| 00:01.24 | TripleFFFF | ap0l1nux asterisk runs on window |
| 00:01.53 | file | [TK]D-Fender: what are you doing home on a Saturday night?!? |
| 00:01.59 | [TK]D-Fender | TripleFFFF : yeah... while emulating linux in a complicated fashion and no proper access to zaptel :/ |
| 00:02.09 | [TK]D-Fender | file : About to head out to play some pool |
| 00:02.20 | TripleFFFF | pool.. hmm been a while |
| 00:02.25 | TripleFFFF | my bday |
| 00:02.27 | TripleFFFF | ;) 33 |
| 00:02.27 | file | [TK]D-Fender: exxxxxxxcellent |
| 00:05.18 | brandini | I have the finest open source voicemail system eva!! |
| 00:05.31 | *** part/#asterisk StewLG (i=user@216-99-218-126.dsl.aracnet.com) |
| 00:45.30 | *** join/#asterisk chiardon (i=Dorphals@pcsp168-254.supercabletv.net.co) |
| 00:47.37 | chiardon | Hello, I have a TE410P connected to two E1s and to two Zhone Channelbanks, every so often I get some channels that 'stay up'. I am not sure if its the CBs or * |
| 00:47.57 | chiardon | How could I discard a physiscal problem?= |
| 00:50.24 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 00:51.52 | hads|home | heh |
| 00:59.49 | brandini | hmmmmmmm, so I *need* a card in the box in order to answer/make calls, or can I do it over a serial modem? |
| 01:00.59 | dlynes_laptop | brandini: you can use an ata instead |
| 01:01.10 | dlynes_laptop | brandini: that's an external box with an fxs port in it |
| 01:01.20 | brandini | yeah... |
| 01:01.36 | dlynes_laptop | but, no |
| 01:01.40 | dlynes_laptop | a serial modem won't cut it |
| 01:01.46 | brandini | so I can't do voip to my * box to the modem |
| 01:02.26 | dlynes_laptop | brandini: http://www.google.ca/search?hl=en&q=define%3Ano&btnG=Google+Search&meta= |
| 01:02.50 | brandini | hah! |
| 01:02.51 | hads|home | lol |
| 01:03.18 | brandini | :P |
| 01:03.25 | brandini | *useless |
| 01:04.35 | dlynes_laptop | brandini: you're talking about a serial port; the serial channel can't handle enough bandwidth for the voice coming from it into your soundcard |
| 01:05.02 | brandini | just enough for computer data |
| 01:05.04 | dlynes_laptop | it's technically incapable of the task, even if asterisk had a driver for it |
| 01:05.05 | brandini | gotcha |
| 01:05.47 | dlynes_laptop | well, actually...not quite |
| 01:05.59 | dlynes_laptop | it can do about 115,200bps - overhead |
| 01:06.14 | dlynes_laptop | assuming you have a 16550AN fifo chip |
| 01:06.36 | dlynes_laptop | but if you're running 16450 or lower, it's only capable of 57,600bps or lower, in a reliable manner |
| 01:06.51 | dlynes_laptop | and it needs to be able to stream raw audio |
| 01:07.03 | dlynes_laptop | 8k raw audio would need 100Kbps |
| 01:07.24 | brandini | gotcha, and isn't it like 16k at least? |
| 01:07.32 | dlynes_laptop | 8k |
| 01:07.40 | dlynes_laptop | 8k == telephone quality |
| 01:07.45 | brandini | I see |
| 01:07.52 | brandini | so that's why we have cards |
| 01:08.00 | dlynes_laptop | or ata's |
| 01:08.15 | dlynes_laptop | the atas talk to asterisk via the network, not serial cable |
| 01:08.35 | brandini | I see |
| 01:09.21 | dlynes_laptop | besides |
| 01:09.28 | dlynes_laptop | do external voice modems even exist? |
| 01:10.19 | dlynes_laptop | brandini: you can also try searching ebay for 'x100p' |
| 01:10.23 | brandini | I figured you know, my Voice modem would do just that... |
| 01:10.24 | dlynes_laptop | brandini: those cards are cheap |
| 01:10.37 | dlynes_laptop | brandini: and they're voice modems with a jumper on them |
| 01:10.48 | dlynes_laptop | brandini: but accordingly, they sound like complete crap, too |
| 01:10.57 | *** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 01:11.15 | brandini | oh but for 15 bucks |
| 01:11.21 | dlynes_laptop | which is why digium stopped selling them |
| 01:11.26 | dlynes_laptop | i guess they probably got too many returns |
| 01:11.42 | dlynes_laptop | brandini: $15 is too expensive for you? |
| 01:11.56 | brandini | no, I was saying *only* 15 bucks it's easier to buy that |
| 01:12.10 | brandini | I wonder if there is a driver for openbsd |
| 01:12.32 | brandini | yup |
| 01:12.40 | dlynes_laptop | anyways |
| 01:12.50 | dlynes_laptop | be happy zaptel even works on freebsd, never mind openbsd :) |
| 01:13.01 | brandini | :) |
| 01:13.57 | dlynes_laptop | asterisk is pretty stable on freebsd, if you only use softphones, atas and fxo gateways |
| 01:15.46 | *** join/#asterisk rnovotny22 (n=rnovonty@71-36-153-135.mpls.qwest.net) |
| 01:20.44 | chiardon | Hello, I have a TE410P connected to two E1s and to two Zhone Channelbanks, every so often I get some channels that 'stay up'. I am not sure if its the CBs or * How could I discard a physical problem? |
| 01:22.26 | |Vulture| | chiardon: you mean the channels stay open and never get a disconnect signal? |
| 01:33.07 | chiardon | |Vulture| -> Yes |
| 01:33.24 | chiardon | |Vulture| -> Even if the phone is properly hanged |
| 01:33.47 | chiardon | Strangely enough when I restart asterisk everything returns to normal |
| 01:45.37 | crochat | Hello ! |
| 01:45.46 | chiardon | Ello |
| 01:46.01 | crochat | Is there any "PlayOgg" like application in Asterisk ? |
| 01:49.15 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 01:51.41 | chiardon | I thinkso |
| 01:51.51 | chiardon | You should take a look at the wiki |
| 01:54.02 | |Vulture| | chiardon: does this happen just on the channel bank or on the E1 too? |
| 01:54.48 | chiardon | only on the CB |
| 01:55.05 | chiardon | and generally on the same channels |
| 01:55.18 | |Vulture| | really... thats interesting |
| 01:55.26 | |Vulture| | I am assuming the banks are for FXS? |
| 01:55.31 | chiardon | Yeap |
| 01:55.47 | chiardon | 48 nice and shiny FXS |
| 01:56.00 | |Vulture| | okay can you try switching a channel with one you never see have the issue with one you do? |
| 01:56.16 | |Vulture| | that would at least narrow it down to a bank/* issue or a phone issue |
| 02:01.50 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:01.50 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:13.27 | *** join/#asterisk terrapen (n=cjs@mauritius.island.nu) |
| 02:13.31 | terrapen | aloha |
| 02:13.48 | terrapen | does anybody know how to get a Polycom to use a different ringtone when a call is transferred to it? |
| 02:17.32 | dlynes_laptop | crochat: why not just use Playback()? |
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| 02:18.00 | *** join/#asterisk sugardave (n=not@cpe-66-68-165-207.austin.res.rr.com) |
| 02:18.14 | dlynes_laptop | crochat: just make sure you have format_ogg_vorbis.so loaded |
| 02:18.15 | crochat | dlynes_laptop: Does Playback() support Ogg format ? |
| 02:18.27 | crochat | mm ok |
| 02:18.33 | dlynes_laptop | crochat: it supports anything you have a format module loaded for |
| 02:18.43 | sugardave | has anyone successfully sent voice to a Google talk client? I think I'm getting close, just no voice :) |
| 02:19.17 | dlynes_laptop | crochat: i've never actually tried playing ogg format files with it, but theoretically, it should work |
| 02:19.56 | crochat | dlynes_laptop: And for the music on hold ? Does it work too ? |
| 02:20.07 | dlynes_laptop | crochat: should, yes |
| 02:20.34 | dlynes_laptop | crochat: you'd have to use mode=files |
| 02:21.05 | *** join/#asterisk tuplink (n=hufft@68-232-82-147.chvlva.adelphia.net) |
| 02:21.16 | tuplink | pbx_extension_helper: No application 'EnumLookup' for extension how do i get enum? |
| 02:22.33 | dlynes_laptop | tuplink: load app_enumlookup.so? |
| 02:22.55 | tuplink | what file do i do that in? |
| 02:23.12 | dlynes_laptop | tuplink: asterisk -rx "load app_enumlookup.so" |
| 02:23.41 | dlynes_laptop | tuplink: or load => app_enumlookup.so in your modules.conf file |
| 02:24.55 | tuplink | Aug 12 22:25:13 WARNING[34645]: loader.c:850 print_and_load: Loading module app_enumlookup.so failed! |
| 02:25.29 | blitzrage | does it exist in /var/lib/asterisk/modules/ ? |
| 02:26.19 | file | or are you using a version where it was deprecated and removed, and replaced by the dialplan function? |
| 02:26.23 | file | and hi blitzrage |
| 02:26.43 | blitzrage | file: !!! |
| 02:26.51 | file | zomg |
| 02:27.00 | blitzrage | omghi! |
| 02:27.05 | file | how is the Madsen Mansion? |
| 02:27.15 | blitzrage | very nice -- I painted yesterday |
| 02:27.24 | file | getting high off the fumes, I see how it is |
| 02:27.38 | blitzrage | nah... plenty of ventilation... and the paint didn't really smell too bad |
| 02:28.12 | tuplink | that file dosnt even exist |
| 02:28.47 | blitzrage | maybe like file said it's a DP function |
| 02:28.51 | blitzrage | do a 'show functions' |
| 02:29.18 | tuplink | what am i looking for in functions |
| 02:29.29 | blitzrage | something enum related |
| 02:29.43 | tuplink | ENUMLOOKUP ENUMLOOKUP(number[|Method-type[|opt ENUMLOOKUP allows for general or specific querying of NAPTR records or counts of NAPTR types for ENUM or ENUM-like DNS pointers |
| 02:29.49 | blitzrage | there you go |
| 02:29.59 | tuplink | well.... whats that mean? |
| 02:30.30 | blitzrage | its a DP function that allows you to perform an ENUM lookup |
| 02:31.49 | tuplink | what is DP? |
| 02:31.53 | blitzrage | dialplan |
| 02:32.01 | tuplink | ok... |
| 02:32.07 | tuplink | how do i call it in the dialplan |
| 02:32.40 | tuplink | exten => _*164X.,4,ENUMLOOKUP(${EXTEN:4}) thats where is get the error |
| 02:32.42 | blitzrage | something like... exten => s,1,Set(RESULT=${ENUMLOOKUP(5195915119)}) |
| 02:32.46 | *** join/#asterisk anthonyl (n=anthony@office.midphase.com) |
| 02:32.54 | anthonyl | ² |
| 02:33.05 | blitzrage | A2A to you too |
| 02:33.27 | tuplink | and i guess RESULT would eql TRUE/FALSE |
| 02:33.39 | blitzrage | I'm not sure what ENUMLOOKUP returns to be honest -- never used it |
| 02:33.56 | blitzrage | I assume it returns an address of some sort |
| 02:34.09 | tuplink | well i dont even think that will work...???... if i show applacation enum? i get nothing |
| 02:34.20 | file | it's not an application, it's a dialplan function |
| 02:34.29 | blitzrage | it's not an application, its a d.... what file said |
| 02:34.32 | file | the application was deprecated and removed, and replaced with the dialplan function |
| 02:34.49 | blitzrage | I love dialplan functions |
| 02:34.50 | tuplink | ok... |
| 02:35.10 | tuplink | im starting to find alot of them.... like callerid stuff |
| 02:35.11 | tuplink | rite |
| 02:35.16 | file | blitzrage: your cell # continues to be cool |
| 02:35.26 | blitzrage | lol |
| 02:35.29 | blitzrage | it sure does :) |
| 02:35.35 | blitzrage | best number ever :) |
| 02:35.42 | blitzrage | well... maybe not ever... but it's up there |
| 02:35.57 | blitzrage | 915195915119 |
| 02:35.58 | tuplink | so i would do somthing like what you said falowed be a gotoif |
| 02:36.21 | blitzrage | tuplink: yah, you'd want to do a check to make sure ENUMLOOKUP actually returned something |
| 02:36.40 | tuplink | ok.... |
| 02:36.59 | tuplink | dose any body actule use enum? |
| 02:37.06 | tuplink | cause im just playing w/ it |
| 02:37.28 | file | I need a bigger whiteboard, or another one |
| 02:37.48 | blitzrage | yes you do!!! |
| 02:37.53 | blitzrage | you can't have too many whiteboards |
| 02:37.58 | blitzrage | I have two, and they are always full |
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| 03:06.13 | sugardave | if anyone has successfully gotten * to send voice to GTalk, I'd like to pick your brain |
| 03:07.40 | |Vulture| | Is anyone working on integration with Exchange 2007 with *? |
| 03:08.30 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.144) |
| 03:13.26 | smackus2 | im back... still having issues with my t1s. |
| 03:14.01 | smackus2 | i have 1 pri from one vendor in port 1 and ports 2,3,4 are pri in a trunkgroup from another t1 provider. |
| 03:14.07 | smackus2 | here are my configs http://pastebin.ca/128619 |
| 03:14.08 | *** join/#asterisk sugardave (n=root@cpe-66-68-165-207.austin.res.rr.com) |
| 03:14.36 | smackus2 | i cannot get anything past chanels 48 to load. |
| 03:14.55 | smackus2 | Aug 12 21:14:45 ERROR[11482]: chan_zap.c:6999 mkintf: Channel 24 is reserved for D-channel. |
| 03:14.55 | smackus2 | Aug 12 21:14:45 ERROR[11482]: chan_zap.c:10288 setup_zap: Unable to register channel '49-96' |
| 03:14.55 | smackus2 | Aug 12 21:14:45 WARNING[11482]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 |
| 03:15.14 | smackus2 | any help would be appreciated, i have been working on this since early this morning. |
| 03:15.44 | |Vulture| | smackus2: can you get just the single PRI up? |
| 03:15.54 | smackus2 | yes |
| 03:16.05 | smackus2 | i can get the single, plus the first t1 of the trunk group |
| 03:16.28 | smackus2 | but if i add the rest of the trunk group into the zapata.conf it will not load |
| 03:17.32 | |Vulture| | hrm I cant help with trunkgroups sorry |
| 03:17.46 | |Vulture| | only done banks + PRI never multi PRI |
| 03:19.04 | |Vulture| | could it be the second "spanmap => 2,2,2" line |
| 03:19.11 | *** join/#asterisk sugardave (n=sugardav@cpe-66-68-165-207.austin.res.rr.com) |
| 03:19.18 | smackus2 | tell me your thoughts on it |
| 03:19.21 | |Vulture| | http://www.voip-info.org/wiki/view/NFAS |
| 03:19.39 | |Vulture| | try changing it to 3,2,2 and the next 4,2,3 just see if it loads |
| 03:20.19 | smackus2 | you mean 1,1,1 3,2,2 4,2,3? |
| 03:20.37 | |Vulture| | well yes and the 2,2,2 line also |
| 03:20.50 | smackus2 | ok... will try it and see |
| 03:21.00 | |Vulture| | because right now you have 2 spanmaps for zapspan 2 |
| 03:22.04 | smackus2 | here is what I tried: http://pastebin.ca/128627 |
| 03:22.07 | smackus2 | did not work |
| 03:23.41 | |Vulture| | okay so you want the 1 provider on its own trunk group right? |
| 03:23.44 | |Vulture| | thats the goal here? |
| 03:24.10 | smackus2 | yeah |
| 03:24.39 | |Vulture| | http://pastebin.ca/128629 |
| 03:24.51 | |Vulture| | try that duno if I am reading the wiki right but that looks correct to me |
| 03:25.56 | smackus2 | still not working |
| 03:26.07 | |Vulture| | really... same error? |
| 03:26.10 | smackus2 | yeah |
| 03:26.40 | |Vulture| | this shouldn't matter but the second trunkgroup should be "trunkgroup=>2,48,72,96" |
| 03:27.04 | smackus2 | the dchan is only on the first t1 |
| 03:27.11 | smackus2 | but i will try that... |
| 03:27.14 | smackus2 | anything at this point |
| 03:27.35 | |Vulture| | yea okay, those are just backup dchans |
| 03:28.39 | |Vulture| | smackus2: can you issue a ztcfg -vv with no error, then attempt a restart of *, see if it errors the same? |
| 03:28.48 | smackus2 | yep |
| 03:29.01 | |Vulture| | hrm stumping me.. |
| 03:29.19 | *** join/#asterisk zx225 (n=root@mytop.ipglobal.net) |
| 03:30.04 | |Vulture| | try taking out the third , in the spanmaps so ex. "spanmap => 1,1" according to the wiki it will auto assign |
| 03:31.31 | smackus2 | which one? |
| 03:31.37 | smackus2 | 1,1 is the first |
| 03:31.50 | smackus2 | so you say the 3rd? so the second of my 2nd trunk group? |
| 03:31.54 | |Vulture| | smackus2: I got to get to bed but scroll down that wiki and there is an example of a similar issue fixed by doomicon |
| 03:32.07 | |Vulture| | Ill pastebin what I mean |
| 03:32.07 | smackus2 | ok, thanks |
| 03:33.30 | |Vulture| | http://pastebin.ca/128636 |
| 03:34.29 | smackus2 | oh, understood |
| 03:34.30 | smackus2 | thanks |
| 03:35.04 | |Vulture| | oo... try in zaptel.conf: bchan=1-23,25-96 |
| 03:35.17 | |Vulture| | that might be it right there |
| 03:35.33 | |Vulture| | in fact I bet thats it |
| 03:36.22 | smackus2 | well, in my case wouldnt it be in zaptel.conf: bchan=1-23,25-47,49-96 cuz of the dcahn on the second t1? |
| 03:36.45 | smackus2 | i should have a dchan on zapport 1 and two |
| 03:37.02 | |Vulture| | ah true |
| 03:37.15 | |Vulture| | damn thought that was it |
| 03:37.21 | smackus2 | thanks though |
| 03:37.26 | |Vulture| | np night |
| 03:37.31 | smackus2 | seems you and i are the only ones on tonight... |
| 03:37.37 | smackus2 | now it is just me |
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| 03:42.52 | shodan | any wifi voip recommendation in the 150-300$usd range ? |
| 03:43.02 | sugardave | anyone using * and GTalk? |
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| 04:33.48 | chiardon | Hello? |
| 04:35.29 | dlynes_laptop | 为？ |
| 04:36.08 | chiardon | I'm trying to find out why do some zap channels stay up when I hang up my phone, I have two zhones connected to a TE410P |
| 04:36.50 | chiardon | Some ppl tell me it may be the channelbanks, however its strange, because when I restart asterisk, everythings works again |
| 04:37.07 | dlynes_laptop | yeah...i seen someone telling you that earlier |
| 04:37.08 | chiardon | (for a while untill the channels will stay up again= |
| 04:37.12 | chiardon | yeah it was me |
| 04:37.12 | chiardon | xD |
| 04:37.29 | chiardon | telling it could be hw |
| 04:37.51 | chiardon | however I did change the cable running from the channelbank to the device |
| 04:37.53 | file | chiardon: Z-Plex? |
| 04:37.57 | chiardon | file yeah |
| 04:38.23 | file | I've read that they aren't that good |
| 04:38.28 | chiardon | and I've also changed the channel |
| 04:38.52 | chiardon | i swapped channels with one that never had the problem and the issue appeared there as well |
| 04:39.09 | dlynes_laptop | file: wouldn't surprise me |
| 04:39.16 | chiardon | this issue happens in extensions with big traffic |
| 04:39.19 | dlynes_laptop | file: I've got some paradyne equipment which is also made by zhone |
| 04:39.48 | dlynes_laptop | file: cheap quality stuff, but unfortunately, it was the only hardware in that price range that would do the job I needed |
| 04:40.02 | *** part/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
| 04:40.29 | dlynes_laptop | when you plug the serial cable into the serial console for it, the power switch doesn't work...crazy ass shit like that |
| 04:41.08 | dlynes_laptop | and paradyne is supposedly good |
| 04:41.13 | dlynes_laptop | but it's all made in china crap now |
| 04:41.42 | file | the popups are eating my soul! :( |
| 04:41.52 | dlynes_laptop | stop using that internet exploder crap then |
| 04:42.19 | dlynes_laptop | ah...traitor :) |
| 04:42.30 | dlynes_laptop | thought all digium employees were required to use linux? :) |
| 04:42.45 | file | the systems @ my desk do |
| 04:42.54 | dlynes_laptop | ah |
| 04:42.56 | file | can't get the wireless to work on my Powerbook |
| 04:43.12 | dlynes_laptop | under linux? |
| 04:43.18 | file | correct |
| 04:43.19 | dlynes_laptop | or under osx? |
| 04:43.20 | dlynes_laptop | ah |
| 04:43.22 | chiardon | So ... any ideas? |
| 04:43.32 | file | the latest 2.6.17 kernel has support for it built in, and it gets detected fine |
| 04:43.36 | file | but I can't get it to associate to an AP |
| 04:43.55 | dlynes_laptop | file: are you using a secret, or a wep key, or unsecured? |
| 04:44.02 | file | unsecured |
| 04:44.26 | dlynes_laptop | Is it an intel ip3945? |
| 04:44.35 | file | it's a broadcom |
| 04:44.39 | dlynes_laptop | eww |
| 04:44.45 | dlynes_laptop | does it have a windows xp driver? |
| 04:45.07 | file | uh, dunno... it's whatever the airport extreme card is |
| 04:45.19 | file | lemme look |
| 04:45.21 | dlynes_laptop | so it probably has a windows xp driver then |
| 04:45.32 | file | bcm43xx based |
| 04:45.46 | dlynes_laptop | if so, you can install the windows xp driver using that windows wrapper layer for wireless cards |
| 04:45.49 | file | 4306 |
| 04:45.54 | dlynes_laptop | you don't actually need wine installed |
| 04:46.23 | dlynes_laptop | because it doesn't use windows to install it; it just uses the info in the windows driver |
| 04:46.49 | dlynes_laptop | file: ndiswrapper.sf.net |
| 04:47.16 | file | I'm Googling to see if there's anything about the driver that's already on here not working |
| 04:47.36 | dlynes_laptop | you'll need to extract the drivers first though, so that ndiswrapper can have access to the raw xp driver file |
| 04:47.51 | dlynes_laptop | file: have you tried iwlist scanning yet? |
| 04:48.00 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 04:48.06 | dlynes_laptop | file: to see if the wireless subsystem even detects your card? |
| 04:48.22 | file | yes, it's detected fine |
| 04:48.36 | file | iwconfig shows it, it's eth1 |
| 04:48.50 | file | oooooooooh |
| 04:48.53 | dlynes_laptop | ? |
| 04:48.57 | file | ifconfig eth1 up |
| 04:49.02 | file | bcm43xx: Error: Microcode "bcm43xx_microcode5.fw" not available or load failed. |
| 04:49.03 | file | :D |
| 04:49.09 | dlynes_laptop | You need the firmware :) |
| 04:49.17 | file | I know, this is progress though |
| 04:49.39 | dlynes_laptop | You need to dump bcm43xx_microcode5.fw in /lib/firmware |
| 04:50.12 | dlynes_laptop | you might also need to run some daemon software for your particular wireless card, too |
| 04:50.28 | dlynes_laptop | that's what i need to do for the intel ipw3945 |
| 04:51.31 | dlynes_laptop | yeah...probably something like: BCM2033-FW.bin |
| 04:51.42 | dlynes_laptop | I think that's a broadcom firmware that comes with 2.6.17.3 |
| 04:52.30 | file | already working on it... |
| 04:52.31 | dlynes_laptop | anyways |
| 04:52.31 | dlynes_laptop | gotta run |
| 04:52.33 | dlynes_laptop | ciao bella |
| 04:52.45 | file | ttyl |
| 04:54.53 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com) |
| 04:59.42 | *** part/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com) |
| 05:02.31 | file | ha, it worked |
| 05:03.59 | sugardave | awesome |
| 05:05.22 | file | and now the movie Antitrust is on, what a perfect way to fall asleep... seeing someone fake code in Java |
| 05:06.01 | denon | Antitrust? never saw it |
| 05:06.03 | denon | what chan? |
| 05:06.10 | file | CTV |
| 05:06.16 | denon | CTV? |
| 05:06.23 | file | it's Canadian |
| 05:06.30 | denon | ah ic |
| 05:06.36 | denon | and british ones |
| 05:06.39 | denon | but no canadian ones |
| 05:06.59 | file | how silly |
| 05:19.34 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 05:24.08 | JT | fake java |
| 05:24.11 | JT | but real gnome |
| 05:24.25 | file | indeed |
| 05:25.12 | chiardon | When I do Zap Show Channel XX I get some info ... |
| 05:25.30 | chiardon | what does the "Owner" and "Real" strings mean? |
| 05:25.50 | benjk | channel XX ? does asterisk support Roman numerals now? |
| 05:26.11 | chiardon | I'm using asterisk 0.7, so everything back there was in roman numerals |
| 05:26.12 | chiardon | hehehe |
| 05:26.17 | chiardon | nah, i lie its 1.2.7 |
| 05:26.47 | chiardon | Anyway |
| 05:26.55 | chiardon | Anybody knows about this? |
| 05:30.19 | docelmo | If you need help you can also join #redbull for help. |
| 05:31.56 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 05:32.39 | Qwell | Is tmobile voicemail still vulnerable to cidnum spoofing? |
| 05:33.26 | *** join/#asterisk twisted[mobile] (n=twisted@pdpc/supporter/active/twisted) |
| 05:33.26 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 05:33.31 | sugardave | it was in 1.2.10, I found by accident |
| 05:33.49 | sugardave | I haven't tried in the recent trunk I'm using |
| 05:37.22 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 05:38.21 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 05:39.24 | twisted[mobile] | lolomgwtfbbq? |
| 05:39.36 | docelmo | Qwell yes I did it to my phone a little while back |
| 05:39.50 | Qwell | docelmo: just require a password on vm? |
| 05:39.54 | Qwell | lame... |
| 05:39.59 | docelmo | yes.. |
| 05:40.02 | docelmo | I set my pass up |
| 05:40.03 | Qwell | oh well |
| 05:40.25 | twisted[mobile] | wheeeee |
| 05:40.44 | Qwell | bbl, hacking phone :D |
| 05:40.47 | twisted[mobile] | so elmo, you gonna be at *con? |
| 05:40.57 | docelmo | yes.. I am speaking this year |
| 05:41.08 | twisted[mobile] | Qwell, try #+*+menu+right dpad |
| 05:41.25 | twisted[mobile] | not all at the same time, one after the other |
| 05:41.35 | twisted[mobile] | it's an old school moto trick to get you into the service menu |
| 05:41.49 | twisted[mobile] | docelmo, cool |
| 05:42.19 | Qwell | twisted[mobile]: k |
| 05:42.33 | twisted[mobile] | Qwell, no guarantee it'll work on newer models |
| 05:42.44 | Qwell | no go |
| 05:42.50 | twisted[mobile] | gawd my bandwidth sucks |
| 05:42.54 | twisted[mobile] | reverse the * and # |
| 05:43.04 | twisted[mobile] | i can't ever get that right |
| 05:43.04 | Qwell | tried |
| 05:43.06 | twisted[mobile] | but you have to do it all within like 2 seconds |
| 05:43.19 | Qwell | oh.. |
| 05:43.49 | twisted[mobile] | i love the fact that my BB has a 3 finger salute :P |
| 05:44.14 | Qwell | heh |
| 05:44.55 | twisted[mobile] | hahahahaha |
| 05:44.59 | twisted[mobile] | what's better though |
| 05:45.08 | twisted[mobile] | is when you look back up at the TV and there's pr0n on |
| 05:45.39 | Qwell | s/when you look back up at the tv and // |
| 05:46.14 | Qwell | btw, whoever at motorola decided that they would go usb for everything...is the man, totally |
| 05:46.26 | twisted[mobile] | ew |
| 05:46.26 | Qwell | plug in the usb cable for my wifes camera...and bam, charging |
| 05:46.29 | twisted[mobile] | usb is so 2005 |
| 05:46.37 | Qwell | better than something proprietary |
| 05:46.46 | twisted[mobile] | oh, for the charger |
| 05:46.50 | denon | you implying bluetooth is better? cause if you like your battery .. its not :) |
| 05:46.52 | twisted[mobile] | yeah i have usb connection for charger |
| 05:46.54 | Qwell | and data transfer |
| 05:47.04 | twisted[mobile] | denon, actually, bt doesn't affect my battery life that much |
| 05:47.11 | twisted[mobile] | i can still go a week using BT and not have to charge |
| 05:47.13 | denon | bluetooth is like a dead short on the battery |
| 05:47.21 | denon | really? using BT how? |
| 05:47.22 | Qwell | new cellphones amaze me, heh |
| 05:47.30 | denon | if I use it for a solid day, its totally dead (1x cdma) |
| 05:47.32 | twisted[mobile] | denon, BT headset, desktop sync, etc |
| 05:47.33 | Qwell | mine has like...42 years of standby |
| 05:47.40 | denon | oh, im talking constant use .. |
| 05:47.40 | Qwell | give or take |
| 05:47.45 | denon | set the phone up next to the laptop and work via vpn |
| 05:48.02 | denon | bt headset or somethin, lasts a long time |
| 05:48.03 | twisted[mobile] | ah, see, that's what i'm trying to get working right now |
| 05:48.04 | file | denon: that's crazy talk |
| 05:48.20 | twisted[mobile] | i have full EDGE here, so I'd like to be able to take advantage of that |
| 05:48.30 | denon | yeah, evdo works pretty well |
| 05:48.38 | denon | though 1x works just fine for ssh and rdp |
| 05:48.42 | denon | and excellent for ICA |
| 05:48.55 | twisted[mobile] | yuck |
| 05:49.01 | twisted[mobile] | grps for ssh == shiat |
| 05:49.10 | twisted[mobile] | type a command, wait 5 minutes |
| 05:49.18 | denon | 1x its just fine |
| 05:49.22 | denon | and its great, when im out camping |
| 05:49.26 | denon | if I need to fix something or such |
| 05:49.30 | twisted[mobile] | well, if you're in the wilderness |
| 05:49.31 | denon | no need to drive to civilization |
| 05:49.36 | twisted[mobile] | i guess it's all you can get |
| 05:49.56 | denon | nah, with an 18dbi directional patch, you'd be surprised |
| 05:50.07 | denon | but cdma is certainly easier |
| 06:02.11 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 06:05.33 | *** join/#asterisk rowter (n=Silver@201.135.9.97) |
| 06:05.49 | rowter | anyone here from venezuela? that might had an experience with R2 |
| 06:10.00 | *** part/#asterisk sugardave (n=sugardav@cpe-66-68-165-207.austin.res.rr.com) |
| 06:13.07 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
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| 06:34.51 | rowter | anyone has seen -- Unicall/10 protocol error. Cause 32776 on a callout in venezuela unicall R2MFC? |
| 06:39.19 | L|NUX | can some one help me with asterfax |
| 06:50.29 | *** join/#asterisk brozer (n=MAGiC@c-69-138-29-18.hsd1.md.comcast.net) |
| 06:50.38 | brozer | hey |
| 06:51.11 | brozer | which sip services allow you to set your DID to what you want |
| 06:52.06 | brozer | ? |
| 06:52.52 | brozer | anyone? |
| 06:53.03 | brozer | recommend any SIP service for asterisk? |
| 06:56.24 | ionix | you mean provider? |
| 06:56.54 | docelmo | plainvoip does |
| 06:57.00 | x86 | brozer: are you talking about outbound Caller ID? |
| 06:57.09 | brozer | yes |
| 06:57.18 | brozer | ANI |
| 06:57.26 | brozer | outgoing |
| 06:57.27 | x86 | brozer: I run a service and allow all of my business plans to set their own ANI |
| 06:57.41 | x86 | brozer: http://www.shellshark.net/voip/ |
| 06:58.10 | brozer | very good |
| 06:58.13 | brozer | can I pm you bro? |
| 06:58.18 | x86 | absolutely |
| 07:01.19 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 07:02.43 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:04.14 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 07:07.36 | *** join/#asterisk livinded (n=livinded@cpe-24-24-186-88.socal.res.rr.com) |
| 07:09.17 | rowter | I have the same error posted here of venezuela http://www.voip-info.org/wiki/index.php?page=Asterisk%20MFC%20R2 |
| 07:09.36 | livinded | is iaxfwd still having problems? |
| 07:23.52 | *** join/#asterisk zedkatuf (n=audela@82-32-57-69.cable.ubr08.azte.blueyonder.co.uk) |
| 07:26.43 | livinded | zedkatuf: are you using fwd via iax? |
| 07:29.43 | rowter | anyone here has unicall working on venezuela? |
| 07:31.54 | rowter | anyone here has work out with CANTV of venezuela? |
| 07:37.53 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
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| 07:50.10 | *** part/#asterisk livinded (n=livinded@cpe-24-24-186-88.socal.res.rr.com) |
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| 08:32.23 | *** join/#asterisk prologic (n=prologic@60-240-230-244-vic-pppoe.tpgi.com.au) |
| 08:33.23 | prologic | http://www.techtopia.com.au/index.php/cPath/36_78 <-- to build an ip-pbx, I'd need an FSX module and a PCI FXO/FXS card right ? |
| 08:33.35 | prologic | Q: What's the difference between FXO and FXS ? |
| 08:34.37 | Qwell | ~fxofxs |
| 08:34.41 | jbot | fxofxs is probably An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 08:35.30 | prologic | ahh |
| 08:35.37 | prologic | so I would in fact need both an FXS and FXO module ? |
| 08:35.39 | prologic | one of each |
| 08:37.41 | L|NUX | any one help me with fax |
| 08:37.41 | L|NUX | ? |
| 08:38.42 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
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| 08:46.44 | *** part/#asterisk prologic (n=prologic@60-240-230-244-vic-pppoe.tpgi.com.au) |
| 08:47.00 | [Airwolf] | If I use a macro inside another macro does the call get returned to the first macro when the second one is completed ? |
| 08:48.58 | *** join/#asterisk brannfenix (i=brannfen@ip68-230-133-70.ri.ri.cox.net) |
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| 08:59.45 | brannfenix | is there anyone around that can help me with my zaptel drivers? |
| 08:59.51 | brannfenix | i have been googling for hours :( |
| 09:00.04 | brannfenix | i get this error line 0: Unable to open master device '/dev/zap/ctl' |
| 09:01.13 | brannfenix | and my modprobe doesnt work either.. i get FATAL: Module zaptel not found. and FATAL: Module wcfxo not found. FATAL: Error running install command for wcfxo |
| 09:01.21 | brannfenix | any ideas would be great |
| 09:02.01 | threat2 | hehe zaptel |
| 09:02.29 | threat2 | brannfenix, have you compiled the drivers? and if so did you use the same version of gcc to compile your kernel and zaptal drivers as well? |
| 09:03.22 | brannfenix | well its a fresh ubuntu 6.06.1 server install |
| 09:03.48 | brannfenix | now i was following the install docs but i think something didnt jive |
| 09:04.07 | brannfenix | i was working with http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation |
| 09:04.41 | brannfenix | now i got to the "modify the EXTRAVERSION statement in Makefile so it matches what you see in: cat /proc/version (see clarification in comments below) " step |
| 09:04.53 | brannfenix | and it started to go off the tracks |
| 09:05.20 | brannfenix | for example when i went to make the drivers i didnt have configure to run the ./configure |
| 09:05.38 | brannfenix | i downloaded the file from the main site so i doubt it was a bad one |
| 09:05.57 | brannfenix | i did the make linux26 and got this warning: WARNING: Symbol version dump /usr/src/linux-source-2.6.15/Module.symversis missing; modules will have no dependencies and modversions. |
| 09:06.02 | brannfenix | but it seemed to work |
| 09:06.45 | threat2 | brannfenix, ubuntu :/ I am a ubuntu user but I have noticed that alot of the "official" packages don't work that good, you may want to apt-get source it and compile build it for your syste,m |
| 09:06.55 | brannfenix | and i dont have a 50-udev.rules |
| 09:07.20 | threat2 | heh |
| 09:07.22 | brannfenix | your thinking just rebuild the whole kernel and give that a try? |
| 09:07.23 | threat2 | ubuntu :) |
| 09:07.34 | brannfenix | i just installed it so i dont have anything vested |
| 09:07.41 | threat2 | rebuild what ever package is giving you grief |
| 09:07.41 | brannfenix | i could careless what distro it is |
| 09:08.00 | brannfenix | thats the problem i dont have a clue what is making all these problems |
| 09:08.03 | threat2 | debian is alot stablier |
| 09:08.14 | threat2 | ubuntu is based off debian unstable |
| 09:08.30 | brannfenix | #1 distro.... bah |
| 09:08.32 | threat2 | of corse debian stable has old packages :S |
| 09:08.43 | brannfenix | i dont want to be bleeding edge i just want to work |
| 09:08.44 | brannfenix | heh |
| 09:08.48 | threat2 | so I guess it isa loose, loose sitch |
| 09:09.11 | threat2 | you give up features for stability, but then again I would prefer stability |
| 09:09.20 | brannfenix | same here |
| 09:09.40 | threat2 | although ubuntu isn't that bad, you probably just need to compile the source packages to resolve your problem(s) |
| 09:09.58 | brannfenix | which source packages th0? |
| 09:10.02 | brannfenix | thats where im lost |
| 09:10.11 | brannfenix | i dont have a clue waht is broken at this point |
| 09:10.15 | brannfenix | it looks like it all doesnt work |
| 09:12.22 | brannfenix | for example... is there any reason i cant do a ./configure on my zaptel-1.2.7 source? |
| 09:14.10 | brannfenix | my gut tells me this is a fair sized problem: WARNING: Symbol version dump /usr/src/linux-source-2.6.15/Module.symvers |
| 09:14.10 | brannfenix | <PROTECTED> |
| 09:14.33 | brannfenix | some people have seen it but there was no clear fix i could find :( |
| 09:42.35 | *** join/#asterisk svenna (n=svenna@p548D1F26.dip0.t-ipconnect.de) |
| 10:08.07 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 10:10.06 | *** join/#asterisk kindor (i=kindor@office.open-ict.nl) |
| 10:10.09 | kindor | heo |
| 10:10.11 | kindor | is there a way to, from the asterisk server, have a (hardware) ip phone call a certain number automaticly? e.g. from an application |
| 10:13.24 | Rawplayer | sure.. you can use AGI for that |
| 10:16.11 | kindor | thanks |
| 10:19.01 | kindor | hey this is cool :0 |
| 10:19.50 | Rawplayer | ofcourse it is ;) |
| 10:20.19 | [Airwolf] | Asterisk = coolness |
| 10:20.29 | neo | except its memory leaks |
| 10:20.31 | neo | :( |
| 10:20.56 | *** join/#asterisk Crshman (n=chatzill@dsl081-035-232.lax1.dsl.speakeasy.net) |
| 10:21.01 | [Airwolf] | And it's bug that it stop executing when you call a macro in a macro |
| 10:21.02 | [Airwolf] | :P |
| 10:21.17 | [Airwolf] | And the lack of good documentation. |
| 10:21.20 | Crshman | hello all, for questions about module code should i ask here or in dev? |
| 10:32.11 | *** join/#asterisk lucasvo (n=lucasvo@wservices.ch) |
| 10:32.13 | lucasvo | hi |
| 10:33.00 | lucasvo | how can I find out which codecs asterisk understand? I haven't compiled it my self and need to find out which calls I can allow |
| 10:38.31 | ltd | asterisk -r |
| 10:38.33 | ltd | show codecs |
| 10:42.14 | sumasuma | ltd: i don't think that is the right command |
| 10:42.40 | sumasuma | show translation gives you the codecs installed |
| 11:15.25 | *** join/#asterisk svenna (n=svenna@p548D1F26.dip0.t-ipconnect.de) |
| 11:22.59 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 11:24.27 | *** join/#asterisk prologic (n=prologic@60-240-230-244-vic-pppoe.tpgi.com.au) |
| 11:24.40 | prologic | Q: Is a PAP2 really needed ? |
| 11:29.55 | lucasvo | sumasuma: ok, show codecs gives me output speech, I have a friend which has allow all. |
| 11:30.16 | lucasvo | sumasuma: but I can't make it use spee |
| 11:32.01 | lucasvo | also my musiconhold is much too silent, how can I make it louder? |
| 11:32.30 | Rawplayer | compile it with -LOUD_NOISE |
| 11:32.37 | Rawplayer | sorry |
| 11:32.41 | Rawplayer | devil on my shoulder |
| 11:32.46 | Rawplayer | :\ |
| 11:33.35 | lucasvo | Rawplayer: any other options? |
| 11:33.45 | Rawplayer | i dont know, sorry |
| 11:34.00 | lucasvo | I read on voip-info.og that one should set mode to mp3 |
| 11:34.06 | lucasvo | but that doesn't work |
| 11:34.13 | lucasvo | mode => file is the only thing that works |
| 11:47.14 | ltd | sumasama: oops, i meant show audio codecs |
| 11:48.29 | ltd | sumasama: show codecs does give you a list of all codecs though, audio, video and image |
| 11:48.44 | FaithX | lucasvo: you can select the volume in moh.conf |
| 11:49.34 | *** join/#asterisk willy1234 (n=icechat5@62.231.36.194) |
| 11:49.45 | willy1234 | hi can u call ur asterisk box by putting its ip address in a sip client? |
| 11:50.17 | willy1234 | i get the pbx saying that its not a valid extension and then it hangs up |
| 11:50.50 | willy1234 | go u have to register an incoming extension and how do i do this? |
| 11:51.00 | FaithX | willy1234: asterisk is a PBX not a phone |
| 11:51.09 | *** join/#asterisk apardo (n=apardo@85-49-131-54.mad2.adsl.uni2.es) |
| 11:51.09 | *** join/#asterisk zaffa (i=zaffa@dsl-98-18-216-83.maxinetti.fi) |
| 11:51.57 | willy1234 | yeah but how do voip comanies use asterisk to recieve calls |
| 11:52.35 | FaithX | willy1234: Hey man, your phones are extensions that asterisk creates and handles |
| 11:53.17 | willy1234 | is it possible to call an extension using the publiv ipaddress |
| 11:53.24 | FaithX | willy1234: read the hitch hikers guide to * or one of the other "start me up" docs |
| 11:53.34 | prologic | can someone tell me what role a PAP2 plays in an IP-PBX (which is what I'm intending to build). Afaik they're to connect standard PTSN phones to IP-based telephony networks (such as one built with Asterisk and a WilcCard TDM400B + FXO/FXS) |
| 11:53.37 | willy1234 | ie 200@ipaddress |
| 11:53.41 | prologic | Is a PAP2 really needed though ? |
| 11:54.03 | FaithX | prologic: Linksys PAP2? |
| 11:54.07 | prologic | *nods* |
| 11:54.22 | prologic | Can't a standard phone be plugged into an FXS neways ? |
| 11:54.29 | prologic | or FXO (I might have mised them up) |
| 11:54.34 | willy1234 | hmm |
| 11:54.37 | FaithX | converst regulare phones 2 sip phones... but the pap2 is a pbx I think |
| 11:55.05 | prologic | well sorry not the pap2 then |
| 11:55.06 | willy1234 | its not a pbx ita an ata |
| 11:55.08 | prologic | the SAP I think |
| 11:55.09 | FaithX | prologic: FXS for phones |
| 11:55.09 | prologic | or something |
| 11:55.24 | prologic | can standard phones be plugged into FXS ports ? |
| 11:55.31 | willy1234 | it has 2 fxs ports |
| 11:55.34 | willy1234 | yup |
| 11:55.36 | FaithX | ok... yes plug pstn phones in they become sip extensions |
| 11:55.50 | prologic | right |
| 11:56.08 | prologic | so I don't need anything except a WildCard TDM400B and an FXO and FXS |
| 11:56.08 | willy1234 | the spa3000 has 1fxo and 1fxs |
| 11:56.35 | FaithX | you configure the sip accounts to authenticate as extensions on your * box or any sip provider |
| 11:56.46 | prologic | I was thinking of the spa2000 |
| 11:56.49 | willy1234 | has anyone used a linksys spa3000? |
| 11:56.58 | prologic | but then I don't need a spa or a pap2 right ? |
| 11:57.15 | FaithX | yup I have one in action |
| 11:57.32 | willy1234 | is it any good? |
| 11:57.51 | willy1234 | i heard the call quality is not great? |
| 11:58.33 | willy1234 | would a tdm400 be any better |
| 11:58.34 | FaithX | yes they are very good ultra configurable but you need a degree in supruaneese to drive the interface or follow some the howtos on the little beasties |
| 11:58.55 | willy1234 | hmmm.... |
| 11:58.56 | prologic | so I'm still confused :) why do I need a SPA if existing PTSN phones can be plugged into FXS ports |
| 11:59.00 | prologic | *sigh* :) |
| 11:59.04 | willy1234 | is it that difficult |
| 11:59.05 | FaithX | Call quality is fine... |
| 11:59.18 | FaithX | prologic: you don't ... if you have FXS ports |
| 11:59.30 | *** join/#asterisk RoyK (n=roy@static-213-115-144-122.sme.bredbandsbolaget.se) |
| 12:00.00 | willy1234 | ok i might order one |
| 12:00.19 | prologic | I suppose you'd only use a PAP2 or SPA to connect to an existing ip-telephony system such as one provided by your isp right ? |
| 12:00.23 | willy1234 | do they take long to configure |
| 12:00.42 | FaithX | spa is a home users type deal... it is what engin give away in AU |
| 12:01.27 | FaithX | It gives you failover... but any of those deals are good for a beginner to play with who doesn't want to fork out for a digium card |
| 12:02.13 | willy1234 | i was thinking of maybe getting a spa 3000 for work to conect the exising phoneline and phone to an asterisk pbx |
| 12:02.44 | prologic | thanks FaithX |
| 12:02.51 | prologic | clears thinks up slighty :) |
| 12:03.37 | FaithX | willy1234: yes that is a solution that works for one line... |
| 12:04.13 | FaithX | spa is very configurable you can configure each port separately or bridge them.... |
| 12:05.09 | willy1234 | we have one line at the minute |
| 12:05.38 | FaithX | one pstn line? |
| 12:05.42 | willy1234 | im thinking of getting the boss to get 2 isdn lines and an isdn card |
| 12:05.58 | willy1234 | one pstn line at the minute |
| 12:06.01 | *** join/#asterisk e-MAC (n=YMironek@82.207.125.177) |
| 12:06.30 | willy1234 | what card would u recomend for 2 isdn cards |
| 12:06.59 | FaithX | I would just buy an x100p for $10 and terminal iax2 account for incomming calls. With the PSTN as a failover. |
| 12:07.24 | FaithX | do a linehunt with the pstn last in line |
| 12:07.49 | FaithX | linehunt from your telco that is. They are typically very cheap |
| 12:08.09 | FaithX | s/terminal/terminate |
| 12:08.35 | RoyK | <PROTECTED> |
| 12:08.51 | prologic | Q: If I wanted multiple internal phones (extensions) would I need more FXS ports ? Or is there some other way to acheive this ? |
| 12:08.55 | willy1234 | is a line hunt when u have multiple lines and one phone number? |
| 12:09.15 | willy1234 | usb soft phones |
| 12:09.16 | FaithX | prologic: that is what IP phones are for |
| 12:09.30 | FaithX | willy1234: wash your mouth out |
| 12:09.39 | willy1234 | why? |
| 12:09.52 | FaithX | usb soft phones... this isn't skype |
| 12:09.53 | prologic | FaithX: ahhh, so just get a hole heap of IP phones ? :) |
| 12:10.01 | FaithX | prologic: that's it |
| 12:10.08 | prologic | excellent :) |
| 12:10.13 | FaithX | that's what asterisk is all about |
| 12:10.26 | willy1234 | ip phones are expensive |
| 12:10.42 | willy1234 | about $150 |
| 12:10.58 | Rawplayer | lol? |
| 12:11.00 | FaithX | willy1234: Well * is a business tool... IP Phones are business tools |
| 12:11.00 | willy1234 | usbpones about $20 |
| 12:11.03 | prologic | Are there any good providers in AU ? |
| 12:11.09 | *** join/#asterisk sip_me (n=ask@80.179.11.31.static.012.net.il) |
| 12:11.14 | prologic | techtopia.com.au is rather expensive and limited in the IP Phones |
| 12:11.14 | FaithX | prologic: are you in AU |
| 12:11.23 | prologic | yeap |
| 12:11.29 | FaithX | Where ? |
| 12:11.49 | zaffa | have anyone been doing outbound caller id hiding (with Zap-channels on E1)? i need to configure few caller ids to be hided... |
| 12:12.09 | FaithX | techtopia.com.au is rather expensive |
| 12:12.13 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 12:12.16 | prologic | Brisbane/QLD |
| 12:12.38 | FaithX | prologic: I could help... depends what you want |
| 12:12.44 | prologic | ahh nice |
| 12:12.59 | FaithX | I could sell you my demo linksys |
| 12:13.07 | prologic | well basically a digium card with 2 fxo/fxs ports, and a bunch of wifi (if possible) ip phones :) |
| 12:13.13 | FaithX | I have a snom in a box I would much prefer to use |
| 12:13.58 | FaithX | zyxel seem to have game stitched there |
| 12:14.04 | FaithX | but man they are expensive |
| 12:14.08 | willy1234 | our business is cheap the will send $40000 on 5 top of the range PCs but only $10-20 on per user on getting a phone extension |
| 12:14.47 | FaithX | willy1234: well that ain't gonna happen ... use idefisk |
| 12:15.01 | willy1234 | im using idefisk |
| 12:15.31 | FaithX | willy1234: bluetooth dongle and headsets |
| 12:15.57 | willy1234 | dam cheap gits, have the cheak of complaining about echo..what do they expect |
| 12:16.16 | willy1234 | good idea |
| 12:16.29 | FaithX | willy1234: fix the echo and they might give you some money to buy some real phones |
| 12:17.18 | FaithX | prologic: well if we could get the chanBT working ... they could all just use their mobile phones as extensions when they walked in the office. |
| 12:17.54 | AndyCap | FaithX: umm, do any phones actually support that? |
| 12:18.05 | FaithX | What? |
| 12:18.15 | willy1234 | chanBT sounds really cool |
| 12:18.31 | willy1234 | how does it work |
| 12:18.39 | AndyCap | FaithX: using the cellphone as a handset as you talk about, not as a audiogw to the gsm network |
| 12:19.09 | FaithX | AndyCap: well yes... if you can get flakey BT to work... |
| 12:19.45 | FaithX | AndyCap: I would much rather prefer an audiogw to the gsm network. |
| 12:20.06 | willy1234 | do u put out ur own gsm base station? |
| 12:20.24 | AndyCap | FaithX: that you can do with a phone. but I haven't seen a phone providing the other way round |
| 12:20.27 | willy1234 | cant find chanBT on google |
| 12:21.01 | AndyCap | chan_ is usually the right way of writing it |
| 12:21.15 | AndyCap | http://www.voip-info.org/wiki/view/Asterisk+Bluetooth+channels |
| 12:22.15 | FaithX | Anyhoo... that's just pipe dreaming at the moment... |
| 12:22.31 | FaithX | AndyCap: that's what MIAX is all about... |
| 12:22.47 | FaithX | if you can get that to work either |
| 12:23.16 | AndyCap | FaithX: vgsm. |
| 12:23.25 | willy1234 | so do the phones need to be bluetooth enabled |
| 12:24.16 | AndyCap | FaithX: but have you seen a phone actually supporting beeing used as a handset? |
| 12:24.59 | FaithX | Yes... |
| 12:25.09 | FaithX | miax but it is flakey |
| 12:25.18 | AndyCap | FaithX: which phone? |
| 12:27.17 | FaithX | SE Z800i , SE Z1010, Nokia 6310i, NEC 616V... but none work ... well... really... bits and pieces... |
| 12:28.10 | FaithX | uses phone as AG |
| 12:28.39 | FaithX | sorry headset profile I think... |
| 12:35.04 | *** join/#asterisk apardo (n=apardo@85-49-130-32.mad2.adsl.uni2.es) |
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| 12:41.01 | PanHat | ? |
| 12:41.56 | PanHat | anyone that can give come advice in geting a HFC ISDN PCI going in Trixbox? |
| 12:43.17 | *** join/#asterisk willy1234 (n=icechat5@62.231.36.194) |
| 12:43.17 | *** join/#asterisk dorel__ (n=liran@212.199.9.246.static.012.net.il) |
| 12:43.41 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:44.46 | FaithX | PanHat: what would you like to know? |
| 12:45.04 | FaithX | You need to run the build script for it at the console menu |
| 12:45.09 | *** join/#asterisk RoyK (n=roy@static-213-115-144-122.sme.bredbandsbolaget.se) |
| 12:45.34 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6F070.versanet.de) |
| 12:45.35 | benjk | use BRIstuff |
| 12:45.42 | PanHat | I have been fighting with this for a while now. I can see the card in the asterisk CLI zap show status |
| 12:45.59 | PanHat | BUT I can not get it configured in the FOP |
| 12:46.46 | PanHat | I have run the update and the script |
| 12:47.32 | FaithX | FOP? |
| 12:47.33 | PanHat | install-ZAPHFC |
| 12:47.45 | PanHat | Flash Operator Panel |
| 12:48.19 | prologic | Q: Where does one get nice voice files from ? Grettings, messages, wnad hwat not... Afaik most TTS engines are still rather terrible :) |
| 12:48.26 | FaithX | I had to configure zaptel.conf by hand |
| 12:48.42 | FaithX | prologic: pass |
| 12:49.10 | FaithX | PanHat: so you have the card working? |
| 12:49.13 | RoyK | benjk: why don't you use visdn or something instead? bristuff is a little ancient... |
| 12:49.30 | benjk | no I use BRIstuff |
| 12:49.43 | FaithX | I use bristuff... |
| 12:49.56 | benjk | more straightforward |
| 12:50.05 | FaithX | it's true my * is stuffed |
| 12:50.21 | prologic | FaithX, pass ? :) |
| 12:50.57 | PanHat | I tried mISDN but got further with BRIstuff |
| 12:51.02 | benjk | also, I don't believe in the proliferation of chan_foo, chan_bar, chan_baz, chan_world_and_dog |
| 12:51.19 | benjk | there should only be one channel type |
| 12:51.42 | benjk | and the technology used should be managed by an abstraction layer library |
| 12:51.52 | benjk | in other words, Unicall |
| 12:52.14 | benjk | until that's ready, I am going to use BRIstuff |
| 12:52.51 | PanHat | I am willing to try visdn but looked a bit long winded to get going |
| 12:53.27 | benjk | I am willing to try any ISDN library as long as it doesn't come with its own channel driver |
| 12:54.11 | benjk | or more precisely, as long as it doesn't need its own channel driver |
| 12:55.56 | PanHat | Any suggestion why I can not see my ISDN trunk in the Flash Operator Panel |
| 12:56.20 | benjk | what channel type is it? |
| 12:56.37 | PanHat | BRI |
| 12:56.45 | benjk | chan_bri ? |
| 12:57.14 | benjk | how do you dial on those? Dial(BRI/g1...) ? |
| 12:57.29 | PanHat | ZAP/g0 |
| 12:57.41 | benjk | so its using BRIstuff then |
| 12:57.49 | PanHat | Yes |
| 12:58.18 | benjk | are the channels up and running? |
| 12:59.12 | PanHat | if i do zap show status I can see the card - I presume that * is happy. Yes 2B + 1 D |
| 12:59.31 | FaithX | vISDN looks very nice |
| 12:59.36 | benjk | zap show status? |
| 12:59.42 | FaithX | BRIstuff works |
| 12:59.46 | benjk | you mean zap show channels |
| 12:59.55 | benjk | what if you do zap show channel 1 |
| 13:00.25 | FaithX | PanHat: so you can use your BRI as ZAP channels? |
| 13:00.37 | benjk | that's the whole point of BRIstuff |
| 13:00.53 | benjk | it uses drivers that mimic a zaptel interface |
| 13:01.04 | benjk | and it uses Libpri as well |
| 13:01.12 | benjk | patched libpri |
| 13:01.24 | PanHat | Looks like I broke it again elier I could do the channel 1 & 2 |
| 13:01.37 | benjk | but it makes BRI channels look just like PRI channels |
| 13:02.03 | FaithX | PanHat: you have to set the zap channels up as trunks if you want them to appear in FOP |
| 13:02.17 | PanHat | yes |
| 13:02.33 | FaithX | and... |
| 13:03.04 | FaithX | PanHat: can you take calls and dial over your card? |
| 13:03.21 | PanHat | no |
| 13:03.47 | benjk | well, there's your problem |
| 13:03.48 | benjk | that's why I asked you to do zap show channel 1 |
| 13:04.53 | PanHat | but I need to see the trunk to define the outgoung route - correct? |
| 13:05.40 | benjk | first you need to get your channels actually up and working |
| 13:05.52 | FaithX | PanHat: what do you get when you run ztconfig |
| 13:06.08 | PanHat | any tools to check this with? |
| 13:06.12 | FaithX | PanHat: as I said I had to set mine up |
| 13:06.23 | FaithX | PanHat: at the console... |
| 13:06.40 | FaithX | ssh into the box |
| 13:07.28 | PanHat | when i do the ztcfg -vvv i can see the 3 channels1,2,3 |
| 13:08.01 | FaithX | what does your zaptel.conf have in? |
| 13:08.09 | FaithX | did you just do the auto config? |
| 13:08.34 | PanHat | yes I ran the auto config |
| 13:08.54 | PanHat | in the zaptel.conf i have: |
| 13:09.04 | PanHat | span=1,1,3,ccs,ami |
| 13:09.04 | PanHat | bchan=1-2 |
| 13:09.04 | PanHat | dchan=3 |
| 13:09.36 | FaithX | and your zapata.conf? |
| 13:10.25 | willy1234 | so can u call the digital reception thing using your public ip address? |
| 13:10.29 | FaithX | in zaptel.conf is your loadzone and default zone correct? |
| 13:10.46 | RoyK | can someone tell me wtf loadzone really is? |
| 13:10.52 | PanHat | i have uncommented include zapata-BRI-HFC.conf |
| 13:11.01 | FaithX | ? |
| 13:11.35 | PanHat | this is at the bottom of the zapata.conf |
| 13:11.48 | lucasvo | I have a problem with agents: |
| 13:11.57 | FaithX | It did not autoconfigure correctly for me... |
| 13:12.01 | lucasvo | agent => 1001,5678,Lucas Vogelsang |
| 13:12.21 | PanHat | what should i see? |
| 13:12.37 | lucasvo | is in agents.conf and in extensions.conf it says: exten=> 2001,1,AgentCallbackLogin(||${CALLERIDNUM}@mycontext) |
| 13:12.41 | FaithX | switchtype = euroisdn |
| 13:12.42 | FaithX | signalling = bri_cpe_ptmp |
| 13:12.42 | FaithX | channel => 1-2 |
| 13:12.56 | FaithX | Now this is for telstra in australia |
| 13:13.08 | PanHat | that is in the zapata-BRI-HFC.conf file |
| 13:13.14 | lucasvo | however I can't login with that data. it says: Please enter the agent number followed by #. I do it and then it says: login incorrect |
| 13:13.45 | FaithX | PanHat: then restart * and you could be a go-er |
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| 13:15.00 | PanHat | have done that but can not show the channel 1/2 |
| 13:15.14 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 13:15.57 | PanHat | if i do zap show status i get: |
| 13:16.01 | PanHat | HFC-S PCI A ISDN card 0 [TE] layer 1 A OK 0 0 0 |
| 13:20.41 | PanHat | when i do a zap show channel 1 i get - Unable to find given channel 1 |
| 13:21.05 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.20) |
| 13:22.56 | FaithX | PanHat: then I guess your zapata.conf is not configured correctly |
| 13:23.07 | FaithX | or your zapata-BRI-HFC.conf |
| 13:24.08 | PanHat | i will move the stuff from zapata-BRI-HFC to zapata.conf and see if that changes anything. |
| 13:24.19 | FaithX | PanHat: ... |
| 13:24.35 | FaithX | Oh you removed the hash from the include? |
| 13:24.57 | PanHat | yes so it should be included |
| 13:24.59 | FaithX | paste up your zapata.conf |
| 13:25.12 | PanHat | busy now |
| 13:25.19 | FaithX | PanHat: no... that is not how the hashes work :-| |
| 13:25.24 | FaithX | the hashes stay |
| 13:25.58 | PanHat | ok me being ... i will put them back |
| 13:27.09 | FaithX | pastebin your zapata.conf and zapata-BRI-HFC.conf |
| 13:27.17 | FaithX | so I can see them |
| 13:27.31 | FaithX | and your zaptel.conf |
| 13:27.45 | FaithX | PanHat: where are you? |
| 13:27.56 | PanHat | pastebin? South Africa |
| 13:29.08 | *** part/#asterisk prologic (n=prologic@60-240-230-244-vic-pppoe.tpgi.com.au) |
| 13:30.47 | FaithX | PanHat: http://sial.org/pbot/ |
| 13:32.31 | PanHat | This content is stored as http://sial.org/pbot/18950. |
| 13:33.52 | FaithX | give me the other 2 files pls |
| 13:34.23 | RoyK | ~pb |
| 13:34.27 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 13:35.04 | PanHat | This content is stored as http://sial.org/pbot/18951. |
| 13:36.42 | FaithX | http://sial.org/pbot/18952 |
| 13:40.00 | FaithX | PanHat: move your file sideways and try mine... |
| 13:40.14 | FaithX | you will need to reload *... |
| 13:40.31 | FaithX | PanHat: assuming that South Africa uses ptmp |
| 13:40.40 | PanHat | just di no I can see the channels |
| 13:40.54 | FaithX | Ok... |
| 13:40.56 | PanHat | just di no I can see the channel 1 & 2 |
| 13:41.26 | FaithX | let me see your zaptel.conf so that your dchannel is correct |
| 13:41.36 | sip_me | Hi, is there any open source SIP/RTP Voip Client that interop with Asterisk? |
| 13:41.39 | FaithX | PanHat: you had PRI dialing in them |
| 13:41.55 | FaithX | sip_me: softphone? |
| 13:42.11 | sip_me | FaithX, Yep |
| 13:42.16 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 13:42.22 | FaithX | yea lots |
| 13:42.34 | delmar | anyone here familiar with MCC ? |
| 13:42.34 | FaithX | sip_me: running linux desktop? |
| 13:43.02 | sip_me | I have both Linux and Win2K. Preferably one that offers both options. |
| 13:43.13 | PanHat | This content is stored as http://sial.org/pbot/18954. |
| 13:43.15 | FaithX | PanHat: Like I said I had to do it zapata.conf by hand |
| 13:43.34 | FaithX | sip_me: xten lite |
| 13:43.58 | sip_me | FaithX, I thought x-lite is commercial. |
| 13:44.07 | sip_me | Is it open source? |
| 13:44.29 | FaithX | sip_me: http://asteriskguru.com/idefisk/free/ |
| 13:44.47 | FaithX | You didn't ask OS |
| 13:45.12 | FaithX | sorry yes you did |
| 13:46.19 | *** join/#asterisk daysmen3 (n=primus@host86-139-118-86.range86-139.btcentralplus.com) |
| 13:46.51 | FaithX | PanHat: you need to change your zone in zaptel.conf to match your local so it will detect properly |
| 13:47.02 | FaithX | other than that you should be set ... |
| 13:49.06 | PanHat | just did that - and restarted |
| 13:50.44 | FaithX | PanHat: have fun |
| 13:51.16 | dorel__ | i know that agi is used to run scripts from within asterisk (like during a call) but can it work the other way? can a script tell asterisk to make a call and do x, y, z? |
| 13:52.23 | PanHat | FaithX: i see both my lines in flash panel but they are red - is this a signaling issue? |
| 13:52.25 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
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| 13:53.30 | RoyK | :þ |
| 13:53.32 | sip_me | FaithX, IDEfisk is freeware but It does not seem to be an Open Source. |
| 13:54.25 | EyeCue | its not :D |
| 13:54.41 | *** join/#asterisk evisu (n=hIRC@bzq-88-155-80-250.red.bezeqint.net) |
| 13:55.15 | sip_me | Can anyone recommend an Open Source softphone that works well with Asterisk? |
| 13:55.51 | sip_me | (One using OSIP and possibly Exosip is a prefered option..) |
| 13:55.52 | RoyK | yakaphone, perhaps |
| 13:56.11 | RoyK | that's iax-based |
| 13:56.55 | RoyK | multiplatform, open source (although I don't know the exact license) |
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| 15:12.49 | zaffa | anyone else had problems with SetCallerPress(prohib)? It seems to work (at least debug messages of PRI seems ok) but the number is still displayed.. Can operator override these settings? |
| 15:13.45 | *** join/#asterisk evilbit (i=hhoffman@gateway/tor/x-29399b4891f664c4) |
| 15:14.27 | evilbit | hi, is anyone using sphinx with asterisk? and if so are you using v2, 3, or 4? |
| 15:19.12 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 15:24.12 | ionix | how can I increment a variable in extensions.conf? |
| 15:24.27 | ionix | something like Set(tout=tout+1) |
| 15:27.22 | evilbit | ionix: something like Set(COUNT=$[${COUNT} + 1]) |
| 15:28.29 | ionix | ok |
| 15:28.37 | ionix | and to print a variable on the CLI? |
| 15:30.20 | evilbit | no idea, I'd prolly read the manual to find out |
| 15:30.27 | Rawplayer | whats a manual? |
| 15:30.47 | evilbit | s/manual/docs/ ;-) |
| 15:31.10 | ionix | oky |
| 15:32.53 | ionix | NoOp(${bleh}) |
| 15:36.52 | *** join/#asterisk liran_ (i=liran@212.199.177.203.static.012.net.il) |
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| 15:45.14 | *** join/#asterisk turth (n=MAGiC@69-171-129-0.clvdoh.adelphia.net) |
| 15:45.41 | turth | hey |
| 15:46.02 | turth | is there anyway to make asterisk register through a socks proxy? |
| 15:46.05 | *** join/#asterisk pbx1 (i=pbx1@netblock-66-245-195-206.dslextreme.com) |
| 15:51.22 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
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| 16:04.59 | *** join/#asterisk ACiDV (n=acidv@modemcable247.11-37-24.mc.videotron.ca) |
| 16:05.47 | ACiDV | Hi... if I get this message 5-10 times per seconds since this morning, does it can be a network problem ? chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission XXX@YYY for seqno 102 (Critical Request) |
| 16:07.04 | ACiDV | agents doesnt receive call since this morning, on a setup with 100 SIP phone (Aastra 480i) and 8xT1 (TE406)... asterisk restarted, nothing changed |
| 16:08.15 | file | it tried to send a packet, and got no acknowledgement that it was received |
| 16:09.30 | turth | is there anyway to make asterisk register through a socks proxy? |
| 16:10.49 | file | turth: nothing is built in for that |
| 16:13.03 | jbalcomb | does the default section of queues.conf work like i hope it does? if i put options in there it'll use it for all my queues? |
| 16:15.05 | sevard | god damnit |
| 16:15.28 | sevard | isn't there a way to define a field in sed to replicate? |
| 16:16.43 | jbalcomb | sevard: are refering to the use of () to save the match to a variable? |
| 16:17.30 | sevard | I've been trying to do this for about 45 minutes but still can't find it in the docs, i swear there was a way to define a field in sed to replicate, like a file with a list of "cp .<paths>" you'd have sed -e s/^...#1$/ |
| 16:18.02 | sevard | like if you have a certian field in the file that you want to replicate |
| 16:19.05 | jbalcomb | sevard: perhaps this is a better job for awk? |
| 16:19.55 | *** join/#asterisk niZon (n=ilt@S0106beefd4cecc3d.wp.shawcable.net) |
| 16:20.10 | sevard | i loathe awk |
| 16:20.41 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 16:23.15 | jbalcomb | hrmm.. vi, grep, sed, and awk are the basic building blocks of all great gurus. :) |
| 16:24.19 | sevard | vi?! |
| 16:24.57 | jbalcomb | umm.. yeah? please tell me you don't use some nub editor like pico, nano, or joe? |
| 16:25.07 | sevard | nano is the shit, dawg. |
| 16:25.30 | DrkShdw | please tell me we're not about to have another nub editor debate :) |
| 16:25.31 | jbalcomb | its named for the size of the brains of people who /need/ it |
| 16:25.36 | sevard | seriously, when i want to edit something i want to edit it, not fly down to florida to take a 8 day course on using a shit ass editor. |
| 16:25.41 | jbalcomb | DrkShdw: has that happened before? ;) |
| 16:26.11 | jbalcomb | sevard: haha.. i'll bet you learned about computers users windows first... |
| 16:26.32 | jbalcomb | s/users/using |
| 16:26.37 | DrkShdw | jbalcomb: if by "happened before" you mean "editor debates have occurred in channels even REMOTELY related to *nix in any way shape or form" then that would be a yes. ;) |
| 16:28.01 | sevard | jbalcomb: When I need to edit a file. I use pico. When I need complicated string editing and fancy replacement I'll use sed. I don't need 36 buffers in an editor where I have to exclusivly switch from interactive and command mode. Reducing the amount of keystrokes is what I'm all about. |
| 16:28.05 | jbalcomb | DrkShdw: yeah, i seem to recollect seeing one once. =)i've had a few BOFH moments because 14 developers require 6 different editors to be install on each of our servers. |
| 16:28.28 | turth | is there asterisk home for windows? |
| 16:28.44 | jbalcomb | turth: yeah, its over in #freepbx though |
| 16:28.50 | DrkShdw | *shrug* I'm of the mindset that no matter what editor I use, it affects nobody else here in any way, so what does it matter ;-) |
| 16:29.05 | benjk | there is no such thing for Windows |
| 16:29.22 | benjk | there is AstWind32 though |
| 16:29.35 | benjk | but its command line based |
| 16:29.43 | DrkShdw | turth: I've read about people using asterisk @home, (now trixbox) in VMWare with varying degrees of success |
| 16:29.45 | jbalcomb | DrkShdw: i can appreciate that point of veiw |
| 16:29.54 | turth | oh so doesnt have an online GUI |
| 16:30.06 | benjk | it has a GUI terminal |
| 16:30.10 | turth | yea ye ai run my pbx in vmware no |
| 16:30.13 | benjk | for the Asterisk console |
| 16:30.54 | benjk | http://www.asteriskwin32.com/ |
| 16:31.31 | DrkShdw | if you want a pretty windows commercial solution, I've used pbxnsip before, for windows.. not too bad |
| 16:31.56 | smackus2 | ok, i am looking for a bidder on something i need done before Monday morning. I have been trying to make queuing work for a few days now. The way I want it to work is like AgentCallbackLogin, but that crashes the system. So I have been trying with AddQueueMember. The problem with that is that I have not been able to make it so that when an agent phone is offhook a call from the queue is not presented to that phone. any takers, PM |
| 16:32.51 | *** join/#asterisk Mike (n=mike@201.112.50.158) |
| 16:32.51 | *** join/#asterisk willy1234 (n=icechat5@62.231.36.194) |
| 16:33.01 | willy1234 | hmmmm... |
| 16:33.24 | willy1234 | so can u call ur asterisk by its ipaddress |
| 16:33.28 | turth | doesnt look as good for windows... So thers no way to make asterisk register through a socks proxy? |
| 16:33.46 | benjk | actuall it does have a GUI manager, but thats not free |
| 16:34.10 | willy1234 | i get invalid extension check number and try again |
| 16:34.27 | willy1234 | please help |
| 16:36.04 | smackus2 | willy1234: what are you trying to do? share your code with us so we can see what you are doing. |
| 16:36.40 | sevard | noes! that's illegal! |
| 16:36.47 | sevard | DMCA DMCA |
| 16:37.39 | willy1234 | i dot a DID number from esms.com and i typed in the ipadress of my server it rings but says thats not a valid extension |
| 16:38.40 | willy1234 | do i have to set an incomming route? |
| 16:39.00 | smackus2 | yes |
| 16:39.07 | willy1234 | i have an any/any set up to the digital receptionest |
| 16:39.23 | smackus2 | once again. share your code |
| 16:39.27 | smackus2 | pastebin.ca |
| 16:39.34 | willy1234 | dont have any code |
| 16:39.56 | smackus2 | then how do you expect asterisk to know what you want that call to do? |
| 16:40.25 | willy1234 | so thats why its comp[laining i guess |
| 16:40.38 | smackus2 | have you set asterisk up? |
| 16:40.45 | willy1234 | yup |
| 16:40.56 | smackus2 | so you have an extensions.conf file then, right? |
| 16:41.03 | willy1234 | yup |
| 16:41.22 | smackus2 | have you entered your extensions in there? where you want that incomming DID to go? |
| 16:42.29 | smackus2 | ? |
| 16:43.52 | willy1234 | So what am i looking for |
| 16:44.06 | willy1234 | i used freepbx to set it up |
| 16:44.26 | turth | can freepbx run through a sock proxy? |
| 16:44.48 | smackus2 | #freepbx |
| 16:45.13 | smackus2 | or http://www.voip-info.org/wiki/ |
| 16:46.05 | willy1234 | http://pastebin.ca/129690 |
| 16:47.02 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 16:47.11 | smackus2 | so you want this DID to ring exten 205? |
| 16:48.34 | willy1234 | no the recepionest but 205 will do for now |
| 16:48.48 | smackus2 | well, you are telling it to go to 205 |
| 16:49.42 | willy1234 | it doesnt go to extension 205 though |
| 16:50.14 | smackus2 | you need to show all of your extension.conf and what I would do is show the output from your CLI> promt while dialing it as well. |
| 16:53.40 | smackus2 | no takers on making my queuing work? I am offering to pay someone to do it, but no one PM'ed me |
| 16:53.52 | sevard | what's wrong now? |
| 16:53.53 | jbalcomb | woot! asterisk MOH + 16 speaker phones makes a great sound system while working on the weekends |
| 16:54.07 | smackus2 | sevard: you asking me? |
| 16:54.10 | sevard | yeha |
| 16:54.13 | sevard | yeah. |
| 16:54.39 | smackus2 | i have the queuing working.. i am adding agents, but if the phone is opff hook, queued calls are still presented to the phone. I need that not to happen. |
| 16:55.00 | smackus2 | the only think i have found that I like is AgentCallbackLogin, but it crashed quite often |
| 16:55.21 | sevard | odd, I've never had AgentCallBackLogin crash on me |
| 16:55.29 | smackus2 | you are using it? |
| 16:55.32 | Qwell | sevard: You're one of the lucky few... |
| 16:55.36 | smackus2 | how many queued calls and agents? |
| 16:55.39 | sevard | was using it, not any more |
| 16:55.51 | sevard | about 10 agents |
| 16:55.58 | sevard | one call queue |
| 16:56.09 | smackus2 | http://bugs.digium.com/view.php?id=6626 |
| 16:56.18 | sevard | odd |
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| 16:57.01 | smackus2 | so i have been trying to make AddQueueMember do what I want it to. |
| 16:57.15 | Qwell | smackus2: haven't figured out group counts? |
| 16:57.23 | smackus2 | no |
| 16:57.36 | Qwell | it's just the GROUPCOUNT function or whatever, and local channels |
| 16:57.57 | Qwell | there will be examples coming from Digium fairly soon, I believe |
| 16:58.02 | smackus2 | are you willing to walk me through it? |
| 16:58.15 | Qwell | I don't know exactly how the group count works |
| 16:59.25 | smackus2 | isnt group count depreciated? |
| 16:59.33 | Qwell | deprecated...and no |
| 16:59.51 | Qwell | (the i makes a huge difference) |
| 17:00.32 | smackus2 | was just reading http://bugs.digium.com/view.php?id=4133 |
| 17:01.14 | Qwell | somebody find me some neat hacks for my motorola v195 :p |
| 17:02.27 | evilbit | anyone using sphinx? if so what version? the voip-info how-to uses v2 but the current release is v4 |
| 17:04.32 | smackus2 | Qwell: do you want to take a crack at making it work? I am paying to get it working. |
| 17:04.36 | smackus2 | out of time on my deadline |
| 17:05.31 | Qwell | well...I don't know if my new employer will let me take on clients today. :) |
| 17:05.42 | Qwell | They basically said "no new clients", but I don't start until tomorrow, technically |
| 17:06.07 | smackus2 | anyone else out there? everyone makes it sound easy... but I can't figure it out. |
| 17:06.09 | smackus2 | need help |
| 17:06.31 | *** join/#asterisk chiardon (i=Dorphals@pcsp168-254.supercabletv.net.co) |
| 17:06.56 | chiardon | hello, anybody knows whats the maximum debug level I can have for a channel? |
| 17:07.04 | Qwell | tell you what.. If I help you, and you just so happen to donate a few bucks to my laptop fund... |
| 17:07.16 | chiardon | ebug level <level> [filename] |
| 17:07.16 | chiardon | <PROTECTED> |
| 17:07.17 | chiardon | is specified, debugging will be limited to just that file. |
| 17:07.25 | smackus2 | pm me |
| 17:07.37 | chiardon | That is if I specify filename debugging will be logged to thef ile? |
| 17:10.27 | quid246 | if you have a large # of SIP/IAX clients, is it more efficient resouce wise to put them in a MySQL DB or flat file? |
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| 17:14.12 | zock | Hi. |
| 17:14.15 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:16.08 | zock | Small question about zaphfc...All seems to be working with 2 hfc-boards...but the machine is damn slow and i get some messages in my syslog: |
| 17:16.15 | zock | Aug 13 19:07:48 asterisk last message repeated 43 times |
| 17:16.15 | zock | Aug 13 19:07:48 asterisk kernel: zaphfc: sync lost, pci performance tnabled. |
| 17:16.15 | zock | Aug 13 19:07:48 asterisk kernel: zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled. |
| 17:16.40 | zock | Both cards have diffrent IRQ's... and i am using acutal stable bristuff. |
| 17:16.46 | florz | zock: try my patch =:-) |
| 17:16.48 | zock | What could be the problem? |
| 17:17.19 | zock | florz: hm, i'll google it right away...one sec. |
| 17:17.52 | florz | zock: http://zaphfc.florz.dyndns.org/ |
| 17:18.03 | zock | florz: http://www.asterisk.li/florz.htm <- That one okay too? |
| 17:19.03 | florz | zock: Well, it links to http://zaphfc.florz.dyndns.org/, too =:-) |
| 17:19.26 | zock | florz: k ... do i need to match the bristuff-version? |
| 17:19.45 | florz | kindof |
| 17:20.09 | zock | florz: i used "bristuff-0.2.0-RC8r" for compiling, but can't find that version of your patch... hm... |
| 17:20.27 | zock | florz: so it's "zaphfc_0.2.0-RC8q_florz-9.1.diff.gz" ? |
| 17:20.41 | florz | yep, that should do |
| 17:20.47 | zock | i'll try. |
| 17:21.49 | zock | florz: 3 hunks fail to patch :/ |
| 17:21.53 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 17:22.16 | florz | Hmm, let's see ... |
| 17:22.44 | zock | patching file zaphfc/Makefile |
| 17:22.44 | zock | Hunk #1 FAILED at 1. |
| 17:22.44 | zock | 1 out of 2 hunks FAILED -- saving rejects to file zaphfc/Makefile.rej |
| 17:22.48 | zock | patching file zaphfc/zaphfc.c |
| 17:22.48 | zock | Hunk #22 FAILED at 772. |
| 17:22.48 | zock | Hunk #26 FAILED at 920. |
| 17:22.49 | zock | 2 out of 26 hunks FAILED -- saving rejects to file zaphfc/zaphfc.c.rej |
| 17:22.49 | zock | patching file zaphfc/zaphfc.h |
| 17:23.20 | zock | Invocation: asterisk:/usr/src/bristuff-0.2.0-RC8r# zcat zaphfc_0.2.0-RC8q_florz-9.1.diff.gz | patch -p1 |
| 17:25.28 | florz | well, wait a sec., I'll merge the changes and create a new patch ... |
| 17:25.43 | zock | florz: :D thanks! |
| 17:32.12 | zock | Another quick configuration question... should be a normal situation... i want callers that wait lets say 30 seconds in queue to be transfered to voicemail automaticaly. I used the timeout-param in the queues.conf, and added some steps to the extensions.conf ... but the calles keeps stuck in the queue and after timeout the same queue-member is ringing again :P ... any suggestions? |
| 17:33.41 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 17:34.21 | *** join/#asterisk xnon (i=xnon@200.82.222.64) |
| 17:34.27 | xnon | hello friends |
| 17:35.15 | *** join/#asterisk svenna (n=svenna@p548D08A5.dip0.t-ipconnect.de) |
| 17:35.15 | xnon | i have a sipura spa 1001 and i want to know how i can reset default values because i dont know what is the user and password for him |
| 17:39.15 | benjk | zock, not finished yet, but this will eventually do what you want ... |
| 17:39.15 | benjk | http://www.sunrise-tel.com/benjk/OriginateCallAPI/call_actions.h |
| 17:39.49 | zock | benjk: Hm? |
| 17:45.29 | florz | zock: there you go ... |
| 17:46.06 | zock | florz: thx, i'll try. |
| 17:47.09 | jbalcomb | Is it possible to adjust the MOH volume? |
| 17:47.26 | benjk | using sox, yes |
| 17:47.38 | benjk | or madplay |
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| 17:47.47 | *** part/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 17:47.50 | jbalcomb | ah, ok |
| 17:52.28 | zock | florz: patching works... just compiling...usage as the zaphfc-module before? |
| 17:53.50 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 17:55.37 | florz | zock: should work - if there are still problems, the web page describes how to tweak/use the "advanced features" of the patched driver |
| 17:57.17 | xnon | i have a SIPURA 1001 and i wann know how i can reset to default values ? |
| 17:57.30 | xnon | i dont know what is the user and password for him |
| 17:57.35 | xnon | anybody know how? |
| 17:57.49 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 17:58.16 | zock | ah cool, the config-problem is also solved. Now everything behaves as i wished :) |
| 17:58.40 | zock | florz: patch seems to work for now. but the problem occured mostly some time after asterisk was started, not from the beginning. |
| 17:58.51 | zock | florz: but i think it solved the problem. thanks! |
| 18:01.27 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 18:08.32 | *** join/#asterisk sugardave (n=sugardav@cpe-70-112-127-158.austin.res.rr.com) |
| 18:09.09 | sugardave | anybody using VoicePulse Connect? I moved my * box to a new location and now I can't receive inbound DID, but I can still call out. |
| 18:10.06 | xnon | i need help for reset to defaults values my sipura 1001 because i cant remember my username and password anybody can help me? |
| 18:14.49 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 18:20.47 | file | xnon: have you tried looking for the manual? |
| 18:25.34 | quid246 | I messaged the guy with how to do it and he didn't tell me whether it worked or not |
| 18:25.38 | xnon | yes |
| 18:25.41 | xnon | of course |
| 18:25.45 | quid246 | did it work? |
| 18:25.53 | xnon | no! |
| 18:25.55 | xnon | :( |
| 18:26.11 | quid246 | is it locked by a provider? |
| 18:26.13 | xnon | i dont know the user name and password to enter de configuration |
| 18:26.43 | xnon | the grandstream handy tone can be reset with a mAC address |
| 18:26.54 | xnon | but sipure 1001 i dont know how! |
| 18:27.03 | quid246 | xnon: Here, I forgot something * * * * 73738 # 1 |
| 18:27.17 | quid246 | try that |
| 18:27.25 | quid246 | and report back |
| 18:28.19 | xnon | friends all be right back |
| 18:32.17 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:41.25 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 18:43.46 | supjigatr | Anyone using a a104d with a circuit from the telco with other DS1's going to local? I'm having trouble accepting timing from telco and applying that timing to all other ports. |
| 18:49.47 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:49.56 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 18:51.30 | supjigatr | Anyone using a a104d with a circuit from the telco with other DS1's going to local? I'm having trouble accepting timing from telco and applying that timing to all other ports. |
| 18:54.26 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 18:54.44 | quid246 | Well I suppose the Sipura reset worked. |
| 18:54.57 | quid246 | what a loser for not even coming back to say thanks |
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| 19:04.00 | *** join/#asterisk rg1_ (n=rg1@www.airlinksystems.com) |
| 19:07.15 | rg1_ | I am setting a caller-id from one asterisk box that is making a call out via IAX2 connection to another asterisk box that has the PRI card in it - the caller-id on the receiving end is NOT getting the caller-id I put on box-1 |
| 19:07.39 | rg1_ | where on the asterisk-box-2-with-PRI would i look to see if there is something that is perhaps overriding that? |
| 19:09.30 | *** join/#asterisk zippytech (n=zippytec@shop.zippytech.com) |
| 19:11.15 | *** join/#asterisk _deg_ (n=deg@200.181.137.62) |
| 19:11.52 | I-MOD | rg1_, try making the iax2 connection between the two boxes a trunk connection |
| 19:12.12 | I-MOD | and make sure you don't have a callerid set in iax.conf on the machine that has the pri |
| 19:12.16 | rg1_ | i-mod - i think they might be setup that way right now, but how would i check/verify that? |
| 19:12.35 | I-MOD | in the peer/user definition, you should have "trunk=yes" |
| 19:12.47 | rg1_ | is that in iax2.conf? |
| 19:12.52 | I-MOD | iax.conf |
| 19:12.59 | rg1_ | on both machines? |
| 19:13.01 | I-MOD | yes |
| 19:13.12 | rg1_ | ok, chcking on that now.... thanks |
| 19:14.05 | zippytech | any one know how to add a delay to outging pot calls to trixbox, i am using a cfg edit of exten => _1NXXNXXXXXX,1,Macro(dialout-trunk,1,w${EXTEN},,) in the extensions_additional.conf file , but it gets replaced on every reload |
| 19:14.31 | zippytech | i add the w right before the $ |
| 19:15.32 | I-MOD | yeah, the w is what you want, but you're going to have to add that in teh trixbox gui so it doesnt get overwritten |
| 19:16.20 | zippytech | i have yet to find a good place for it even used phpmyadmin |
| 19:16.49 | I-MOD | try #freepbx for help w/ options on the gui |
| 19:17.28 | zippytech | thanks |
| 19:17.50 | supjigatr | Does anyone know if the sangoma a104d has a signal clock on the card? |
| 19:18.20 | I-MOD | i would hope/assume so, but i've never used a sangoma card |
| 19:20.10 | I-MOD | supjigatr, how do you have your zaptel.conf setup? (pastebin.ca) |
| 19:20.12 | supjigatr | It would seem it doesn't and you can define each port as local or source for a specific port. |
| 19:20.55 | supjigatr | Well it 1 0 0 0 which I take to get clock from first port (Telco) and output to my PBX's ports 2,3,4 but I'm having trouble with any modem calls. |
| 19:23.35 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 19:24.57 | supjigatr | I-MOD: does that sound right? |
| 19:25.16 | supjigatr | zaptel syncs from port 1 and all other use local synced clock? |
| 19:25.46 | QbY | Is my understanding of the "canreinvite" correct? I have two UA's registered with OpenSER, I have created a simple DIAL(SIP/) extension so that when UA #1 dials 211 it will actually DIAL UA #2 -- I have canreinvite=yes set, so therefore Asterisk should set up the call directly between the UAs and free up the channels? |
| 19:26.22 | supjigatr | canreinvite should make a direct connection. |
| 19:26.33 | file | a direct RTP connection |
| 19:27.53 | I-MOD | signalling still passes through asterisk |
| 19:28.13 | I-MOD | because asterisk is not a proxy |
| 19:28.17 | QbY | ah. |
| 19:28.57 | supjigatr | But no rtp channel is created. |
| 19:29.08 | supjigatr | It hands off the call direct. |
| 19:29.19 | QbY | so it will still show in the "show channels" |
| 19:29.24 | supjigatr | Nope. |
| 19:29.24 | QbY | or is that an RTP channel |
| 19:29.30 | file | yes, it will |
| 19:29.36 | supjigatr | As RTP? |
| 19:29.38 | I-MOD | in "show channels", it is the signalling connection |
| 19:29.43 | file | the call is still going through Asterisk for signalling, but the audio will go direct |
| 19:29.46 | supjigatr | But no RTP |
| 19:29.49 | supjigatr | Right. |
| 19:30.08 | QbY | how can i find out for sure where the rtp is? |
| 19:30.14 | supjigatr | ethereal |
| 19:31.32 | QbY | or easier, i could just shut down asterisk and see if the call stays |
| 19:31.53 | file | Asterisk will/should terminate the dialogs/calls |
| 19:32.22 | *** join/#asterisk zeppelin_ (n=zeppelin@201.89.132.250) |
| 19:34.45 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 19:36.27 | jbalcomb | is there no way to play a message for the caller before you put them in a queue by using just the queues.conf or Queue() function? |
| 19:42.04 | x86 | Playback() |
| 19:42.07 | x86 | Queue() |
| 19:42.11 | x86 | in your dialplan |
| 19:46.29 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 19:46.53 | *** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br) |
| 19:47.38 | quid246 | hmm, is there a way to change add/remove the fields * prints in the regular CDR (not CDR-CUSTOM)? |
| 19:48.05 | *** join/#asterisk Assid (i=assid@203.115.83.213) |
| 19:51.02 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 19:51.10 | TripleFFFF | whats MAX length of a INTL # ?> |
| 19:51.13 | TripleFFFF | 14 ? |
| 19:53.23 | *** join/#asterisk liran_ (i=liran@212.199.177.203.static.012.net.il) |
| 19:55.05 | *** join/#asterisk razu (n=razu@87-119-182-130.tll.elisa.ee) |
| 19:56.09 | lucasvo | exten => 1000,1,Dial(SIP/lucasvo,20,r) |
| 19:56.09 | lucasvo | exten => 1000,2,MusicOnHold() |
| 19:56.10 | lucasvo | exten => 1000,3,Hangup |
| 19:56.46 | lucasvo | ^^^^ Why isn't the music being played when it tries to dial? |
| 19:57.43 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 19:58.02 | quid246 | lucas: drop the 2nd command and put m into your dial command |
| 19:58.43 | quid246 | exten => 1000,1,Dial(SIP/lucasvo,20,r,m) |
| 19:58.49 | lucasvo | oh |
| 19:59.02 | quid246 | let me know how it works |
| 19:59.30 | lucasvo | doesn't |
| 20:00.08 | lucasvo | I still get the signal |
| 20:00.11 | lucasvo | no music |
| 20:00.25 | file | exten => 1000,1,Dial(SIP/lucasvo,20,rm) |
| 20:00.28 | file | although |
| 20:00.30 | file | r overrides the m |
| 20:00.33 | file | so you can only have one or the other |
| 20:01.50 | *** join/#asterisk da_monumental_1 (n=da_monum@adsl-074-246-252-251.sip.ilm.bellsouth.net) |
| 20:01.53 | lucasvo | it just mutes it, but no music |
| 20:02.37 | lucasvo | file: suggestion? |
| 20:02.39 | file | and your MOH is setup properly? |
| 20:02.49 | Rawplayer | where can i read up about AGI coding/ |
| 20:02.59 | Rawplayer | and what i can exactly do with it |
| 20:03.00 | lucasvo | file: well if I play it manully yes |
| 20:03.06 | lucasvo | file: MusicOnHOld() works |
| 20:03.14 | Rawplayer | i mean when i want to add users to my system, i need to use agi right? |
| 20:03.22 | file | lucasvo: what's the current Dial line you are using? |
| 20:03.27 | Rawplayer | i can also use some system scripts |
| 20:03.30 | Rawplayer | or something like that |
| 20:03.34 | Rawplayer | with some bash |
| 20:03.42 | lucasvo | file: exten => 1000,2,Dial(SIP/lucasvo,20,m) |
| 20:03.58 | file | Rawplayer: add users? no, not so much... AGI is for allowing you to do things using an outside script/program in the dialplan |
| 20:04.44 | Rawplayer | file: well |
| 20:04.51 | Rawplayer | how would you handle that? |
| 20:04.56 | file | how would I handle what? |
| 20:04.56 | QbY | supjigatr & file: my canreinvites seem to be performing as desired.. no rtp coming from asterisk.. one other question, with no rtp, what is the load on the Asterisk server? I should be able to set up tons of calls this way, versus OpenSER? |
| 20:04.58 | Rawplayer | with some perl/bash scripts? |
| 20:05.08 | file | you can use whatever you want |
| 20:05.09 | Rawplayer | adding users with a website |
| 20:05.31 | file | QbY: OpenSER will always be able to handle more if you just use it, since it's just doing proxying of signalling |
| 20:06.02 | file | Rawplayer: you could use realtime and link it to a database, or make flat files out of a database, whatever you want |
| 20:06.18 | QbY | file. i totally agree, however i'm looking at a lot of the other features and scripting that is easier with * |
| 20:06.31 | Rawplayer | file: but how would i get the database into asterisk? |
| 20:06.44 | QbY | Rawplayer.. AGI |
| 20:06.52 | QbY | my favorite method |
| 20:06.58 | TripleFFFF | anyone know Paul Vernon ? |
| 20:07.03 | Rawplayer | i just mentioned that |
| 20:07.06 | file | Rawplayer: Asterisk has realtime which can link it to the database for user authentication and user accounts |
| 20:07.19 | Rawplayer | QbY: its not possible with agi |
| 20:07.25 | file | Rawplayer: or you can write your own thing to take the contents of a database and produce a regular flat file |
| 20:07.39 | file | Rawplayer: either using a background process, or the #exec option in configs |
| 20:07.59 | Rawplayer | hmm |
| 20:08.00 | Rawplayer | ok |
| 20:08.08 | Rawplayer | well thats basicly the problem |
| 20:08.15 | Rawplayer | that coding isnt the problem |
| 20:08.22 | file | you have everything you need to be able to do this, you just need to research/read/learn |
| 20:08.24 | Rawplayer | but from db to asterisk was confusing me |
| 20:10.53 | Qwell | ! |
| 20:10.56 | Qwell | tmobile customer service ROCKS |
| 20:11.05 | file | Qwell: do tell |
| 20:11.12 | Qwell | I've been a customer for A DAY, and they're giving me a SIM unlock code :) |
| 20:11.20 | file | yay |
| 20:11.25 | Qwell | I told them I'm going overseas in a few weeks :D |
| 20:11.29 | TripleFFFF | wow |
| 20:11.30 | TripleFFFF | lol |
| 20:11.33 | TripleFFFF | whats the code |
| 20:11.35 | TripleFFFF | ;) |
| 20:11.37 | file | Qwell: celebrate! |
| 20:11.37 | Qwell | it's per phone |
| 20:11.48 | TripleFFFF | oh.. what phone.. got 2-3 unlockrs here |
| 20:11.50 | TripleFFFF | you need cable |
| 20:11.51 | TripleFFFF | ;) |
| 20:12.01 | Qwell | I have a cable. It's just USB :D |
| 20:12.17 | Qwell | Motorola + tmobile == <3 |
| 20:12.23 | quid246 | I dunno... SIM cards are a dirty business... always full of shady guys, like phone cards. |
| 20:12.32 | Qwell | quid246: yeah, no doubt |
| 20:12.46 | quid246 | ie. guy who was aprt of the Madrid train bombins, ran some kind of SIM hack shop or someting |
| 20:12.52 | quid246 | bombings |
| 20:13.26 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 20:18.12 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 20:20.33 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 20:21.29 | *** join/#asterisk da_monumental_1 (n=da_monum@adsl-074-246-252-251.sip.ilm.bellsouth.net) |
| 20:22.29 | *** join/#asterisk zedkatuf (n=zedkatuf@82-32-57-69.cable.ubr08.azte.blueyonder.co.uk) |
| 20:24.14 | zippytech | is there a comand i can run from shell to dial and play a file |
| 20:24.18 | zippytech | or just dial i number |
| 20:25.00 | zippytech | i want to setup a script to call me and tell me my servers are down |
| 20:25.05 | sevard | you want Asterisk to call an extension and play a file? |
| 20:25.30 | sevard | You should look into call files, they get dropped into /var/spool/asterisk/outgoing and make calls based on the options you put into the callfile |
| 20:25.31 | zippytech | yes or a over pot trunk |
| 20:26.54 | zippytech | have tried sms and email to cell phones but takes to long |
| 20:27.25 | sevard | I have a script that generates a call file every 2 hours from 11 am to 6 pm that calls my cell phone and says Move Your Car |
| 20:27.28 | sevard | so I don't get a ticket :) |
| 20:27.33 | sevard | m-f |
| 20:27.40 | zippytech | hmm |
| 20:27.57 | sevard | there are simple examples of call files at voip-info.org |
| 20:28.36 | zippytech | cool i am on my way thanks |
| 20:28.40 | sevard | np |
| 20:30.19 | TripleFFFF | Qwell got a GSM card to asterisk ? |
| 20:31.27 | Qwell | not yet...but I'll be able to soon :) |
| 20:33.48 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 20:34.21 | sevard | grr, this is Exactly why I hate GUIs, gmplayer doesn't have the option to -zoom |
| 20:34.54 | E-bola | use vlc |
| 20:35.01 | sevard | ahahahahhahahahahahahaha |
| 20:37.58 | *** join/#asterisk DeVilSoulBlacK (n=devilsou@gea-gye-internet.telconet.net) |
| 20:38.07 | marl | hi, can anyone help me with an IAX problem when trying to use the Digum IAXy phone adapter? conf files at http://pastebin.ca/129960 problem ocurs anytime i try to get the IAXy to register with * :( please can anyone see anything ive done wrong, there is also an error listed at hte bottom of the paste, is this error ignorable or is it posibly the cause of the IAX problems? |
| 20:38.14 | *** join/#asterisk ball (n=ball@dialup-4.158.0.225.Dial1.Chicago1.Level3.net) |
| 20:38.53 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 20:38.58 | ball | hello generalhan |
| 20:39.18 | generalhan | whats up all ... i just broke something and it looks bad .. i need some help ! |
| 20:39.22 | sevard | heh |
| 20:39.26 | generalhan | lol |
| 20:39.27 | generalhan | this sux |
| 20:39.30 | generalhan | SOOOO glad its sunday |
| 20:39.47 | sevard | dork. |
| 20:39.49 | trelane | marl, quite a bit's wrong, you call the iaxy "username" and in iax.conf you call it [iaxy]. There's no connector block for [username] |
| 20:40.06 | trelane | generalhan, I havn't seen a question, was there one? |
| 20:40.21 | generalhan | i did a reboot and my TE210 started blinking red ... and i was getting "loader.c:554 load_modules: Loading module chan_zap.so failed!" |
| 20:40.41 | generalhan | so i reloaded libpri and zaptel and reloaded and the system said that all my channels were configured and ready |
| 20:40.50 | trelane | generalhan, anything else that might give us a hint? (have you run ztcfg to check and make sure that nothing's changed?) |
| 20:40.57 | marl | so user in the iaxy.conf file should equal the [] in the iax.conf file? |
| 20:40.58 | generalhan | but asterisk wont start ... and now there are no lights AT all on the TE |
| 20:41.08 | lucasvo | I have music on hold working but it's a little bit too quiet. How can I make it louder? (I am currently using mode: files, it's the only one that worked) |
| 20:41.09 | trelane | generalhan, reboot the machine and reload hte driver? |
| 20:41.38 | trelane | marl, right... and if you use "username" as your username and "password" as your password a mob of torch bearing sysadmins will hunt you down |
| 20:41.56 | generalhan | trelane" http://generalhan.pastebin.ca/129965 |
| 20:41.59 | trelane | lucasvo, mode=mp3 vs quietmp3, take a look at musiconhold.conf at voip-info.org |
| 20:42.00 | ball | Does anyone make reasonably good WiFi sip-phones that work with Asterisk? |
| 20:42.01 | generalhan | my ztcfg -vvvvvvvvvvvvvvv |
| 20:42.01 | sevard | and kill you. |
| 20:42.19 | marl | trelane, lol, dont worry, i just edited that so the real user/pass wasnt pasted :) thats what i get for RTFMing :( |
| 20:42.25 | lucasvo | trelane: the only mode that worked was files, I tried mp3 already |
| 20:42.51 | trelane | generalhan, check /etc/zaptel.conf, I think your config is borked Your second D channel is out of place |
| 20:43.01 | *** join/#asterisk Trystan` (i=tslee@c-67-186-230-12.hsd1.ut.comcast.net) |
| 20:43.04 | trelane | generalhan, better yet, paste it |
| 20:43.34 | marl | any ideas on the error that i get on the iax reload command? |
| 20:43.44 | trelane | marl, where's the error? |
| 20:44.01 | marl | at the bottom of the paste bin |
| 20:44.14 | trelane | aah looks like your iaxy firmware is corrupt, redownload asterisk and cp it? |
| 20:44.19 | generalhan | trelane: http://generalhan.pastebin.ca/129970 <<<----------- /etc/zaptel.conf |
| 20:44.54 | generalhan | thats how ive always had it |
| 20:45.05 | generalhan | and its been running ok (until today) for about 6 months |
| 20:45.19 | ball | hello Trystan` |
| 20:45.33 | trelane | generalhan, you've got a dual-span pri, and a wctdm4xxp? |
| 20:45.36 | marl | would that cause any other problems with IAX? and can i just cp the IAX firmware, or would i need to re-install the entire * install? current asterisk = Asterisk SVN-trunk-r7230 |
| 20:45.41 | generalhan | trelane: yes |
| 20:45.44 | Trystan` | hello |
| 20:45.47 | generalhan | but only one PRI going into the dual card |
| 20:46.09 | trelane | generalhan, which port? |
| 20:46.13 | generalhan | but when first setting it up it wouldnt let me config the TDM unless i accounted for ALL the possible chans on the TE so thats why its defined |
| 20:46.15 | generalhan | !st |
| 20:46.17 | generalhan | 1st rather |
| 20:47.10 | lucasvo | trelane: any Ide? |
| 20:47.16 | lucasvo | *idea |
| 20:47.17 | trelane | hrm, generalhan has ztcfg always failed with the error on channel 53? |
| 20:47.39 | trelane | lucasvo, yes, you have the wrong version of mpg123 installed, go to your asterisk directory and "make mpg123" |
| 20:48.02 | generalhan | trelane: yes ,,, its for a fax machine ... but its not plugged in so thats why there is no device |
| 20:48.17 | trelane | generalhan, the fxo/fxs will configure whether it's plugged in or not |
| 20:48.22 | generalhan | hmm |
| 20:48.25 | lucasvo | trelane: I don't have a makefile for mpg123 |
| 20:48.39 | trelane | lucasvo, it's not a seperate makefile |
| 20:48.55 | trelane | in the root of your asterisk source directory "make mpg123" |
| 20:48.58 | lucasvo | trelane: I installed packages from the ubuntu mirror |
| 20:49.31 | trelane | lucasvo, build from source, or find the proper version of mpg123, and install it (There should be docs somewhere as to which one it is) |
| 20:50.06 | trelane | generalhan, I think you've had a long standing problem that's just gotten worse, though I don't know exactly what it is, but I'm betting that chan_zap.so will load if you fix it, and I'm pretty certain it is a configs issue |
| 20:50.25 | trelane | generalhan, go grab a copy of genzaptelconf, back up your configs and run it, see what you get and compare it with what you had |
| 20:50.38 | *** join/#asterisk MoutaPT (n=MoutaPT@a213-22-40-120.cpe.netcabo.pt) |
| 20:50.47 | generalhan | well .. the TDM lights are all on .. but the TE has none now |
| 20:50.56 | generalhan | http://generalhan.pastebin.ca/129977 and i fixed the chan 53 thing |
| 20:51.07 | ball | hello MoutaPT |
| 20:51.09 | generalhan | when i try to load asterisk with "safe |
| 20:51.36 | generalhan | _asterisk" i get an error "Asterisk died with code 1. |
| 20:51.47 | trelane | generalhan, I still think you have a broken config, pastebin your /etc/asterisk/zapata.conf? |
| 20:51.54 | trelane | file:poke? |
| 20:51.59 | file | eh? |
| 20:53.01 | trelane | file, assuming generalhan's zapata.conf is sane he's got a problem beyond my ability to fix and it's causing a crash on startup, figured I'd poke you |
| 20:53.35 | file | hey look, clear channels instead of foggy ones |
| 20:53.38 | generalhan | http://generalhan.pastebin.ca/129981 |
| 20:54.24 | generalhan | i know my configs look odd because of the grouping ... but back when i first set this up with the TE and the TDM i had to do some weird things to make it work right |
| 20:55.06 | trelane | looks odd but not broken, you've stumped me, might be a bug but your configs seem sane |
| 20:55.09 | generalhan | its the whole lights off on the TE210 that is freaking me out ... thats soo weird |
| 20:55.25 | trelane | generalhan, confirm the module's loaded? |
| 20:55.33 | trelane | (rebuild zaptel modprobe -r and modprobe) |
| 20:55.35 | generalhan | wow that is the first time any one has refered to me as sane before |
| 20:55.43 | trelane | generalhan, don't let it go to your head |
| 20:55.47 | generalhan | lol |
| 20:55.51 | generalhan | whats the -r |
| 20:55.53 | generalhan | for remove ? |
| 20:55.54 | trelane | remove |
| 20:55.55 | trelane | right |
| 20:55.56 | generalhan | ok |
| 20:56.05 | trelane | generally you tend to get lights after modprobe and ztcfg |
| 20:56.09 | generalhan | i JUST rebuilt libpri and zaptel ... lemme try the remove |
| 20:56.13 | trelane | ooh |
| 20:56.24 | trelane | that could cause the problem |
| 20:56.51 | generalhan | not removing first ? |
| 20:57.07 | generalhan | ok well i cant get it removed it says that its in use ... but asterisk is not running ? |
| 20:57.08 | trelane | libpri was compiled against a different zaptel |
| 20:57.13 | trelane | ooh |
| 20:57.19 | trelane | and asterisk compiled against a different zaptel too |
| 20:57.26 | Assid | great.. microsoft just entered f1 .. |
| 20:57.30 | Assid | this is gonna be interesting |
| 20:57.41 | trelane | do this: recompile zaptel and install, recompile libpri and install, and recompile asterisk and install, in that order |
| 20:57.44 | generalhan | well i will remove them all and then rebuild again then modprobe all over |
| 20:57.51 | trelane | right |
| 20:58.01 | generalhan | butr i cant remove it cause it says its in use |
| 20:58.06 | generalhan | should i rebuild anyway ? |
| 20:58.20 | trelane | yeah do those steps in order and follow it with the old shutdown -r now |
| 20:58.34 | generalhan | k 1 min |
| 20:58.39 | generalhan | thanks for the help |
| 20:58.53 | trelane | no problem, I'm heading out (have to take someone to the airport) |
| 21:09.57 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 21:10.00 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 21:10.45 | ball | hello DrkShdw, Syrus_ |
| 21:11.41 | *** join/#asterisk Flauto (n=HP_Owner@adsl-75-3-169-144.dsl.chcgil.sbcglobal.net) |
| 21:11.55 | Flauto | hi all |
| 21:11.59 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 21:12.44 | generalhan | trelane: i hope you get this and youre not gone already ... THANK YOU SOOOO MUCH .. the recompile and the reboot fixed it all !!!!!! THANK YOU THANK YOU !! |
| 21:13.09 | ball | hello Faulto |
| 21:13.10 | DrkShdw | hi ball |
| 21:13.14 | ball | Flauto* |
| 21:13.26 | Flauto | i am using mandriva 2006 now and i was using mandriva 2005 |
| 21:13.53 | Flauto | after i did the upgrade, webvmail stopped working |
| 21:14.35 | TripleFFFF | is it possible to do .. macro(blah)¯oblah? |
| 21:14.39 | TripleFFFF | i mean hmm |
| 21:16.21 | Flauto | 1hi ball |
| 21:16.49 | Flauto | do you use webvmail? ball |
| 21:17.22 | ball | Flauto: No, sorry. |
| 21:17.30 | Flauto | that is okay ball |
| 21:18.09 | *** join/#asterisk logicwrath (n=adsfasdf@c-68-60-121-112.hsd1.mi.comcast.net) |
| 21:18.27 | ball | hello logicwrath |
| 21:18.46 | logicwrath | hi |
| 21:19.27 | ball | What's a decent basic business phone these days? Speakerphone would be nice, as would the option of a headset. One number is fine, Caller ID would be nice. SIP, naturally. |
| 21:20.21 | logicwrath | can i use meetme without having ztdummy running if I dont have any cards installed? |
| 21:20.36 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:20.37 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:20.40 | *** join/#asterisk dflow (i=pch@h82-143-131-199-static.e-wro.net.pl) |
| 21:20.43 | ball | hello russellb |
| 21:20.47 | ball | hello dflow |
| 21:20.58 | hads | You need some sort of timing, either a card or ztdummy |
| 21:21.22 | logicwrath | is that why i am getting that is not a valid conference number? |
| 21:21.35 | hads | Probably not |
| 21:22.11 | logicwrath | i feel its configured correctly -minus ztdummy |
| 21:22.36 | logicwrath | i tried using freepbx and i also tried configuring it myself by hand |
| 21:22.41 | logicwrath | same results |
| 21:23.35 | Flauto | you need ztdummy for meetme for sure |
| 21:23.40 | Flauto | or a card |
| 21:24.04 | logicwrath | any other apps that require ztdummy? |
| 21:24.09 | *** part/#asterisk dflow (i=pch@h82-143-131-199-static.e-wro.net.pl) |
| 21:24.31 | Flauto | not that i aware of |
| 21:28.37 | Trystan` | first question so bare with me.. got it all installed, making calls out, and around the office no issues, trying to setup hold music but with format_mp3 I either get a segfault or message about no class default, with mpg123 it says its starting and stopping the hold but the phone that is holding hears nothing |
| 21:30.11 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 21:32.53 | *** join/#asterisk asyouare405 (i=MyIdent@alexelf09.rutgers.edu) |
| 21:32.56 | asyouare405 | hello channel |
| 21:33.07 | asyouare405 | can anyone recommend a good guide to starting with asterix |
| 21:35.05 | hads | ~thebook |
| 21:35.06 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:35.08 | Juggie | 1) spell it correctally, its asterisk. |
| 21:35.32 | ball | hello asyouare405 |
| 21:37.03 | Nivex | wasn't asterix a cartoon character? |
| 21:37.20 | hads | Yes |
| 21:37.23 | ball | Asterix the Gaul I think |
| 21:43.02 | MoutaPT | hi does any one here has already test googletalk connection with asterisk? my jabber account from Asterisk server doesn't register... don't know why.. |
| 21:43.19 | TripleFFFF | lol |
| 21:44.42 | *** part/#asterisk Trystan` (i=tslee@c-67-186-230-12.hsd1.ut.comcast.net) |
| 21:45.55 | MoutaPT | TripleFFF: experienced with Asterisk and Googletalk? |
| 21:46.28 | *** join/#asterisk jarg (n=jarg@dsl-201-129-100-218.prod-infinitum.com.mx) |
| 21:47.15 | ball | hello jarg |
| 21:47.51 | jarg | hello ball |
| 21:56.51 | *** join/#asterisk beyond (n=beyond@201-1-179-189.dsl.telesp.net.br) |
| 21:57.12 | ball | hello beyond |
| 21:57.26 | beyond | ball, hello dude |
| 21:59.07 | MoutaPT | nobody here tested googletalk and asterisk? |
| 22:06.44 | MoutaPT | any one in the channel? |
| 22:07.05 | *** part/#asterisk MoutaPT (n=MoutaPT@a213-22-40-120.cpe.netcabo.pt) |
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| 22:21.26 | *** part/#asterisk ball (n=ball@dialup-4.158.0.225.Dial1.Chicago1.Level3.net) |
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| 23:03.35 | *** join/#asterisk mcreedjr (n=mcreedjr@68.72.158.2) |
| 23:04.15 | mcreedjr | Hi all, I'm still having some problems with echo at the beginning of an answered call. I've setup echo cancelling in zapata.conf, and ran fxotune (the new SVN one) and am still having the problem. Suggestions? |
| 23:04.32 | mcreedjr | Hardware is Digium TDM400 series |
| 23:08.02 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 23:10.14 | *** join/#asterisk Un1x (n=x@CPE001731208485-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 23:10.30 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 23:12.06 | russellb | mcreedjr: contact support@digium.com |
| 23:12.37 | mcreedjr | russellb: I had thought about that, will it require a contract, or do they provide the support on the card? |
| 23:12.51 | russellb | free hardware support |
| 23:13.12 | mcreedjr | russellb: sweet. i will do that then. i'm just impatient sometimes :) |
| 23:13.49 | russellb | they also provide services for 24x7 support if you're that impatient ;) |
| 23:14.42 | mcreedjr | russellb: good to know heh. |
| 23:20.41 | *** join/#asterisk MoutaPT (n=MoutaPT@a213-22-40-120.cpe.netcabo.pt) |
| 23:21.11 | MoutaPT | hi does any one got gtalk working fine with asterisk? i'm in troubles with jabber register from my * box |
| 23:21.29 | delmar | Is there anyone here familiar with the MCC application ... billing software for use with Asterisk? |
| 23:21.49 | MoutaPT | only few tests with A2billing... |
| 23:22.14 | anthonyl | freshmeat.net has some voip accouting/billing tools |
| 23:22.29 | *** join/#asterisk ESCulapio__ (n=ESCulapi@151stb68.codetel.net.do) |
| 23:22.30 | anthonyl | ined #asterisk |
| 23:22.43 | delmar | mcreedjr, hey when you call Digium.. you can ring them via VoIP using the "demo" extension that comes in the Asterisk configs |
| 23:22.44 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 23:23.27 | mcreedjr | delmar: thanks, but i blew away all the default configs and started from scratch.. pretty sure that got nixed |
| 23:23.28 | delmar | mcreedjr, i can't say I had many echo issues with the TDM 400...in fact.. it cured all my problems... I know a great deal about echo :P |
| 23:23.49 | delmar | mcreedjr, just add a new extension for Digium |
| 23:23.54 | MoutaPT | google for IAX/500 ... digium |
| 23:24.07 | mcreedjr | delmar: i might do that.. |
| 23:24.11 | MoutaPT | i think you will find it |
| 23:24.19 | delmar | mcreedjr, u know how to do that ok? |
| 23:24.27 | mcreedjr | delmar: yeah, thanks. |
| 23:24.36 | delmar | mcreedjr, ok great. |
| 23:24.44 | MoutaPT | got the exten mcreedjr? |
| 23:24.59 | delmar | mcreedjr, so.. I think i missed the description of your echo problem... how about before you call them .. give us a good description... |
| 23:25.34 | delmar | mcreedjr, what device/client are you using to talk to Asterisk? |
| 23:26.18 | delmar | mcreedjr, I gather you are using a TDM400 with at least 1FXO port on a PSTN running .. latest Asterisk and Zaptel? |
| 23:26.26 | Lyfe | has anyone else experience d problems with mixmonitor, where it will cause asterisk to segfault if the call doesn't get completed (eg, the caller hangs up while the callee is ringing) |
| 23:29.00 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 23:30.03 | mcreedjr | delmar: mostly right... Digium with 4 FXO. Zaptel 1.2.6, asterisk 1.2.9. Linksys SIP phones. Echo is a problem at the beginning of the call for 2-10 seconds. |
| 23:30.56 | mcreedjr | delmar: echo training and cancelling is enabled in zapata.conf. fxotune (new SVN version) has been run and it returning a less than 5% echo ratio. |
| 23:31.33 | mcreedjr | delmar: i think the gains may be to blame, but i can't find the milliwatt test tone number on my CO switch. i'm prank calling people as we speak :) |
| 23:32.20 | delmar | mcreedjr, lol |
| 23:32.46 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:33.02 | delmar | mcreedjr, fxotune works by dialing a number that will cause the line to be silent, then it generates tones on the line to detect echo |
| 23:33.30 | delmar | mcreedjr, the latest zaptel is 1.2.7 and asterisk is 1.2.10 |
| 23:33.31 | mcreedjr | delmar: well i did find the silent number on my CO switch and used that in fxotune, but that doesn't help me set the gains |
| 23:33.51 | delmar | mcreedjr, your gains should start at 0.0 for both |
| 23:34.04 | mcreedjr | delmar: i understand, but i run gentoo and i prefer to run things out of portage, neither of those two releases are in portage yet |
| 23:34.16 | delmar | mcreedjr, so you are saying that the FXO ports are all attacked to a CO switch? |
| 23:34.25 | delmar | mcreedjr, what type of switch? |
| 23:34.38 | mcreedjr | delmar: no idea, but yes they are connected to the PSTN |
| 23:35.02 | delmar | mcreedjr, have you tried anything other than the linksys phones? |
| 23:35.21 | mcreedjr | delmar: no, i guess not, i could try an Xten or something |
| 23:35.58 | mcreedjr | delmar: or i do have an fxs port, i could try that also |
| 23:36.09 | delmar | mcreedjr, if you do, use a headset and use x-lite on a windows box. |
| 23:36.36 | delmar | mcreedjr, I have encountered some wierd issues with linux audio drivers mixing sound output back into the mic channel. |
| 23:36.40 | *** join/#asterisk sugardave (n=sugardav@cpe-70-112-127-158.austin.res.rr.com) |
| 23:36.48 | mcreedjr | delmar: alright... i suppose it could just be sidetone on the linksys phones maybe the gain is set too high on the mic or somethign |
| 23:36.53 | delmar | mcreedjr, there you go. the FXS port is another idea. |
| 23:37.11 | delmar | mcreedjr, also.. in zapata.conf have you tried echotraining=800 ? |
| 23:37.28 | delmar | mcreedjr, have you made any changes to the zaptel code? |
| 23:37.29 | mcreedjr | delmar: didn't think of that... the fxo port would certainly rule that out. as for zapata, no echotraining=yes |
| 23:37.37 | mcreedjr | delmar: wouldn't dare :) |
| 23:37.53 | delmar | mcreedjr, intel or amd system? |
| 23:38.54 | mcreedjr | delmar: ermm.. via |
| 23:39.02 | mcreedjr | delmar: it's a mini-itx system |
| 23:39.07 | delmar | mcreedjr, what CPU |
| 23:39.17 | mcreedjr | mcreedjr: nehemiah C3 |
| 23:39.26 | mcreedjr | delmar: i think thats how you spell it |
| 23:39.30 | MoutaPT | does any one got success gtalk+Asterisk? |
| 23:39.55 | delmar | MoutaPT, nope, but how hard can it be.. Asterisk and Jabber work great. |
| 23:40.22 | MoutaPT | yeah, but i can't get My jabber account from asterisk to get registred |
| 23:40.41 | MoutaPT | also from CLI i only test jabber show connected |
| 23:40.56 | delmar | mcreedjr, try the echotraining=800 thing and see how that goes. don't forget when you make changes you need to shutdown Asterisk and start it again, not just reload. |
| 23:40.57 | MoutaPT | and it says my user is disconnected |
| 23:41.01 | mcreedjr | MoutaPT: there was an article on the front page of voip-info.org about that the other day, did you find it? included example configs and what not |
| 23:41.19 | MoutaPT | yes i've followed that example |
| 23:41.23 | delmar | mcreedjr, try to rule out that it is not more a problem with the SIP client. test a few alternatives. this is good ammo when you call Digium |
| 23:41.28 | MoutaPT | but not yet working... |
| 23:41.28 | mcreedjr | delmar: yeah, i discovered that the hard way... couldn't figure out why my echo training settings weren't chaning :) |
| 23:41.51 | mcreedjr | delmar: i appreciate it, i don't like sounding stupid. good things for me to try. |
| 23:41.54 | delmar | mcreedjr, ok. but once you have done all that.. there are a couple of other things.... |
| 23:42.00 | mcreedjr | delmar: i appreciate it |
| 23:42.09 | mcreedjr | delmar: want me to try these first? |
| 23:42.45 | delmar | mcreedjr, im gonna head off and get a coffee and get into some work so make note of this also... take a look in your zaptel source dir, and take a look at ztconfig.h |
| 23:42.59 | MoutaPT | delmar any guess to debug why my jabber asterisk account didn't register to gtalk? |
| 23:43.03 | delmar | mcreedjr, in there, you will see where you can change the type of echo-can being used |
| 23:43.08 | mcreedjr | delmar: aye... as an FYI, my current echocanceller is mark2 |
| 23:43.31 | mcreedjr | delmar: please excuse my terrible spelling today... i'm tired i guess. |
| 23:43.37 | delmar | MoutaPT, not really dude. I installed a jabber thing a while back jsut to see how it would work and havent looked again since. |
| 23:43.59 | MoutaPT | that's what i'm doing:) did it work for u? |
| 23:44.01 | delmar | mcreedjr, yeah i think mark2 might be the default. just a asec |
| 23:44.31 | mcreedjr | delmar: i believe greater than 1.2.1 in zaptel, it's kb1 actually. |
| 23:44.45 | mcreedjr | delmar: i read somewhere that mark2 was a good choice |
| 23:44.58 | delmar | mcreedjr, not ztconfig.h sorry.. its zconfig.h |
| 23:45.29 | delmar | mcreedjr, ok so in that file... there is a section where you comment or uncomment the stuff to enable/disable the echocan |
| 23:45.53 | delmar | mcreedjr, actually.. |
| 23:45.56 | *** join/#asterisk caloi (n=caloi@cpe-24-58-22-17.twcny.res.rr.com) |
| 23:45.57 | delmar | mcreedjr, mark2 is not correct |
| 23:46.03 | MoutaPT | sorry guys, just a question, all this issues about echo you r talking is for pots line? |
| 23:46.06 | delmar | mcreedjr, im sure I use KB |
| 23:46.08 | delmar | mcreedjr, brb |
| 23:46.10 | mcreedjr | delmar: i'm familiar with this process, go on.. |
| 23:46.18 | mcreedjr | MoutaPT: yes, for a POTS line. |
| 23:46.38 | delmar | mcreedjr, yeah dude.. here is mine... default... #define ECHO_CAN_KB1 |
| 23:46.47 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 23:46.50 | delmar | MoutaPT, yes. pots lines are bloody useless. |
| 23:47.05 | mcreedjr | delmar: right, i realize that, i manually changed it to mark2 :) i'll recompile with kb1 instead. |
| 23:47.07 | MoutaPT | ok ok, just asking because one week a go i got echo in sip calls, but i think that was Volume adjustment for handsets of Linksys SPA941 |
| 23:47.16 | delmar | soon as I can get wireless, I will drop my pots, and have VoIP DID's only. |
| 23:47.30 | mcreedjr | MoutaPT: fancy that, i'm using SPA942s, i bet thats it... |
| 23:47.38 | MoutaPT | even that i got surprise for this hardware (almost cisco :))) ) got ECHO |
| 23:47.43 | MoutaPT | i couldn't believe |
| 23:47.45 | delmar | mcreedjr, I found that KB1 worked better for me... and believe me.. I pissed around with it lots. |
| 23:47.59 | MoutaPT | my plantronics with DSP USB was muche better sound with xlite |
| 23:48.00 | wunderkin | delmar, you'll want to make sure you have a good connection first |
| 23:48.03 | mcreedjr | delmar: lol good deal, switching to KB1 is easy enough |
| 23:48.05 | delmar | mcreedjr, I know little about gentoo but.. I dont understand why you cant just use the latest zaptel? |
| 23:48.19 | delmar | wundaboy, of course. |
| 23:48.24 | mcreedjr | delmar: i could, it's just not done within the package management then |
| 23:48.43 | mcreedjr | delmar: i'm lazy and for deps and stuff, it's easiest to use whats in the portage tree |
| 23:48.56 | MoutaPT | mcreedjr: did you get echo between to sip extensions? |
| 23:49.10 | mcreedjr | MoutaPT: no, only between SIP and PSTN |
| 23:49.14 | MoutaPT | i got it, and this hardware was 150$ each.. |
| 23:49.26 | delmar | wundaboy, but im prepared to put up with a little bit of shit for it. I mean.. paying NS$43/month for a pots i dont use.. and only need for DSL .. another $60/month... when I can get NZ DID numbers for about NZ$10/month... i dont need the damn pots/voice line service |
| 23:49.51 | *** part/#asterisk MoutaPT (n=MoutaPT@a213-22-40-120.cpe.netcabo.pt) |
| 23:49.56 | delmar | mcreedjr, I use debian. and i do NOT use the package management at all when it comes to Asterisk |
| 23:49.57 | mcreedjr | delmar: lol i agree, this analog BS is driving me nuts. |
| 23:50.17 | mcreedjr | delmar: right, i have that option, but once i go that route, it's difficult to go back |
| 23:50.30 | delmar | mcreedjr, I remember when I first got an FXO card... X100 clone. fuckin hopeless. |
| 23:50.35 | mcreedjr | delmar: do you know of major echo-can enhancements in 1.2.8? |
| 23:50.47 | delmar | mcreedjr, I put it on the fire.. .im not joking.. I watched it burn |
| 23:51.01 | delmar | mcreedjr, got the TDM400 and never looked back. |
| 23:51.09 | mcreedjr | delmar: lol yeah, i hear those X100s are crap. the TDM400 stuff for low end seems to be alright |
| 23:51.18 | delmar | mcreedjr, nope. are they planning some good echo-can in 1.2.8? |
| 23:52.02 | mcreedjr | delmar: err, i meant to ask if you knew if they was any major improvments in 1.2.7 for echo-can. meaning, would the upgrade really be that beneficial for this specific problem? |
| 23:53.00 | delmar | mcreedjr, im still running 1.2.5 on this system.. only because I havent got around to upgrading..... running 1.2.7 on a customer box I just installed... also has TDM400... im not sure if theres alot of difference. |
| 23:53.15 | delmar | mcreedjr, but when you call digium... if you are running their latest stuff... yet more ammo. |
| 23:53.27 | mcreedjr | delmar: agreed. |
| 23:53.34 | delmar | mcreedjr, installing the new modules is as easy as doing make clean && make install in the dir. |
| 23:53.50 | delmar | mcreedjr, then rmmod wctdm && rmmod zaptel |
| 23:53.55 | delmar | mcreedjr, then poke them back in |
| 23:54.00 | delmar | mcreedjr, ztcfg -vvv |
| 23:54.07 | delmar | mcreedjr, re-tune perhaps.. for good measure |
| 23:54.14 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 23:54.22 | delmar | mcreedjr, using the fxotune that comes with the new zaptel |
| 23:54.42 | delmar | mcreedjr, set the echo thingies.. fire up asterisk... test... call Digium.. moan at them :P |
| 23:54.46 | mcreedjr | delmar: yeah, ive been using that, got it out of SVN, with the new -d option... cool stuff |
| 23:54.55 | delmar | mcreedjr, you better hurry tho. Digium will be closed for the day shortly. |
| 23:55.11 | mcreedjr | delmar: do they offer free hardware phone support? |
| 23:55.13 | delmar | mcreedjr, not seen that.. what does -d do for ya? |
| 23:55.52 | mcreedjr | delmar: dumps the input vs. output waves. lets you graphically depict echo on the line :) |
| 23:56.02 | delmar | mcreedjr, well.. this isnt so much an installation issue.. but then .. it is... the card isnt working right from the get go.... mostly installation trubleshooting is free. I think this counts.. but im not entirely sure. |
| 23:56.11 | mcreedjr | delmar: lookie here: http://www.voip-info.org/wiki/view/Asterisk+fxotune |
| 23:56.29 | mcreedjr | delmar: mmm pretty graphs |
| 23:56.46 | delmar | mcreedjr, ooo nice |
| 23:56.52 | delmar | mcreedjr, ive not seen that before |
| 23:57.01 | delmar | mcreedjr, runs in a shell or on X? |
| 23:57.21 | mcreedjr | delmar: shell, and dumps the data to .csv can graph in excel or something |
| 23:57.34 | mcreedjr | delmar: pretty sure he used excel to make those graphs |
| 23:57.46 | delmar | mcreedjr, ah ok |
| 23:57.54 | delmar | mcreedjr, so.. you can tune on the fly? |
| 23:58.26 | delmar | mcreedjr, ill take a look sometime. so you did this with yours? and have a flat-line like the graph shows? |
| 23:58.32 | mcreedjr | mcreedjr: nearly, i suppose. you can certainly see before and after results. |
| 23:58.51 | mcreedjr | delmar: after running it, yes, a flat line, similar to what he has. |
| 23:59.07 | mcreedjr | delmar: my echo ratio went from 37% to 2% |
| 23:59.35 | delmar | mcreedjr, good. more ammo. so .. you could now assume.. the card is tuned as best it can be... so .. the software echo-can is not doing the job |
| 23:59.51 | mcreedjr | delmar: or it's excess side tone on the sip phone |