digium.com/pipermail/asterisk-users/2004-November/074362.html seems to indicate that they do, just a bit convoluted |
| 22:23.49 | doughecka_ | or does it immedietly show soemthing? |
| 22:24.00 | doughecka_ | and whats the voltage range that the phone supports... |
| 22:24.01 | doughecka_ | ;) |
| 22:24.10 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 22:24.52 | harryvv | 90 volts |
| 22:25.19 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 22:25.35 | modulus_ | jbot OLED? |
| 22:25.38 | doughecka_ | harryvv: really? |
| 22:25.50 | modulus_ | jbot OLED is Organic Light Emitting Diode |
| 22:25.51 | jbot | okay, modulus_ |
| 22:27.04 | doughecka_ | harryvv: so it would accept 60 volts ok? |
| 22:27.23 | harryvv | I dont think so what country are you in doug? |
| 22:27.32 | doughecka_ | US |
| 22:27.56 | harryvv | nope its 90 volts then |
| 22:28.11 | doughecka_ | cause the phone says 48V |
| 22:28.17 | doughecka_ | so I assume thats the minimum |
| 22:28.19 | ManxPower | most phones will work at much lower voltages. |
| 22:28.26 | doughecka_ | but I wonder if it takes higher |
| 22:28.33 | *** join/#asterisk sudhir492 (~sudhir@4.7.59.232) |
| 22:28.35 | ManxPower | Um 48V is the non-ringing voltage, 90V is the ringing voltage. |
| 22:28.44 | harryvv | Manx, you probebly know more then me. What is it 90 vac for rining voltage? |
| 22:28.48 | harryvv | okay |
| 22:29.02 | harryvv | 48 is the voice ac then? |
| 22:29.12 | sudhir492 | when I do ztcfg -v, I get the following error |
| 22:29.17 | sudhir492 | ....Notice: Configuration file is /etc/zaptel.conf |
| 22:29.17 | sudhir492 | line 8: Unable to open master device '/dev/zap/ctl' |
| 22:29.38 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 22:29.40 | ManxPower | 48VDC I think. I'm sure there's a billion pages on google with the info |
| 22:29.42 | doughecka_ | ManxPower: its a cisco 7960 |
| 22:29.47 | harryvv | im sure |
| 22:29.48 | harryvv | ;) |
| 22:29.54 | doughecka_ | 7940* |
| 22:30.06 | redder86 | jaiger: and it works :-) |
| 22:30.30 | ManxPower | Cisco phone would be PoE and that IS 48vdc |
| 22:30.33 | [cc]smart | i cimpiled ztdummy and modprobed it, but i still get MOOH stutter. do i need to "enable" ztdummy usage somewhere in config files ? |
| 22:30.35 | harryvv | btw, what does safe asterisk do always chech sip iax connection and other asterisk releated services and restart them if thay go down? |
| 22:31.14 | NoOS | can you use callwaiting with a BRI and a CAPI card? |
| 22:31.16 | doughecka_ | ManxPower: ok |
| 22:31.30 | doughecka_ | ManxPower: but would it accept higher voltages ok? |
| 22:31.37 | doughecka_ | surely they have somesort of protection inside |
| 22:31.42 | harryvv | doug dont bet on it. |
| 22:31.46 | ManxPower | doughecka_, I doubt it. |
| 22:31.49 | doughecka_ | hm |
| 22:31.55 | [cc]smart | NoOS: feture of chan_capi are described in the source package |
| 22:31.55 | harryvv | fry some transistors with higher voltages. |
| 22:32.05 | doughecka_ | unless they use a voltage regulator |
| 22:32.10 | harryvv | sure |
| 22:32.19 | harryvv | unless thay do have that circuitry |
| 22:32.45 | *** join/#asterisk Matt-E- (~matto@66-224-125-137.atgi.net) |
| 22:33.10 | harryvv | btw, does a asterisk recompile wipe out the modified conf files in /etc/asterisk I backed up mine just in case. |
| 22:33.20 | doughecka_ | no |
| 22:33.23 | doughecka_ | unless you do make samples |
| 22:33.28 | harryvv | okay |
| 22:33.39 | harryvv | forgot about that one ;) |
| 22:34.13 | [cc]smart | nobody knows if ztdummy needs to be enabled in configs to be recognized ? or is it all set once the module is loaded ? |
| 22:34.56 | doughecka_ | [cc]smart: yes, but it needs USB |
| 22:34.58 | doughecka_ | do you have usb |
| 22:34.59 | ManxPower | [cc]smart, ztdummy does not need any configuration |
| 22:35.50 | [cc]smart | i have kernel 2.6 and understood from description that then, it uses the new timers |
| 22:35.57 | doughecka_ | oh |
| 22:36.05 | sudhir492 | :q |
| 22:36.11 | [cc]smart | ? |
| 22:36.13 | sudhir492 | oops |
| 22:37.04 | clint_ | Folks: Anyone know how to get Asterisk to generate a real busy signal? |
| 22:37.14 | [cc]smart | new kernel timers that is, to generate zapata time |
| 22:37.33 | [cc]smart | that wrong ? |
| 22:38.03 | [cc]smart | clint_: Where ? On ext. line ? |
| 22:38.08 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 22:38.59 | clint_ | an analog phone hanging off of a channel bank hanging of of a 4 port digium card. |
| 22:39.17 | clint_ | when a line is busy, I get a 'reorder' (fast busy) tone. |
| 22:39.41 | clint_ | This is fine for most things, but I've got some older (dialogic based) stuff that doesn't recognize this as a busy tone. |
| 22:39.55 | [cc]smart | if i got that right, then this is tonezone in /etc/zaptel.conf no ? |
| 22:40.52 | clint_ | I suppose, but I don't know what I'm looking for - does asterisk differentiate between far end congestion (user busy) and middle of the line congestion (all circuits are busy) when determining what tone to play? |
| 22:41.53 | [cc]smart | next |
| 22:41.55 | [cc]smart | :) |
| 22:41.55 | *** join/#asterisk devel (~devel@wiggum.digitalcoven.com) |
| 22:41.58 | clint_ | ... And if so, where would I change that... |
| 22:42.14 | clint_ | (yeah, that's what I was feeling :) ) |
| 22:42.31 | [cc]smart | but that should be testable |
| 22:42.46 | clint_ | I was really hoping not to have to go code diving, as I'm not the world's greatest developer :) |
| 22:42.47 | [cc]smart | could possibly have two cases. |
| 22:43.04 | [cc]smart | try a local registered softphone that is busy and check busy tonw |
| 22:43.12 | clint_ | standby.. |
| 22:43.18 | [cc]smart | then try non registered one via lan and check busytone |
| 22:44.06 | [cc]smart | so somebody knows how ztdummy behaves on kernel 2.6 ? the hi frequency timers do/don't work ? |
| 22:44.25 | clint_ | ok... short test... sip phone to pstn busy - fast busy |
| 22:44.33 | *** join/#asterisk devel (~devel@wiggum.digitalcoven.com) |
| 22:44.36 | clint_ | pstn to sip phone busy - fast busy |
| 22:44.51 | clint_ | hmmm... that's wierd. |
| 22:44.54 | [cc]smart | sip phone is local regostered |
| 22:45.00 | clint_ | yes. |
| 22:45.14 | clint_ | (breaking that) |
| 22:45.25 | [cc]smart | and the busy to pstn was due to receiving set busy or due to channel busy ? |
| 22:45.36 | clint_ | Set busy. |
| 22:45.44 | [cc]smart | then it's the same case |
| 22:45.47 | clint_ | So it would appear that asterisk does not differentiate. |
| 22:45.54 | [cc]smart | no, doesn't |
| 22:46.07 | clint_ | ... That it is telling the network, via the PRI, hey, I've got no path to there, rather than hey, that guy's on the phone.... |
| 22:46.07 | [cc]smart | cause the difference would have arosen, when all channels were busy at the time |
| 22:46.21 | [cc]smart | in your case, both times the set was busy or did i get that wrong ? |
| 22:46.24 | modulus_ | mary rose sat on a pin, mary rose. |
| 22:46.37 | clint_ | Yep, busy'ed the destination set.... |
| 22:46.39 | clint_ | ... |
| 22:46.47 | modulus_ | jbot jbot? |
| 22:46.48 | jbot | somebody said jbot was the shipboard computer, but you may call me eddie if it helps you relax |
| 22:46.59 | [cc]smart | otsnack |
| 22:47.04 | [cc]smart | ~botsnack |
| 22:47.04 | jbot | aw, gee, [cc]smart |
| 22:47.14 | clint_ | But even dialing in from outside, the phone company gave me a fast busy, leading me to believe that asterisk told it that it didn't have a channel avail as opposed to that the user was busy. |
| 22:47.47 | [cc]smart | can you identify if the call was set up on your phones display ? |
| 22:47.53 | clint_ | Now I really don't know what I'm looking for.... |
| 22:47.57 | clint_ | Let's see.. |
| 22:49.27 | harryvv | you know...I knew some day that discrete electronic silicon chips would be replaced by light chips and mabey intel has done it? Its in the news. |
| 22:50.25 | clint_ | Ok, on the SIP phone, it looks like the call is immediately rejected - no talk path opened. |
| 22:50.50 | clint_ | on the analog phone (chan bk to asterisk) I get inband signalling of the fast busy, asterisk (debug) says congestion... |
| 22:51.00 | clint_ | SIP debug shows a 503 |
| 22:51.24 | clint_ | Is 503 (svc unavail) the correct way to indicate subscriber busy? |
| 22:52.43 | [cc]smart | dunno, from here it takes someone with experience :) |
| 22:53.41 | clint_ | Dammit, man, that's why I come to you! :) |
| 22:53.41 | hajekd | Looks like VoipJet has issues with DNS. |
| 22:54.03 | hajekd | Their DNS is down...;) |
| 22:54.05 | loud | yep |
| 22:54.07 | harryvv | fun |
| 22:54.15 | *** join/#asterisk brokep (brokep@basthard.com) |
| 22:54.27 | hajekd | I was just about to give them a try...fun |
| 22:54.44 | brokep | hi, everybody. i have some troubles with Realtime, anyone care to make a small effort and help me out? |
| 22:54.45 | harryvv | my * is fukared. can somone verify me what ls -la /dev/zap/channel is? |
| 22:55.13 | iMediax | huh? |
| 22:55.41 | brokep | anyone seen "pbx.c:783 pbx_find_extension: No such switch 'Realtime'" before? |
| 22:55.44 | [cc]smart | 196, 254 |
| 22:55.52 | clint_ | Looks, btw, like 486 is the expected sip response (Busy Here) to indicate that the subscriber is successfully contacted, but on the phone or otherwise busy. |
| 22:56.13 | clint_ | So for a more general question: |
| 22:56.23 | clint_ | Does anyone's asterisk give a busy signal ever? |
| 22:56.43 | clint_ | ... or a SIP 486 or similar as opposed to a 5xx series sip response when a user is busy? |
| 22:56.43 | afrosheen | harryvv: on my box it's asterisk:asterisk with a billion files |
| 22:57.11 | ^Fenris | I have a one POTS line connected to my * box that is acting as a fax machine, however, when I send a fax to it it makes my VOIP line ring when its receiving the fax, how can I make it stop doing that? |
| 22:57.13 | clint_ | And if so, what very basic thing am I farking up? |
| 22:57.18 | [cc]smart | harryvv: 196, 254 |
| 22:57.35 | tzanger | hmm did digium discontine the T100P? |
| 22:57.38 | tzanger | it's no longer on their site |
| 22:57.49 | tzanger | makes sense as they already have teh TE110P |
| 22:57.53 | afrosheen | ebay didn't discontinue it :) |
| 22:57.55 | brokep | anyone seen "pbx.c:783 pbx_find_extension: No such switch 'Realtime'" before? |
| 22:58.35 | harryvv | tzanger, you pretty familliar with what asterisk/zap permissions are set to? I have had some issues since last night with it. |
| 22:58.42 | Matt-E- | is it better to use a ip phone or to buy like a sipura 2000 and use analog phones? |
| 22:58.43 | tzanger | root.root |
| 22:58.57 | harryvv | you would think. its root:root for mine to. |
| 22:59.34 | hajekd | silly voipjet, they have both DNS on the same network... |
| 22:59.50 | [cc]smart | root:root 660 |
| 22:59.53 | afrosheen | harryvv: i've got AMP installed, everything runs under the asterisk user |
| 23:00.01 | harryvv | whats amp? |
| 23:00.11 | afrosheen | asterisk manager portal, amp.voxbox.ca |
| 23:00.15 | hardwire | A Male Pe*** |
| 23:00.22 | afrosheen | pearl? |
| 23:00.31 | hardwire | ugh |
| 23:00.40 | afrosheen | pence |
| 23:00.43 | hardwire | sucking chicky noodle soup through this little latte cup hole is .. hard |
| 23:01.09 | harryvv | well anway this problem came up as a issue of asterisk not able to read /dev/zap and showing this in /var/log/asterisk/messages |
| 23:01.12 | shido6 | amp is a good idea, but not finished |
| 23:01.20 | *** part/#asterisk brokep (brokep@basthard.com) |
| 23:01.28 | afrosheen | shido6: it's come a long way in a short time |
| 23:01.41 | tzanger | harryvv: uh... it should be working |
| 23:01.53 | afrosheen | and ryan + jason are working hard on it, the forums are very active on sf |
| 23:02.15 | harryvv | tzanger yea it should take a look at the cli errors http://pastebin.ca/5983 |
| 23:02.48 | harryvv | It was setup to load auto when the system boots. |
| 23:02.53 | tzanger | harryvv: are your zaptel hardware modules loaded? Did you run ztcfg ? |
| 23:03.01 | harryvv | it should be let me check |
| 23:03.04 | [cc]smart | ~Fenris: some stuff down on http://www.voip-info.org/wiki-Asterisk+fax |
| 23:03.10 | [cc]smart | ^Fenris: some stuff down on http://www.voip-info.org/wiki-Asterisk+fax |
| 23:03.51 | harryvv | tz... well i guess thay were not. thats weird. |
| 23:04.29 | hardwire | wiki wiki |
| 23:04.30 | hardwire | wah |
| 23:04.31 | hardwire | wah |
| 23:04.32 | harryvv | What I really want it to do is load everything on booting the system say as a electroical brown out. |
| 23:04.45 | harryvv | let me reboot it see what happens. |
| 23:11.32 | harryvv | tzanger init.d or some other service is not loading the modules in debian. |
| 23:11.35 | tessier_ | Question... I want _011. to go to one carrier but _01184. to go to another |
| 23:11.46 | tessier_ | Does asterisk evaluate the dialplan in top down order? |
| 23:11.50 | tessier_ | So I can put the more specific route first? |
| 23:11.57 | `Sauron | sort of, maybe |
| 23:11.57 | tessier_ | I am always confused on this point. |
| 23:12.09 | tessier_ | Sort of? What is the proper sure-fire way to handle this then? |
| 23:12.14 | `Sauron | Actually, I don't know. I'd make 2 different contexts, and include them in the order you want |
| 23:12.22 | `Sauron | [carrier1] |
| 23:12.30 | dsmouse | tessier_: if it doesn't, you can use gotoif |
| 23:12.31 | `Sauron | exten => _01184. |
| 23:12.33 | `Sauron | blahblah |
| 23:12.35 | tessier_ | And they are guarenteed to get ordered properly? |
| 23:12.36 | `Sauron | [carrier2] |
| 23:12.41 | `Sauron | exten => _011. |
| 23:12.42 | tessier_ | if you include contexts in order? |
| 23:12.43 | `Sauron | blahblah |
| 23:12.48 | `Sauron | then in your main context, do |
| 23:12.52 | `Sauron | include => carrier1 |
| 23:12.55 | `Sauron | include => carrier2 |
| 23:15.08 | *** join/#asterisk obiyoda (~Jared@70-58-164-219.bois.qwest.net) |
| 23:17.31 | tessier_ | That seems rather kludgy |
| 23:17.40 | *** join/#asterisk bjohnson_ (~bjohnson@jecinc.tor.istop.com) |
| 23:17.42 | [cc]smart | zttest gives me about 95% accuracy and asterisk CLI says: Warning, flexible rate not heavily tested! |
| 23:17.47 | [cc]smart | whats going on ? |
| 23:17.55 | `Sauron | that's the recommended way to force order of evaluation |
| 23:18.02 | `Sauron | the warning has nothing to do with the dialplan change |
| 23:18.26 | PatrickDK | heh, that is the only way to do it, is using two different includes |
| 23:18.57 | *** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.rr.com) |
| 23:19.00 | PatrickDK | bkw has a good write up on that |
| 23:19.36 | |Vulture| | Anyone here know the name of that program that can run clients on w2k3 and linux, and links to a server to notify that they are online and services are active? |
| 23:19.41 | `Sauron | I wonder if there's a reason they haven't adopted a standard best-match algorithm |
| 23:19.59 | `Sauron | I guess it's because best-match doesn't handle cases such as "equal match" very well |
| 23:20.19 | PatrickDK | hmm, I think it has to do with matching as you dial |
| 23:20.43 | PatrickDK | not really too sure |
| 23:21.00 | PatrickDK | it could also do with the timelimit includes |
| 23:21.13 | harryvv | well /etc/modules.conf should be loading wcfxo and zaptel on my system when its a cold reboot. |
| 23:21.15 | PatrickDK | some peoples configuration, best match could totally change how it works |
| 23:21.33 | PatrickDK | harryvv, you need to run ztcfg too |
| 23:22.22 | harryvv | it has that included after each module entery |
| 23:22.27 | *** join/#asterisk welby (~welby@solas.plus.com) |
| 23:24.13 | [cc]smart | vulture: might think of sjphone ? |
| 23:25.49 | terrapen | i was playing air guitar at my desk |
| 23:25.55 | terrapen | <--- spaz |
| 23:26.11 | |Vulture| | [cc]smart: no I am looking for a solution to know if my servers crash or the inet goes off there |
| 23:26.24 | *** join/#asterisk stonefly (~stonefly@toby.stoneflytech.com) |
| 23:26.32 | redder86 | can callgroup be set in the dialplan? For example, I don't want *all* calls to be pickup-able, but just some. |
| 23:32.02 | riksta | don't suppose anyone here is good at java swing? i could use some newbie help |
| 23:33.18 | stonefly | I feel like an idiot, but here goes.... On FC3, modprobe.conf has alias char-major-196 ztdummy and install ztdummy /sbin/modprobe --ignore-install ztdummy && /sbin/ztcfg, but it doesn't autoload when asterisk starts.... I've also tried zaptel.init, but it won't load ztdummy either... what is the recomened way to autoload ztdummy? |
| 23:33.21 | srt | riksta: whats your prob? |
| 23:33.24 | CoaxD | Can any of you folks think of a simple way to blacklist a phone number based on callerid - without doing something w/ AGI? |
| 23:33.42 | CoaxD | i.e. you send them to a special bitbucket, etc |
| 23:34.04 | stonefly | CoaxD, the exgirlfriend example, or is that agi? I don't remember.... |
| 23:34.23 | stonefly | CoaxD, couldn't you just do some extension logic based on callerid? |
| 23:34.28 | riksta | srt: i wrote this v basic swing http://eugeneciurana.com/pastebin/pastebin.php?show=5358 but i cant figure out why i have a big gap on the right |
| 23:35.16 | *** join/#asterisk purplebob (~don@206-230-187-185.sugardog.com) |
| 23:35.51 | CoaxD | stonefly: that was what i was asking, dude ;) |
| 23:36.08 | CoaxD | stonefly: Asterisk isn't real good at if/then/else logic within extensions.conf |
| 23:36.15 | CoaxD | stonefly: Some of it is possible, but.. |
| 23:37.04 | visik7 | CoaxD there is a kind of tcl way to write a dialplan instead of extension.conf |
| 23:37.08 | visik7 | somewhere |
| 23:37.17 | CoaxD | visik: It is called AGI |
| 23:37.31 | visik7 | :) |
| 23:37.32 | CoaxD | visik: You can use either perl or php |
| 23:37.47 | stonefly | CoaxD, look for the exgirlfriend example |
| 23:37.48 | visik7 | it wasn't a question |
| 23:37.53 | CoaxD | as i said in my original statement, I don't want to use AGI. I don't get it. |
| 23:38.03 | CoaxD | stonefly: Hmm. Oay |
| 23:38.04 | *** join/#asterisk thefallen (PolarBear@thefallen.user) |
| 23:38.05 | CoaxD | er okay |
| 23:38.28 | purplebob | how can you have asterisk dial an zap interface, and if its busy, goto a a specific voicemail box. Right now if its busy, it just hangs up. I have goto busy and noanswer defined in the dialplan entry. it just doesn't follow to them.. |
| 23:38.37 | *** part/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 23:38.40 | mikegrb | check the wiki |
| 23:38.54 | purplebob | I am using the example off the wiki. |
| 23:39.01 | CoaxD | http://www.voip-info.org/wiki-Asterisk+rollout+tips |
| 23:39.04 | CoaxD | heh |
| 23:39.58 | CoaxD | Ahhh. EASY |
| 23:39.59 | stonefly | the wiki's helped me out so much! |
| 23:40.12 | CoaxD | exten => 8005551212/4085551212,1,Congestion |
| 23:40.23 | harryvv | anyone here have a x100p and can reboot there system and it loads the zap drivers message me. |
| 23:41.10 | PatrickDK | harryvv, why? |
| 23:41.14 | PatrickDK | it's really easy to do |
| 23:41.23 | PatrickDK | modprobe wcfxo; ztcfg; asterisk |
| 23:41.25 | PatrickDK | done |
| 23:41.41 | harryvv | not when im not here and there is a brown out. |
| 23:42.03 | PatrickDK | well, script it |
| 23:42.13 | harryvv | i want this system to reboot and reload those modules. ive just modified modules.conf and it should work..hopfully now. |
| 23:42.33 | shido6 | check ur pm, harryvv |
| 23:42.36 | stonefly | Patrick, add alias char-major-196 wcfxo and install wcfxo /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg to modprobe.conf |
| 23:42.38 | purplebob | I think the command he is looking for is chmod -R 000 / |
| 23:43.18 | PatrickDK | stonyfly, my system doesn't have a problem |
| 23:43.24 | obiyoda | riksta did you get your question figured out? |
| 23:43.24 | PatrickDK | something about how harry is doing it |
| 23:43.28 | purplebob | then userdel -r root |
| 23:43.34 | shido6 | oh come on |
| 23:43.36 | stonefly | PatrickDK, my bad.... |
| 23:43.37 | shido6 | do a make config |
| 23:43.39 | shido6 | and call it a day |
| 23:43.45 | stonefly | I meant haryw.... |
| 23:45.12 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 23:45.47 | *** join/#asterisk bjohnson_ (~bjohnson@jecinc.tor.istop.com) |
| 23:45.57 | srt | riksta: hm dont really understand what you are trying to accomplish ;) |
| 23:46.00 | bjohnson_ | is anyone able to get to voipjet.com? |
| 23:46.01 | PBXtech | there shouldnt be a an issue with 1 span as 5ess and another as ni2 |
| 23:46.05 | purplebob | but seriously. I have the busy, and noanswer lines defined. When the extension is called the zap dial goes out. then exits non 0. when its busy. it doesn't go to the vmail box as its supposed to. |
| 23:46.22 | harryvv | http://pastebin.ca/5986 /usr/src/zaptel make config error. |
| 23:46.31 | srt | riksta: but line 52 adds a JPanel that doesnt make sense there |
| 23:47.04 | srt | riksta: you add three panels to the GridLayout(2,1) - that wont work |
| 23:49.52 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
| 23:50.43 | JonR800 | bjohnson: not i |
| 23:51.47 | redder86 | can callgroup be set in the dialplan? |
| 23:52.19 | CoaxD | BTW, stonefly, the ex-girlfriend feature is exactly what i needed. Thanks. |
| 23:52.19 | PBXtech | i have a 5ess span and i cant get my second span going its ni2, i think its talking 5ess to it. I have the switchtype seperated for the two.. any idea |
| 23:52.53 | stonefly | CoaxD, np |
| 23:54.35 | riksta | srt, am i looking to change that to 3,1 |
| 23:56.26 | srt | riksta: what do you need that empty panel for? |
| 23:57.05 | srt | if it should be below the "hello" button line 29 must be 3,1 |
| 23:57.10 | riksta | i meant to put a hello button in it |
| 23:57.22 | srt | a second hello button? |
| 23:57.27 | riksta | no just one |
| 23:57.29 | riksta | i think i added it too soon |
| 23:58.22 | srt | if you remove line 47 you get the button array in the upper 1/4th, the hello button below and the lower half is empty |
| 23:58.30 | riksta | srt: i found the problem, line 47! |
| 23:58.32 | riksta | doh |
| 23:58.33 | riksta | thanks man |
| 23:58.40 | riksta | i just found it as you did |
| 23:59.02 | srt | k :) |
