| 00:03.16 | rhiliam | Fender: I am assuming like this -> http://pastebin.ca/658975 |
| 00:05.07 | Yourname` | Hi. When it's supposed to play a message, it doesn't.. and CLI gives me this error: Aug 15 20:04:50 NOTICE[2735]: res_musiconhold.c:533 monmp3thread: Request to schedule in the past?!?! -> What's the beef? |
| 00:06.37 | neverblue | is there a part, USB male on one end to a 'normal' head phone female, but with two females, I was wanting to use it for double-jacking for training purposes? |
| 00:06.38 | [TK]D-Fender | rhiliam, Yes, like that. |
| 00:06.44 | rhiliam | Thanks |
| 00:06.48 | rhiliam | will give it a whirl. |
| 00:07.01 | [TK]D-Fender | neverblue : a USB **SPLITTER**? |
| 00:07.05 | JT | neverblue: i have no idea what you just asked |
| 00:07.08 | neverblue | correct |
| 00:07.21 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583527.dsl.bell.ca) |
| 00:07.32 | JT | usb... headphones... what do you want? |
| 00:07.35 | neverblue | a USB splitter, but not for two USB headsets, but two 'regular' headsets |
| 00:07.36 | boch | if i cant rely in channel vars when h extension is called, how can i identify the channel to bill ? |
| 00:07.36 | [TK]D-Fender | neverblue : I cannot conceive of such a thing. |
| 00:07.52 | neverblue | http://www.ncix.com/products/index.php?sku=23306&vpn=980445-0403&manufacture=Logitech |
| 00:08.08 | [TK]D-Fender | neverblue : Its nearly ludicrous. 2 PC's aren't supposed to be able to operate on the same bus and every device is unique |
| 00:08.32 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 00:08.47 | JT | neverblue: yeah it's usb, you can't just split it, especially for the one application |
| 00:08.58 | JT | why get a usb headset? |
| 00:09.03 | [TK]D-Fender | neverblue : Put down the crack pipe! (c) JerJer |
| 00:09.22 | [TK]D-Fender | neverblue : ChanSpy <---- |
| 00:11.25 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 00:13.16 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 00:13.17 | neverblue | 2 PCs? |
| 00:13.45 | neverblue | what gave you the impression I was talking about 2 PCs |
| 00:13.46 | neverblue | lol |
| 00:15.38 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 00:15.38 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:16.28 | [TK]D-Fender | neverblue : USB headset aren't some magical thing you can daisy-chain and spy on. |
| 00:16.39 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 00:16.39 | JT | neverblue: lol rofl, anyway, magical idea, not happening |
| 00:17.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:26.06 | Yourname` | Hi [TK]D-Fender |
| 00:26.11 | Yourname` | Can you please help me out here? |
| 00:26.26 | [TK]D-Fender | Yourname`, WITH? |
| 00:26.29 | carrar | LIFE!!! |
| 00:26.46 | Yourname` | [TK]D-Fender: When it's supposed to play a message, it doesn't.. and CLI gives me this error: Aug 15 20:04:50 NOTICE[2735]: res_musiconhold.c:533 monmp3thread: Request to schedule in the past?!?! -> What's the beef? |
| 00:28.22 | [TK]D-Fender | Yourname`, "it" ? |
| 00:28.45 | carrar | They sell "it" on eBay don't they? |
| 00:29.04 | Yourname` | [TK]D-Fender: According to the dialplan, right when this error happens.. asterisk is supposed to be playing musiconhold |
| 00:29.15 | Yourname` | lol carrar, are you a bot? |
| 00:29.30 | [TK]D-Fender | Yourname`, thats justdave mpg123 spewing off useless warning |
| 00:29.31 | carrar | Yourname`, hi, yes I am a bot! |
| 00:29.31 | JT | no, he sounds a lot less botlike than you :) |
| 00:29.54 | carrar | heh |
| 00:30.29 | Yourname` | [TK]D-Fender: Aaah, hmm, so that message has nothing to do with the music on hold not playing? |
| 00:30.51 | Yourname` | JT: Bitter again because I prefer to be helped by [TK]D-Fender and not you? |
| 00:30.58 | *** join/#asterisk robh71 (n=robh71@host-65-124-86-25.entouch.net) |
| 00:31.15 | JT | Yourname`: no, it's not even a real problem |
| 00:31.19 | [TK]D-Fender | Yourname`, not in and of itself. |
| 00:31.54 | Yourname` | [TK]D-Fender: What can it be, can you please help in figuring this out as to why the musiconhold isn't playing? |
| 00:31.59 | s0ck | morning chaps. |
| 00:32.14 | *** join/#asterisk MindTheGap_ (n=iote@bhe201062195037.res-com.wayinternet.com.br) |
| 00:32.40 | [TK]D-Fender | Yourname`, show me that your MoH files are in the right state, in the right place, and that your runing the right version of mpg123. |
| 00:33.37 | JT | Yourname`: clearly i was not the only one irritated by your question asking abilities |
| 00:34.05 | MindTheGap_ | [OT] night everyone... does anyone knows of a sip/xmpp client thas runs on windows and macos and offers support of QOS settings? |
| 00:34.41 | Yourname` | JT: Well, clearly you're the only mouthy one. If someone else has a problem in me not typing "cn u help me????????" and rather trying to ask as concise a question .. they'd probably be frothing like you too. |
| 00:34.55 | *** join/#asterisk Steven_elvisda (n=Elvisda@202.47.107.60) |
| 00:34.56 | Yourname` | [TK]D-Fender: That's a clue, I'm going to look around and get back to you. |
| 00:35.26 | JT | Yourname`: mouthy? you're the one who accused another channel member of being a bot. |
| 00:37.02 | Yourname` | JT: I have reasons for that. The replies were almost instantaneous, and were usually followed by a counter-question -> usually bots behave like this. But then again, I was polite enough to ask, wasn't I? |
| 00:37.43 | carrar | ERROR, trivial crap buffer overflowing |
| 00:37.56 | Yourname` | lol I get the picture carrar, I stop now. |
| 00:38.48 | fujin | blah |
| 00:38.50 | JT | Yourname`: i didn't see any behaviour that'd typically trigger a bot |
| 00:38.52 | fujin | don't use mpg123 for MOH |
| 00:39.07 | Yourname` | fujin: What's good? |
| 00:39.11 | fujin | native |
| 00:39.22 | [TK]D-Fender | JT, Yourname` : I'm not going to ask you two to agree. |
| 00:39.30 | [TK]D-Fender | JT, Yourname` : I'm not going to ask you two to agree to DISAGREE either. |
| 00:39.53 | [TK]D-Fender | JT, Yourname` : But at the very least as far as this naggin is concerned... |
| 00:40.03 | Steven_elvisda | good morning [TK]D-Fender |
| 00:40.08 | [TK]D-Fender | JT, Yourname` : I'm gonna ask you to put a sock in it :D |
| 00:40.15 | JT | you can ask |
| 00:40.17 | JT | so anyway |
| 00:40.27 | [TK]D-Fender | PEACE! |
| 00:40.43 | Yourname` | fujin: I don't think I'm using mpg123 either, don't remember configuring it. But I'm going to look for it and get back here soon, heh |
| 00:40.44 | fujin | Yourname`: format_mp3 from the asterisk-addons package |
| 00:40.50 | Yourname` | ah! |
| 00:40.53 | fujin | Or transcode your mp3s to the format you are playing over your PSTN |
| 00:40.54 | JT | Yourname`: also it is very rude to ask for specific channel member's help if it's not with regards to an issue you've previously had a dialogue with them about |
| 00:40.59 | Yourname` | Thanks fujin , looking into it right now. |
| 00:41.01 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 00:41.04 | JT | [TK]D-Fender: do you enjoy being singled out for questions? ;) |
| 00:41.04 | fujin | (that's what I do, all of my MOH is transcoded to uLAW |
| 00:41.25 | fujin | Yourname`: imaging trying to decode 30 mp3's, and then transcode them all to xlaw |
| 00:41.31 | [TK]D-Fender | JT : You just singled me out just thinking I know everything about myself! Dear God will it never end?!?! |
| 00:41.54 | JT | zomg! |
| 00:42.35 | Yourname` | JT: Now that's a very understandable para you just typed. See, politely telling me what's rude and what's not makes me oblige. If that was the case, I wouldn't have explicitly asked for [TK]D-Fender. But then, [TK]D-Fender has helped me in God knows how many issues.. which is why I automatically default to his help. Anyway, if its rude, now I know. I'll try not to. |
| 00:43.14 | JT | okay |
| 00:43.20 | tengulre | morning, JT |
| 00:43.31 | JT | hi tengulre |
| 00:43.45 | tengulre | JT: how to cancel the iax2 echo. ;(, I crazy for it |
| 00:43.58 | fujin | tengulre: how to english? |
| 00:44.01 | JT | tengulre: iax2 shouldn't be echoing |
| 00:44.03 | fujin | I'm crazy for it. |
| 00:44.14 | JT | tengulre: what is on each end of this iax2 link? |
| 00:44.30 | fujin | Yourname`: generally, singling people out for a response in a community ran support channel is a Bad Idea |
| 00:44.42 | tengulre | fujin: I 'm sorry , I known only a little english! ;( |
| 00:44.50 | fujin | There's a good document somewhere on google about "Asking Questions" |
| 00:44.56 | JT | ~question |
| 00:44.57 | jbot | well, question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 00:45.14 | fujin | ah |
| 00:45.21 | fujin | That's it :) |
| 00:45.40 | Yourname` | fujin: I didn't know that, see. If someone had really taken the time to tell me that.. I'd happily apologize and oblige. Do I look hostile to you lol.. |
| 00:45.46 | fujin | specifically, |
| 00:45.46 | fujin | post a personal e-mail to somebody who is neither an acquaintance of yours nor personally responsible for solving your problem |
| 00:45.46 | JT | ESR may be a crackhead, but sometimes he's very right :) |
| 00:45.51 | tengulre | JT: really? oh, NO! I think iax2 is not suite as a softphone, because it have echo. ;( |
| 00:46.05 | tengulre | s/have/has |
| 00:46.14 | JT | tengulre: what |
| 00:46.25 | JT | tengulre: WHAT ARE YOU CONNECTING TO with IAX2? |
| 00:46.29 | fujin | lol |
| 00:46.29 | fujin | fail |
| 00:46.44 | fujin | really? oh, NO! |
| 00:46.46 | fujin | qft |
| 00:46.50 | JT | now now |
| 00:46.54 | tengulre | JT: <IAX2 softphone> <---> <Asterisk box> <----> <IAX2 softphone> |
| 00:47.02 | JT | which softphone |
| 00:47.11 | tengulre | JT: Iaxcomm |
| 00:47.18 | [TK]D-Fender | JT : about being right : "Here lies the body of Michael O'Day, |
| 00:47.18 | [TK]D-Fender | who died maintaining the right of way. |
| 00:47.18 | [TK]D-Fender | He was right, dead right, as he sailed along, |
| 00:47.18 | [TK]D-Fender | but he's just as dead as if he'd been wrong." |
| 00:47.24 | JT | are you using a headset on each end? |
| 00:47.42 | tengulre | JT: how to setting headset? |
| 00:47.51 | fujin | lol. |
| 00:48.02 | JT | tengulre: do you use a headset at each end? (yes, no) |
| 00:48.06 | tengulre | fujin : do u lol to me? |
| 00:48.16 | fujin | yes, I lol to you. |
| 00:48.23 | tengulre | JT: I don't know what's headset? |
| 00:48.29 | fujin | bool bUsingHeadset = Usingheadset(); |
| 00:48.30 | tengulre | fujin : Fuck u! ;( |
| 00:48.53 | tengulre | JT: I think I no |
| 00:49.08 | tengulre | fujin : in source code? |
| 00:49.23 | fujin | yes, check if you are using a headset in the source code. That makes lots of sense. |
| 00:49.29 | JT | tengulre: headphones that go over your ear, and also has a boom microphone |
| 00:49.40 | fujin | s/crings/cringe/ |
| 00:50.15 | fujin | I'd give up while you're ahead, tbh |
| 00:50.23 | tengulre | OK, I checking it now. JT, fujin thank you! |
| 00:50.30 | fujin | No problem! |
| 00:50.36 | tengulre | don't lol to me later! OK? |
| 00:51.04 | tengulre | 你好！ |
| 00:51.06 | JT | tengulre: OR are you using normal computer speakers and desktop microphone? |
| 00:51.23 | tengulre | JT: yes. |
| 00:51.29 | JT | tengulre: of course you have echo. |
| 00:51.35 | tengulre | ?? |
| 00:51.37 | JT | tengulre: stop using desktop speakers. |
| 00:51.37 | tengulre | why? |
| 00:51.38 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 00:51.56 | JT | because audio from your speakers feed straight into the microphone |
| 00:51.58 | tengulre | JT: but I have only desktop speakers! |
| 00:52.04 | JT | you must buy headsets to use a softphone |
| 00:52.10 | JT | then you cannot use a softphone. |
| 00:52.20 | fujin | if you're using desktop speakers and a microphone, you will get echo |
| 00:52.25 | fujin | such are the laws of physics |
| 00:52.27 | JT | you are experiencing direct end-user accoustic echo |
| 00:52.30 | tengulre | headsets==头设置? |
| 00:52.36 | fujin | what the hell was that? |
| 00:52.39 | JT | i don't know what that means |
| 00:52.41 | fujin | even my utf-8 client couldn't draw them |
| 00:52.43 | coppice | if you want to use a speakerphone you need a proper speakerphone that mutes or echo cancels |
| 00:52.55 | coppice | tengulre: yes |
| 00:53.14 | fujin | http://images.google.co.nz/images?q=headset&ie=UTF-8&oe=UTF-8&rls=org.mozilla:en-GB:official&client=firefox-a&um=1&sa=N&tab=wi |
| 00:53.22 | JT | http://www.joytech.net/images/products/56-1-xbox360_headset.jpg |
| 00:53.22 | [TK]D-Fender | tengulre, headsets are dirt cheap. |
| 00:53.24 | fujin | That's a headset. |
| 00:53.25 | JT | ^headset |
| 00:53.39 | *** join/#asterisk djs_2_6 (n=djstillm@cpe-075-182-081-167.nc.res.rr.com) |
| 00:53.59 | coppice | fujin: fix your client :-) |
| 00:54.12 | tengulre | JT: lol, it's headset, yes, I using headset. |
| 00:54.52 | JT | on both sides? |
| 00:54.52 | tengulre | JT: yes. |
| 00:55.14 | fujin | coppice: might be the font I'm drawing with |
| 00:55.18 | tengulre | JT: when I turn down the mic volume , the echo is down. |
| 00:55.27 | JT | using both the speaker and micrphone function of the headsets?.... IF you have desk speakers, they should be on MUTE |
| 00:55.33 | JT | i see |
| 00:55.41 | JT | tried any other softphones? |
| 00:55.46 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 00:55.50 | tengulre | JT: yes, I have not desktop speakers. |
| 00:55.51 | coppice | fujin: what he entered was simplified chinese |
| 00:56.04 | JT | tengulre: ok |
| 00:56.12 | fujin | ah, win. |
| 00:56.16 | tengulre | JT: thanks, I try it now |
| 00:56.23 | JT | tengulre: idefisk |
| 00:56.25 | tengulre | coppice: where are u ? |
| 00:56.48 | coppice | HK |
| 00:57.02 | tengulre | JT: I developt myself softphone with iax2client libaray. |
| 00:57.16 | JT | tengulre: ok, so for a TEST, TRY idefisk |
| 00:57.20 | JT | see if it echos |
| 00:57.38 | tengulre | coppice: 我在大陸！ 西安妳聽說過嗎？ |
| 00:57.54 | tengulre | JT: OK ,thanks again. :-D |
| 00:58.00 | carrar | nihongo! |
| 00:58.06 | coppice | I guessed that. you wouldn't used simplified if you were elsewhere :-) |
| 00:58.23 | fujin | carrar: chinese, not japanese |
| 00:58.26 | coppice | carrar: er, no |
| 00:58.45 | carrar | only one kanji came through that I recognised |
| 00:58.59 | JT | it's NOT kanji |
| 00:59.01 | JT | duh |
| 00:59.15 | tengulre | coppice: what working do u doing? |
| 00:59.27 | carrar | japanese kanji took their set from the chinese so some of it is the same |
| 00:59.35 | coppice | I think most of that is valid Kanji as well as Hanzi |
| 00:59.53 | carrar | 大 |
| 00:59.57 | carrar | thats all I saw |
| 01:00.07 | carrar | pretty basic one |
| 01:00.31 | coppice | but a biggy :-) |
| 01:01.06 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:01.08 | tengulre | 我非常喜歡asterisk == (I very like asterisk), 哈哈哈！！！==(Hahaha!!!!) |
| 01:01.45 | carrar | heh |
| 01:02.03 | coppice | 我非常中國女人 :-) |
| 01:02.15 | tengulre | coppice: are you chinese? |
| 01:02.16 | coppice | 我非常喜歡中國女人 :-) |
| 01:02.36 | boch | if i cant rely in channel vars when h extension is called, how can i identify the channel to bill ? or the CHANNEL var is _ALWAYS_ present ? |
| 01:02.45 | coppice | 我是英國人 |
| 01:02.51 | fujin | take it to #china |
| 01:02.57 | fujin | I don't care for your filthy language |
| 01:03.20 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:03.31 | tengulre | coppice: 我給妳介紹一個中國女人 ＝＝（ I introduce a chinese woman to u) |
| 01:03.49 | tengulre | coppice: haha.. |
| 01:03.55 | coppice | I have one, thanks |
| 01:04.14 | tengulre | coppice: ?? I married a chinese woman? |
| 01:04.24 | tengulre | coppice : s/I/you |
| 01:04.37 | coppice | 是旳 |
| 01:05.07 | Yourname` | fujin: Just used the files mode!! And it works :) |
| 01:05.31 | Yourname` | [TK]D-Fender: Just used the files mode and it worked.. apparently there's this problem with mpg123, bla bla.. and files mode usually fixes it.. and it did. |
| 01:05.35 | Yourname` | Thanks to both of you. |
| 01:06.14 | *** join/#asterisk MindTheGap_ (n=iote@bhe201062195037.res-com.wayinternet.com.br) |
| 01:06.26 | tengulre | coppice: 恭喜你！ 中国女人是非常善良和勤劳的！！ 呵呵！ |
| 01:06.46 | [TK]D-Fender | Yourname`, No, there usually ISN'T a problem with pmg123, and files doesn't "fix it", it REPLACES it :) |
| 01:06.49 | tengulre | coppice:你是工程师？ |
| 01:07.05 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 01:07.47 | coppice | 是旳 |
| 01:08.37 | fujin | Yourname`: of course it works |
| 01:08.46 | fujin | transcoding your mp3s to the audio codec you're using works better |
| 01:08.49 | fujin | (less load) |
| 01:09.09 | fujin | I wouldn't want to be decoding mp3 and then transcoding it to ulaw,speex,g729, anything.. really. |
| 01:09.16 | fujin | especially more than 1 call at a time :P |
| 01:09.21 | Yourname` | fujin: Meaning like maybe use a raw player? |
| 01:09.34 | fujin | no, use files mode, but a codec that asterisk understands |
| 01:09.44 | Yourname` | Ahh.. and not mp3. |
| 01:09.48 | fujin | (i.e.; use SoX to convert your mp3 to ULAW, if you're using a ULAW-codec phoen system) |
| 01:09.51 | Yourname` | Hmm, I'm trying to install Sox |
| 01:09.54 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 01:09.59 | [TK]D-Fender | Yourname`, MP3 is FINE, fujin is just being PARANOID |
| 01:10.02 | Yourname` | Yes sir, I'm using a ulaw system |
| 01:10.05 | Yourname` | LOL |
| 01:10.13 | fujin | uh, not it's not |
| 01:10.16 | fujin | I did a load test the other day |
| 01:10.25 | fujin | 30 concurrent MOH callers, load of 2.0 on the box |
| 01:11.02 | Yourname` | Hmm, that sounds exactly like what we'll be.. |
| 01:11.10 | fujin | sox -V x.mp3 -t ul -r 8000 -b -c 1 x.ulaw |
| 01:11.21 | fujin | (for asterisk compatible playback) |
| 01:11.25 | fujin | as you can imagine, this cuts down on cycles |
| 01:11.30 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:11.41 | fujin | unless you're running like a dual-quad-Xeon, I wouldn't want to be using format_mp3 for a high-volume callcentre |
| 01:11.48 | fujin | from the testing that I've done, anyway; |
| 01:11.54 | Yourname` | sox -V *.mp3 -t ul -r 8000 -b -c 1 x.ulaw |
| 01:11.59 | Yourname` | For all my mp3s in there. hehe |
| 01:12.05 | fujin | that won't work |
| 01:12.06 | fujin | but yeah |
| 01:12.16 | fujin | you'll have to use a for i in x; do y; done; |
| 01:12.57 | fujin | you won't need to change moh.conf either, files just usese the applicable codec for the file (and if you're already using ulaw for everything, nothing will need transcoding) |
| 01:13.34 | [TK]D-Fender | fujin, If you have 30 people on hold, you need to get poeple to stop slacking and start taking calls! |
| 01:13.45 | fujin | That's hardly a solution to the problem |
| 01:13.51 | fujin | and I disagree with that statement also |
| 01:14.13 | [TK]D-Fender | fujin, Sure it is... faster response = happier customer = lower load on your system. Win-win! |
| 01:14.24 | [TK]D-Fender | fujin, So limited you are! |
| 01:14.33 | fujin | hrmph |
| 01:14.41 | fujin | while, logically your statement is correct |
| 01:14.53 | fujin | it is not a solution to the problem of decoding/transcoding mp3->ulaw |
| 01:15.23 | fujin | 30 people on hold is also a good simulation for 30 concurrent *calls* |
| 01:15.24 | [TK]D-Fender | fujin, I never proposed it to be. You're the one misrepresenting everything ;) |
| 01:15.38 | fujin | which is why we did it |
| 01:15.58 | fujin | going to retest on Friday with ulaw encoded MOH |
| 01:16.07 | [TK]D-Fender | fujin, And technically yreah I agree, encoding your MoH to the same codec as your majority uses and running Native MoH makes sense :) |
| 01:16.08 | fujin | which is obviously more similar to a two-way ulaw call |
| 01:16.22 | fujin | [TK]D-Fender: absolutely |
| 01:16.37 | fujin | unfortunately the person who specced the Asterisk boxes I've inherited wasn't the smartest |
| 01:16.42 | fujin | especially for a callcentre for 10k customers |
| 01:17.15 | *** join/#asterisk Byan (n=nullstri@c-71-227-62-198.hsd1.mi.comcast.net) |
| 01:17.35 | Byan | hey, I am looking to buy one or two X100P's |
| 01:17.42 | fujin | Intel(R) Pentium(R) D CPU 3.00GHz / 1024mb ddr2-533 / raid1 sata |
| 01:17.50 | Byan | they used to be litered all over ebay... |
| 01:17.53 | JT | Byan: not really advisable |
| 01:18.02 | Byan | anyone know where they can find really cheap ones |
| 01:18.06 | [TK]D-Fender | Byan, They are shit cards. |
| 01:18.09 | [TK]D-Fender | ~ygwypf |
| 01:18.10 | jbot | hmm... ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 01:18.22 | Byan | I understand that they suck... |
| 01:18.28 | Byan | but I only wanna experiment |
| 01:18.32 | JT | Byan: and are no longer made |
| 01:18.34 | JT | use softphones |
| 01:18.39 | fujin | google.com |
| 01:19.12 | *** join/#asterisk saftsack (n=saftsack@pD9E05B78.dip.t-dialin.net) |
| 01:19.16 | Byan | JT: and, how do you prupose I connect the PBX up the phone line then? |
| 01:19.24 | [TK]D-Fender | Byan, Agreed. Go download the ZOIPER soft-phone for testing & learning |
| 01:19.45 | [TK]D-Fender | Byan, thats not "testing" or "learning *" thats "production use of my line" |
| 01:19.55 | JT | ^^ |
| 01:20.11 | Byan | [TK]D-Fender: thats testing and learning how to do that stuff with my line that is free and never used |
| 01:20.35 | fujin | get a tdm400p or something |
| 01:20.38 | fujin | with one card in it |
| 01:20.44 | [TK]D-Fender | Byan, 0 cost phone line? Neato. I'll take 1000 :) |
| 01:20.45 | fujin | those Tiger card |
| 01:20.58 | Byan | [TK]D-Fender: well, it's not free, it's included |
| 01:21.19 | [TK]D-Fender | Byan, this story gets better every time I hear it! Continue! |
| 01:21.29 | Yourname` | fujin: You're right. Currently it's on mp3 and the load is 2+ |
| 01:21.34 | Byan | >_> |
| 01:21.37 | Yourname` | fujin: Going to be changing it now. |
| 01:21.45 | Byan | fujin: those are too expensive |
| 01:21.50 | fujin | Get a better job |
| 01:21.54 | fujin | sell more drugs |
| 01:21.58 | fujin | I dunno :) |
| 01:22.21 | Byan | fujin: I have hte money.. I just ain't gonna spend that much on it |
| 01:22.23 | fujin | Yourname`: yeah, the preliminary testing that I did (30calls) showed that they were inefficient for such a use |
| 01:22.23 | Byan | it's not worth it |
| 01:22.31 | fujin | well then investigate other alternatives |
| 01:22.32 | JT | Byan: get a linksys SPA-3102 |
| 01:22.38 | fujin | ^^. |
| 01:22.39 | JT | ~cheap |
| 01:22.40 | jbot | somebody said cheap was a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 01:22.45 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 01:22.53 | fujin | spa3102 is a pstn->sip device |
| 01:23.04 | Yourname` | fujin: The calls were 50+ though, in the queue. But load avg'd around 2.1 |
| 01:23.05 | [TK]D-Fender | SPA-3102 is a considerably better choice than a X100P |
| 01:23.18 | fujin | Yourname`: how did you manage to simulate 50 calls? |
| 01:23.27 | fujin | hell - the only way I could do that was to get everyone to pick up their phone! |
| 01:23.35 | JT | fujin: sipp |
| 01:23.49 | fujin | does sipp actually negotiate a "call"? |
| 01:23.52 | fujin | like, brings up rtp |
| 01:23.54 | fujin | bridges, etc? |
| 01:24.07 | Byan | JT: still over budget >_> |
| 01:24.10 | JT | yes |
| 01:24.19 | JT | Byan: then adjust your budget |
| 01:24.23 | JT | it is in fantasy land |
| 01:24.49 | [TK]D-Fender | Byan, You know what just do whatever you're going to do and be prepared to live with it. |
| 01:25.00 | [TK]D-Fender | Byan, got any real questions for us? |
| 01:25.51 | [TK]D-Fender | :) |
| 01:26.07 | Byan | coulda just.. said.. no.. |
| 01:26.21 | Byan | I know very well they are sucky cards >_> |
| 01:26.31 | [TK]D-Fender | Byan, we know you don't want to spend a penny more than you think you have to or want to and I'm going to move on. |
| 01:27.29 | Byan | whatever |
| 01:27.39 | [TK]D-Fender | Byan, Then you are either a masochist, and idiot, or an optimist (often another name for one or both of the the former two) |
| 01:27.41 | [TK]D-Fender | hehe |
| 01:27.50 | [TK]D-Fender | Byan, SO.. what can we do for you now :) |
| 01:28.11 | [TK]D-Fender | Byan, Some people ARE happy with their X100P's... maybe you will too. |
| 01:29.15 | mvanbaak | it's a bad card |
| 01:29.20 | mvanbaak | I tried it as well |
| 01:29.27 | Yourname` | fujin: sox: Failed reading fpm-calm-river.mp3: Do not understand format type: mp3 :( |
| 01:29.36 | mvanbaak | even plain ulaw over my isdn dialin connection works better |
| 01:29.47 | [TK]D-Fender | Yourname`, you need to install asterisk-addons for MP3 support |
| 01:30.15 | Yourname` | [TK]D-Fender: But asterisk was so far playing .mp3s in the mohmp3 directory..? |
| 01:30.49 | [TK]D-Fender | Yourname`, wait.. thats a SOX error.. where are you getting that? |
| 01:30.55 | Byan | I mean, really, this is suppose to be a way to utilize the line spending little money.. I am spending way too much lately... and why am I still talking |
| 01:31.18 | Yourname` | [TK]D-Fender: In /var/lib/asterisk/mohmp3 when I do: sox -V *.mp3 -t ul -r 8000 -b -c 1 *.ulaw |
| 01:31.38 | mvanbaak | you need sox with mp3 support |
| 01:31.56 | [TK]D-Fender | Yourname`, Yay, more apps to add support for! |
| 01:32.28 | mvanbaak | I'm off to bed |
| 01:32.29 | mvanbaak | latero |
| 01:32.41 | Yourname` | mvanbaak: !! |
| 01:32.53 | Yourname` | mvanbaak: How do I do it? I don't see anywhere on that website that tells me about it.. |
| 01:32.55 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 01:33.08 | Yourname` | [TK]D-Fender: Is that a good thing? hehe |
| 01:33.20 | *** part/#asterisk bintut (n=bintut@203.125.63.150) |
| 01:33.35 | [TK]D-Fender | Byan, maybe you're trying to find someone to help you accept that you don't want to be unhappy with a planned purchase. You say you are rightfully concerned about the purchase you are planning yet are preparing to go through with it anyways. |
| 01:34.27 | [TK]D-Fender | Byan, So you either want our acceptance or to help you overcome the hudle of accepting a larger expense to ensure your long-term happiness with your solution. |
| 01:34.54 | [TK]D-Fender | I should Op @ #drphil ...... |
| 01:35.04 | Byan | but... the thing is gonna be used for like.. |
| 01:35.09 | Byan | well |
| 01:35.16 | Byan | 12-9 |
| 01:35.20 | Byan | 3 months.. |
| 01:35.26 | Byan | I can't justify spending over $30 |
| 01:35.37 | [TK]D-Fender | 3 months? Then what? |
| 01:35.54 | Byan | then, I don't have the line |
| 01:36.00 | [TK]D-Fender | Byan, ... and THEN what? |
| 01:36.07 | Byan | uh, I don't use a line.. |
| 01:36.13 | Byan | all on my cell at that point |
| 01:36.26 | [TK]D-Fender | Byan, no other PST of any kind? |
| 01:36.30 | [TK]D-Fender | PSTN* |
| 01:36.34 | Byan | yep |
| 01:36.51 | [TK]D-Fender | Byan, Why don't you go full-cell right now? |
| 01:37.19 | Byan | I am full cell right now.. |
| 01:37.24 | Byan | as I said, the line is included |
| 01:37.33 | [TK]D-Fender | Byan, Then why buy ANYTHING for your land line? |
| 01:37.45 | Byan | for the hell of it |
| 01:37.46 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 01:37.53 | [TK]D-Fender | Byan, Sounds like its already defunct. why spend a single penny on it in that case? |
| 01:38.05 | Byan | cause I wanna mess around with asterisk? |
| 01:38.06 | [TK]D-Fender | Byan, Well you've got the best of reasons right there.... |
| 01:38.32 | [TK]D-Fender | Byan, and once your land-line goes bye-bye, thats it for * huh? |
| 01:38.58 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:38.58 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 01:39.02 | Byan | unless I get a SIP service |
| 01:39.14 | Byan | I wouldn't be using PSTN again |
| 01:39.19 | [TK]D-Fender | Byan, And the odds on that would be? |
| 01:39.35 | [TK]D-Fender | Byan, PSTN does not mean an analog line. |
| 01:39.39 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 01:39.41 | Byan | oh |
| 01:39.46 | Byan | what does it mean? |
| 01:39.49 | [TK]D-Fender | Byan, assumed wrong :) |
| 01:39.51 | blitzrage | it could be BRI, PRI, etc... |
| 01:39.54 | JT | public switched telephone network |
| 01:39.58 | [TK]D-Fender | Byan, Analog line is one WAY to get to the PSTN |
| 01:40.01 | blitzrage | anything that connects to the public switched telephone network |
| 01:40.09 | JT | pstn is the global phone network |
| 01:40.23 | Byan | oh |
| 01:40.31 | magic_hat | hey everyone. the first few seconds of our greeting message are getting cut off on inbound calls. Anyone have an idea why that might be? It happens on external calls, but not on ones that go through our VOIP. here's a log: http://pastie.caboo.se/88114 |
| 01:40.32 | [TK]D-Fender | Byan, ISDN PRI / BRI, CAS T1, SS&, to name a few |
| 01:40.33 | blitzrage | a 'switched' network, as opposed to a packetized (i.e. Internet) network |
| 01:40.42 | Byan | [TK]D-Fender: I got ya now |
| 01:40.44 | JT | not any switched network |
| 01:40.51 | JT | THE network :) |
| 01:40.51 | Byan | well, if I got SIP, I'd be using IP phones.. and such.. |
| 01:40.57 | blitzrage | no, not any switched network :) |
| 01:41.03 | Byan | all digital.. |
| 01:41.05 | [TK]D-Fender | Byan, So what are the realistic odds of you using * with say an ITSP afterwards? |
| 01:41.27 | Byan | define ITSP |
| 01:41.45 | [TK]D-Fender | ~itsp |
| 01:41.45 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 01:41.46 | blitzrage | Internet Telephony Service PRovider |
| 01:42.02 | blitzrage | lol |
| 01:42.06 | [TK]D-Fender | Byan, a VoIP provider to place and/or receive calls |
| 01:42.22 | Byan | uhm, pretty good eventually.. |
| 01:42.31 | [TK]D-Fender | Byan, GOOD! Here : http://www.telephonydepot.com/product_p/105-054-312.htm |
| 01:42.54 | [TK]D-Fender | Byan, This will leyt you use to analog LINE (3 month life) with * for $70.95 |
| 01:43.11 | [TK]D-Fender | Byan, in ADDITION, it will let you use 1 analog PHONE as a SIP phone with * |
| 01:43.34 | Byan | [TK]D-Fender: and why would I want an analog phone? |
| 01:43.36 | [TK]D-Fender | Byan, So you don't have to use some shitty headset with * after (or during) this process |
| 01:44.11 | [TK]D-Fender | Byan, YDo you want to use a crappy headset chaining you to your desk to make a call, or use an analog phone you likely anready OWN? |
| 01:45.03 | magic_hat | lol, please say shitty headset! |
| 01:45.07 | Byan | I wanna use an IP phone.. |
| 01:45.11 | Byan | like I already own |
| 01:45.18 | [TK]D-Fender | Byan, Oh, what do you have now? |
| 01:45.51 | Byan | an F1000 |
| 01:46.05 | [TK]D-Fender | Byan, ICK... but I guess that counts... |
| 01:46.24 | Byan | [TK]D-Fender: remember I said my budget was low =) |
| 01:46.43 | magic_hat | http://pastie.caboo.se/88114 anyone? |
| 01:47.47 | [TK]D-Fender | Byan, Well the SPA-3102 is reusable in so many different ways (like letting you leach off someone elses lines after your done with yours) so as to be able to recoup the value, or use a wired normal phone. |
| 01:47.56 | [TK]D-Fender | Byan, in the end its all up to your. |
| 01:47.58 | [TK]D-Fender | you* |
| 01:48.34 | Byan | well, see |
| 01:48.38 | Byan | I could pay for that |
| 01:48.41 | Byan | but on the other hand.. |
| 01:49.26 | Byan | years from now when I actually need a home phone.. things will probably be either A. better B. cheaper |
| 01:49.38 | blitzrage | rmayorga: is your first name Ray? |
| 01:49.41 | SplasPood | this conversation makes my head hurt |
| 01:50.10 | [TK]D-Fender | magic_hat, Put your Wait AFTER your answer for 2s |
| 01:50.23 | rmayorga | blitzrage: no, is Rene |
| 01:50.38 | blitzrage | rmayorga: oh ok... for some reason I was trying to figure out why I knew your name :) |
| 01:50.44 | blitzrage | but apparently I didn't |
| 01:50.45 | blitzrage | heh |
| 01:50.46 | Byan | that and..... another outside device.. to stick places.. I already got too many of those.. |
| 01:50.47 | NightMonkey | Byan: I seconf [TK]D-Fender, I got a GrandStream HT-488, ok but no callerID from PSTN, then got a SPA-3102, and joy was accomplished. That, and got a GrandStream BT-200 SIP phone, and use Twinkle on my lappy, and my "playing around" learning kit is complete. |
| 01:51.28 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 01:51.32 | [TK]D-Fender | Byan, well your X100p idea has a) problems likely b) low redeployability. Pick your poison. |
| 01:51.37 | NightMonkey | s/seconf/second/ |
| 01:51.43 | NightMonkey | wow |
| 01:51.53 | NightMonkey | Smart bot! |
| 01:52.25 | Byan | NightMonkey: yeah, but.. how much did you pay? |
| 01:53.07 | Byan | s/y/n/g |
| 01:53.13 | Byan | xD |
| 01:53.14 | NightMonkey | Byan: About $140, IIRC. Could go cheaper with a BT-100 phone. |
| 01:53.34 | Byan | NightMonkey: yeah, 4 times my budget =) |
| 01:53.43 | NightMonkey | Byan: OK, good luck. |
| 01:54.03 | [TK]D-Fender | Byan, $70.95 <- |
| 01:54.54 | blinky42 | Byan, buy a tdm400p on ebay, play for x months then sell it on ebay. total cost < 30 in the end. |
| 01:54.56 | magic_hat | TKD-Fender: do I want answer, ringing, wait? or ringing, answer, wait? |
| 01:55.11 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 01:55.31 | Byan | blinky42: meh, I'll never do that.. I'll be too lazy to resell it |
| 01:55.41 | asterisknerds | <PROTECTED> |
| 01:55.43 | [TK]D-Fender | magic_hat, Answer, wait, no ringing. |
| 01:55.50 | [TK]D-Fender | asterisknerds you don't say! |
| 01:56.08 | JT | the log bot dies again |
| 01:56.11 | Yourname` | Damn, finding libmp3lame IS a hard thing! |
| 01:56.39 | CoffeeKid | google.com? :) |
| 01:56.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:57.15 | CoffeeKid | search for lame, when compiled it makes libmp3lame |
| 01:57.45 | *** join/#asterisk techman97_andy (n=me@75-134-128-138.dhcp.roch.mn.charter.com) |
| 01:57.48 | techman97_andy | evening all |
| 01:58.10 | techman97_andy | ok, so I have a PRI and am getting inband signalling that I need Asterisk to recognize. It's not now. the card is a TE212P. I |
| 01:58.17 | techman97_andy | I've been looking all over and can't find the option to turn on. |
| 01:58.19 | techman97_andy | any hints? |
| 01:58.22 | magic_hat | TKD-Fender: thanks. |
| 01:58.35 | techman97_andy | hey [TK]D-Fender - I'm back. Couldn't find the option to do this. |
| 01:58.56 | JT | techman97_andy: what sort of inband signalling/ |
| 01:59.23 | magic_hat | TKD -- is there any way to get the call to ring while it's going thru teliax? The ringing actually doesn't start til it hits my * server. |
| 01:59.32 | techman97_andy | basically, this number I'm calling, when you call it from a cell phone / land line / etc, you get a slow busy. Asterisk calling it over the PRI just rings and rings and rings until the remote side disconnects the call |
| 01:59.48 | [TK]D-Fender | magic_hat, why would you want it to ring? waste of time. |
| 01:59.53 | techman97_andy | watching the pri debug, you can see the PRI telling Asterisk there is inband info available, but Asterisk doesn't do anything with it |
| 01:59.56 | *** join/#asterisk jmesquita (n=jmesquit@200.162.229.225.user.ajato.com.br) |
| 02:00.28 | magic_hat | TKD-Fender... I dunno, because it takes like five second to get from placing the call to hitting our * box. it sounds weird. |
| 02:01.02 | JT | techman97_andy: are you talking about call supervision? |
| 02:01.25 | techman97_andy | I don't know the term for it...I just need it to pass through whatever is happening to the extension...=P |
| 02:01.43 | [TK]D-Fender | magic_hat, You shouldn't HAVE a pause for an ITSP. |
| 02:01.59 | techman97_andy | 90% of my calls work fine with normal PRI hangup codes...these other ones have this inband signalling....and from what my telco tells me, that inband crap is like busy tones, operator voice messages, etc. |
| 02:02.21 | JT | techman97_andy: can you hear the inband audio? |
| 02:02.24 | magic_hat | TKD-Fender: hmm. Teliax ticket, then? |
| 02:02.31 | techman97_andy | no, all I get is ringing from Asterisk. |
| 02:02.42 | techman97_andy | but the PRI debug says there is inband available, but just doesn't do crap with it |
| 02:02.44 | magic_hat | + |
| 02:03.02 | techman97_andy | I have my zapata.conf set to "priindication=inband", still nothing |
| 02:05.38 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 02:06.51 | techman97_andy | any ideas anyone? |
| 02:07.01 | JT | techman97_andy: show us your dial string |
| 02:07.50 | techman97_andy | ZAP/g1/<my number here>|2000|trWw |
| 02:08.01 | JT | .... |
| 02:08.04 | JT | remove r |
| 02:08.27 | JT | never use the r dial option |
| 02:08.32 | techman97_andy | is the "r" responsible for my inband problem/ |
| 02:08.33 | techman97_andy | ? |
| 02:08.34 | JT | i think it means "retarded" |
| 02:08.35 | JT | yes. |
| 02:09.15 | techman97_andy | trying |
| 02:09.44 | techman97_andy | SON OF A BITCH...rofl |
| 02:10.11 | techman97_andy | I can't believe that was the problem |
| 02:10.12 | JT | don't use unnecessary dial options |
| 02:10.13 | techman97_andy | =P |
| 02:10.39 | techman97_andy | JT, you the man. |
| 02:10.43 | JT | i doubt t or w are needed either |
| 02:10.47 | techman97_andy | you just saved my ass a bunch of moolah. |
| 02:10.51 | techman97_andy | =D |
| 02:10.52 | JT | W maybe |
| 02:11.11 | techman97_andy | next time you're in MN, hit me up - I owe you a beer...hehehe |
| 02:11.20 | JT | haha that might be a while |
| 02:11.25 | techman97_andy | well, offer open. |
| 02:11.25 | techman97_andy | rofl |
| 02:11.31 | JT | haven't been to the US yet |
| 02:14.53 | *** join/#asterisk remmo (n=junk@202.1.119.80) |
| 02:15.37 | pigpen | What does this mean? [Aug 15 21:13:34] NOTICE[17513]: chan_sip.c:14758 handle_request_register: Registration from '<sip:6241@172.20.192.254>' failed for '172.20.194.28' - Device does not match ACL |
| 02:15.59 | blitzrage | you have a permit/deny statement that is too restrictive |
| 02:16.11 | pigpen | in the sip.cfg? |
| 02:16.15 | pigpen | sorry...sip.conf |
| 02:16.15 | blitzrage | sip.conf |
| 02:16.18 | blitzrage | yes |
| 02:16.21 | pigpen | hmm.. |
| 02:16.27 | blitzrage | look at the IP address, and look at your statements |
| 02:16.33 | pigpen | k. |
| 02:16.46 | pigpen | just added about 70 phones to the existing 80 or so. |
| 02:16.56 | pigpen | there is allot going on. |
| 02:17.03 | pigpen | thanks for the pointer. |
| 02:17.24 | blitzrage | probably sitting in a different subnet or something |
| 02:19.22 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:19.33 | pigpen | hmm... I have 14 subnets. |
| 02:19.36 | pigpen | :) |
| 02:20.27 | blitzrage | well maybe you can start by showing your permit/deny statements for that extension :) |
| 02:21.31 | pigpen | Well, this is embarrassing...I don't have neither! |
| 02:21.44 | pigpen | Just host=dynamic |
| 02:22.10 | pigpen | I will review the wiki for the sip.conf |
| 02:22.17 | blitzrage | I'm going to bed |
| 02:22.24 | pigpen | I don't blame you. |
| 02:22.29 | *** join/#asterisk dammit2006 (i=HydraIRC@dsl-220-235-198-14.nsw.westnet.com.au) |
| 02:29.34 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 02:29.57 | [TK]D-Fender | pigpen, you put "172.20.194.28" as a USER ID in there. |
| 02:30.01 | [TK]D-Fender | ^^^^^^^^^^ |
| 02:30.26 | pigpen | in the phone config ? |
| 02:30.41 | [TK]D-Fender | pigpen, Yes. What phone? Let me guess.. a Polycom... |
| 02:30.48 | pigpen | :-P |
| 02:31.45 | pigpen | reg.1.auth.userId="6234" |
| 02:32.14 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 02:32.35 | [TK]D-Fender | pigpen, the field labeled "reg.x.address" doe not mean the host/ip. It actually means the USER NAME. |
| 02:32.55 | [TK]D-Fender | (due to timeout I assumed your answer was "yes") |
| 02:32.57 | pigpen | ah, not "6234" |
| 02:33.30 | pigpen | Here is the first part of the line: <reg reg.1.displayName="6234" reg.1.address="6234" reg.1.label="6234" reg.1.type="private" reg.1.lcs="" reg.1.thirdPartyName="6234" reg.1.auth.userId="6234" reg.1.auth.password="letmein!" reg.1.server.1.address="172.20.192.254" reg.1.server.1.port="5060" reg.1.server.1.transport="DNSnaptr" reg.1.server.2.transport="DNSnaptr" |
| 02:33.41 | pigpen | Sorry for the length...but heh... |
| 02:34.39 | pigpen | You know, it also may be a mix, that the phone config files was based on sip 2.1.1 where the phones are running 2.0.3 |
| 02:34.49 | pigpen | I am in the process of rebooting and updating them all. |
| 02:34.53 | [TK]D-Fender | there you go, thats good |
| 02:35.22 | pigpen | yeah...I know there are quite a few differences between 2.0.3 and 2.1.1 |
| 02:35.31 | pigpen | I already took it in the teeth on that one. |
| 02:36.01 | pigpen | You know that bell thing I was doing, I decided just to do a distinctive ring instead...since they never really answer. |
| 02:37.28 | *** join/#asterisk Corydon76-dig (i=pink@pdpc/supporter/sustaining/Corydon76-home) |
| 02:37.29 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 02:38.46 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 02:38.56 | ZaVoid | sup all |
| 02:39.04 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:41.11 | ZaVoid | anyreason why insecure=very and asterisk still sends 407 proxy auth requried? |
| 02:41.59 | JT | allowguest=yes |
| 02:42.45 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:43.05 | ZaVoid | in the sip peers? |
| 02:43.10 | *** join/#asterisk d3wayne (n=fluffhea@c-71-228-186-75.hsd1.al.comcast.net) |
| 02:43.10 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 02:43.11 | ZaVoid | should i remove the insecure? |
| 02:43.37 | ZaVoid | i have allowguest=yes in my [general] |
| 02:43.59 | *** part/#asterisk Corydon76-dig (i=pink@pdpc/supporter/sustaining/Corydon76-home) |
| 02:44.22 | *** join/#asterisk anthony[ (n=anthony@fl-71-49-118-147.dhcp.embarqhsd.net) |
| 02:46.24 | anthony[ | Hi |
| 02:46.34 | anthony[ | If I am changing providors, I already have a setup on my server |
| 02:46.46 | anthony[ | What file do I need to edit to have asterisk connect? |
| 02:47.32 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 02:47.57 | ZaVoid | yeah meh SIP/2.0 407 Proxy Authentication Required |
| 02:48.19 | ZaVoid | probably sip.conf anderiv |
| 02:48.21 | ZaVoid | anthony |
| 02:48.28 | ZaVoid | JT anythouhts on the proxy auth? |
| 02:49.26 | anthony[ | I checked sip.conf |
| 02:49.27 | anthony[ | nothing |
| 02:49.29 | anthony[ | aix? |
| 02:49.40 | anthony[ | iax.conf? |
| 02:50.17 | ZaVoid | possible |
| 02:50.34 | [TK]D-Fender | anthony[, We're only psychic on TUESDAYS. |
| 02:51.06 | *** part/#asterisk dammit2006 (i=HydraIRC@dsl-220-235-198-14.nsw.westnet.com.au) |
| 02:51.47 | ZaVoid | hey fender :) |
| 02:51.52 | ZaVoid | hows your night going |
| 02:52.32 | [TK]D-Fender | meh |
| 02:52.58 | [TK]D-Fender | refreshing all the software on my laptop (winxp/ubuntu), and otherwise just killing time |
| 02:53.42 | ZaVoid | nice |
| 02:53.43 | ZaVoid | [bob] |
| 02:53.43 | ZaVoid | type=peer |
| 02:53.44 | ZaVoid | insecure=very |
| 02:53.44 | ZaVoid | host=1.2.3.4 |
| 02:53.44 | ZaVoid | context=jane |
| 02:53.47 | fujin | ~pb |
| 02:53.48 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:53.48 | ZaVoid | drives me nuts that won't work |
| 02:53.55 | ZaVoid | yeah i flodded sorry |
| 02:54.06 | ZaVoid | still drives me nuts every time aterisk sends back 407 |
| 02:54.14 | [TK]D-Fender | ZaVoid, that BEGINS to resemble what I'd ask for. |
| 02:54.34 | ZaVoid | thats the whole sip peer |
| 02:55.03 | ZaVoid | strange thing is i never see it hitting that context ever.. soonas invite comes in it goes 407!! |
| 02:55.34 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 02:56.04 | asterisknerds | <PROTECTED> |
| 03:04.05 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:05.16 | CoffeeKid | hello all |
| 03:05.51 | CoffeeKid | i'm doing a transfer from (using agi) of two channels to a queue... but the second channel thats transfered to the queue loses its ability to talk (but can still listen)... |
| 03:06.17 | CoffeeKid | well, loses its ability to talk after its then transfered again to an agent. |
| 03:06.18 | *** join/#asterisk MdeP (n=mdep@219-91-22-190.adsl.tie.cl) |
| 03:08.48 | *** join/#asterisk mtaht4 (n=m@12.161.178.66) |
| 03:10.11 | anthony[ | What file do I need to edit to have asterisk connect? |
| 03:10.16 | anthony[ | If I am changing providors, I already have a setup on my server |
| 03:12.01 | CoffeeKid | anthony[, using a t1? |
| 03:14.30 | *** join/#asterisk Corydon76-dig (i=four@pdpc/supporter/sustaining/Corydon76-home) |
| 03:14.30 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 03:16.19 | Mavvie | configure works. |
| 03:17.23 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 03:17.51 | Juggie | coffee, can still listen to what, moh? can still hear the agent when they answer, etc? |
| 03:18.00 | *** join/#asterisk Olobola (n=chatzill@74.95.13.57) |
| 03:18.38 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 03:19.03 | AJaymn | Anyone know where there is a how-to for installing Zaptel 1.4.10 on kernel 2.6+ |
| 03:19.37 | Juggie | ./configure; make install |
| 03:19.44 | tengulre | AJaymn: read the README. |
| 03:21.29 | CoffeeKid | juggie, I'm trying to do an attended transfer using AGI... so to old the channels I put both callers in a Queue, then make the one caller call the transfered caller... |
| 03:21.31 | anthony[ | CoffeeKid a t1? |
| 03:21.37 | anthony[ | what does that have to do with my Q? |
| 03:22.06 | CoffeeKid | when the transfered to caller answers, they can hear everything but can't talk. |
| 03:22.49 | Mavvie | it compiles! |
| 03:23.59 | CoffeeKid | anthony[ , sorry, was confused about your question. |
| 03:25.27 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:27.08 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 03:30.47 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 03:30.58 | JerJer | has anyone seen this b4? |
| 03:30.59 | JerJer | [Aug 15 23:20:18] WARNING[9300]: codec_g729.c:210 g729tolin_framein: Invalid data (4 bytes at the end) |
| 03:31.00 | JerJer | [Aug 15 23:20:18] WARNING[9300]: translate.c:199 framein: g729tolin did not update samples 0 |
| 03:32.24 | JerJer | they say silence suppression is turned off |
| 03:35.14 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-126-140-84.dsl.irvnca.pacbell.net) |
| 03:38.23 | tengulre | does the asterisk support this way, Zap channel(20lines) ----> Queue----> IAX2 client(10 lines)? |
| 03:39.33 | [TK]D-Fender | tengulre, yup |
| 03:39.47 | tengulre | ?? |
| 03:39.58 | [TK]D-Fender | tengulre, you asked,I answered. |
| 03:40.10 | tengulre | yup? |
| 03:41.09 | tengulre | [TK]D-Fender: I dont like agent mode! |
| 03:42.35 | JT | !! |
| 03:44.02 | *** join/#asterisk jmesquita (n=jmesquit@200.162.229.225.user.ajato.com.br) |
| 03:44.11 | fujin | don't use agents, it's dumb. |
| 03:52.33 | tengulre | fujin: I agree u! :-D |
| 03:55.13 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 03:55.22 | AJaymn | I think that the zaptel hardware you have on your system is: |
| 03:55.37 | AJaymn | this is all i get when i try to compile zaptel... it never says it found anything |
| 03:55.43 | asterisknerds | <PROTECTED> |
| 03:57.00 | JT | AJaymn: what the hell, it finds stuff now? |
| 03:58.38 | AJaymn | ha?! |
| 03:58.39 | AJaymn | lol |
| 03:58.53 | AJaymn | im trying to install zaptel-1.4.4 |
| 03:58.59 | AJaymn | on a 2.6 kernel! |
| 03:59.14 | JT | AJaymn: i've always loaded the driver(s) i needed myself and configured zaptel myself |
| 03:59.31 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:59.32 | AJaymn | all i want is ztdummy :( |
| 03:59.38 | AJaymn | i just want the conference rooms to work |
| 03:59.45 | *** join/#asterisk dguido (n=dguido@c-69-140-135-20.hsd1.md.comcast.net) |
| 03:59.45 | JT | so wait |
| 03:59.54 | AJaymn | wait for what? |
| 04:00.10 | JT | you're telling me your system has no zaptel hardware, and you're wondering why it's not autodetecting any zaptel hardware? |
| 04:00.26 | AJaymn | shouldnt it say ztdummy? |
| 04:00.53 | JT | how does it detect hardware that doesn't exist? |
| 04:01.06 | AJaymn | i figured it would say ztdummy :P |
| 04:01.24 | JT | i doubt it |
| 04:01.29 | JT | seeing as it isn't |
| 04:02.36 | AJaymn | oh shutup :P |
| 04:03.12 | AJaymn | how do i get the conference rooms to work then? it keeps says that is not a valid conference #.... as if ztdummy isnt loaded |
| 04:03.19 | Juggie | JT = John Todd? |
| 04:03.22 | JT | no |
| 04:03.30 | Juggie | confusing |
| 04:03.31 | tzafrir_laptop | AJaymn, that is the output of the script xpp/utils/zaptel_hardware |
| 04:03.39 | JT | who is this john todd guy/ |
| 04:03.40 | tzafrir_laptop | It has been fixed in SVN |
| 04:03.45 | Juggie | an asterisk guy |
| 04:03.51 | JT | i see |
| 04:04.14 | tzafrir_laptop | anyway, if you just need ztdummy you don'r need zaptel hardware |
| 04:05.16 | AJaymn | tzafrir_laptop Not a thing. |
| 04:05.31 | AJaymn | [root@chi utils]# ./zaptel_hardware |
| 04:05.31 | AJaymn | [root@chi utils]# |
| 04:06.01 | tzafrir_laptop | right. You just want ztdummy so no need for hardware |
| 04:06.06 | AJaymn | thought you had to compile zaptel to get the conference rooms to work |
| 04:06.09 | tzafrir_laptop | sorry for the confusion |
| 04:06.38 | JT | AJaymn: you need zaptel to use MeetMe |
| 04:06.46 | tzafrir_laptop | The zaptel init.d script should load ztdummy if it can't find hardware |
| 04:07.08 | AJaymn | so what am i doing wrong?! :( |
| 04:07.18 | AJaymn | 1.4 is so confusing :P |
| 04:09.36 | tzafrir_laptop | What do you see wrong? What distro do you use? |
| 04:10.02 | AJaymn | fedora core 6 |
| 04:10.25 | tzafrir_laptop | try: mv /etc/default/zaptel /etc/sysconfig/zaptel |
| 04:10.44 | AJaymn | then reboot? |
| 04:10.46 | tzafrir_laptop | but before that: what's the output of: zttest |
| 04:10.50 | tzafrir_laptop | no reboot needed |
| 04:11.27 | AJaymn | Unable to open zap interface: No such file or directory |
| 04:14.15 | AJaymn | Unable to open zap interface: No such file or directory |
| 04:17.06 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.180.56) |
| 04:24.18 | [TK]D-Fender | AJaymn, "modprobe zaptel" |
| 04:24.25 | [TK]D-Fender | AJaymn, "modprobe ztdummy" |
| 04:24.32 | [TK]D-Fender | AJaymn, "ztcfg -vvvv" |
| 04:24.37 | [TK]D-Fender | AJaymn, then start * |
| 04:24.47 | sevard | [TK]D-Fender: don't you get tired of saying the same stuff, over and over? |
| 04:24.48 | JT | the modprobe zaptel bit is optional is it not |
| 04:24.56 | [TK]D-Fender | AJaymn, I'd personally suggest doing all this BEFORE even recompiling * |
| 04:25.12 | [TK]D-Fender | sevard, load chan_brokenrecord.so |
| 04:25.30 | sevard | excellent |
| 04:25.39 | [TK]D-Fender | JT : in my world "optional" applies only if you want it to WORK. |
| 04:26.02 | JT | [TK]D-Fender: i've always found that loading the kernel module for the relevant driver calls zaptel |
| 04:26.10 | JT | unless it's a buggy release |
| 04:27.23 | AJaymn | FATAL: Error inserting ztdummy (/lib/modules/2.6.22.1-32.fc6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 04:27.33 | *** join/#asterisk r0b- (i=rob@adsl-76-235-210-74.dsl.klmzmi.sbcglobal.net) |
| 04:28.13 | fujin | anyone familiar with sip-tester (sipp) |
| 04:28.17 | fujin | and/or how to use it with asterisk? |
| 04:28.42 | r0b- | configure: error: *** termcap support not found |
| 04:28.58 | r0b- | i checked apt-get for termcap and it came up negative |
| 04:29.16 | tzafrir_laptop | libncurses-dev |
| 04:29.26 | tzafrir_laptop | r0b-, work in progress script: |
| 04:30.16 | r0b- | ty |
| 04:30.17 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 04:30.17 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:30.23 | tzafrir_laptop | http://updates.xorcom.com/astribank/bristuff/1.4/bristuff-current/prereq.sh |
| 04:30.28 | tzafrir_laptop | wget it |
| 04:30.31 | tzafrir_laptop | and then: |
| 04:30.38 | tzafrir_laptop | ./prereq.sh test |
| 04:30.54 | r0b- | hang on |
| 04:30.58 | tzafrir_laptop | It goes for a rather maximalistic approach |
| 04:31.00 | JT | AJaymn: make sure it's actually compiled against your current running kernel |
| 04:31.56 | AJaymn | JT thats the problem.. because im trying to compile 1.4 on a 2.6 kernel theres so many conflicting howtos.. |
| 04:32.16 | r0b- | wabam! |
| 04:32.17 | r0b- | it works |
| 04:32.18 | r0b- | :) |
| 04:32.28 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 04:32.32 | JT | AJaymn: how hard can it be? |
| 04:32.44 | JT | everyone uses 2.6 these days btw, especially for asterisk |
| 04:33.24 | antimoof | the biggest problem I'd gotten with compiling asterisk is with the zaptel module, cuz my stoooopid linux has a bogus include problem. |
| 04:33.50 | AJaymn | http://www.voip-info.org/wiki/index.php?page=Asterisk+timer+ztdummy |
| 04:34.47 | [TK]D-Fender | AJaymn, ANCIENT |
| 04:35.35 | [TK]D-Fender | AJaymn, untar it. "./configure" , "make menuselect" (make sue to SAVE as you quit), "make", "make install", "make samples" |
| 04:36.04 | JT | [TK]D-Fender: you talking about zaptel or asterisk? |
| 04:36.10 | r0b- | make takes awhile |
| 04:36.45 | AJaymn | sorry to be stupid TK but what does make sue to save as you quit mean? :P |
| 04:37.01 | JT | i think he's talking about compiling asterisk |
| 04:37.11 | AJaymn | oh make sure! |
| 04:37.13 | AJaymn | nevermind :) |
| 04:37.13 | JT | first of all, you need zaptel compiled properly against your current kernel |
| 04:37.23 | JT | sounds like it's compiled against the wrong kernel |
| 04:37.30 | AJaymn | ok so im inside menuselect now and the only thing thats selected is ztdummy |
| 04:37.42 | JT | that sounds ok |
| 04:38.13 | AJaymn | whats the command to see what current kernel is running? |
| 04:38.33 | JT | uname -a |
| 04:39.01 | r0b- | how long does the "make" take to finish |
| 04:39.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:39.09 | sheppard | weeks |
| 04:39.12 | AJaymn | WARNING: "rtc_register" [/usr/src/zaptel/ztdummy.ko] undefined! |
| 04:39.12 | AJaymn | WARNING: "rtc_unregister" [/usr/src/zaptel/ztdummy.ko] undefined! |
| 04:39.17 | r0b- | im serious |
| 04:39.25 | sheppard | so am I |
| 04:39.28 | r0b- | lol |
| 04:39.31 | JT | r0b-: on what hardware/ |
| 04:39.41 | sheppard | asterisk took a week to compile on my 486/16 w/ 4MB of ram |
| 04:39.46 | JT | haha |
| 04:39.50 | r0b- | P4 2.8 with 100MB ram |
| 04:39.51 | *** join/#asterisk lbow (n=lbow@41-195-77-226.access.uunet.co.za) |
| 04:39.56 | JT | you should've crosscompiled |
| 04:40.05 | JT | r0b-: shouldn't take more than 15-20mins |
| 04:40.14 | JT | then again |
| 04:40.15 | r0b- | well its Virtual Machine |
| 04:40.16 | sheppard | JT: I had 80 boxes running in a distcc cluster |
| 04:40.19 | sheppard | still took a week |
| 04:40.23 | JT | i usually only use Xeons to compile asterisk |
| 04:40.36 | JT | sheppard: sounds like you have something defective occuring |
| 04:40.39 | JT | r0b-: ... |
| 04:40.41 | sheppard | hahahaha |
| 04:40.47 | JT | r0b-: no wonder it takes forever |
| 04:40.53 | sheppard | yeah, defective non-existant cluster of 486's ;) |
| 04:40.56 | JT | r0b-: what type of virtual machine? |
| 04:40.59 | r0b- | well its done now |
| 04:40.59 | JT | sheppard: haha |
| 04:41.03 | r0b- | its Debian |
| 04:41.13 | JT | r0b-: no, what virtualisation scheme? |
| 04:41.21 | r0b- | VMWare Workstation |
| 04:41.21 | sheppard | got a nickle on vmware |
| 04:41.27 | r0b- | lol |
| 04:41.28 | sheppard | woohoo, I won a nickle |
| 04:41.31 | r0b- | :P |
| 04:41.38 | JT | ah |
| 04:41.42 | JT | slow |
| 04:41.57 | JT | don't expect it to perform well for asterisk |
| 04:42.18 | r0b- | wheres the asterisk configs? |
| 04:42.29 | JT | did you make samples? |
| 04:42.36 | JT | /etc/asterisk |
| 04:42.38 | r0b- | no |
| 04:42.38 | JT | ~thebook |
| 04:42.38 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:42.38 | r0b- | :P |
| 04:42.52 | sheppard | I love my xeons |
| 04:43.04 | sheppard | spooled a 300mb print job in about 4 seconds |
| 04:43.21 | JT | nice |
| 04:43.23 | sheppard | and it rasterizes images in photoshat very quickly |
| 04:43.56 | AJaymn | JT ok did the ./configure" , "make menuselect" (make sue to SAVE as you quit), "make", "make install" do i just do a reboot? |
| 04:44.11 | JT | you don't need to reboot |
| 04:44.17 | JT | AJaymn: was this zaptel ro asterisk |
| 04:44.19 | JT | or |
| 04:44.25 | AJaymn | zaptel |
| 04:44.39 | JT | so try to modprobe ztdummy again |
| 04:44.40 | [TK]D-Fender | AJaymn, At the end do the rest of what I told you |
| 04:44.50 | [TK]D-Fender | AJaymn, "modprobe zaptel", etc |
| 04:45.01 | AJaymn | SAME THIGN!!! FATAL: Error inserting ztdummy (/lib/modules/2.6.22.1-32.fc6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 04:45.16 | JT | what does uname -a say? |
| 04:45.27 | JT | also |
| 04:45.27 | [TK]D-Fender | AJaymn, Did you do "modprobe zaptel" FIRST? |
| 04:45.29 | AJaymn | 2.6.22.1-32.fc6 #1 SMP Wed Aug 1 15:15:33 EDT 2007 i686 i686 i386 GNU/Linux |
| 04:45.45 | JT | did you compile your kernel yourself, or is it stock? |
| 04:46.02 | AJaymn | modprobe zaptel just comes back to the prompt no errors |
| 04:46.13 | [TK]D-Fender | AJaymn, do ztdummy now |
| 04:46.16 | JT | try ztdummy now |
| 04:46.17 | AJaymn | stock. |
| 04:46.40 | AJaymn | [root@chi zaptel-1.4.4]# modprobe zaptel |
| 04:46.40 | AJaymn | [root@chi zaptel-1.4.4]# modprobe ztdummy |
| 04:46.40 | AJaymn | FATAL: Error inserting ztdummy (/lib/modules/2.6.22.1-32.fc6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 04:47.19 | r0b- | how can i make a user for asterisk |
| 04:47.55 | JT | AJaymn: what is the output of ls -la /lib/modules/2.6.22.1-32.fc6/misc/ztdummy.ko ? |
| 04:48.23 | AJaymn | -rw-r--r-- 1 root root 109155 Aug 15 23:42 /lib/modules/2.6.22.1-32.fc6/misc/ztdummy.ko |
| 04:48.31 | JT | (and what is your current system time? |
| 04:48.47 | AJaymn | Wed Aug 15 23:48:36 PDT 2007 |
| 04:49.08 | JT | ok, sounds like it did recompile, it just doesn't work |
| 04:49.15 | JT | does dmesg have anything to say about it |
| 04:49.51 | AJaymn | Zaptel Version: 1.4.4 |
| 04:49.52 | AJaymn | Zaptel Echo Canceller: MG2 |
| 04:49.52 | AJaymn | ztdummy: Unknown symbol rtc_register |
| 04:49.52 | AJaymn | ztdummy: Unknown symbol rtc_unregister |
| 04:49.52 | AJaymn | ztdummy: Unknown symbol rtc_control |
| 04:52.29 | *** join/#asterisk lione_1234567 (n=lione@ita.dsl-comm.vsi.ru) |
| 04:53.04 | thansen|laptop | if I have my choice between sip and iax...which is a better choice? |
| 04:53.52 | AJaymn | JT does it have something to do with enabling RTC in the kernel? |
| 04:53.58 | JT | possibly |
| 04:54.07 | JT | thansen|laptop: sip has the most widespread support |
| 04:54.10 | AJaymn | how the $$$$ do i do that?:P |
| 04:54.53 | thansen|laptop | JT: this is my connection with a provider..they support both sip and iax...which should I use? |
| 04:55.13 | JT | thansen|laptop: up to you, i'd say sip |
| 04:55.21 | r0b- | can someone help me create a user in asterisk |
| 04:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 04:55.36 | thansen|laptop | JT: sounds good, thanks! |
| 04:57.13 | AJaymn | JT: http://rafb.net/p/SQIYP658.html |
| 04:57.13 | lione_1234567 | Hello. I run /etc/init.d/zaptel status and it returns 176:check_zaptel_timing: not found What does this mean? |
| 04:58.09 | lione_1234567 | platfom ubuntu serv Feisty Fawn |
| 04:58.46 | JT | AJaymn: try modprobe crc-ccitt |
| 04:58.51 | JT | then try modprobe zaptel |
| 04:59.10 | AJaymn | root@chi zaptel-1.4.4]# modprobe crc-ccitt |
| 04:59.10 | AJaymn | [root@chi zaptel-1.4.4]# modprobe zaptel |
| 04:59.10 | AJaymn | [root@chi zaptel-1.4.4]# |
| 04:59.44 | JT | sweet now you can ztcfg -vv |
| 05:00.02 | AJaymn | 0 channels configured. |
| 05:00.06 | r0b- | can someone help me? |
| 05:00.08 | JT | hooray |
| 05:00.17 | AJaymn | hooray for what?! :P |
| 05:00.31 | JT | for the fact it should work now |
| 05:01.04 | AJaymn | compiling? |
| 05:01.45 | r0b- | why wont my asterisk account work! |
| 05:02.08 | lione_1234567 | done modules loaded but mesage still same |
| 05:02.13 | JT | AJaymn: it's compiled, zaptel is running, asterisk time |
| 05:02.29 | r0b- | grr! |
| 05:02.33 | lione_1234567 | I check with lsmod |
| 05:02.36 | sheppard | lol |
| 05:02.45 | r0b- | google isnt freacking working! |
| 05:02.46 | sheppard | full build of asterisk done in 1m22s |
| 05:02.47 | sevard | . |
| 05:03.00 | *** join/#asterisk pkunkra (n=chris@cpe-74-73-28-89.nyc.res.rr.com) |
| 05:04.13 | AJaymn | [root@chi zaptel-1.4.4]# modprobe zaptel |
| 05:04.13 | AJaymn | [root@chi zaptel-1.4.4]# modprobe ztdummy |
| 05:04.13 | AJaymn | FATAL: Error inserting ztdummy (/lib/modules/2.6.22.1-32.fc6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 05:04.16 | AJaymn | still getting that?! |
| 05:04.26 | r0b- | FUCK! |
| 05:05.20 | r0b- | asterisk is being a pain in my butt! |
| 05:05.22 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 05:05.28 | fujin | hi sorry |
| 05:05.31 | lione_1234567 | In fact apt-get install zaptel didn't do it correct |
| 05:05.34 | fujin | anyone used SIPp with asterisk? |
| 05:05.44 | lione_1234567 | so my one is compiled |
| 05:06.07 | lione_1234567 | what should I do to run them correct |
| 05:07.23 | JT | AJaymn: wtf did you do, did you reboot? |
| 05:07.31 | lione_1234567 | 176:check_zaptel_timing: not found |
| 05:08.42 | lione_1234567 | Do averybody install zaptel with no problem for kernel 2.6? |
| 05:08.57 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 05:09.39 | r0b- | GRRRR!!!!!!!!!!!1 |
| 05:09.47 | r0b- | im about to break this computer |
| 05:10.11 | r0b- | why wont this friggen extension work1 |
| 05:10.36 | AJaymn | JT no |
| 05:11.18 | JT | AJaymn: i thought you told me it loaded okay, what is the reason for loading it again? |
| 05:11.45 | r0b- | why wont anyone help me? |
| 05:11.53 | fujin | You're doing it wrong. |
| 05:12.37 | r0b- | this is pissing me off |
| 05:13.05 | JT | r0b-: |
| 05:13.08 | JT | ~question |
| 05:13.08 | jbot | rumour has it, question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 05:13.34 | AJaymn | hehe |
| 05:13.42 | r0b- | GRRRRRR! |
| 05:13.58 | JT | r0b-: take your whingeing elsewhere |
| 05:14.13 | r0b- | i just want to know how to add a bloody user to Asterisk |
| 05:14.29 | AJaymn | read the howto on asterisk.org |
| 05:14.35 | r0b- | i did |
| 05:14.38 | r0b- | theres nothing |
| 05:14.53 | fujin | read sip.conf.examples |
| 05:14.57 | fujin | for adding a sip user |
| 05:14.59 | r0b- | im a noob btw |
| 05:15.04 | r0b- | i want IAX |
| 05:15.04 | fujin | No shit. |
| 05:15.14 | fujin | then read iax2.conf.example |
| 05:15.18 | AJaymn | lol |
| 05:15.42 | fujin | they both contain clear examples of how to configure users, friends and peers, and whatever the equivalent in IAX2 is |
| 05:15.45 | JT | r0b-: read the book yet? |
| 05:15.47 | JT | ~thebook |
| 05:15.48 | jbot | somebody said thebook was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:16.12 | fujin | or that :P |
| 05:16.19 | fujin | so, anyone used SIPp to test asterisk? |
| 05:16.22 | fujin | if so, how the hell did you do it |
| 05:17.01 | AJaymn | JT: http://bugs.digium.com/view.php?id=5236 |
| 05:17.39 | fujin | have you built the dependencies for it? |
| 05:17.45 | fujin | and/or linking against your kernel? |
| 05:17.50 | fujin | looks like you're doing it wrong. |
| 05:18.15 | AJaymn | fujin: who are you talking to? |
| 05:18.34 | JT | AJaymn: so what on earth does that have to do with your problem? |
| 05:19.00 | JT | AJaymn: i asked you a question before, please answer it |
| 05:19.09 | JT | instead of sending lots of URLs |
| 05:19.30 | fujin | AJaymn: I'm talking to you |
| 05:19.34 | fujin | (duh?) |
| 05:19.36 | AJaymn | <JT> AJaymn: try modprobe crc-ccitt |
| 05:19.36 | AJaymn | <JT> then try modprobe zaptel |
| 05:19.36 | AJaymn | <AJaymn> root@chi zaptel-1.4.4]# modprobe crc-ccitt |
| 05:19.36 | AJaymn | <AJaymn> [root@chi zaptel-1.4.4]# modprobe zaptel |
| 05:19.36 | AJaymn | <AJaymn> [root@chi zaptel-1.4.4]# |
| 05:19.37 | AJaymn | <JT> sweet now you can ztcfg -vv |
| 05:19.38 | AJaymn | <AJaymn> 0 channels configured. |
| 05:19.41 | AJaymn | you said... |
| 05:19.42 | AJaymn | i said.. |
| 05:19.44 | JT | AJaymn: stop that |
| 05:19.47 | JT | NO |
| 05:19.53 | AJaymn | but you never said to modprobe ztdummy there. |
| 05:19.54 | JT | no more pasting massive chunks of text |
| 05:20.22 | JT | fine |
| 05:20.37 | AJaymn | when i try to modprobe ztdummy thats when i get the error |
| 05:20.49 | AJaymn | <JT> AJaymn: try modprobe crc-ccitt |
| 05:20.49 | AJaymn | <JT> then try modprobe zaptel |
| 05:20.49 | AJaymn | <AJaymn> root@chi zaptel-1.4.4]# modprobe crc-ccitt |
| 05:20.49 | AJaymn | <AJaymn> [root@chi zaptel-1.4.4]# modprobe zaptel |
| 05:20.49 | AJaymn | <AJaymn> [root@chi zaptel-1.4.4]# |
| 05:20.50 | AJaymn | <JT> sweet now you can ztcfg -vv |
| 05:20.52 | AJaymn | <AJaymn> 0 channels configured. |
| 05:20.54 | AJaymn | shit |
| 05:20.55 | AJaymn | sorry |
| 05:21.27 | r0b- | ok i added myself but still no go |
| 05:21.57 | JT | r0b-: awesome, and why are you telling us this? |
| 05:22.01 | AJaymn | r0b- you ever get a SIP exten to work yet? |
| 05:22.09 | r0b- | no i havent |
| 05:22.19 | r0b- | its making me mad |
| 05:22.31 | AJaymn | get a SIP ext to work first... |
| 05:22.33 | JT | i don't fucking care if it's making you mad |
| 05:22.34 | AJaymn | then try IAX |
| 05:22.45 | JT | do you epect us to give you special treatment if you're mad? |
| 05:22.47 | JT | expect |
| 05:26.01 | sevard | special treatment == /ignore $nick |
| 05:26.08 | r0b- | screw it |
| 05:26.15 | fujin | just piss off, seriously |
| 05:26.18 | fujin | quit while you're ahead |
| 05:26.20 | fujin | you clearly suck at everything |
| 05:26.27 | r0b- | fuck you asswhipe |
| 05:26.30 | fujin | you can't even make an extension |
| 05:26.42 | r0b- | this is new to me fuck nut |
| 05:26.50 | AJaymn | read the howtos r0b- |
| 05:26.54 | fujin | go reinstall windows or something |
| 05:26.56 | sevard | r0b-: I'm a contractor, I'd gladly help you. |
| 05:27.01 | r0b- | bite me |
| 05:27.01 | fujin | install addons to firefox |
| 05:27.05 | fujin | something more of your skill level |
| 05:27.05 | AJaymn | heh |
| 05:27.11 | r0b- | this channel is clearly assholes |
| 05:27.18 | sevard | r0b-: if you'd like to discuss easy methods, /msg me. |
| 05:27.19 | fujin | yup |
| 05:27.20 | fujin | kbai |
| 05:27.24 | AJaymn | do you know how to insert the Winblows Install CD? |
| 05:27.30 | fujin | keke |
| 05:27.41 | fujin | "oh noes, asterisk doesn't autoconfigure everythign like windows" |
| 05:27.49 | fujin | what's this? configuration optiosn? |
| 05:27.58 | antimoof | mumble files mumble honey. |
| 05:28.09 | *** join/#asterisk nighty^ (n=nighty@p3132-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 05:30.05 | JT | now now |
| 05:30.10 | JT | everyone was a newbie once |
| 05:30.26 | JT | which is fine, as long as you give the channel more to work on than "I AM GETTING REALLY ANGRY" |
| 05:30.40 | r0b- | ive tried stating my problem |
| 05:30.42 | AJaymn | you have to admit JT atleast some of us try before telling the whole world 'Im a dumb ass' |
| 05:31.27 | mog | why is everyone so testy tonight |
| 05:31.32 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:31.56 | JT | r0b-: all you kept saying was sip didn't work and you were angry |
| 05:32.05 | JT | unless i missed something |
| 05:32.07 | r0b- | IAX! u dumbass |
| 05:32.15 | antimoof | any of you happen to know if the patch from sangoma to zaptel will be folded into the source tree anytime soon? |
| 05:32.16 | JT | sip, iax, whatever |
| 05:32.18 | mog | rob- you are about to be kicked |
| 05:32.22 | mog | calm down |
| 05:32.26 | JT | there is no need to be abusive, r0b- |
| 05:32.32 | r0b- | mog i am calm |
| 05:32.32 | AJaymn | ok r0b- JT has dont nothign to you to call him a dumbass |
| 05:32.38 | mog | no you arent |
| 05:32.42 | antimoof | mm, /ignore. |
| 05:32.48 | jarod14 | hi guys |
| 05:32.53 | mog | hi jarod14 |
| 05:33.05 | jarod14 | ;o) |
| 05:33.29 | RypPn | lol |
| 05:33.30 | mog | ugh |
| 05:33.30 | sevard | muahahaha |
| 05:33.36 | flenders | ahhaha |
| 05:33.37 | mog | oh well cant save em all |
| 05:33.40 | sevard | somebody needs happy pills. |
| 05:33.42 | JT | sounded calm to me |
| 05:33.50 | JT | ;) |
| 05:34.01 | sevard | He messaged me for help, I offered him contracting services, he flipped a nut. |
| 05:34.12 | sevard | I HAVE TO PAY TO GET SOMEBODY TO FIX THIS FOR ME?! |
| 05:34.29 | mog | ugh its over, how was everyone's day? |
| 05:34.30 | flenders | not fix... install |
| 05:34.51 | sevard | mog: my day was amazing! how ws yours |
| 05:34.56 | mog | grand grand |
| 05:35.24 | sevard | JT: "Calmer than you are. Calmer than you are." |
| 05:35.51 | jarod14 | mog, my day is just starting 7:30am here in paris |
| 05:36.20 | mog | get back in bed then |
| 05:36.25 | mog | 730 am is far to early |
| 05:36.42 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 05:36.48 | jarod14 | bed is far away I'm still at work |
| 05:37.00 | jarod14 | I meant already at work |
| 05:37.12 | mog | you should install a bed at work then |
| 05:37.30 | jarod14 | huhu |
| 05:37.37 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 05:37.40 | flenders | I went to paris once to work on a client, and EVERYONE in their IT department would get to work before 8AM |
| 05:37.43 | lione_1234567 | aha make make install |
| 05:37.47 | lione_1234567 | bed |
| 05:38.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:38.25 | mog | my wife drives me to work in the morning, so i really dont wakeup till i get out of the car |
| 05:39.23 | lione_1234567 | gud wife |
| 05:39.36 | lione_1234567 | lucky |
| 05:40.26 | mog | yeah she is pretty awesome |
| 05:40.46 | jarod14 | lol mog, i can imagine what you look like in the car |
| 05:40.58 | jarod14 | it should be close to a zombi |
| 05:41.06 | mog | i just pull the lever on the chair all the way back |
| 05:41.25 | mog | is nice |
| 05:47.17 | mog | well its late here everyone play nice |
| 05:47.18 | mog | gnite |
| 05:47.40 | lione_1234567 | where are you from? |
| 05:47.59 | dlynes | lione_1234567: he's from digium |
| 05:48.18 | mog | im from texas, i live at digium these days |
| 05:49.00 | mog | and when not there huntsville alabama |
| 05:49.38 | lione_1234567 | got it |
| 05:49.59 | lione_1234567 | gud night |
| 05:50.36 | *** join/#asterisk docelm0 (n=vircuser@c-76-99-154-203.hsd1.de.comcast.net) |
| 05:55.22 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 05:55.52 | asterisknerds | <PROTECTED> |
| 06:02.05 | *** join/#asterisk af_ (n=getsmart@81-174-46-119.dynamic.ngi.it) |
| 06:02.53 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 06:04.18 | kiscokid | I have a question about what is going to happen when I add a Sangoma card to my system -- what will the channel names be called? |
| 06:04.54 | JT | what do you mean? |
| 06:05.06 | *** join/#asterisk Jabeeds (n=jabeeds@15.107.dsl.mel.iprimus.net.au) |
| 06:05.39 | kiscokid | well if I add an FXO card I get a zap/1 channel |
| 06:06.11 | kiscokid | will I get pri/1 through pri/23?" |
| 06:07.38 | JT | no |
| 06:07.41 | JT | it's still zap |
| 06:08.05 | kiscokid | so I'll get zap/1 through z/23? |
| 06:08.44 | JT | depending on what else is on your system and how you configure it, yes |
| 06:09.40 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 06:10.26 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 06:10.26 | Jabeeds | Hell All, I have a problem I am hoping someone can help me with. I have an Asterisk Server, that has a registration with another provider. I then have 3x extensions hooked up to my asterisk box. Now my Box is NOT behind nat, it is on a RealWorld IP Address. Two of the extensions are NAT'ed, one is not all NAT settings are correct. Any of the extensions can call any of the other extensions perfectly, however when a call comes from my termi |
| 06:10.44 | kiscokid | when I make the dial plan try to dial out using the pri do I have to try zap/1, zap/2, ... zap/23 in order? |
| 06:12.28 | AJaymn | Jabeeds your message is cut off.. |
| 06:12.38 | Jabeeds | Where at? |
| 06:12.42 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 06:15.21 | Jabeeds | AJaymn: Where did the message cut off? |
| 06:15.30 | AJaymn | call comes from my termi |
| 06:15.57 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 06:16.02 | Jabeeds | from my termination provider, it forwards to its destination, but the calling party cannot hear any voice. However if put on hold, can hear hold music, and can hear DTMF keypresses. What am I doing wrong? |
| 06:16.29 | AJaymn | 1way audio? |
| 06:16.30 | mosty | i have a problem with dtmf, when the caller presses the hash/pound button the sound is not heard properly at the other end, there tone is only played for a very short period, then it appears to be cut out by the pbx. what could be causing this? |
| 06:17.02 | Jabeeds | mostly: # is configured as transfer? |
| 06:17.07 | Jabeeds | Yes, one way audio |
| 06:17.09 | mosty | Jabeeds, is your asterisk server behind nat? |
| 06:17.29 | Jabeeds | well, one way audio on Xlite (Nat'ed) |
| 06:17.47 | JT | kiscokid: zap groups |
| 06:17.59 | Jabeeds | No audio on the other extensions. |
| 06:18.02 | mosty | jabeed: but is your asterisk server behind a different nat? |
| 06:18.10 | Jabeeds | No. |
| 06:18.29 | Jabeeds | Its on a realworld public IP address |
| 06:18.58 | mosty | Jabeeds, run a a packet capture util like wireshark on the asterisk server, see if you can see the RTP packets coming in with the correct source/dest addresses |
| 06:20.41 | Jabeeds | I'll give it a go now. Thanks |
| 06:21.14 | kiscokid | JT: thanks |
| 06:22.12 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 06:25.24 | Jabeeds | mostly: I forgot to mention that outgoing calls are fine... |
| 06:26.18 | mosty | Jabeeds, so on incoming calls your outgoing audio is not arriving at the other end? |
| 06:26.30 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.121) |
| 06:27.55 | Jabeeds | on the exten configured with xlite I get one way audio, but on the others i get only music and keypress tones |
| 06:28.10 | Jabeeds | if any extensions dials any other extension they are fine |
| 06:28.10 | mosty | what others? |
| 06:28.35 | Jabeeds | and if any extensions dials the outside world audio is fine both ways on that |
| 06:28.37 | JT | Jabeeds: sorry i missed it, your sip endpoints are behind nat are they? |
| 06:28.39 | Jabeeds | other extensions |
| 06:28.45 | Jabeeds | 2/3 are |
| 06:28.57 | JT | Jabeeds: in sip.conf, canreinvite=no |
| 06:29.00 | JT | then do a sip reload |
| 06:29.03 | JT | then try again |
| 06:29.06 | Jabeeds | tries it |
| 06:29.09 | Jabeeds | *tried |
| 06:29.17 | Jabeeds | had no effect |
| 06:29.26 | JT | it's set to no in general and is not overridden by a sip peer/friend? |
| 06:29.34 | Jabeeds | thats correct |
| 06:29.37 | mosty | Jabeeds, and obviously you have nat=yes in sip.conf? |
| 06:29.44 | Jabeeds | correct |
| 06:30.37 | mosty | how did the packet capture go? did you see the RTP traffic arriving at the asterisk server correctly? |
| 06:30.49 | Jabeeds | Installing wireshark now |
| 06:31.20 | mosty | tshark -i eth0 host <yourphonesipaddr> |
| 06:31.53 | Jabeeds | the nat'ed address or the public address |
| 06:31.58 | Jabeeds | public address i would imagine? |
| 06:32.20 | mosty | the public ip is what you should see if everything is working |
| 06:33.27 | Jabeeds | Am getting a whole lot of packets |
| 06:33.33 | Jabeeds | any way to specify port range? |
| 06:34.51 | mosty | i'm not sure. i normally just tell everyone else to stop making calls when i test something like this |
| 06:34.59 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:35.15 | Jabeeds | yeah but I have an ssh session to the box also from that IP |
| 06:35.36 | Jabeeds | I had like 2800 packets in 20 seconds without making a call lol |
| 06:36.06 | Jabeeds | I'll output to a file then grep it |
| 06:36.54 | mosty | just pipe tshark to grep -v SSH |
| 06:37.06 | mosty | then you won't be flooded with ssh packets |
| 06:38.41 | Jabeeds | RTP data looks good |
| 06:40.24 | mosty | how is your asterisk server terminating the call? and can you setup an echo test in your dialplan (that will prove one way or the other if rtp is coming/going correctly to/from asterisk)? |
| 06:41.15 | Jabeeds | Yeah |
| 06:41.19 | Jabeeds | Echo test is fine |
| 06:41.34 | Jabeeds | have bidirectional audio calling any other extension too |
| 06:41.58 | mosty | where do the incoming calls come from? |
| 06:41.59 | Jabeeds | and I have setup the incoming to route to echo test aswell, and when call from PSTN that works too |
| 06:42.03 | Jabeeds | PSTN |
| 06:42.09 | Jabeeds | via another server |
| 06:42.21 | mosty | pstn -> zap -> sip ? |
| 06:42.32 | Jabeeds | I am not 100% sure |
| 06:42.41 | Jabeeds | its a call termination provider |
| 06:42.55 | mosty | you said this was incoming calls |
| 06:43.03 | Jabeeds | yes |
| 06:43.11 | Jabeeds | incoming from the provider |
| 06:43.19 | Mavvie | is there a tool to "visualize" the output of "sip debug" ? |
| 06:43.41 | mosty | jabeeds: so a SIP call comes in from this provider, and you send it on to x-lite? |
| 06:43.53 | Jabeeds | Yes, or to an ATA. |
| 06:44.05 | Jabeeds | Xlite gives me one way audio, the ATA gives me nothing |
| 06:44.19 | mosty | jabeeds: ok, now to a packet trace on the asterisk box for packets to/from the sip provider |
| 06:44.22 | Jabeeds | Except keypress tones and MOH |
| 06:45.36 | Jabeeds | What do you know... No RTP packets |
| 06:46.45 | Jabeeds | Wait... So how do I have one way audio with no RTP packets? |
| 06:47.55 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 06:48.19 | Jabeeds | Ok, that was the IP of their proxy |
| 06:48.44 | Jabeeds | I did another broad trace. There is RTP data there |
| 06:49.25 | mosty | you've found where the problem is then at least |
| 06:49.34 | Jabeeds | How so? |
| 06:49.58 | mosty | you've found that the rtp packets are not being sent to the upstream sip server |
| 06:50.03 | Jabeeds | The data is there I just did the initial trace to their proxy not sip server |
| 06:50.37 | mosty | two seperate ip's? |
| 06:50.44 | Jabeeds | yes |
| 06:51.49 | mosty | perhaps it should be going to the proxy then |
| 06:52.15 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 06:52.18 | mosty | play with the proxy settings in sip.conf |
| 06:53.17 | Jabeeds | This stil leaves me wondering why the echo test works fine if set to pickup incoming calls |
| 06:55.35 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 06:56.05 | asterisknerds | <PROTECTED> |
| 06:56.46 | mosty | jabeeds: are your asterisk server and sip clients on the same network? |
| 06:56.59 | Jabeeds | mosty: No |
| 06:57.22 | mosty | ok, then does your asterisk server have a public ip address? |
| 06:57.33 | Jabeeds | yes |
| 06:57.49 | mosty | and it's not behind nat |
| 06:58.36 | mosty | and you calls between sip clients and asterisk work fine, and calls between the upstream provider and asterisk also work fine |
| 06:58.54 | Jabeeds | correct |
| 06:59.11 | mosty | and you have canreinvite=no in sip.conf? |
| 06:59.20 | Jabeeds | correct |
| 06:59.34 | Jabeeds | and in each peer's config |
| 07:00.05 | mosty | see if you can compare a packet trace for incoming -> echo test versus incoming -> local sip client |
| 07:00.51 | FlatFoot | good morning all |
| 07:01.45 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 07:04.18 | *** join/#asterisk docelmo (n=vircuser@c-76-99-154-203.hsd1.de.comcast.net) |
| 07:04.52 | *** join/#asterisk remmo (n=junk@202.1.119.80) |
| 07:07.48 | Jabeeds | mosty: What am I comparing? |
| 07:07.53 | Jabeeds | I have both dumps |
| 07:08.34 | mosty | you basically want to see how the rtp streams compare in terms of src/dest addresses |
| 07:09.05 | mosty | because incoming -> echo test works, and incoming -> local sip client doesn't |
| 07:09.44 | Jabeeds | They are pretty well identical dumps |
| 07:09.53 | Jabeeds | source is where the source should be |
| 07:10.02 | Jabeeds | destination is where the destination should be |
| 07:11.30 | Jabeeds | only visible difference is the RTP port |
| 07:11.45 | Jabeeds | 10000 - 12000 on incoming to echo |
| 07:12.21 | mosty | what does your rtp.conf has as start and end ports? |
| 07:12.51 | Jabeeds | 18000 server side -> client |
| 07:12.58 | Jabeeds | 65000 client -> server |
| 07:13.19 | Jabeeds | rtp.conf states 10000-20000 |
| 07:13.38 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:14.05 | mosty | strange. i have to go, good luck |
| 07:14.28 | Jabeeds | Anyone else want a crack? |
| 07:23.27 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:25.55 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 07:26.20 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:29.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:30.00 | *** join/#asterisk lbow (n=lbow@vc-196-207-41-251.3g.vodacom.co.za) |
| 07:34.49 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 07:34.49 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:38.06 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 07:42.44 | *** join/#asterisk ReD-MaN (i=daemon@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
| 07:48.56 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 07:51.58 | *** join/#asterisk arezik (n=arezik@tech.cas.ip-anywhere.net) |
| 07:55.49 | zeeesh | trying to configure realtime asterisk "switch => Realtime/mycontext@realtime_ext |
| 07:55.49 | zeeesh | " |
| 07:56.06 | zeeesh | could not understand this statement "switch => Realtime/mycontext@realtime_ext |
| 07:56.06 | zeeesh | " |
| 08:04.08 | *** join/#asterisk waptaxi (n=waptaxi@45.151-224-87.telenet.ru) |
| 08:04.12 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 08:04.21 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 08:04.51 | asterisknerds | <PROTECTED> |
| 08:06.35 | Uatec | Hey, has anyone here heard of GCI COM? |
| 08:09.21 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 08:16.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:18.34 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 08:18.40 | jeremy_g | morning |
| 08:18.43 | *** join/#asterisk lbow (n=lbow@196.212.27.90) |
| 08:28.41 | Sweeper | Uatec: that sounds painful |
| 08:28.45 | Sweeper | COM is bad enough as it is |
| 08:29.07 | WilliamK | morning |
| 08:30.41 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 08:32.39 | Uatec | it's a telecoms company |
| 08:38.21 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 08:39.16 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 08:39.45 | *** join/#asterisk kkn088 (n=kikoun@88-138-6-52.adslgp.cegetel.net) |
| 08:42.20 | santoshr | how can i set a variable from inside and application which will be available to extensions.conf. and no i cannot pass it to the application as an argument |
| 08:46.22 | santoshr | any ideas guys |
| 08:49.54 | *** join/#asterisk shinao1 (n=shinao1@41.219.195.25) |
| 08:51.20 | jeremy_g | santoshr:rephrase your question please |
| 08:51.55 | jeremy_g | santoshr:i have been watching too much porn and inside seem to have a wrong conotation for me at least |
| 08:54.36 | santoshr | i have a custom application. What i want is to set a variable which should be available in extensions.conf |
| 08:55.55 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 08:55.56 | jeremy_g | hmm |
| 08:56.17 | asterisknerds | <PROTECTED> |
| 08:57.02 | JT | santoshr: global variables, or astdb, or odbc |
| 08:57.12 | santoshr | global variables |
| 08:57.28 | jeremy_g | santoshr:i dont know but there must exist some way of setting the channel variable from the apps. |
| 08:57.29 | jeremy_g | hmm |
| 08:57.40 | jeremy_g | JT:global |
| 08:58.02 | JT | jeremy_g: they are all options that i listed. |
| 08:58.09 | jeremy_g | but astdb should be equally acceptable |
| 08:58.26 | jeremy_g | he just then has to retrieve the value at the start of extensions.conf |
| 09:00.35 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 09:00.57 | *** join/#asterisk hank (n=hank@leonardo.netwichtig.de) |
| 09:00.58 | axscode | whats a good sip phone from cisco? |
| 09:00.59 | hank | hi there |
| 09:01.06 | axscode | any model please.. and cost effect also. |
| 09:02.56 | hank | we are looking for a simple way for an queue member to see wether he is currently in the queue or not. a windows app (trayicon perhaps?) would be ok, some website or a way to light a snom phones led when the phone is in the queue. any recommendations? |
| 09:04.51 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 09:06.59 | *** join/#asterisk E-bola (n=bola@cpe-76-179-4-233.maine.res.rr.com) |
| 09:07.21 | E-bola | My hdd is failing, how do i backup my purchased 729 licenses? |
| 09:10.46 | creativx | good morning JT! |
| 09:22.50 | *** join/#asterisk frederickp2 (n=fp@modemcable161.125-203-24.mc.videotron.ca) |
| 09:23.45 | *** part/#asterisk frederickp2 (n=fp@modemcable161.125-203-24.mc.videotron.ca) |
| 09:23.57 | shinao1 | hi guys.. i wonder if anyone has integrated asterisk with any Panasonic PBXs.. specifically the KXTA-616 |
| 09:24.15 | shinao1 | id like to be able to do that |
| 09:24.59 | shinao1 | any tips, or anything to watch out for? |
| 09:29.25 | *** join/#asterisk engrxyz (i=1000@host81-150-247-250.in-addr.btopenworld.com) |
| 09:31.14 | jeremy_g | E-bola:damn |
| 09:34.37 | shinao1 | hi guys.. i wonder if anyone has integrated asterisk with any Panasonic PBXs.. specifically the KXTA-616 |
| 09:34.44 | shinao1 | any tips, or anything to watch out for? |
| 09:34.48 | shinao1 | id like to be able to do that |
| 09:39.33 | Merlin83b | shinao1: We saw that 10 minutes ago. It would seem nobody has anything to say on the matter, please don't repeat yourself. |
| 09:49.46 | *** join/#asterisk Ebola (n=Ebola@host86-138-7-229.range86-138.btcentralplus.com) |
| 09:52.17 | jeremy_g | shinao1:please repeat yourself. Merlin83b is on drugs, ignore him. |
| 09:52.32 | jeremy_g | we are still listening.. |
| 09:52.53 | jeremy_g | panasonic yeyyyY |
| 09:55.22 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 09:55.52 | asterisknerds | <PROTECTED> |
| 09:56.07 | shinao1 | soory guys |
| 09:56.14 | shinao1 | jeremy_g: |
| 09:56.51 | Daviey | Hmm.. can anybody help me find why outbound calls on a SIP trunk are taking upto 5 seconds for audio to start, ports are fine and inbound works fine.. Any ideas? |
| 09:59.35 | JT | hello creativx |
| 09:59.47 | JT | axscode: why cisco? |
| 10:03.37 | shinao1 | jeremy_g: im trying to integrate asterisk and a Panasonic PBX |
| 10:03.49 | shinao1 | so that i can achieve interoffice calling |
| 10:04.00 | shinao1 | sorry i had to go beat up my boss :) |
| 10:04.57 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 10:05.05 | JT | shinao1: how does it connect to the pstn, digital or analogue? |
| 10:05.18 | shinao1 | it doesnt |
| 10:05.32 | shinao1 | its strictly an intercom for now |
| 10:06.36 | shinao1 | its a Panasonic KXTA-616 |
| 10:06.48 | shinao1 | has 6COs, 24 extensions |
| 10:07.15 | jeremy_g | asterisknerds2:time to exit the channel cause: excess flood :D |
| 10:07.15 | JT | sorry |
| 10:07.30 | JT | you have a pbx you use just as an intercom? |
| 10:07.54 | shinao1 | THEY do.. they havent connected it to anything yet |
| 10:08.05 | JT | hrm ok |
| 10:08.29 | jeremy_g | shinao1:oh okay so you trying to integerate a panasonic pbx. thats some real new information you just gave. |
| 10:08.34 | jeremy_g | hmm |
| 10:08.45 | jeremy_g | shinao1:what does it look like |
| 10:08.55 | *** join/#asterisk onats (n=julian@122.53.139.111) |
| 10:08.58 | shinao1 | hey i already said that jeremy_g :D |
| 10:08.59 | *** part/#asterisk E-bola (n=bola@cpe-76-179-4-233.maine.res.rr.com) |
| 10:09.01 | JT | shinao1: so it's an analogue pbx? |
| 10:09.02 | shinao1 | the PBX? |
| 10:09.14 | shinao1 | one sec i can send you a link to see it... |
| 10:09.47 | shinao1 | http://www.smarthomeuae.com/productsdetails.php?productid=218&pcategory=16001 |
| 10:09.52 | shinao1 | thats what it looks like |
| 10:10.07 | onats | hello, can anyone guide me to a fresh install on an ubuntu box? |
| 10:10.11 | jeremy_g | shinao1:oh damn, it wont work |
| 10:10.35 | JT | ugh, evil analogue |
| 10:11.03 | shinao1 | jeremy_g: why not? |
| 10:11.21 | jeremy_g | shinao1:because it is white in color. asterisk is not designed to support white colored pbxes |
| 10:11.34 | shinao1 | :D |
| 10:11.40 | shinao1 | are you for real? |
| 10:11.56 | onats | i;d like to get all the bells and whistles of asterisk now.. |
| 10:12.24 | jeremy_g | shinao1:the reason is more than believable. one of the developers had issues with his gf who loved the white color. they broke and that guy deliberately took the decision not to support any white colored pbx |
| 10:12.45 | jeremy_g | the rest of the team had to respect his feelings |
| 10:12.57 | shinao1 | :D wow... |
| 10:13.04 | shinao1 | i GOTTA be a developer too |
| 10:13.11 | shinao1 | that is so DAMN COOL!!! |
| 10:13.12 | onats | hi guys, hello? |
| 10:13.22 | shinao1 | so uh, what can i do? |
| 10:13.31 | jeremy_g | hi onats, you look new. |
| 10:13.35 | onats | i am... |
| 10:13.43 | onats | new to this channel, and to asterisk |
| 10:14.04 | onats | hello to you guys |
| 10:14.11 | jeremy_g | shinao1:i say you should send him some gifts or order a pizza at least and say something that convinces him that the world is beyond black and white |
| 10:14.45 | shinao1 | pizza wont help |
| 10:14.46 | jeremy_g | onats:whats bothering you |
| 10:14.56 | jeremy_g | shinao1:yeah..very smart |
| 10:14.59 | shinao1 | on a guy with issues that big |
| 10:15.14 | shinao1 | so, uh... what can i do then? |
| 10:15.26 | jeremy_g | shinao1:try painting it to black or sth |
| 10:15.36 | jeremy_g | and then show the pic to that guy |
| 10:15.43 | jeremy_g | with model and brand.. |
| 10:15.45 | onats | jeremy_g, well i'd like to play with an asterisk install, but would like to get all the progs that's included in asterisknow, except the OS |
| 10:16.09 | shinao1 | i sincerely hope this isnt some elaborate prank jeremy_g |
| 10:16.12 | jeremy_g | onats:you sound like a real experienced user |
| 10:16.19 | onats | jeremy_g, nope.. newbie |
| 10:16.37 | onats | never seen/touched an asterisk install |
| 10:16.39 | jeremy_g | shinao1: elaborate prank |
| 10:16.50 | shinao1 | i was dying here |
| 10:17.04 | JT | it was so hard to reserve comment :P |
| 10:17.10 | jeremy_g | shinao1:he will then send you the driver for that model |
| 10:17.32 | shinao1 | i may have to go shoot my boss again |
| 10:17.34 | onats | also planning to purchase a zapmicro pci card.. will that work fine? |
| 10:17.35 | shinao1 | one sec.. |
| 10:17.43 | shinao1 | yeah hes dead now for sure |
| 10:17.51 | JT | onats: what on earth is that? |
| 10:18.54 | shinao1 | so seriously,now, jeremy_g how do i integrate these things? |
| 10:19.29 | jeremy_g | shinao1:what do you mean by integeration... explain what kinda a setup you would like to have.. |
| 10:19.36 | JT | shinao1: either through the CO ports or handset ports |
| 10:19.45 | JT | using analogue interfaces |
| 10:19.55 | jeremy_g | shinao1:you might like to shoot yourself before you say anything to your boss |
| 10:20.12 | onats | wait, are you guys working for digium/asterisk? |
| 10:20.17 | shinao1 | i want to have the PBX extensions being able to use the asterisk as their PBX |
| 10:20.29 | onats | http://www.zapmicro.com/ |
| 10:21.13 | onats | anyone? |
| 10:21.37 | jeremy_g | shinao1:asterisk is not developed by mere beer drinkers at whose hand it could suffer. its a mature industry grade pbx that has stood the test of time and beyond such limitations. |
| 10:21.44 | JT | onats: never ever heard of them |
| 10:21.51 | shinao1 | what i need is IVR, and interoffice calling.. i hope to have a * at the other office |
| 10:21.55 | jeremy_g | shinao1:go shag yourself for believing what i said. |
| 10:21.55 | onats | JT, what cards do you use? digium? |
| 10:22.00 | shinao1 | those are my basic needs |
| 10:22.03 | JT | sometimes |
| 10:22.06 | JT | or sangoma |
| 10:22.11 | JT | or gateways/ATAs |
| 10:22.16 | onats | which is more expensive? |
| 10:22.20 | onats | sangoma or digium? |
| 10:22.56 | shinao1 | you see, the most i ever did with asterisk is setting up SIP users on a single box |
| 10:23.02 | JT | they're about the same |
| 10:23.17 | shinao1 | so now im elected to setup IVR and interoffice calling between two locations |
| 10:23.42 | shinao1 | only thing is, they already use the Panasonic at one location |
| 10:23.45 | jeremy_g | shinao1:thats some rigged election. how cud they select a wise guy like u, you should be selling turkeys |
| 10:24.05 | shinao1 | i agree totally |
| 10:24.09 | jeremy_g | :D |
| 10:24.10 | shinao1 | :D |
| 10:24.20 | jeremy_g | are you idiot or pretending hard to be like one |
| 10:24.31 | onats | JT, can i private message you? |
| 10:24.34 | jeremy_g | shinao1:whats stopping you from doing what you want? asterisk does it all |
| 10:24.42 | JT | onats: maybe |
| 10:24.50 | shinao1 | i know it does jeremy_g |
| 10:24.57 | onats | JT, would like to ask some info re. initial installation on ubuntu |
| 10:25.04 | onats | if you're ok w/ that |
| 10:25.05 | shinao1 | but the guys arent ready to give up their existing PBX |
| 10:25.07 | jeremy_g | shinao1:yeah i know you know :D |
| 10:25.21 | JT | onats: you can't ask that on channel? |
| 10:25.28 | jeremy_g | shinao1:configure your box to make it work with it |
| 10:25.35 | jeremy_g | and demo the proof of concept |
| 10:25.39 | shinao1 | thats the onion |
| 10:25.51 | shinao1 | i need to make it work with it but i dont know how |
| 10:25.59 | jeremy_g | shinao1:btw for last couple of minutes you havent repeated your banner message 'has anywork integerated...' do that |
| 10:26.01 | jeremy_g | :D |
| 10:26.12 | shinao1 | :D |
| 10:26.15 | jeremy_g | shinao1:paint the box as i suggested. |
| 10:26.22 | Mavvie | Two asterisks together works... Three asterisks together doesn't work. Anybody interested in hearing this story? |
| 10:26.23 | jeremy_g | thats the only way for you. |
| 10:26.32 | Mavvie | (or shall I shove it towards -dev?) |
| 10:26.54 | jeremy_g | Mavvie: are these boxes using sip registeration? |
| 10:27.00 | shinao1 | come on now jeremy_g |
| 10:27.08 | Mavvie | jeremy_g: It's SIP related yes. |
| 10:27.24 | Mavvie | jeremy_g: phone -> asterisk1 -> asterisk2 -> CCM works. |
| 10:27.32 | shinao1 | who is this guy??? :D |
| 10:27.41 | Mavvie | jeremy_g: phone -> asterisk1 -> asterisk2 -> asterisk3 -> CCM totally crashes the SIP session. |
| 10:27.48 | shinao1 | well im prepared to be come on or even more till i get what i need |
| 10:28.08 | jeremy_g | Mavvie: why? |
| 10:28.53 | Mavvie | jeremy_g: Honest answer: NFI. I have a simple proof of concept of it |
| 10:28.56 | jeremy_g | shinao1:keep wishing.. dont work..waste our time |
| 10:29.25 | *** join/#asterisk Fl1p (n=david@port-83-236-208-174.static.qsc.de) |
| 10:29.28 | Mavvie | jeremy_g: It's a long story, let me write up a webpage with it. |
| 10:29.56 | Mavvie | jeremy_g: it has to do with SIP and the canreinvite option. |
| 10:30.14 | jeremy_g | Mavvie:yeah write that web page and ping. |
| 10:30.16 | shinao1 | hoo-kay.. time to bite the bullet |
| 10:30.32 | jeremy_g | Mavvie: :) i am also blaming that. |
| 10:31.17 | jeremy_g | Mavvie: got the sip log? |
| 10:31.34 | Mavvie | jeremy_g: Got everything, now merging it together. |
| 10:31.51 | Mavvie | jeremy_g: even wrote a little ngrep parser to display something funky like http://www.mavetju.org/~edwin/two-hops.txt |
| 10:32.43 | jeremy_g | whats the problem |
| 10:32.54 | jeremy_g | set the reinvite= to same value on all the * boxes |
| 10:33.16 | jeremy_g | does CCM send a reinvite bypassing the intermediate * boxes |
| 10:33.18 | Mavvie | jeremy_g: it's all "canreinvite=yes". It doesn't trash with =no |
| 10:33.36 | Mavvie | jeremy_g: Give me 15 minutes, and I'll contact you again with the full story. |
| 10:33.41 | *** part/#asterisk hank (n=hank@leonardo.netwichtig.de) |
| 10:33.54 | jeremy_g | ok |
| 10:35.23 | JT | Mavvie: why do you have five billion asterisk servers between the phone and ccm? |
| 10:35.57 | Mavvie | JT: because we get calls from the internet which exactly experience this SIP-session-being-trashed problem |
| 10:36.47 | JT | why don't you use a sip proxy? |
| 10:36.51 | Mavvie | JT: the calls come from a softswitch, which forwards it to a softswitch, which forwards it to our asterisk insatnce, which forwards it to the CCM. |
| 10:38.10 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 10:48.33 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.77) |
| 10:50.09 | Daejeo1 | anyone tried motorola Ojo IP phone with asterisk? http://www.bestwebbuys.com/electronics/Motorola_Ojo_Personal_Video_Phone_PVP1000_-_IP_video_phone-4740802.html?isrc=e-search |
| 10:52.06 | *** join/#asterisk m4sh` (n=mAsH@87.19.178.148) |
| 10:55.24 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 10:55.40 | jeremy_g | Daejeo1:i did see a demo of it at von |
| 10:55.53 | jeremy_g | Daejeo1:a sexy girl they had |
| 10:55.54 | asterisknerds | <PROTECTED> |
| 10:55.56 | jeremy_g | uhhhh |
| 10:56.16 | Daejeo1 | jeremy_g: i did not |
| 10:56.22 | *** join/#asterisk kkn088 (n=kikoun@88-138-6-52.adslgp.cegetel.net) |
| 10:58.21 | Daejeo1 | jeremy_g: URL? |
| 10:59.10 | *** join/#asterisk lbow (n=lbow@dsl-146-5-187.telkomadsl.co.za) |
| 11:00.38 | jeremy_g | Daejeo1:von is a conference. i was physically present there. |
| 11:00.47 | Daejeo1 | oh |
| 11:00.56 | JT | jeremy_g: it's a pesky log bot isn't it? |
| 11:02.05 | jeremy_g | JT:it beers all characteristics of such an object as you envisage |
| 11:02.19 | JT | jeremy_g: eh? |
| 11:02.32 | jeremy_g | jeremy_g:yeah it is |
| 11:02.39 | jeremy_g | :) |
| 11:03.34 | *** join/#asterisk m4sh` (n=mAsH@87.19.178.148) |
| 11:05.41 | pkunkra | hey, i'm trying to find the build instructions for how they make the asterisk now iso. a makefile, srpm, anything? |
| 11:05.50 | pkunkra | can someone point me the right way? |
| 11:07.05 | jeremy_g | pkunkra:you do debian |
| 11:07.53 | *** join/#asterisk nick125 (n=nick@unaffiliated/nick125) |
| 11:07.56 | nick125 | Hello |
| 11:09.01 | JT | jeremy_g: you drunk? |
| 11:09.05 | pkunkra | jeremy_g, no. i'm actually trying to do it on fedora |
| 11:09.13 | pkunkra | or centos, either is fine. |
| 11:10.03 | jeremy_g | pkunkra:but centos is for girls |
| 11:10.49 | pkunkra | jeremy_g: eh? how so? |
| 11:11.04 | pkunkra | i find it quite suitable for men too. |
| 11:11.16 | pkunkra | i like it quite a bit |
| 11:11.56 | jeremy_g | pkunkra:that was just on a light note |
| 11:12.25 | jeremy_g | pkunkra:does your live cd come with gcc already? |
| 11:12.25 | pkunkra | hah. you seemed very serious about that. ;-) |
| 11:13.25 | pkunkra | jeremy_g, i'm not running off a live cd but i do have gcc installed |
| 11:15.46 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 11:16.18 | jeremy_g | pkunkra:i asked does your live cd come with gcc? |
| 11:16.48 | jeremy_g | JT:no |
| 11:17.57 | pkunkra | jeremy_g: i don't know. i'd have to boot something in qemu to find out. |
| 11:18.38 | jeremy_g | pkunkra: :) ok |
| 11:20.19 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 11:20.36 | flujan | hi all. :D |
| 11:25.18 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:26.44 | jeremy_g | pkunkra:does qemu work from console (non X) |
| 11:28.06 | pkunkra | pkunkra, i haven't tried it. but i believe you can ask it to stay in text mode. |
| 11:28.22 | pkunkra | you're screwed if it switches to graphical though. |
| 11:28.22 | pkunkra | hmmm |
| 11:28.37 | pkunkra | you can definately do a serial console only through text mode |
| 11:31.49 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 11:34.13 | pkunkra | oh, nice. DAG has asterisk in it. |
| 11:37.52 | anthony[ | I have asterisk setup |
| 11:38.01 | anthony[ | But I want to use a different providor |
| 11:38.07 | anthony[ | What file would I have to edit? |
| 11:39.15 | mvanbaak | anthony[: depends wether it uses sip or iax |
| 11:39.57 | anthony[ | IAX |
| 11:40.04 | anthony[ | (I believe) |
| 11:40.08 | anthony[ | how can I find out? |
| 11:40.17 | anthony[ | I didn't set this up, so I don't know much about it |
| 11:41.21 | *** join/#asterisk Rahail (n=rahail@c-68-43-176-199.hsd1.mi.comcast.net) |
| 11:42.14 | Rahail | any one exprince with a2billing |
| 11:42.36 | pkunkra | a2billing? |
| 11:42.55 | Rahail | its prepaid billing platform for asterisk |
| 11:43.31 | *** join/#asterisk blackhole (n=pkakkar@unaffiliated/blackhole) |
| 11:44.08 | mvanbaak | anthony[: if you dialout, does it do Dial(IAX2/ or Dial(SIP/ |
| 11:44.28 | anthony[ | hang on |
| 11:45.43 | anthony[ | right now this account is disconnected |
| 11:45.46 | anthony[ | so I can't see this |
| 11:45.49 | anthony[ | any other way to tell? |
| 11:47.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:49.00 | anthony[ | ? |
| 11:54.39 | *** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 11:55.34 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 11:56.04 | asterisknerds | <PROTECTED> |
| 11:56.26 | KpoH | wctdm24xxp: Unknown symbol pci_module_init, what I miss ? |
| 11:56.29 | anthony[ | can someone help me please? |
| 11:57.49 | *** join/#asterisk rexile (i=elixer@65.207.74.18) |
| 12:02.18 | mvanbaak | brb |
| 12:05.08 | *** join/#asterisk chris_1 (n=chris@ng1.kurtkrenn.com) |
| 12:06.09 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 12:07.41 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 12:07.41 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:08.50 | Daviey | Arg! Is anybody else having problems of canreinvite=no not working? |
| 12:09.00 | Daviey | in * 1.4? |
| 12:09.16 | Mavvie | Daviey: yes |
| 12:09.27 | Mavvie | well, maybe. |
| 12:09.57 | Mavvie | jeremy_g: http://www.mavetju.org/~edwin/asterisk-sip-reinvite.html. Now uploading the SIP traces. |
| 12:10.05 | Daviey | ie, Packet2Packet bridging happening when it shouldn't? |
| 12:10.43 | Daviey | Mavvie: i'm finding it in SVN-branch-1.4-r78146 |
| 12:13.08 | Daviey | I'm allowing others to call $(exten)@myip, all extensions are set to canreinvite=no - but bridging is still attempted? |
| 12:13.35 | Mavvie | Daviey: if you say "canreinvite=no", all the RTP traffic will go via that asterisk device. |
| 12:13.45 | Daviey | and that's what i want |
| 12:13.51 | Daviey | as the * is behind a NAT |
| 12:16.24 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 12:17.33 | Daviey | internet -> Firewall (NAT) -> Asterisk -> extensions ; means canreinvite should not be happening to ext's, surely |
| 12:23.40 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:28.34 | *** join/#asterisk Yoe (n=wouter@samba.grep.be) |
| 12:29.26 | *** join/#asterisk saftsack (n=saftsack@217.224.121.195) |
| 12:30.15 | Yoe | Hi -- I would like Asterisk to make multiple phones ring when a certain extension is dialed, but when someone picks up the phone on one of the endpoints, have the call be transferred to that phone only |
| 12:30.22 | Yoe | (i.e., not a conference call or some such) |
| 12:30.37 | creativx | dial() |
| 12:30.38 | Mavvie | See Dial() |
| 12:30.56 | Mavvie | Dial(SIP/something&Zap/something&IAX/something) |
| 12:31.06 | Yoe | ah, thanks |
| 12:32.04 | KpoH | I'm using 2.6.22 and than i try modprobe wctdm24xxp I get Unknown symbol pci_module_init |
| 12:32.29 | KpoH | is it something with kernel, or I just miss some setting in kernel? |
| 12:34.48 | Mavvie | [~] root@tardis>nm /home/edwin/asterisk/zaptel/1.2/wctdm24xxp.ko | grep pci |
| 12:34.52 | Mavvie | doesn't show pci_module_init |
| 12:35.02 | Mavvie | that's with a 2.6.17 kernel. |
| 12:35.23 | KpoH | i'm using zaptel 1.4.4 |
| 12:35.42 | [TK]D-Fender | KpoH: I have head MANY reports of 2.6.22 not working with Zaptel & Wanpipe. Roll-back to 2.6.20 seems to cure it |
| 12:35.58 | *** join/#asterisk mellentm (n=thansen@85.235.250.110) |
| 12:37.14 | KpoH | that's odd :( |
| 12:38.47 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 12:43.23 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 12:43.23 | *** mode/#asterisk [+o mog] by ChanServ |
| 12:44.15 | anthony[ | If I am changing providors, what file would I have to edit? |
| 12:45.16 | blackhole | Hey people join #kannel and #a2blling |
| 12:45.43 | JT | blackhole: why? |
| 12:46.11 | jkiff | anthony[: That depends entirely on what kind of "providor" you had/will have. |
| 12:46.23 | anthony[ | teliax |
| 12:46.30 | jkiff | Then iax.conf. |
| 12:46.31 | blackhole | JT, Because there is no one there and most of guys here would have knowlege of kannel and a2billing, so lets make those rooms usuable too! |
| 12:46.34 | anthony[ | im going to use teliax, just under a different account |
| 12:47.22 | JT | blackhole: weird, i'd say almost NO-ONE here has kannel and a2billing knowledge |
| 12:47.46 | blackhole | JT, So you mean almost no one in freenode have. |
| 12:48.00 | JT | in #asterisk anyway |
| 12:50.21 | *** part/#asterisk blackhole (n=pkakkar@unaffiliated/blackhole) |
| 12:50.24 | *** join/#asterisk blackhole (n=pkakkar@unaffiliated/blackhole) |
| 12:50.28 | *** part/#asterisk blackhole (n=pkakkar@unaffiliated/blackhole) |
| 12:50.28 | *** join/#asterisk blackhole (n=pkakkar@unaffiliated/blackhole) |
| 12:50.54 | *** part/#asterisk blackhole (n=pkakkar@unaffiliated/blackhole) |
| 12:54.30 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 12:56.00 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 12:56.03 | asterisknerds | <PROTECTED> |
| 12:57.17 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 12:59.27 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:59.47 | Uatec | hi |
| 13:00.24 | Uatec | i've added an entry in to my ISA server for sip, forwarding TCP on 5060 to my asterisk box, but i still can't register my SIP device to my asterisk box |
| 13:00.28 | Uatec | the registration times out |
| 13:00.34 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:00.46 | Uatec | does anybody know how it's supposed to be configured? |
| 13:00.46 | JT | well that's a surprise (not really) |
| 13:00.51 | Uatec | if there are any weird ins and outs...? |
| 13:01.03 | JT | yes, you need to use the right protocol |
| 13:01.16 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 13:01.17 | KpoH | [TK]D-Fender: how about 2.6.21? module not works too? |
| 13:03.47 | Uatec | well JT, i added the SIP protocol in isa |
| 13:03.57 | JT | tcp |
| 13:04.00 | JT | what would that acheive |
| 13:04.08 | shinao1 | hey [TK]D-Fender : its me from last night.. you were asking how many free ports on the Panasonic PBX i was talking about |
| 13:04.40 | Uatec | ISAs SIP rule is TCP 5060 Inbound and UDP 5060 Rceive send |
| 13:04.58 | JT | tcp will do nothing |
| 13:05.05 | JT | also, you need to allow RTP too |
| 13:05.40 | Uatec | oh |
| 13:05.45 | JT | and ISA does NAT as well as a pile of turds emulates a masterpiece |
| 13:06.12 | tzanger | JT: dpeends on the critic |
| 13:06.21 | JT | true ;) |
| 13:07.27 | De_Mon | hurm, I Know I can look this up, but what is the default user/pass for polycom 320/330s? |
| 13:07.36 | JT | Polycom 456 |
| 13:07.59 | De_Mon | yay |
| 13:08.24 | *** join/#asterisk guillote_GNU (n=bancaria@host222.201-253-44.telecom.net.ar) |
| 13:09.14 | *** join/#asterisk skyphyr (n=alanj@host81-157-11-138.range81-157.btcentralplus.com) |
| 13:09.31 | [TK]D-Fender | Uatec: SIP = 5060 UDP <------- |
| 13:09.41 | [TK]D-Fender | shinao1: Got an ANSWER now? :) |
| 13:10.06 | shinao1 | yes i do [TK]D-Fender |
| 13:10.08 | shinao1 | :) |
| 13:10.21 | shinao1 | they have all 6COs free on the unit |
| 13:10.25 | KpoH | [TK]D-Fender: I found this https://bugs.launchpad.net/ubuntu/+source/linux-restricted-modules-2.6.22/+bug/121978 |
| 13:10.29 | shinao1 | they arent using it for nothing! |
| 13:10.53 | KpoH | regard pci_module_init |
| 13:10.58 | *** join/#asterisk mjmarrio (n=mike@219-90-234-106.ip.adam.com.au) |
| 13:11.08 | mjmarrio | hello all |
| 13:11.12 | rexile | Does anyone here have a Sangoma A200 or A400? |
| 13:11.18 | [TK]D-Fender | shinao1: CO = port for your to plug a line from the TELCO on? |
| 13:11.26 | [TK]D-Fender | rexile: Yes, what about it? |
| 13:11.37 | rexile | rexile? damn it. hold on. |
| 13:11.41 | mjmarrio | I was wondering if I can map an incoming line to an intenal extension? |
| 13:11.46 | shinao1 | yes [TK]D-Fender |
| 13:11.50 | shinao1 | thats right |
| 13:12.00 | shinao1 | they arent using it at all |
| 13:12.03 | Uatec | [TK]D-Fender, oh, ok |
| 13:12.11 | [TK]D-Fender | shinao1: Ok, thats 6 x CO one ONE side. And on the OTHER? |
| 13:12.19 | elixer | ok, that's better. |
| 13:12.42 | elixer | [TK]D-Fender: i am trying to get the A400 working on a Dell PowerEdge 2950 with CentOS 5.0 |
| 13:13.12 | elixer | [TK]D-Fender: linux 2.6.18 |
| 13:13.15 | mjmarrio | Do I have to map an incoming line to an extension via the ${CHANNEL} var? |
| 13:13.23 | creativx | mjmarrio: no |
| 13:13.35 | elixer | [TK]D-Fender: and the device shows up, its able to configure itself, etc. |
| 13:13.36 | [TK]D-Fender | mjmarrio: That var is deprecated and NO. |
| 13:13.46 | elixer | [TK]D-Fender: everytime i shut the machine down, i get a kernel panic |
| 13:13.53 | mjmarrio | I have three incoming lines and want to make sure calls on one line go through to one particualr extension |
| 13:13.54 | [TK]D-Fender | mjmarrio: You need to learn how to use the dialplan & contexts |
| 13:14.00 | creativx | first you tell asterisk in what context to to place your inbound calls mjmarrio |
| 13:14.10 | mjmarrio | yes |
| 13:14.14 | elixer | [TK]D-Fender: and yes, i know i shouldn't shut the machine down :) |
| 13:14.26 | [TK]D-Fender | mjmarrio: Set that line into a context of its own and do what you want with it there |
| 13:14.30 | shinao1 | [TK]D-Fender: 24 extensions .. all taken up by users phones |
| 13:14.37 | [TK]D-Fender | elixer: Ok, I can't advise for your problem. |
| 13:14.39 | shinao1 | on the other side |
| 13:14.41 | mjmarrio | is that the only way? |
| 13:14.42 | elixer | [TK]D-Fender: ah, ok. |
| 13:14.46 | elixer | [TK]D-Fender: thanks anyway |
| 13:15.01 | [TK]D-Fender | elixer: If it installed fine and runs fine and only causes a kernel panic on shutdown there is nothing I can advise personally. |
| 13:15.11 | mjmarrio | ok thanks |
| 13:15.39 | elixer | [TK]D-Fender: well we haven't actually tested that the card works yet, the guy in requisitions bought the wrong modules (FXO instead of FXS) so we can't really test |
| 13:15.45 | [TK]D-Fender | shinao1: You tell me ONLY CO ports on (A), and that all PHONE ports on (B) are taken. You have not told me about anything AVAILABLE on (B) yet. |
| 13:16.06 | [TK]D-Fender | elixer: Guess you don't actually have a problem yet then ;) |
| 13:16.07 | elixer | [TK]D-Fender: i'm going to try and get a decent crash dump and send it to sangoma and see if they can come up with something |
| 13:16.16 | elixer | [TK]D-Fender: well, maybe not "technically" ;-) |
| 13:17.18 | elixer | [TK]D-Fender: getting the new modules today though, so maybe that will just magically fix the panics |
| 13:17.48 | shinao1 | [TK]D-Fender: if by available you mean "free", unplugged to any devices, then theres nothing available there |
| 13:17.52 | elixer | this is in stark contrast to setting up the digium TE220 |
| 13:17.59 | [TK]D-Fender | shinao1: no CO ports? |
| 13:18.07 | elixer | i plugged it in, brought up the server, installed zaptel 1.4.4 and it just works |
| 13:18.09 | elixer | simple as that. |
| 13:18.14 | [TK]D-Fender | shinao1: If so I gues you're SCREWED. |
| 13:19.39 | shinao1 | sc.. screwed..? |
| 13:19.47 | shinao1 | ..really? |
| 13:19.49 | elixer | yes. f**cked |
| 13:19.50 | elixer | heh |
| 13:19.55 | elixer | wow |
| 13:19.57 | elixer | i'm an idiot |
| 13:19.59 | elixer | f*cked |
| 13:20.00 | elixer | heh |
| 13:20.17 | [TK]D-Fender | shinao1: You go and tell me that on one side you have no ports to plug anything into! What are you expecting? |
| 13:20.32 | shinao1 | :D a miracle |
| 13:20.54 | [TK]D-Fender | shinao1: The impossible I do immediately, miracles require 24hrs notice. |
| 13:21.12 | elixer | heh |
| 13:21.22 | KpoH | [TK]D-Fender: I've replace in wctdm24xxp.c:3522 s/pci_module_init/pci_register_driver/ and module start to work |
| 13:23.26 | shinao1 | [TK]D-Fender: lets say i can free up one of those extension ports... does that help? |
| 13:23.32 | jeremy_g | [TK]D-Fender:hehe |
| 13:26.18 | [TK]D-Fender | shinao1: Do I actually have to answer that? ..... |
| 13:26.23 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 13:26.31 | Uatec | AHAH |
| 13:26.32 | [TK]D-Fender | shinao1: CO would be cheaper |
| 13:26.33 | Uatec | i think it's working |
| 13:26.35 | Uatec | on thing is |
| 13:26.49 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.169.196) |
| 13:26.55 | hi365 | can i set (or boost) the volume of MOH? |
| 13:27.47 | nick125 | Hmm, I'm having a bit of a problem with getting my asterisk server to register with my provider, Vitelity. I keep getting these errors in the console: [Aug 16 07:27:15] WARNING[15819]: chan_sip.c:7379 transmit_register: Probably a DNS error for registration to xxxxxx@x.x.x.x, trying REGISTER again (after 20 seconds) |
| 13:27.57 | nick125 | I've tried it with IPs and hostnames |
| 13:28.00 | Uatec | Aug 16 14:27:51 ERROR[5490]: chan_sip.c:6652 register_verify: Peer 'asterisk1' is trying to register, but not configured as host=dynamic |
| 13:28.03 | Uatec | Well no DUH! |
| 13:28.10 | Uatec | of course it's not configured as host=dynamic |
| 13:28.17 | Uatec | i have host=<THE IP> |
| 13:28.35 | JT | Uatec: then what are you trying to register to it? no duh! |
| 13:28.45 | JT | s/what/why/ |
| 13:28.52 | *** join/#asterisk MindTheGap (n=iote@c9505ffe.bhz.virtua.com.br) |
| 13:28.55 | shinao1 | ok [TK]D-Fender thanks anyway |
| 13:29.14 | shinao1 | thanks for listening at least :) |
| 13:29.21 | [TK]D-Fender | shinao1: But yes, if you free up at least ONE kind of port you will be able to do something. |
| 13:29.42 | [TK]D-Fender | shinao1: No CO's free on (B)? |
| 13:30.12 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 13:30.17 | shinao1 | the unit has 6 CO ports grouped together, and 24 extension ports grouped seperately |
| 13:30.18 | [TK]D-Fender | shinao1: And is it harder/more expensive to do so? |
| 13:30.37 | shinao1 | it will be more expensive [TK]D-Fender |
| 13:31.03 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:31.03 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:31.04 | shinao1 | at this point really, id just like to know what to do for next time |
| 13:31.18 | shinao1 | if i have another opportunity |
| 13:31.21 | shinao1 | :) |
| 13:31.22 | [TK]D-Fender | shinao1: So 6 CO ports in the system. ALL in use? |
| 13:31.33 | shinao1 | None in use |
| 13:31.42 | JT | he's always said none in use |
| 13:31.42 | [TK]D-Fender | shinao1: !??! |
| 13:31.46 | shinao1 | none at all in use |
| 13:31.49 | JT | it's a fancy intercom |
| 13:31.55 | shinao1 | verily |
| 13:32.01 | [TK]D-Fender | shinao1: You have 24 PHONES and no lines to the telco in use? |
| 13:32.31 | shinao1 | yes [TK]D-Fender |
| 13:32.54 | [TK]D-Fender | shinao1: I asked if you have any CO ports free and you answer "the unit has 6 CO ports grouped together, and 24 extension ports grouped seperately" which doesn't answer me at all. |
| 13:33.35 | [TK]D-Fender | shinao1: I'd swear you were being purposefully evasive to my very blunt questions. |
| 13:33.47 | shinao1 | sorry i was describing the PBX |
| 13:33.56 | shinao1 | there are no CO ports in use at all |
| 13:34.07 | shinao1 | im sorry if it looks like that, i have no reason to evade |
| 13:34.18 | JT | shinao1: are there plans to use those ports? |
| 13:34.38 | shinao1 | not right now, |
| 13:34.53 | shinao1 | they have a seperate telephone that they use for business |
| 13:34.58 | JT | i see |
| 13:34.59 | [TK]D-Fender | shinao1: so lets try this from the BEGINNING and LABEL them ok? How many CO ports AVAILABLE on (A) and then on (B)? |
| 13:36.02 | shinao1 | [TK]D-Fender: http://www.smarthomeuae.com/productsdetails.php?productid=218&pcategory=16001 |
| 13:36.07 | shinao1 | thats the PBX |
| 13:36.16 | [TK]D-Fender | shinao1: NOT THE ANSWER I ASKED FOR |
| 13:36.29 | shinao1 | i havent seen it physically so im working by what i hear on the phone |
| 13:36.49 | [TK]D-Fender | shinao1: You are continuosly running this in CIRCLES. |
| 13:36.53 | shinao1 | we have a total of 6 CO ports avaiable on A |
| 13:36.59 | shinao1 | nothing available on B |
| 13:37.05 | shinao1 | sorry about that |
| 13:37.28 | [TK]D-Fender | B has NO ports for CO on it? Or it DOES, and they are all TAKEN? |
| 13:37.47 | [TK]D-Fender | shinao1: CLARIFY <------- |
| 13:38.35 | shinao1 | :( ok |
| 13:38.44 | shinao1 | no ports for CO in it! |
| 13:38.49 | shinao1 | no ports |
| 13:39.42 | [TK]D-Fender | freaky. At which point you said that it DOES have PHONE ports, but none AVAILABLE, correct? |
| 13:40.32 | shinao1 | yes |
| 13:40.38 | shinao1 | thats right |
| 13:42.03 | [TK]D-Fender | shinao1: Ok, can those ports support boring plain phones, or only digital sets? |
| 13:43.09 | shinao1 | digital sets only |
| 13:43.22 | skyphyr | hi all - I'm using names for voicemail boxes and so anywhich start with a flag have the flags stripped - is there some way to tell asterisk there are no flags? |
| 13:43.59 | skyphyr | or should I just put flags in that way it'd always work? |
| 13:44.16 | [TK]D-Fender | shinao1: so you CAN'T plug a normal home phone on there if you wanted to in place of a digital set? |
| 13:45.06 | skyphyr | ahh wait - will try using the pipe so it expects 1.4 format |
| 13:46.34 | shinao1 | hey [TK]D-Fender i cant confirm that right now but i think one should be able |
| 13:46.35 | shinao1 | to |
| 13:46.39 | *** part/#asterisk mellen (n=thansen@85.235.250.110) |
| 13:47.08 | shinao1 | it DOES support regular phones as well as digital |
| 13:47.41 | De_Mon | me hands [TK]D-Fender a machette and walks away |
| 13:49.27 | shinao1 | this is definitely not a good day for me |
| 13:49.33 | flujan | guys, I have x-lite running with ulaw/alaw on my network |
| 13:49.37 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:49.38 | shinao1 | i shall walk away slowly now |
| 13:49.54 | flujan | No I am trying a eyebeam with the codecs g729 ulaw and alaw enabled. |
| 13:50.01 | *** part/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com) |
| 13:50.07 | [TK]D-Fender | shinao1: Final summary : (A) has 6 CO ports on there system and NONE of them are in use. (B) ONLY has 24 PHONE ports ALL of which are in use and you think you might be able to free some up. It also has NO CO ports, avaiable or otherwise. Correct? |
| 13:50.16 | flujan | when I try to place a call between eyebeam an x-lite asterisk connects the calls but mute both channels... |
| 13:50.46 | shinao1 | wait a second.. i think i get where the confusion is! |
| 13:50.59 | shinao1 | im not describing for two sites, just one! |
| 13:51.09 | Daejeo1 | [TK]D-Fender: ping |
| 13:51.11 | shinao1 | im describing only one PBX! |
| 13:51.15 | flujan | Asterisk should not recognize that the channels have a codec in common and make it work with the common codec? |
| 13:51.23 | [TK]D-Fender | shinao1: You you are feeding all of your information through http://www.willitblend.com/ |
| 13:52.06 | [TK]D-Fender | shinao1: You've been asking how to bridge 2 PBX's together and you've only been telling me about ONE?!?!WTF!?!?!? |
| 13:52.23 | shinao1 | no! |
| 13:52.36 | [TK]D-Fender | shinao1>im describing only one PBX! <------------ |
| 13:52.40 | shinao1 | im not trying to bridge two pbxs together! |
| 13:52.42 | [TK]D-Fender | shinao1>no! <----------------- |
| 13:52.52 | tzanger | wuh oh |
| 13:52.56 | [TK]D-Fender | shinao1: You said place free calls to other branch <------ |
| 13:52.57 | tzanger | someone pissed off tk |
| 13:52.58 | shinao1 | hold up please dont be mad |
| 13:53.00 | shinao1 | yes |
| 13:53.08 | [TK]D-Fender | shinao1: Yuo are so remarkably unclear its inhuman. |
| 13:53.11 | shinao1 | but the other branch has nothing there at all |
| 13:53.14 | shinao1 | im sorry |
| 13:53.18 | shinao1 | im sorry im sorryu |
| 13:53.34 | shinao1 | site one has a PBX site B has nothing at a |
| 13:53.35 | shinao1 | all |
| 13:53.41 | [TK]D-Fender | shinao1: If order to make 2 sides talk to each other you need to know exactly whats on EACH FRIGGEN SIDE |
| 13:53.46 | De_Mon | flujan No? |
| 13:53.55 | shinao1 | there is absolutely nothing on B |
| 13:54.01 | De_Mon | flujan asterisk should not, not recognize they have a codec in common... |
| 13:54.10 | shinao1 | damn i see now |
| 13:54.38 | [TK]D-Fender | shinao1: So new recap : (A) has a 24/6 system using 24/0 (uses NONE of its CO ports and is used as a glorified intercom). |
| 13:54.41 | De_Mon | its not mad, just ready to kill someone |
| 13:54.45 | De_Mon | *hes |
| 13:54.46 | shinao1 | i was actually hoping to use asterisk with just sip clients at site B |
| 13:54.47 | [TK]D-Fender | shinao1: is THIS correct? |
| 13:54.56 | shinao1 | very correct |
| 13:55.07 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 13:55.08 | shinao1 | (B) has nothing installed |
| 13:55.09 | [TK]D-Fender | shinao1: How many phones do you want at site (b)? |
| 13:55.12 | flujan | De_Mon: So which is the point in configuring a lot of codecs on the softphone? |
| 13:55.15 | [TK]D-Fender | WANT <- |
| 13:55.24 | shinao1 | well there going to be like 4 phones |
| 13:55.30 | shinao1 | 4 for now |
| 13:55.33 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 13:55.42 | De_Mon | flujan from my limited understanding, the call will not connect if the phones can not determine a codec to use. |
| 13:55.49 | [TK]D-Fender | shinao1: You able to put a Linux box to run * over there? |
| 13:55.56 | shinao1 | yes i can |
| 13:56.00 | shinao1 | i intend to |
| 13:56.03 | asterisknerds | <PROTECTED> |
| 13:56.04 | De_Mon | determine/negotiate/ |
| 13:56.11 | [TK]D-Fender | shinao1: Ok, NOW we're getting somewhere. |
| 13:56.22 | flujan | hum... so the problem is with the phones right? |
| 13:56.33 | [TK]D-Fender | shinao1: What kind of phones do you want? Are these for office users at their desks? |
| 13:56.54 | De_Mon | flujan try a simpler test, can you call * directly and talk to the echo test from both phones? |
| 13:57.03 | shinao1 | office users at their desks [TK]D-Fender |
| 13:57.25 | shinao1 | so for (B) i want to drop IP phones there |
| 13:57.38 | shinao1 | everything runs over the LAN |
| 13:57.38 | anonymouz666 | anyone knows the difference between PAP2-NA and PAP2-TNA? |
| 13:57.52 | [TK]D-Fender | shinao1: Naturally each site has a decent broadband connection, right? |
| 13:58.14 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 13:58.29 | shinao1 | they are linked with a very high quality point-to-point WIFI link |
| 13:58.40 | shinao1 | i have 5 Mbps on that |
| 13:58.50 | [TK]D-Fender | shinao1: For this empty site : |
| 13:58.50 | shinao1 | THEY, actually |
| 13:58.57 | [TK]D-Fender | shinao1: 4 x http://www.telephonydepot.com/product_p/105-058-330.htm |
| 13:59.10 | [TK]D-Fender | shinao1: 4 x http://www.telephonydepot.com/product_p/105-058-203.htm |
| 13:59.25 | [TK]D-Fender | shinao1: for your Panasonic PBX side : |
| 13:59.26 | shinao1 | checking it out now.. |
| 13:59.50 | *** join/#asterisk masterisk (n=mascool@70.88.122.206) |
| 14:00.03 | [TK]D-Fender | shinao1: 1-3 these DUAL PORT ATAs : http://www.telephonydepot.com/product_p/105-054-212.htm |
| 14:00.27 | elixer | [TK]D-Fender: i have these two styrafoam cups and this long piece of string... and i was thinking... |
| 14:00.28 | elixer | heh |
| 14:00.28 | [TK]D-Fender | shinao1: Dependsing on how many calls you want to be able to carry |
| 14:00.44 | [TK]D-Fender | elixer: Only works on TIN!!!! how many times do I have to say it!??!?!?! |
| 14:00.51 | MindTheGap | morning all... is it possible to separate zap channels? example, i have 16 zap channels but i want Zapscan to cycle channels 2 to 10 only, and reserve the other channels for private calls. |
| 14:01.15 | shinao1 | :D |
| 14:01.20 | shinao1 | murder by irc |
| 14:01.22 | shinao1 | cool |
| 14:01.38 | elixer | [TK]D-Fender: damn, you're right. i'll update the wiki. |
| 14:01.41 | JT | [TK]D-Fender: so i thought tin was too expensive and made some cups from sound tile foam |
| 14:01.44 | [TK]D-Fender | shinao1: "Reach Out And Kill Someone" (Welcome to the NEW AT&T) |
| 14:01.51 | shinao1 | yeah ok? why do you recommend the SPAs? have to ask |
| 14:02.02 | [TK]D-Fender | JT : Foam is a sound INSULATOR |
| 14:02.12 | shinao1 | so where does * fit in all this? |
| 14:02.18 | JT | yeah but i read on the forums it was good |
| 14:02.30 | [TK]D-Fender | shinao1: you need a FXS ATA to plug into those CO ports and the Linksys ones work very well. |
| 14:02.37 | elixer | JT: only with the latest drivers |
| 14:02.45 | shinao1 | i see... hmmm |
| 14:02.47 | [TK]D-Fender | shinao1: how this fits (diagram) : |
| 14:03.36 | *** join/#asterisk umop3plsdn (n=blah@rrcs-72-43-56-75.nys.biz.rr.com) |
| 14:04.06 | [TK]D-Fender | shinao1: Panasonic PBX ---> 4 CO ports (2 SPA's) ---> {Wireless link} ---> (other side) ---> Asterisk Server ----> (local lan) ----> Polycom IP phones. |
| 14:04.37 | shinao1 | i seee.... |
| 14:04.38 | [TK]D-Fender | shinao1: this way if your link is down your remote office with your * server can still talk amongst themselves |
| 14:04.55 | shinao1 | cant i just have 2 * servers? |
| 14:05.07 | shinao1 | i was going to build that in , because... |
| 14:05.08 | [TK]D-Fender | shinao1: You are effectively giving THEM the PBX and your Panasonic is more like the "guest" in this setup. |
| 14:05.11 | [TK]D-Fender | shinao1: No need. |
| 14:05.12 | JT | you could, just lacks efficiency |
| 14:05.29 | shinao1 | the side with the PBX is the main office |
| 14:05.38 | [TK]D-Fender | shinao1: AND lacks efficiency and adds and additional point of failure |
| 14:05.56 | [TK]D-Fender | shinao1: Scary that this empty remote place is getting the BETTER setup, isn't it? :) |
| 14:05.58 | shinao1 | and if there will be PSTN calls coming in, it will be thru the side with the PBX |
| 14:06.06 | [TK]D-Fender | shinao1: this who project should cost < $1000 |
| 14:06.30 | shinao1 | ok |
| 14:06.53 | *** join/#asterisk Strom_M (n=strom@bas5-montreal28-1177917181.dsl.bell.ca) |
| 14:07.01 | shinao1 | i see.. really below budget! |
| 14:07.10 | shinao1 | :D thanks a million |
| 14:07.19 | shinao1 | uh,,, thousand really |
| 14:07.20 | shinao1 | :D |
| 14:07.46 | [TK]D-Fender | shinao1: $753.5 to be precise. (including the 100$ PC this project deserves) |
| 14:08.01 | *** join/#asterisk nephfl (n=no@wsip-68-110-130-57.ga.at.cox.net) |
| 14:08.39 | shinao1 | aah.. |
| 14:08.53 | nephfl | how can i switch the order of the lines within a trunk that my system chooses to dial out on |
| 14:09.23 | [TK]D-Fender | nephfl: Change the GROUP that the channels belong to and dial out using that group |
| 14:09.37 | shinao1 | at the expense of being totally stupid, if i need to put in a PSTN line, it still goes into the Panasonic, right? |
| 14:09.59 | x86 | shinao1: you can run a PSTN line directly to asterisk |
| 14:10.08 | [TK]D-Fender | shinao1: Panasonic people will choose an outside line (which is faked by the SPA), and dial an ext on the other side. |
| 14:10.13 | [TK]D-Fender | shinao1: You can do that, sure. |
| 14:10.26 | nephfl | so there is no way to command it to change how it selects from the group, i just need to put each line in its own group and select the order i want? |
| 14:10.27 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:10.38 | [TK]D-Fender | shinao1: Whether your panasonic will let calls coming in from your * side go OUT those lines depends on your system |
| 14:10.52 | [TK]D-Fender | shinao1: I don't know how well it lets you do that. |
| 14:11.06 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:11.36 | [TK]D-Fender | nephfl: No such thing as "order" really. A channel can be a member of MULTIPLE groups simultaneously if you want. |
| 14:11.39 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 14:11.55 | shinao1 | ok... |
| 14:12.27 | shinao1 | lemme think in this [TK]D-Fender |
| 14:12.40 | shinao1 | thanks a lot for the framework and pricing |
| 14:12.57 | [TK]D-Fender | shinao1: Since you never USED the CO ports on that panasonic before I'm guessing that box is as dumb as can be. It probably has no Auto-attendant, no VM or anything does it? |
| 14:13.08 | shinao1 | nothing at all |
| 14:13.25 | shinao1 | can i get * on the other side to take care of that? |
| 14:14.39 | shinao1 | even IVR? |
| 14:14.40 | [TK]D-Fender | shinao1: Not really. |
| 14:14.45 | shinao1 | in this setup? |
| 14:14.49 | [TK]D-Fender | shinao1: that panasonic is really a dead-end |
| 14:15.01 | shinao1 | i wish i could junk it.. |
| 14:15.19 | [TK]D-Fender | shinao1: it starts cheap and then you'd better hope your nostrils can grow to 10x their normal size... |
| 14:15.20 | shinao1 | i was hoping i could get them to buy a Astribank instead |
| 14:15.29 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:15.31 | [TK]D-Fender | shinao1: wouldn't help |
| 14:15.52 | [TK]D-Fender | shinao1: thing is you can't integrate it into that setup for VM/etc. Analog is just too dumb. |
| 14:16.10 | [TK]D-Fender | shinao1: And I would personally never use that product you mentioned. |
| 14:16.23 | shinao1 | wow.. ok... what would you use then? |
| 14:16.30 | shinao1 | recommendation? |
| 14:17.50 | [TK]D-Fender | shinao1: well first there is NOTHING to do with your existing system. Its a complete dead-end. |
| 14:18.01 | *** join/#asterisk eject_ck (n=eject@85.223.182.86) |
| 14:18.01 | [TK]D-Fender | shinao1: recommendation : SELL IT and start over |
| 14:18.40 | shinao1 | :D not my decision to make.. but id like to really |
| 14:18.44 | shinao1 | too much the hassle |
| 14:18.50 | shinao1 | thanks a lot [TK]D-Fender |
| 14:19.00 | shinao1 | i have to go strangle my boss now |
| 14:19.16 | eject_ck | Hi all, I'm trying call via asterisk to Linksys SPA-3102 from X-Lite (want use VoIP-to-PSTN) but I cannot input PIN when SPA ask it. Can anybody help? |
| 14:19.17 | [TK]D-Fender | shinao1: PM |
| 14:19.19 | De_Mon | shinao1 tell him your way under budget and would like to expand to project to include replacing the existing pbx too |
| 14:19.34 | De_Mon | err panasonic (whatever it is) |
| 14:19.38 | elixer | shinao1: an axe is more affective |
| 14:19.40 | shinao1 | and if i dont strangle him properly i dont get a raise |
| 14:20.02 | shinao1 | nah elixer its less personal |
| 14:20.39 | shinao1 | already tried that |
| 14:20.47 | shinao1 | he was gonna sue for unprofessionalism |
| 14:20.50 | shinao1 | :D |
| 14:20.53 | shinao1 | cheers guys |
| 14:21.28 | shinao1 | thanks again |
| 14:21.35 | Byan | shinao1: whatcha selling? |
| 14:23.39 | *** join/#asterisk jamos (n=jamie@dsl-58-6-20-178.wa.westnet.com.au) |
| 14:23.47 | jamos | Hi All |
| 14:24.34 | jamos | Does anyone know of any good guides for getting a TDM400 working on Fedora Core 6? I cant get it to compile |
| 14:24.37 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 14:25.42 | [TK]D-Fender | jamos: I've heard MANY problems with 2.6.22. Roll-back to 2.6.20 |
| 14:28.31 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 14:28.36 | ZaVoid | morning |
| 14:28.37 | [TK]D-Fender | eject_ck: www.voxilla.com <- check out the forums. Generally it should never be asking for a PIN. |
| 14:28.54 | ZaVoid | is it possible to selectively turn on asterisk realtime based on the incoming IP of an invite? |
| 14:29.03 | [TK]D-Fender | eject_ck: Normal setup has its dialplan accept anything * gives it and just dials out. |
| 14:29.12 | [TK]D-Fender | ZaVoid: No. |
| 14:29.15 | ZaVoid | damn |
| 14:29.23 | ZaVoid | that was my problem from last night i figured it out |
| 14:29.43 | ZaVoid | my invite is coming in and asteirsk real time is deciding what context to put it in because my invite has a known account # in it |
| 14:29.59 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:33.58 | jamos | Ah-k, thanks |
| 14:34.26 | *** part/#asterisk jamos (n=jamie@dsl-58-6-20-178.wa.westnet.com.au) |
| 14:35.15 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 14:35.29 | *** join/#asterisk el_critter (n=chatzill@190.74.100.35) |
| 14:35.56 | *** join/#asterisk Yarq (n=Nwm@88.250.219.95) |
| 14:36.03 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 14:36.32 | Yarq | Hello |
| 14:36.36 | Yourname` | Hi. Would anyone be as kind to help me out with using sipp, please? |
| 14:36.59 | Yarq | would anyone be as kind to tell me what that error message means [Aug 16 17:31:13] NOTICE[16322]: chan_sip.c:14758 handle_request_register: Registration from '<sip:bora@89.149.232.50>' failed for '88.250.219.95' - Device does not match ACL |
| 14:37.08 | tzanger | Yarq: that's easy |
| 14:37.19 | tzanger | you have permit/deny and the registration from the deivce matching it does not match the ACL requirements |
| 14:37.39 | Yarq | I'm trying to use a softphone |
| 14:37.41 | tzanger | i.e. you have something like deny=0/0;permit=192.168.1.55/32 and the device is actually 192.168.1.54 |
| 14:38.26 | Yarq | hmm tzanger: where do i set that option in sip.conf ? |
| 14:39.04 | tzanger | look in the entry for that IP; it should be obvious, as you will have had to declare an ACL for that message to be emitted |
| 14:41.09 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 14:42.18 | Yarq | http://www.pastebin.ca/659760 |
| 14:44.05 | *** join/#asterisk Uatec_ (n=uatecuk@adsl.ntsols.com) |
| 14:45.10 | [TK]D-Fender | Yarq: remove your "permit", and "username=bora" |
| 14:46.19 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 14:46.20 | Yarq | ok but this time I get no response on the console |
| 14:47.02 | [TK]D-Fender | Yarq: SIP debug on and please pastebin the attempt. |
| 14:49.36 | *** join/#asterisk DarylVOIP (n=daryl@host-24-225-239-34.patmedia.net) |
| 14:49.54 | Yarq | http://pastebin.ca/659764 |
| 14:51.49 | Uatec_ | Why my asterisk box tries to make a call, using iax, to our other asterisk box, i get back "No authority found" |
| 14:52.15 | tzanger | hahaha |
| 14:52.16 | tzanger | I just read an analogy for power factor that compares it to beer foam |
| 14:52.40 | Mercestes | Uatec_, I think that means it couldn't find the context it is supposed to be talking to |
| 14:52.52 | Yourname` | ~sipp |
| 14:52.53 | jbot | i guess sipp is a free Open Source test tool / traffic generator for the SIP protocol available from http://sipp.sourceforge.net/. If you really want to know how many channels your Asterisk box can do, learn how to utilize this program. |
| 14:53.44 | Mercestes | Is that safe to run during business hours? |
| 14:53.45 | [TK]D-Fender | Yarq: Password pretty clearly seems wrong... |
| 14:54.17 | Uatec_ | Mercestes, that's really weird |
| 14:54.21 | DarylVOIP | Hey all - anyone up for helping me figure out a nasty IF/REGEX thing in extensions.conf? I'm trying to set a static number for CALLERID(num) only if the one provided is not a valid NANP. |
| 14:54.22 | Uatec_ | infact, i don't think it does |
| 14:54.38 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-84ca8e15ba40b8f8) |
| 14:54.46 | *** join/#asterisk lirakis (n=eric@69.24.142.1) |
| 14:54.56 | Uatec_ | becuase if i Dial(IAX2/ast2/773@random) and random doesn't exist, i get a different error |
| 14:55.11 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 14:55.19 | Yarq | [TK]D-Fender |
| 14:55.24 | Yarq | its not with the password |
| 14:55.35 | Yarq | it was with the ip I havent selected a specific ip to bind asterisk on |
| 14:55.41 | asterisknerds | <PROTECTED> |
| 14:55.54 | Yarq | because of that It was using the ip 47. not 50. |
| 14:56.03 | Mercestes | Uatec_, then it probably means that it failed to authenticate then. IAX2 is really weird. You kinda have to follow the example verbatim to make it work. |
| 14:56.16 | Mercestes | asterisknerds, Did you mean every word of that? |
| 14:58.50 | Yarq | another question has anyone been successful |
| 14:58.56 | Yarq | in setting up wm6 voip with asterisk |
| 15:01.55 | Sweeper | when I'm doing echo testing, playing ivr messages should be a good enough test, shouldn't it? |
| 15:01.55 | [TK]D-Fender | DarylVOIP: I gave you a complete way to test the validity of NANPA jsut the other day.... |
| 15:02.16 | [TK]D-Fender | Sweeper: No, a human voice where audio can loop back <- |
| 15:02.38 | keith4 | [TK]D-Fender: do you remember everyone you talk to, or do you just have a 7 day chat log? :-P |
| 15:02.42 | [TK]D-Fender | Sweeper: What card? |
| 15:02.49 | Sweeper | tdm400p |
| 15:02.54 | [TK]D-Fender | keith4: All commited to memory. |
| 15:03.00 | keith4 | impressive! |
| 15:03.04 | [TK]D-Fender | Sweeper: Pray..... |
| 15:03.09 | [TK]D-Fender | Sweeper: How old is it? |
| 15:03.13 | Sweeper | a week |
| 15:03.26 | [TK]D-Fender | Sweeper: Go contact Digium to get your HPEC keys for it |
| 15:03.35 | [TK]D-Fender | Sweeper: You'll want that option open. |
| 15:06.32 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 15:07.00 | Sweeper | [TK]D-Fender: what info do I need to have for them? |
| 15:07.50 | *** join/#asterisk saftsack (n=saftsack@pD9E079C3.dip.t-dialin.net) |
| 15:07.54 | [TK]D-Fender | Sweeper: Should have your invoice handy |
| 15:07.59 | [TK]D-Fender | Sweeper: Go call |
| 15:08.17 | Sweeper | uh, apparently the card didn't get bought straight from digium |
| 15:08.26 | keith4 | what is an HPEC key? |
| 15:08.27 | Sweeper | (don't ask me why..." |
| 15:08.58 | elixer | keith4: echo cancellation |
| 15:09.00 | [TK]D-Fender | Sweeper: Doesn't have to be |
| 15:09.31 | elixer | keith4: specifically High Performance Echo Cancellation |
| 15:09.38 | keith4 | sound enticing |
| 15:09.42 | keith4 | why does it require a key? |
| 15:09.57 | elixer | keith4: its not free? |
| 15:10.13 | elixer | keith4: http://ftp.digium.com/pub/telephony/hpec/README |
| 15:10.29 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 15:11.14 | keith4 | they don't happen to have a "check my warranty" page, do they? |
| 15:12.15 | elixer | keith4: no clue. my knowledge of HPEC goes all the way back to when i googled "HPEC key" 6 1/2 minutes ago. |
| 15:12.40 | keith4 | ah |
| 15:12.50 | keith4 | well, don't tease me like that by acting all knowledgable |
| 15:14.00 | jcolp | Sweeper: do you have your card's serial number? |
| 15:15.17 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 15:15.19 | Sweeper | jcolp: don't, sadly. it's already installed on location :/ |
| 15:15.38 | Sweeper | gonna double check my fxotune stuff, and if not, send someone to crack the box |
| 15:17.07 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:17.07 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:18.29 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 15:19.45 | Sweeper | what's that command that loops what you say back to you? |
| 15:19.58 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 15:22.44 | [TK]D-Fender | Sweeper: "show application echo" |
| 15:22.59 | [TK]D-Fender | Sweeper: Of course thats pretty much BACKWARDS from what you're trying to do :) |
| 15:23.20 | *** join/#asterisk joeTSUNAMI (n=nnnjoeTS@unaffiliated/joetsunami) |
| 15:23.45 | *** join/#asterisk thansen|laptop (n=thansen@74-36-210-143.dr01.hmdl.id.frontiernet.net) |
| 15:24.01 | Uatec_ | Mercestes, it's not fair, i have it working one way, but not the other |
| 15:24.17 | Uatec_ | i even copied the iax.conf section from the working one to the broken one |
| 15:24.23 | Mercestes | Uatec_, I agree, it's not fair |
| 15:24.36 | Sweeper | [TK]D-Fender: well, if I can put enough delay on it, I should be able to do decent testing |
| 15:24.43 | *** part/#asterisk joeTSUNAMI (n=nnnjoeTS@unaffiliated/joetsunami) |
| 15:24.55 | Mercestes | Iax.conf could benefit greatly from a syntatical upgrade. |
| 15:25.07 | Mercestes | Something far more simple like, [me] and [them] |
| 15:25.29 | Mercestes | Or a nice .msi installre. |
| 15:25.36 | Mercestes | s/installre/installer/ |
| 15:25.38 | *** join/#asterisk Yarq (n=Nwm@88.250.219.95) |
| 15:25.47 | *** join/#asterisk joeTSUNAMI (n=nnnjoeTS@unaffiliated/joetsunami) |
| 15:28.20 | Uatec_ | oh dear |
| 15:28.24 | Uatec_ | now it doesn't work either way :'( |
| 15:28.33 | Yarq | allow sip inbound ? |
| 15:28.43 | Yarq | is there an option like that |
| 15:29.21 | [TK]D-Fender | Yarq: 401 means the user/pass is wrong. Maybe you did your client in ALL CAPS or something silly, etc. |
| 15:29.50 | [TK]D-Fender | Yarq: Oh wait.. you had an IP problem for that before, didn't you? Bind issue... |
| 15:29.56 | [TK]D-Fender | Yarq: Whats this related to? |
| 15:30.50 | Yarq | well I configured my softphone but now trying to configure my wm6 voip sip phone |
| 15:30.53 | Yarq | which comes with wm6 |
| 15:30.57 | Yarq | windows mobile |
| 15:31.35 | skyphyr | I'm getting this error when dialing an extension of a phone that is registered. You can also call out from that phone - any idea how I can track the error down? [Aug 16 16:30:05] WARNING[13904]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 15:31.37 | Yarq | http://pastebin.ca/659801 |
| 15:31.55 | skyphyr | would giving the phones static IPs help? |
| 15:32.05 | Mercestes | skyphyr, that pretty much means, literally, that there is no network route to that phone. |
| 15:32.21 | Mercestes | skyphyr, try running an mtr to tha tphones ip address as it's listed in sip show peers |
| 15:33.04 | [TK]D-Fender | skyphyr: Try pastebinning the whole CALL with SIP debug & verbose 10. That error alone can mean ANYTHING. |
| 15:33.06 | [TK]D-Fender | ~pb |
| 15:33.07 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:33.08 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 15:35.08 | *** part/#asterisk joeTSUNAMI (n=nnnjoeTS@unaffiliated/joetsunami) |
| 15:36.34 | *** join/#asterisk Bombe (n=droden@wing.pterodactylus.net) |
| 15:36.57 | Mercestes | [TK]D-Fender, not a very linux-esque error, then, is it? |
| 15:37.00 | Bombe | Hi. |
| 15:37.20 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:37.53 | *** join/#asterisk alin` (n=user@193.226.173.50) |
| 15:37.57 | Bombe | When I put a user into a meetme conference using the 'm' flag (muting him) can I still receive DTMF tones via the AGI background script? |
| 15:38.26 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 15:38.39 | alin` | I made a transcodec for RTP packets, and I hear a noise on the background. Can somebody give me a suggestion, please? |
| 15:38.48 | Mercestes | [TK]D-Fender, Linux is usually more helpful with their errors. "Encountered Error 3: Device not registered. This is usually due to an incorrect password, or NAT traversal errors. Sip.conf examples can be found at "Http://www.here.com" and NAT settings can be found at "Http://www.there.com" You may find online help for your error at IRC: #asterisk. Would you like a cup of tea with that?" |
| 15:39.31 | [TK]D-Fender | Mercestes: No the usual "Would you like fries with that, sir?" works for me ;) |
| 15:39.41 | Mercestes | Now, *MICROSOFT* on the other hand, is more, "Encountered error 0xf8o8ea0d |
| 15:40.28 | Bombe | That address is not dword-aligned and thus highly unlikely to be featured in an error message like this. :) |
| 15:40.56 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 15:41.19 | Mercestes | Bombe, The rest of it was truncated because the window was too little. |
| 15:41.32 | Bombe | :) |
| 15:41.39 | *** part/#asterisk alin` (n=user@193.226.173.50) |
| 15:41.50 | *** join/#asterisk alin` (n=user@193.226.173.50) |
| 15:42.29 | *** part/#asterisk alin` (n=user@193.226.173.50) |
| 15:42.33 | *** join/#asterisk alin` (n=user@193.226.173.50) |
| 15:42.42 | Mercestes | alin`, I missed you. |
| 15:43.08 | [TK]D-Fender | Mercestes: Try again! |
| 15:43.11 | alin` | Mercestes: ?? |
| 15:43.55 | Mercestes | sigths? |
| 15:44.50 | Mercestes | alin`, you left. and I missed you, and then you came back, and I was happy, and then you left again, and I was disappointed, and then you came back again, and I was vindicated. |
| 15:45.10 | Mercestes | [TK]D-Fender, quite a lisp there, big boy. :) |
| 15:45.59 | alin` | I use ERC and I changed a variable in Emacs, and I killed the #asterisk buffer, and I reopened it. |
| 15:46.10 | Mercestes | alin`, that's what I said. |
| 15:46.35 | alin` | Mercestes: ok, but how can I solve the problem of noise? |
| 15:46.45 | Mercestes | alin`, /ignore me would probably work. |
| 15:46.47 | Yarq | pff this time no sound |
| 15:46.52 | Mercestes | alin`, Oh, your not talking about *me* are you? |
| 15:47.01 | Mercestes | alin`, what noise? |
| 15:47.15 | alin` | noise of sound. |
| 15:47.28 | Mercestes | alin`, you made a transcodec? |
| 15:47.31 | alin` | yes |
| 15:47.37 | alin` | of RTP |
| 15:49.05 | *** join/#asterisk el_critter (n=chatzill@190.74.100.35) |
| 15:49.05 | Mercestes | define, transcodec. |
| 15:49.26 | alin` | ??? |
| 15:49.49 | Yarq | no sound on my sip :S |
| 15:49.52 | Mercestes | would that be a cross dressing protocol? Or a protocol designed to...transcode to other protocols>? |
| 15:49.56 | blitzrage | anyone use a mediatrix 4124 with asterisk? Doesn't really look like a lot of configuration that I can do to this device. It registers, but doesn't reply to INVITEs. I think it's the remote ends network blocking the invites from getting to the mediatrix 4124 |
| 15:50.15 | blitzrage | just making sure there isn't some magic switch I've missed :D |
| 15:50.41 | alin` | Mercestes: I reveive RTP packs in a codec, and I send RTP with the same voice into another transcodec |
| 15:52.05 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 15:52.06 | Yarq | :( |
| 15:52.17 | Mercestes | alin`, that's trascode. |
| 15:54.42 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:55.16 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 15:56.00 | magic_hat | anyone know what dtmfmode setting I need to get key touches working on interactive menus? |
| 15:56.05 | magic_hat | with teliax? |
| 15:56.58 | tzafrir_laptop | teliax use sip? |
| 15:57.05 | magic_hat | yes. |
| 15:59.26 | Bombe | Does anybody know whether I can still read DTMF tones from the AGI background channel belonging to a meetme user with the 'm' flag? |
| 16:01.56 | [TK]D-Fender | Bombe: have you considered TRYING?! |
| 16:02.53 | Bombe | Indeed, I have. I just don't know whether I'm doing something wrong or whether Asterisk really doesn't receive/parse any DTMF tones. |
| 16:03.35 | [TK]D-Fender | Bombe: have the same device enter without the "m" flag. if it works, then there's your answer. |
| 16:04.23 | Bombe | Which means that a muted user does not have any possibility to enter DTMF tones to adjust volume or leave a conference or something. |
| 16:04.26 | Bombe | What a pity. |
| 16:05.53 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 16:07.01 | [TK]D-Fender | Bombe: leave? |
| 16:07.18 | [TK]D-Fender | Bombe: ... there IS no exit strategy!!! |
| 16:07.55 | hi365 | can i set (or boost) the volume of MOH? |
| 16:08.03 | *** part/#asterisk alin` (n=user@193.226.173.50) |
| 16:08.47 | [TK]D-Fender | hi365: mpg123 yes, other externals, probably, Native = no (all can be done by using SOX to mod the priginal files) |
| 16:08.57 | [TK]D-Fender | original* |
| 16:09.15 | hi365 | [TK]D-Fender: guess werll go that rout then. thanks |
| 16:09.30 | [TK]D-Fender | hi365: just increase your base files. |
| 16:10.22 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 16:11.42 | *** join/#asterisk luke-jr|work (n=luke-jr@adsl-76-194-177-177.dsl.ksc2mo.sbcglobal.net) |
| 16:11.51 | *** join/#asterisk trustinfo-tb (n=trust@AStrasbourg-156-1-59-71.w81-51.abo.wanadoo.fr) |
| 16:11.54 | trustinfo-tb | hi |
| 16:11.56 | luke-jr|work | how do you deal with people who insist on spelling * "asterix"? |
| 16:12.08 | trustinfo-tb | ftp://ftp.digium.com/pub/zaptel/b410p/mISDN-1_1_3.tar.gz doesn't work |
| 16:12.08 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 16:12.27 | [TK]D-Fender | luke-jr|work: ridicule and name calling naturally! |
| 16:13.31 | elixer | luke-jr|work: these are people with high school educations? |
| 16:14.14 | elixer | (for the record, i _barely_ have one of those) |
| 16:15.25 | luke-jr|work | [TK]D-Fender: hm, will it get me in trouble? |
| 16:15.28 | luke-jr|work | this is a coworker |
| 16:15.36 | Mercestes | Are they in #asterisk? |
| 16:15.44 | luke-jr|work | no |
| 16:15.48 | Mercestes | Then probably. |
| 16:16.02 | Mercestes | Here, you could probably get away with it. I riducle and name-call people all day long in here. |
| 16:16.10 | Mercestes | From the moment they say "hi." |
| 16:16.49 | [S]Mercestes | Brar |
| 16:16.51 | elixer | hi |
| 16:17.25 | [S]Mercestes | elixer, moron! |
| 16:17.38 | *** join/#asterisk Daemon5150 (n=daemon@snapon.execulink.net) |
| 16:17.41 | elixer | but i want to know about asterix |
| 16:17.42 | elixer | ! |
| 16:17.48 | [S]Mercestes | lol |
| 16:17.58 | blitzrage | asterix is a cartoon... |
| 16:18.13 | blitzrage | or were you being wrong on purpose? :) |
| 16:18.34 | [TK]D-Fender | luke-jr|work: Tell him to show some IQ and actually spell this right. I'm using a GUI at the office here for * (ScopServ). Our idiot ex-head-of customer service kept calling is SScop. Jittery ass..... |
| 16:18.53 | blitzrage | s/*/asterisk/ |
| 16:18.57 | Daemon5150 | Hi all. This may be an FAQ but I can't find the answer I'm looking for... Is there any way to get Asterisk to authenticate INVITE's with an error 401 instead of an error 407? |
| 16:19.04 | [TK]D-Fender | blitzrage: That vid you posted.... OMG too much. Where do you find that crap? |
| 16:19.14 | elixer | blitzrage: if i didn't know, i'd be wondering why no one was in #asterix |
| 16:19.15 | elixer | :0 |
| 16:19.22 | blitzrage | heh |
| 16:19.25 | elixer | s/:0/:)/ |
| 16:19.28 | elixer | heh |
| 16:19.35 | [TK]D-Fender | Daemon5150: 401 isn't an authenticate, its a REFUSAL to do so :) |
| 16:19.50 | blitzrage | [TK]D-Fender: :D |
| 16:20.41 | Daemon5150 | 401 is Unauthorized, and is supposed to include a WWW-Authenticate challenge |
| 16:20.41 | blitzrage | [TK]D-Fender: friend of mine posted it, then I just clicked the share button... hilarious :) |
| 16:20.50 | Daemon5150 | My softswitch can't deal with 407, it needs 401 |
| 16:20.56 | blitzrage | lol |
| 16:20.59 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 16:21.03 | blitzrage | that's a bad softswitch then :) |
| 16:21.17 | blitzrage | 407 is needed when you authenticate an INVITE |
| 16:21.26 | Daemon5150 | hah, it's a CLASS4/CLASS5 tandem and local access switch |
| 16:21.33 | blitzrage | that doesn't make it good :) |
| 16:22.09 | [T]ank | i am trying to figure out how to use atxfer. I have uncommented atxfer in features.conf and have added exten => s,n,Set(DYNAMIC_FEATURES=atxfer) to my extension macro. is that all i should have to do? |
| 16:22.47 | [S]Mercestes | Daemon5150, which softswitch is it? |
| 16:23.05 | Daemon5150 | Perhaps, but if Asterisk isn't a proxy (as is commonly discussed), then why is it using proxy authentication? |
| 16:23.13 | Daemon5150 | The softswitch is a Metaswitch |
| 16:23.22 | [S]Mercestes | I've heard of that one. |
| 16:23.37 | [S]Mercestes | I know some guys who have an asterisk hooked to it so I know it can be dnoe. |
| 16:23.52 | [TK]D-Fender | [T]ank: Make sure to pass the appropriate options to Dial. |
| 16:24.02 | [S]Mercestes | s/dnoe/done/ |
| 16:24.08 | Daemon5150 | I know it can be done using insecure=very, but I want the Asterisk end to authenticate INVITEs |
| 16:24.16 | skyphyr | hi Mercestes and [TK]D-Fender - sorry got dragged away for other issues - it seems the phones with issues all say unspecified now |
| 16:24.27 | skyphyr | with sip show peers I mean |
| 16:24.32 | [S]Mercestes | skyphyr, so they are *NOT* registered as you lead me to believe? |
| 16:24.44 | skyphyr | if I look on the phones web page |
| 16:24.48 | elixer | [S]Mercestes doesn't like when people lie to him |
| 16:24.48 | skyphyr | it believes it still is |
| 16:25.00 | [T]ank | [TK]D-Fender: by options do you mean like tT for the transfer? |
| 16:25.07 | skyphyr | I didn't check on asterisk as I didn't not how :$ |
| 16:25.08 | [TK]D-Fender | [T]ank: yes. |
| 16:25.14 | [S]Mercestes | skyphyr, Stop now. |
| 16:25.15 | [S]Mercestes | ~book |
| 16:25.16 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:25.16 | [T]ank | yeah, i have that too |
| 16:25.22 | [S]Mercestes | ~wikis |
| 16:25.22 | jbot | hmm... wikis is http://www.voip-info.org |
| 16:25.24 | [TK]D-Fender | skyphyr: Enable "sip debug" and PASTEBIN something. |
| 16:25.24 | [S]Mercestes | ~docs |
| 16:25.25 | jbot | methinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 16:25.26 | [TK]D-Fender | !pb |
| 16:25.28 | [S]Mercestes | read all that first. |
| 16:25.31 | skyphyr | I have this book - read it a couple of years ago |
| 16:25.36 | skyphyr | so it's all faded |
| 16:25.37 | [S]Mercestes | read it again. |
| 16:25.40 | [S]Mercestes | and then come back |
| 16:25.48 | skyphyr | alrighty - thanks for the help :-) |
| 16:25.51 | [S]Mercestes | np |
| 16:25.54 | elixer | and then unlearn all of the outdated stuff |
| 16:25.57 | elixer | heh |
| 16:25.59 | [S]Mercestes | but, that was why you were getting "no route to host" they are not registered. |
| 16:26.03 | Daemon5150 | So is it the case that Asterisk can't authenticate INVITE's with a 401 challenge, like many other SIP devices do?? |
| 16:26.24 | [S]Mercestes | Daemon5150, I thought 401 was an error, not a challenge. |
| 16:26.24 | skyphyr | could this be because the boss went with the crappy Bugetone 100s that I didn't want to get? |
| 16:26.31 | [S]Mercestes | ~gs |
| 16:26.32 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:26.34 | [S]Mercestes | ~phones |
| 16:26.34 | jbot | [phones] http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 16:26.46 | sheppard | ~thebook |
| 16:26.47 | jbot | i guess thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:26.47 | [S]Mercestes | skyphyr, very possible. |
| 16:27.06 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 16:27.15 | Daemon5150 | From RFC2616: 401 is Unauthorized, and "the response MUST include a WWW-Authenticate field containing a challenge..." |
| 16:27.36 | [S]Mercestes | ok |
| 16:28.03 | Daemon5150 | IMHO Polycom is crap, I've been working with then for 2 years and they're buggy as all heck |
| 16:28.14 | [S]Mercestes | Daemon5150, then update your firmware. |
| 16:28.19 | Yourname` | Hi, following this link: http://www.das-asterisk-buch.de/stable/applikationen-waitforsilence.html, I've tried using WaitForsilence(2500,1,30) -- and for some reason it isn't working, what am I doing wrong? The way I interpret it is, wait for 2.5 seconds of silence, once and if the silence doesn't happen within 30 seconds.. move on. |
| 16:28.22 | datachomper | I <3 my polycom 501's |
| 16:28.44 | Daemon5150 | We do, but sometimes the upgrade reintroduces bugs. eg. some bugs in 1.6.2 were resolved in 1.6.3 & 4 but were reintroduced in 1.6.5+ |
| 16:29.20 | [S]Mercestes | Daemon5150, Well, given the global success rate of Polycom here in this community you just knocked about 20 points off of your IQ, I'd wager. |
| 16:29.28 | [S]Mercestes | Daemon5150, and use the latest firmware. |
| 16:29.29 | Daemon5150 | I could go on & on about them. eg. in the XML config if you aren't case-exact on some settings then they don't take effect |
| 16:29.58 | Daemon5150 | HAh, I'm not worried about points |
| 16:30.06 | [S]Mercestes | I gathered that. |
| 16:30.15 | elixer | Yourname`: what version of asterisk are you using? |
| 16:30.21 | Yourname` | 1.2.23 |
| 16:30.23 | [S]Mercestes | What phones do you prefer? Out of curiosity? |
| 16:30.25 | Daemon5150 | 2.0.1 is working OK for us and we're not touching them |
| 16:30.38 | Yourname` | elixer: 1.2.23 |
| 16:30.43 | elixer | Yourname`: the third argument isn't supported in 1.2.23 |
| 16:30.58 | Yourname` | elixer: 1.2.24 maybe? |
| 16:31.02 | Daemon5150 | We've also heard good things about the Aastra's, but we're doing alot of BLC and PacketCable stuff that works way better than most SIP endpoints |
| 16:31.08 | elixer | Yourname`: i doubt its in 1.2 at all |
| 16:31.26 | [TK]D-Fender | Daemon5150: 2.0.1 = old |
| 16:31.35 | Yourname` | elixer: Oh :S |
| 16:31.37 | elixer | Yourname`: at the top of that link you pasted, it says this "Interner Hilfetext zu dieser Applikation in Asterisk 1.4:" |
| 16:31.39 | Daemon5150 | Snom is OK but the interface is kinda crappy |
| 16:31.48 | Daemon5150 | lots of features though |
| 16:31.57 | elixer | Yourname`: but i have no idea what that says. i did translate the "1.4" part to english though... |
| 16:32.13 | Yourname` | elixer: lol, but it does have 1.2 as well |
| 16:32.41 | Daemon5150 | ya 2.0.1 is kinda dated but it's the first stable release since 1.6.2 and we don't have time to keep re-testing every time we upgrade |
| 16:32.53 | Daemon5150 | sorry, it's 2.0.3 we're on not 2.0.1 |
| 16:32.55 | elixer | Yourname`: it does, but I am looking at the code for the latest in the 1.2 branch, and the third argument is not supported |
| 16:33.33 | elixer | Yourname`: the timeout arg is only available in 1.4 |
| 16:33.40 | elixer | Yourname`: you could try to backport |
| 16:33.50 | Yourname` | elixer: What do you mean? |
| 16:34.23 | elixer | Yourname`: you could add the argument to 1.2. copy the changes in the 1.4 branch. its not necessarily simple, but i am sure its possible. |
| 16:34.50 | Yourname` | elixer: whoa, lol.. that's ok, I think I might mess up the system.. |
| 16:35.08 | elixer | Yourname`: alrighty. |
| 16:35.45 | Yourname` | elixer: Thanks though.. but is that possible in any other way to do in 1.2 |
| 16:35.47 | Yourname` | ? |
| 16:36.15 | elixer | Yourname`: i won't waste your time - the simple answer is "i don't know" |
| 16:36.46 | Yourname` | elixer: Thank you. :) |
| 16:37.14 | elixer | Yourname`: yup |
| 16:38.20 | elixer | Yourname`: for future reference, unless its a bug fix, new features are not added within release branches. |
| 16:38.28 | elixer | Yourname`: (typically) |
| 16:38.33 | Yourname` | elixer: Gotcha! |
| 16:38.52 | elixer | Yourname`: so if function Foo doesn't work in 1.2.9.1, it won't work in 1.2.23 (unless it was deemed a 'bug') |
| 16:39.04 | elixer | Yourname`: ok, good luck and god speed. |
| 16:39.13 | Yourname` | elixer: Thank you, lol |
| 16:39.32 | Yourname` | elixer: While I have your attention, do you know much about sipp? |
| 16:39.44 | elixer | Yourname`: i know enough to get my very simple setup working. |
| 16:39.50 | elixer | Yourname`: ask away though |
| 16:40.38 | [T]ank | i am having trouble hearing people on my zap connected analog phone. Here are my configs: http://pastebin.ca/659884 |
| 16:40.39 | [T]ank | I have tried adjusting the gain... i have taken it up to 100 and still hear no difference. I also see no difference in "/usr/src/zaptel-1.4.2.1/ztmonitor 1 -vv" |
| 16:40.45 | [T]ank | am I doing it correctly? |
| 16:41.16 | Yourname` | elixer: I have a queue, extension 20. And I'd like to fill 20 up with calls from sipp. And maybe even overflow. So, I ran: sipp -sn uac -d 500 -s 20 -i 127.0.0.1 127.0.0.1 -l 10 -trace_err --> and all I see is INVITEs being sent in sip debug, and no calls in the queue at all. |
| 16:41.55 | elixer | ohhhhhhhhh, sipP (thought it was a typo) |
| 16:41.57 | elixer | heh |
| 16:42.02 | elixer | yeah, i'm no good to you again :) |
| 16:44.25 | Yourname` | haha |
| 16:44.26 | Yourname` | It's all good. |
| 16:44.46 | tzafrir_laptop | [T]ank, a gain of more than 16 is generally a bad idea |
| 16:45.47 | elixer | should this work: `svn diff http://svn.digium.com/branches/1.2/apps/app_waitforsilence.c http://svn.digium.com/branches/1.4/apps/app_waitforsilence.c` |
| 16:45.48 | elixer | ? |
| 16:46.01 | tzafrir_laptop | try adding 3 (dB) at a time, starting from 0. Thegain is set in /etc/asterisk/zapata.conf, and is actively set when Asterisk reloads the configuration (of chan_zap) |
| 16:47.00 | [T]ank | tzafrir_laptop: I cant seem to get anything to make an effect. |
| 16:47.22 | [T]ank | I have made the changes and restarted asterisk. |
| 16:47.26 | tzafrir_laptop | [T]ank, pastebin your zapata.conf |
| 16:47.39 | [T]ank | that was the link i added |
| 16:47.44 | [T]ank | let met get it again... hang on |
| 16:47.51 | [T]ank | http://pastebin.ca/659884 |
| 16:48.38 | *** join/#asterisk lbow (n=lbow@41-195-77-226.access.uunet.co.za) |
| 16:48.52 | *** part/#asterisk lirakis (n=eric@69.24.142.1) |
| 16:49.17 | tzafrir_laptop | you set it on the FXO port alone, right? |
| 16:49.45 | [T]ank | thought is was supposed to be on the fxs port |
| 16:49.55 | Yarq | Does anyone know a place that has cheep outbound services ? |
| 16:50.13 | tzafrir_laptop | fxsks signalling => fxo port |
| 16:50.46 | [T]ank | oh, yeah... always confuses me... So I need to move it to my fxo-ks section? |
| 16:50.51 | tzafrir_laptop | try removing that last channel=> line, restart asterisk. You'll still have dialtone in the FXS port, but the FXO port won't answer calls |
| 16:51.14 | tzafrir_laptop | (that was only for demonstration, of course) |
| 16:52.03 | Yarq | (an easy way to create a dialplan?) |
| 16:52.29 | tzafrir_laptop | if you move it to the first one, it will also the second one. |
| 16:53.20 | Daejeo1 | laptop |
| 16:53.30 | tzafrir_laptop | At least until you put a rxgain=0 |
| 16:53.34 | tzafrir_laptop | txgain=0 |
| 16:53.36 | tzafrir_laptop | after it |
| 16:53.59 | [T]ank | tzafrir_laptop: yeah.. moving it to the fxo section was the correction. Thank you. |
| 16:57.02 | Sweeper | damnit. |
| 16:57.06 | Sweeper | oslec not working :/ |
| 16:58.55 | [T]ank | ok... now back to atxfer. |
| 16:59.02 | [T]ank | cant get it to work. |
| 17:00.10 | [TK]D-Fender | [T]ank: Pastebin your configs, and the CLI output of your failed attempt at verbose 10, and some confirmation that DTMF is functional for the deice you're testing with. |
| 17:00.12 | tzafrir_laptop | Sweeper, what problems do you have? |
| 17:00.15 | [TK]D-Fender | device* |
| 17:01.02 | Sweeper | tzafrir_laptop: well, at the moment, it seems oslec isn't actually functioning |
| 17:01.26 | Sweeper | I cat'd /proc/oslec/info during a call and got told it wasn't monitoring anything |
| 17:01.48 | tzafrir_laptop | zap show channel NNN |
| 17:01.56 | luke-jr|work | anyone recommend a decent softphone for Windoze? |
| 17:01.59 | tzafrir_laptop | does it show that the echo canceller is used? |
| 17:02.04 | luke-jr|work | if any exist |
| 17:02.10 | tzafrir_laptop | sip? iax? |
| 17:02.30 | tzafrir_laptop | kiax? |
| 17:02.33 | Sweeper | Echo Cancellation: 128 taps unless TDM bridged, currently ON |
| 17:02.37 | Sweeper | that's what I've got |
| 17:02.57 | Sweeper | but at the same time, nothing in oslec |
| 17:03.46 | tzafrir_laptop | the oslec panel script mentions somethingabout enabling anddisabling oslec. Any idea what that is? |
| 17:04.06 | coppice | oslec seems to work pretty well. I hope i can coax david into spending the time to fill in the remaining holes :-) |
| 17:04.07 | Sweeper | nope |
| 17:04.18 | elixer | luke-jr|work: we use CounterPath's eyeBeam here |
| 17:04.42 | luke-jr|work | elixer: is it gratis? |
| 17:04.43 | Sweeper | I've got it loaded into the kernel, as well as having patched and reinstalled zaptel |
| 17:04.56 | elixer | luke-jr|work: no |
| 17:04.57 | Sweeper | luke-jr|work: eyebeam is not, x-lite, the reduced version, is |
| 17:05.28 | Sweeper | tbh, you're not going to get a much better softphone than eyebeam, imo |
| 17:06.11 | [T]ank | [TK]D-Fender: extensions.conf: http://pastebin.ca/659927 ; features.conf: http://pastebin.ca/659929 ; cli> output on an inbound call: http://pastebin.ca/659932 |
| 17:06.31 | wunderkin | i tried oslec once and like my second call or so was all distorted |
| 17:06.39 | [T]ank | it did not display any output when i did dtmf... but I called my bank and successfully used their IVR |
| 17:07.14 | [TK]D-Fender | [T]ank: Do a local test with Read |
| 17:07.26 | luke-jr|work | Sweeper: is it easy to configure? |
| 17:08.39 | [TK]D-Fender | [T]ank: The rest looks just fine (and very clean for the record) |
| 17:09.11 | Sweeper | luke-jr|work: yea. I've even gotten it to take remote provisioning, although it took some hacking |
| 17:09.28 | [TK]D-Fender | luke-jr|work: Zoiper has an edge on X-lite : native transfer capability + IAX/SIP |
| 17:09.34 | luke-jr|work | Sweeper: I mean from a dumb "asterix" end user >.> |
| 17:09.58 | Sweeper | luke-jr|work: oh, yea, the manual config is very straightforward. plug in a few values, hit go |
| 17:12.07 | Sweeper | tzafrir_laptop: I found that control panel thing....but still nothing |
| 17:12.20 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 17:12.58 | Sweeper | if you're going free, might as well check out zoiper |
| 17:18.21 | Nivex | zoiper on the mac keeps telling me a newer version is available, but it's the same version I'm running |
| 17:18.21 | Yarq | I need a prepaid calling solution any suggestions ? |
| 17:18.24 | [T]ank | [TK]D-Fender: http://pastebin.ca/659947 |
| 17:18.54 | blitzrage | Yarq: consultant |
| 17:19.35 | Yarq | outbound trunk |
| 17:19.51 | [TK]D-Fender | [T]ank: Ok, I'm all out of ideas... |
| 17:21.39 | *** join/#asterisk FFForever (n=chris@adsl-69-228-52-74.dsl.irvnca.pacbell.net) |
| 17:21.50 | FFForever | can i run asterisk on my home server? |
| 17:22.13 | FFForever | i normally just use it for files but i want to expand it to run a telephone system also |
| 17:25.55 | [TK]D-Fender | FFForever: Typically, sure. |
| 17:26.09 | FFForever | how many modems will i need? |
| 17:26.10 | AJaymn | Anyone know hot to enable RTC in the kernel? :P |
| 17:26.52 | *** part/#asterisk Daemon5150 (n=daemon@snapon.execulink.net) |
| 17:27.21 | luke-jr|work | FFForever: ... modems? |
| 17:27.31 | FFForever | for the computer to answer the line? |
| 17:27.36 | luke-jr|work | one cable or DSL modem, ideally |
| 17:27.48 | luke-jr|work | FFForever: ew, landline |
| 17:27.50 | FFForever | i meant for the phone line |
| 17:28.07 | [S]Mercestes | FFForever, Yea, don't use modems. google FXO cards |
| 17:28.17 | AJaymn | Copper!? |
| 17:28.30 | luke-jr|work | [S]Mercestes: FXO cards are far more expensive than they shoudl be ;) |
| 17:28.34 | FFForever | sorry for my idiotness ive never done anything like this b4 |
| 17:28.53 | luke-jr|work | FFForever: I'd get rid of the landline :) |
| 17:28.58 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:28.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:28.59 | [S]Mercestes | luke-jr|work, Modems suck as FXO cards far more than is tolerable. |
| 17:29.04 | FFForever | its for a small shop |
| 17:29.29 | luke-jr|work | and you need a landline why? |
| 17:29.32 | [S]Mercestes | Of course, I also suggest against using residential broadband for trying to carry voice. |
| 17:29.38 | FFForever | the 1800 number |
| 17:29.43 | luke-jr|work | [S]Mercestes: ... |
| 17:29.49 | FFForever | i just want a voicemail system..... |
| 17:29.53 | luke-jr|work | FFForever: port the 800 number to IP |
| 17:30.08 | FFForever | we get the number from att |
| 17:31.27 | *** join/#asterisk sakic (n=sakic@adsl-146-182-113.clt.bellsouth.net) |
| 17:31.45 | FFForever | so can asterisk answer the landline?, also can it have something like if the other line is busy press 2 to wait for a rep after like 2mins it will just say sorry no one is here to answer your call, please press 1 to leave a message for...., or 2 to leave a message for..... |
| 17:32.17 | [S]Mercestes | FFForever, only if you have 2 lines. |
| 17:32.23 | FFForever | we do :) |
| 17:32.29 | FFForever | we have 3actually..... |
| 17:32.43 | [S]Mercestes | In fact, case, on call 4 they will just get a busy signal |
| 17:32.54 | luke-jr|work | FFForever: att sucks |
| 17:33.05 | [S]Mercestes | All carriers suck. |
| 17:33.09 | [S]Mercestes | some just suck less than others. |
| 17:33.14 | luke-jr|work | indeed |
| 17:33.14 | FFForever | well line 3 is for reserve...., its for dial outs only....... |
| 17:33.16 | [S]Mercestes | but never by much |
| 17:33.22 | FFForever | so only 2 can be called :) |
| 17:33.36 | luke-jr|work | FFForever: why do you use a landline, again? |
| 17:33.37 | [S]Mercestes | FFForever, if you are paying for 3 lines it might be best just to invest in a PRI |
| 17:34.08 | luke-jr|work | bah, X-lite isn't working either |
| 17:34.12 | Sweeper | woohoo |
| 17:34.16 | FFForever | well he likes landlines.... |
| 17:34.35 | FFForever | what hardware will i need and how much am i looking at spending for what i need? |
| 17:35.04 | [S]Mercestes | 4 port FXO card |
| 17:36.24 | FFForever | how much should i be spending on it $600? |
| 17:37.39 | luke-jr|work | so any idea why X-lite and SJPhone both fail registration without sending a password? |
| 17:38.09 | [S]Mercestes | [S]Mercestes, The Digum 4 port FXO card. |
| 17:38.23 | FFForever | is there any cheaper one? |
| 17:38.24 | Sweeper | tzafrir_laptop: turns out the updated zapata modules didn't get installed over the existing ones :) |
| 17:38.48 | Sweeper | luke-jr|work: what's the problem? |
| 17:39.06 | tzafrir_laptop | hmmm... now how do we push oslec into Zaptel? |
| 17:39.24 | luke-jr|work | Sweeper: X-lite sends a REGISTER, Asterisk denies it with a WWW-Authenticate, and X-lite just gives up (sends nothing more) |
| 17:39.30 | coppice | you don't. oslec is GPL |
| 17:39.53 | luke-jr|work | and Zaptel isn't? |
| 17:40.44 | Sweeper | derp |
| 17:40.59 | Sweeper | you know, you CAN have non-gpl modules for asterisk if you want |
| 17:41.05 | FFForever | would this work?, http://cgi.ebay.com/4-Port-FXO-FXS-Card-Asterisk-VOIP-PCI-IP-PBX-TrixBox_W0QQitemZ120151919398QQihZ002QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 17:41.11 | Sweeper | you just can't DISTRIBUTE them as a bundle :) |
| 17:41.17 | coppice | nothing in zaptel or asterisk compatible with GPL code |
| 17:41.20 | [T]ank | [TK]D-Fender: wasnt pressing digits fast enough.... .5 seconds is the time out. got atxfer working. thank you. |
| 17:41.24 | luke-jr|work | Sweeper: nope |
| 17:41.38 | Sweeper | FFForever: maybe. if you've not broken any mirrors lately, or seen any black cats |
| 17:41.50 | [S]Mercestes | Don't buy from ebay. |
| 17:41.51 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:41.59 | [S]Mercestes | ~cheap |
| 17:41.59 | jbot | somebody said cheap was a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 17:42.03 | AJaymn | Is there a howto for installing zaptel i keep getting an error and finding it needs RTC enabled in the kernel |
| 17:42.32 | Sweeper | AJaymn: maybe you need to recompile your kernel with RTC? |
| 17:42.42 | Sweeper | just, you know, a guess.... |
| 17:42.43 | FFForever | [S]Mercestes, what would u recommend on me getting for a 2line card? |
| 17:42.46 | [S]Mercestes | AJaymn, make menuconfig under /usr/src/linux and /rtc and install it as a kernel module. |
| 17:42.54 | [S]Mercestes | FFForever, 4 line. |
| 17:42.58 | FFForever | from where? |
| 17:43.05 | Sweeper | there's always the ATA route :D |
| 17:43.11 | FFForever | ata route? |
| 17:43.15 | [S]Mercestes | FFForever, Digium. Could check out voip-supply |
| 17:43.18 | luke-jr|work | Sweeper: ATA won't work for incoming landlines |
| 17:43.25 | Sweeper | luke-jr|work: err, sure it will |
| 17:43.25 | AJaymn | [S]Mercestes the link for /usr/src/linux isnt there... |
| 17:43.31 | Sweeper | just need to buy an fxo :P |
| 17:43.35 | [S]Mercestes | AJaymn, aren't you using gentoo or a gentoo variant? |
| 17:43.43 | AJaymn | Fedora core 6 |
| 17:43.48 | [S]Mercestes | =/ |
| 17:43.51 | AJaymn | ya |
| 17:43.54 | Sweeper | bahah |
| 17:43.57 | Sweeper | good luck~ |
| 17:44.02 | [S]Mercestes | ~wglwat |
| 17:44.03 | jbot | well, wglwat is well, good luck with all that |
| 17:44.05 | FFForever | is a landline digital or analog? |
| 17:44.13 | [S]Mercestes | FFForever, I hope it's analog. |
| 17:44.27 | luke-jr|work | FFForever: analog |
| 17:44.44 | AJaymn | depends.. do you say 101011001010 when you talk? heh |
| 17:44.53 | luke-jr|work | :) |
| 17:45.22 | FFForever | witch tdm do i get??? |
| 17:45.31 | [S]Mercestes | 410p I think |
| 17:45.34 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 17:45.38 | [S]Mercestes | but..I'm really bad with the whole numbering thing. |
| 17:45.41 | Sweeper | well |
| 17:45.43 | FFForever | AJaymn, sometimes :), depending on who i am talking to :D |
| 17:45.51 | [S]Mercestes | could be 440p or wahtever. The one with four FXO modules in it |
| 17:45.56 | Sweeper | you get a 400p, with the appropriate modules |
| 17:46.48 | FFForever | modules??? |
| 17:46.56 | Sweeper | yea |
| 17:47.01 | [S]Mercestes | Yea, you can have FXO or FXS, and they are little cards you pop onto a bigger card. |
| 17:47.17 | [S]Mercestes | so if you wanted to have an anlogue testline then you would get a 431p or something. 3 FXO and 1 FXS |
| 17:47.21 | [S]Mercestes | ~fxofxs |
| 17:47.21 | jbot | [fxofxs] 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. |
| 17:47.34 | FFForever | ..., i just want the cheapest......... and the one that will work the best |
| 17:47.41 | [S]Mercestes | FFForever, pick one. |
| 17:48.01 | Sweeper | TOO MANY DECISIONS |
| 17:48.09 | Sweeper | MAKE THEM FOR ME, JBOT |
| 17:48.19 | Sweeper | jbot: cheap or good |
| 17:48.24 | FFForever | witch one of these do i get they seem to have a different number on the online buying page...... |
| 17:48.26 | FFForever | http://www.digium.com/en/wheretobuy/digiumdirect/products.php?category_id=17 |
| 17:48.26 | [S]Mercestes | ~8ball Cheap or good? |
| 17:48.27 | jbot | Yes. |
| 17:48.37 | Sweeper | XD |
| 17:48.40 | [S]Mercestes | Yea. |
| 17:48.46 | [S]Mercestes | ~8ball Am I a sexy fiend? |
| 17:48.47 | jbot | Sure. Yeah, exactly. |
| 17:49.16 | Sweeper | ~8ball will asterisk ever be able to handle over 200 calls? |
| 17:49.17 | jbot | Absolutely. |
| 17:49.30 | [S]Mercestes | I vote TDM04B |
| 17:49.36 | [TK]D-Fender | ~ball Sweeperdoesn't have a clue does he? |
| 17:49.38 | FFForever | what is the difference on fxo and fxs? |
| 17:49.38 | [S]Mercestes | But you probably need a TDM13B |
| 17:49.43 | [TK]D-Fender | ~8ball Sweeperdoesn't have a clue does he? |
| 17:49.43 | jbot | I'm not sure. |
| 17:49.45 | [S]Mercestes | ~fxofxs |
| 17:49.46 | jbot | from memory, fxofxs is 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. |
| 17:50.12 | Sweeper | that's because I'm quantum! |
| 17:50.18 | coppice | ~8ball is George Bush the greatest leader the US has ever known? |
| 17:50.19 | jbot | Unsure. |
| 17:50.27 | FFForever | so i just need fxo ports??? |
| 17:50.36 | luke-jr|work | probably both |
| 17:50.38 | Yourname` | Hello, anyone with any knowledge of sipp, please? |
| 17:50.56 | luke-jr|work | you only need FXS if you want to use an analog phone |
| 17:51.27 | FFForever | nope :) |
| 17:51.33 | [S]Mercestes | Then just FXO |
| 17:53.31 | *** join/#asterisk anthm (n=anthm@mb10736d0.tmodns.net) |
| 17:53.31 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:54.06 | *** join/#asterisk MrMister2 (n=mrmister@195-23-105-142.net.novis.pt) |
| 17:54.08 | Corydon76-vcch | ~8ball is it 5 o'clock somewhere? |
| 17:54.09 | jbot | I'm not sure. |
| 17:54.53 | luke-jr|work | FFForever: so you have an IP phone? |
| 17:55.03 | FFForever | no |
| 17:55.03 | coppice | ~8ball is it 4:54 somwhere? |
| 17:55.04 | jbot | Please ask again. |
| 17:55.14 | luke-jr|work | FFForever: so you don't have any phones at all |
| 17:55.37 | FFForever | i have landlines.... |
| 17:55.42 | luke-jr|work | landlines are lines |
| 17:55.43 | luke-jr|work | not phones |
| 17:55.45 | FFForever | bah stop making my head hurt (LOL) |
| 17:56.03 | FFForever | i have a phone i connect to the line :) |
| 17:56.09 | luke-jr|work | that's an analog phone |
| 17:56.13 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 17:56.15 | luke-jr|work | and you'll need a FXS to connect that to Asterisk |
| 17:56.21 | luke-jr|work | or ATA |
| 17:56.30 | FFForever | i have an extra phone slot :) |
| 17:56.36 | FFForever | my head hurts lol |
| 17:56.39 | [TK]D-Fender | FFForever: Lets rewind everything, this is going in circles. |
| 17:56.41 | luke-jr|work | that won't connect it to asterisk |
| 17:56.47 | [TK]D-Fender | FFForever: What do you want to DO? |
| 17:56.58 | [TK]D-Fender | luke-jr|work: Leave him to me :) |
| 17:57.22 | FFForever | setup a voicemail system and some nice on hold music and a menu for something... |
| 17:58.04 | [TK]D-Fender | FFForever: Ok, you have some analog lines, right? How many? |
| 17:58.19 | FFForever | 2 inbound lines |
| 17:58.24 | luke-jr|work | 3 |
| 17:58.37 | FFForever | 1 for outbound |
| 17:58.44 | [TK]D-Fender | FFForever: Normall lines from your telco aren't just "incoming". A line is a line is a line. Calls go BOTH ways. |
| 17:58.54 | [TK]D-Fender | FFForever: So, 2 analog lines? |
| 17:59.02 | FFForever | 3 but we only use 2 |
| 17:59.12 | [TK]D-Fender | FFForever: What is the 3rd one doing? |
| 17:59.35 | [Airwolf] | Evening. I'm trying to compile asterisk-addons on Openbsd, but I'm getting a few compile error's. I solved them all but one and I was wondering if someone knows the answer. I posted the error and the c code line here: http://pastebin.returning.net/41 |
| 17:59.48 | FFForever | nothing really |
| 17:59.59 | [TK]D-Fender | FFForever: Do you WANT to make use of it? |
| 18:00.27 | MrMister2 | Hi. I'm having trouble getting a perl script to work with Asterisk Call Manager. http://pastebin.ca/659984 Anyone willing to give a hand? |
| 18:00.35 | FFForever | nope |
| 18:01.12 | MrMister2 | I'm a Linux and Asterisk newbie so it's most certainly something _very_ simple but I'm not seeing it :( |
| 18:01.49 | luke-jr|work | FFForever: so you're paying for a line you never want to use? |
| 18:02.01 | FFForever | so we need yeah |
| 18:02.06 | [TK]D-Fender | FFForever: So you just want to leave that line there doing nothing at all? Strange, but OK, lets move on. |
| 18:02.19 | FFForever | my boss is weird |
| 18:02.23 | FFForever | (lol) |
| 18:02.31 | [TK]D-Fender | FFForever: I prefer to think "retarded". |
| 18:02.50 | luke-jr|work | FFForever: now how many analog phones? |
| 18:02.52 | [TK]D-Fender | FFForever: So, moving on. You'd like * to be your PBX and answer calls with an IVR, etc, right? |
| 18:03.03 | *** join/#asterisk tails__ (n=tails@thief.white-box.net) |
| 18:03.05 | [TK]D-Fender | luke-jr|work: hold off on that front. |
| 18:03.08 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-221-219.socal.res.rr.com) |
| 18:03.56 | FFForever | for about 2 lines how much memory should we need? |
| 18:04.17 | [S]Mercestes | Less than you should ahve to worry about. |
| 18:04.18 | [TK]D-Fender | FFForever: forget about that right now.... lets concentrate on what you need to DO. |
| 18:04.25 | [TK]D-Fender | FFForever: that will determine the rest. |
| 18:04.38 | [TK]D-Fender | FFForever: So please confirm my last question. |
| 18:04.48 | FFForever | PBX????, IVR???? |
| 18:05.42 | [TK]D-Fender | FFForever: PBX = phone system. IVR = touch-tone menu to process calls, etc. |
| 18:06.17 | [TK]D-Fender | FFForever: IVR : "Welcome to XYZ, for sales press 1, etc..." |
| 18:08.31 | FFForever | bk |
| 18:08.36 | *** join/#asterisk fiber0pti (i=fiber0pt@216.31.101.41) |
| 18:08.47 | fiber0pti | Does anyone know the dates of the exhibit hall at Astricon this year? |
| 18:09.05 | FFForever | both |
| 18:09.08 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 18:09.11 | FFForever | sorry for the long reply |
| 18:09.22 | FFForever | explaining this stuff to my boss.... |
| 18:09.35 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.169.196) |
| 18:09.41 | blitzrage | tzanger: ping?! |
| 18:10.11 | blitzrage | fiber0pti: you could check astricon.net... should be the middle days of the conference |
| 18:10.40 | fiber0pti | sure.. it's just a little vague to plan airline and hotel tickets around without details |
| 18:11.32 | blitzrage | it's September 24-28 |
| 18:11.53 | blitzrage | exhibits should be the last 3 days |
| 18:12.30 | *** join/#asterisk red9012 (n=marc3234@206-248-134-209.dsl.teksavvy.com) |
| 18:12.30 | MrMister2 | Anyone worked with perl and Asterisk Call Manager and willing to give a hand? It's the simples script you can get, just makes a call, nothing more. |
| 18:12.30 | [TK]D-Fender | FFForever: Ok, How many phones will you be needing? |
| 18:12.48 | red9012 | ok so I need to modify the options offered in voicemailmain app. any way to do this? |
| 18:12.57 | blitzrage | fiber0pti: but for information I'd email lking@sokol-associates.com |
| 18:13.17 | fiber0pti | thanks |
| 18:13.18 | *** join/#asterisk saftsack (n=saftsack@p57A746AC.dip.t-dialin.net) |
| 18:13.28 | red9012 | I only need the voicemail playback, and record voicemail greeting option. Can I remove the other options? |
| 18:13.33 | FFForever | froms powered from the system 0 |
| 18:13.59 | FFForever | phones**** |
| 18:14.19 | FFForever | so we have to get a splitter for the new lines that go to the new system? |
| 18:14.40 | [TK]D-Fender | FFForever: You have flown off on a whacked-out tangent |
| 18:14.52 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 18:15.04 | [TK]D-Fender | FFForever: How many phones will you be wanting to use with this new setup? |
| 18:15.11 | FFForever | 2 |
| 18:15.27 | FFForever | we only use 2lines here.... |
| 18:17.49 | *** join/#asterisk Tako-san (n=Tako-san@24.108.162.254) |
| 18:17.54 | *** join/#asterisk Corydon76-home (i=silver@pdpc/supporter/sustaining/Corydon76-home) |
| 18:17.54 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 18:20.42 | FFForever | [TK]D-Fender, u there? |
| 18:20.58 | [TK]D-Fender | FFForever: Taht was not assumed anywhere. So you have 2 lines and you only need 2 phones? |
| 18:21.14 | [TK]D-Fender | brb |
| 18:21.20 | FFForever | wait i have to plug the phones also into the system??? |
| 18:21.45 | FFForever | and ok |
| 18:25.03 | [S]Mercestes | FFForever, Yes. Exactly how do you expect to gte from the system to the phones? |
| 18:25.10 | FFForever | yeah.... |
| 18:25.16 | FFForever | we just need 1 phone now :) |
| 18:25.23 | FFForever | we are going to get some new cordless phones :D |
| 18:25.45 | [S]Mercestes | .... |
| 18:25.50 | [S]Mercestes | how many phones do you want to end up with? |
| 18:25.57 | FFForever | 1 :) |
| 18:26.10 | [S]Mercestes | Then why do I care if you are giong to get some new cordless phones? |
| 18:26.18 | [S]Mercestes | ...and when you do get them, exactly what do you plan on plugging them into? |
| 18:26.30 | FFForever | the new system i am going 2 build :) |
| 18:26.33 | FFForever | i get it now :) |
| 18:27.07 | FFForever | so should i get a TDM22B? |
| 18:27.08 | [A]Mercestes | It would be best if you planned out your expansion now... |
| 18:27.22 | FFForever | 2 green ports and 2 red ports should work for us right? |
| 18:27.32 | [A]Mercestes | How many total PHONES do you want to have when you are completely done with your system??? |
| 18:27.48 | luke-jr|work | FFForever: you should go IP, IMO |
| 18:28.03 | FFForever | cant....... |
| 18:28.09 | *** join/#asterisk knarfly (n=knarfly@c-98-203-55-196.hsd1.fl.comcast.net) |
| 18:28.24 | FFForever | i am only the techie the boss likes the current setup and i like my paycheck (lol) |
| 18:28.28 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 18:28.47 | knarfly | can anyone help me diagnose my zaptel problem? |
| 18:30.57 | [A]Mercestes | Your current setup doesn't include a PBX |
| 18:32.14 | FFForever | mine? |
| 18:32.19 | [A]Mercestes | FFForever, correct |
| 18:32.24 | *** join/#asterisk techie (n=techie@adsl-68-127-126-144.dsl.frsn02.pacbell.net) |
| 18:33.09 | FFForever | *goes nuts* *lol* what setup should i get then? |
| 18:33.14 | FFForever | just tell me what i need and i will get it :) |
| 18:33.20 | [A]Mercestes | A clue. |
| 18:33.32 | FFForever | basically we just want after the 5th ring go to the new voice mail system |
| 18:33.47 | [A]Mercestes | Then buy an answering machine. |
| 18:33.58 | FFForever | we want on hold music and some other nice things |
| 18:33.59 | FFForever | :) |
| 18:34.08 | sevard | [A]Mercestes: have you ever heard of this? a client is asking me for a feature he said he saw at a blackhat convention in 05, where you speak into the telephone, it does speach to text and "maps your voice", then it does some processing, then you pick up the phone and use their "voice map" (like voice synthesis in text to speach) to speak like them. |
| 18:34.45 | sevard | I know AT&T labs has a product exactly like this, except they use professionals and very high end audio recording equpitment and hours hand mapping |
| 18:34.58 | sevard | but he says he saw an automated system that worked fine with low phone quality audio |
| 18:37.01 | [A]Mercestes | sevard, sounds highly illegal. =/ |
| 18:37.41 | sevard | [A]Mercestes: from my research, it CAN be used illegaly, although it hasn't been. and afaict it's patented. |
| 18:37.50 | [A]Mercestes | sevard, but basically, that's possible to do a voice to text conversion, and then use text to speech and use a "voice" that sounds like someone else. The end user wouldn't be required at that poitn however. |
| 18:38.06 | [A]Mercestes | sevard, doing it on the fly would be difficult at best. |
| 18:38.50 | [A]Mercestes | FFForever, breathe for a second..... |
| 18:39.05 | *** join/#asterisk nephfl (n=no@wsip-68-110-130-57.ga.at.cox.net) |
| 18:39.08 | FFForever | i am :) |
| 18:39.13 | [A]Mercestes | FFForever, How many working phones do you want to have? |
| 18:39.27 | FFForever | 2 |
| 18:39.30 | sevard | [A]Mercestes: it sounds pretty damn tough |
| 18:39.31 | FFForever | thats all :) |
| 18:39.34 | nephfl | i have one extension that is going direct to voicemail saying they are on the phone, when they are not on the phone, anybody know why that would happen? |
| 18:39.37 | [A]Mercestes | FFForever, how many simultanious calls do you want to be able to handle? |
| 18:39.48 | nephfl | polycom voip phones |
| 18:39.51 | FFForever | 2 |
| 18:39.52 | [A]Mercestes | nephfl, Check for Do Not Disturb. |
| 18:40.02 | nephfl | i checked DND on the phone |
| 18:40.25 | [A]Mercestes | FFForever, then you can do it with a 22P, 2 analog PHONES on FXS and two analog LINES on FXO. |
| 18:40.34 | nephfl | fop shows the connection as green |
| 18:40.45 | nephfl | hints shows the connection as free |
| 18:40.46 | FFForever | 22B or 22P? |
| 18:40.47 | [A]Mercestes | FFForever, *OR* you can get some IP phones and run them over ethernet, and just have two FXO ports. |
| 18:40.51 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:41.05 | [A]Mercestes | FFForever, pretty sure P but I don't know for certain. |
| 18:41.09 | nephfl | i have rebooted the server and the phone |
| 18:41.10 | [A]Mercestes | nephfl, can you call out from them? |
| 18:41.14 | nephfl | yes |
| 18:41.23 | FFForever | http://www.digium.com/en/wheretobuy/digiumdirect/products.php?category_id=17 i don't see a 22P on here |
| 18:41.39 | [A]Mercestes | nephfl, pastebin your dialplan for this phone and a nice "sip show peer <peer>" |
| 18:41.58 | [A]Mercestes | FFForever, close enough |
| 18:42.08 | FFForever | ok :) |
| 18:42.59 | FFForever | [A]Mercestes, we plan on getting a new cordless phone system (now) how would that work with the system? |
| 18:43.14 | FFForever | it will have 3 phones 2 the 1base phone |
| 18:43.24 | [A]Mercestes | FFForever, When you get your card....hook a couple wires to your FXS port.... |
| 18:43.25 | FFForever | only the base needs the phone cable |
| 18:43.34 | nephfl | it was working and i havent changed the dialplan and the sip show peer is the same as the others |
| 18:43.41 | [A]Mercestes | FFForever, then wet your ears a little bit with some dilectic grease.... |
| 18:43.48 | [A]Mercestes | FFForever, take the wires, put them to your ears...then call yourself. |
| 18:43.54 | FFForever | LMAO |
| 18:44.13 | [A]Mercestes | Why do you need a new cordless phone system?? |
| 18:44.37 | [A]Mercestes | Describe this Cordless Phone System first... |
| 18:44.39 | FFForever | the old one is dieing :) |
| 18:44.46 | FFForever | and we need a new one anyways |
| 18:44.59 | [A]Mercestes | just describe it. |
| 18:45.08 | [TK]D-Fender | FFForever: FFForever>we are going to get some new cordless phones :D <- |
| 18:45.11 | [A]Mercestes | nephfl, I believe you. Pastebin please. |
| 18:45.14 | FFForever | well the one we are getting comes with 1 base and 2 cradles for charging |
| 18:45.23 | [TK]D-Fender | FFForever: Now you've gone and picked stuff on all sorts of assumptions. |
| 18:45.30 | [A]Mercestes | FFForever, That's not a cordless phone "system" |
| 18:46.01 | [TK]D-Fender | FFForever: those home mult-phone kits are USELESS in your new setup. that CANNOT operate independently. |
| 18:46.12 | FFForever | really? |
| 18:46.16 | [TK]D-Fender | FFForever: And forget about "on-hold" type stuff for the most-part. |
| 18:46.33 | [A]Mercestes | and most other features |
| 18:46.40 | FFForever | why? |
| 18:46.58 | [TK]D-Fender | FFForever: Stupid analog phones can't tell * to put the call on hold. |
| 18:47.05 | FFForever | why would the multi-phone system home kit be useless with the new setup? |
| 18:47.20 | [TK]D-Fender | FFForever: its treated like 1 stupid phone <----- |
| 18:47.30 | [TK]D-Fender | FFForever: Get over it and throw that idea right out the windows |
| 18:47.38 | FFForever | crap... |
| 18:47.47 | [TK]D-Fender | FFForever: they cann't be independant extensions. They were never designed to be. |
| 18:48.19 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 18:49.07 | FFForever | can you recommend some independent cordless phones we can get? |
| 18:49.09 | nephfl | i think it was just the featurecode dnd not the phone dnd |
| 18:49.35 | [A]Mercestes | nephfl, glad I could help |
| 18:49.43 | [TK]D-Fender | FFForever: Just remember that "hold" will not be a practical reality. |
| 18:50.00 | [TK]D-Fender | FFForever: At which point the phone choice is up to you. |
| 18:50.16 | [A]Mercestes | Of course, with only 2 lines, hold is an impracticality anyways. |
| 18:53.16 | FFForever | ok new info :) |
| 18:53.28 | FFForever | when they call it goes straight to a recorded message/menu |
| 18:53.59 | FFForever | 1 for address, 2 for hours, 3 for operator..., when they press 3 it rings the phones :) |
| 18:54.01 | *** join/#asterisk MindTheGap (n=iote@mail.lpj.com.br) |
| 18:55.08 | datachomper | How do I terminate a call from the CLI? |
| 18:55.14 | datachomper | I've got a hung line that I need to kill |
| 18:55.15 | luke-jr|work | FFForever: useless new info |
| 18:55.25 | FFForever | i just need to know how to do it :) |
| 18:55.38 | luke-jr|work | FFForever: it has no effect on your hardware requirements |
| 18:55.46 | luke-jr|work | as for how |
| 18:56.15 | luke-jr|work | 1 => Playback(address); 2 => Playback(hours); 3 => Dial(...); |
| 18:56.26 | FFForever | dial??? |
| 18:56.36 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 18:57.16 | AJaymn | would i have any issues installing Astersik on CentOS 4.3 ? |
| 18:57.22 | [TK]D-Fender | FFForever: Before doing anything else you clearly have no clue about * at all. Go downlaod and read THE BOOK. |
| 18:57.24 | [TK]D-Fender | ~book |
| 18:57.25 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:57.27 | hmmhesays | you shouldn't |
| 18:57.32 | De_Mon | FFForever is this for zap or sip/zip |
| 18:57.34 | FFForever | what book? |
| 18:57.38 | [TK]D-Fender | FFForever: Then download and install * and play around with a soft-phone for a bit. |
| 18:57.41 | Sweeper | AJaymn: there's a wiki page about it |
| 18:57.44 | [TK]D-Fender | FFForever: Linked above |
| 18:57.45 | De_Mon | oops |
| 18:57.49 | FFForever | soft-phone??? |
| 18:57.51 | luke-jr|work | [TK]D-Fender: imo, the book sucks :) |
| 18:57.53 | Sweeper | AJaymn: http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+CentOS |
| 18:57.54 | De_Mon | datachomper zap or sip/iax? |
| 18:58.05 | [TK]D-Fender | FFForever: used skype before? |
| 18:58.24 | [TK]D-Fender | FFForever: or MSN's voice chat? |
| 18:58.28 | Sweeper | luke-jr|work: it's evangelical more than anything, I find it very appropriate that people call it "The Book" :D |
| 18:58.37 | luke-jr|work | Sweeper: it doesn't use AEL |
| 18:58.45 | De_Mon | ahahahahaha |
| 18:58.47 | Sweeper | AEL is nasty :P |
| 18:58.51 | luke-jr|work | AEL > .conf |
| 18:58.59 | FFForever | yeah :) |
| 18:58.59 | datachomper | De_Mon sip |
| 18:59.01 | FFForever | skype |
| 18:59.02 | Sweeper | adhearsion > AEL > .conf :) |
| 18:59.07 | FFForever | skype can call asterisk? |
| 18:59.09 | De_Mon | datachomper soft hangup <tab> |
| 18:59.09 | luke-jr|work | wtf is adhearsion? |
| 18:59.17 | luke-jr|work | FFForever: no, Skype sucks too much for that |
| 18:59.17 | Sweeper | http://adhearsion.com/ |
| 18:59.19 | elixer | its when something sticks to something else |
| 18:59.24 | datachomper | De_Mon Thanks bro |
| 18:59.25 | [TK]D-Fender | FFForever: Well Skype is a softphone using its own service and protocol. Same thing though. |
| 18:59.31 | De_Mon | datachomper for zap its destroy... something |
| 18:59.39 | [TK]D-Fender | FFForever: Lets you use your computer + headset as a phone for testing & learning. |
| 18:59.41 | FFForever | is there any free soft-phone that can call the asterisk? |
| 18:59.47 | De_Mon | my skype uses a normal telephone |
| 18:59.47 | Sweeper | FFForever: lots |
| 19:00.00 | [TK]D-Fender | FFForever: http://www.zoiper.com/ |
| 19:00.20 | *** join/#asterisk sysreq (n=sysreq@197.36-ppp.3menatwork.com) |
| 19:00.33 | [TK]D-Fender | FFForever: there. Now you need to get a CLUE. Fo download everything and get to work. You aren't ready to even introduce this to your boss right now. |
| 19:00.53 | FFForever | he introduced it to me.... |
| 19:00.54 | *** join/#asterisk pejo_ (n=peter@136.240.13.217.in-addr.dgcsystems.net) |
| 19:00.55 | luke-jr|work | FFForever: you at least know how to use Linux, right? |
| 19:00.58 | luke-jr|work | LOL |
| 19:01.03 | FFForever | what is linux? |
| 19:01.08 | FFForever | ;) |
| 19:01.11 | luke-jr|work | ............ |
| 19:01.13 | FFForever | j/k i am on ubuntu :D |
| 19:01.26 | [TK]D-Fender | FFForever: Oh... so you STILL have no clue then little GUI boy... |
| 19:01.42 | FFForever | i love the terminal :D |
| 19:05.02 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:05.55 | FFForever | [TK]D-Fender, do i have to read all the pages...., BAH really long book (LOL) |
| 19:06.28 | [TK]D-Fender | FFForever: Don't start whining now, there's plenty more to come. |
| 19:06.50 | FFForever | *falls over....* |
| 19:06.59 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 19:07.40 | Deeewayne | FFForever: read the book. Its free, easy to read, and will answer almost all the questions you've been asking |
| 19:08.49 | [TK]D-Fender | FFForever: If you are looking for a complete hand-holding seminar on how to deploy *, my rates are very accessable.... |
| 19:09.19 | hmmhesays | haha |
| 19:09.20 | Sweeper | FFForever: mine are probably more so than his ;) |
| 19:09.58 | FFForever | [TK]D-Fender, what are ur rates?, anyone else would also like to give rates (fixed price no hourly bs) |
| 19:10.11 | Sweeper | fixed price? ha |
| 19:10.19 | Sweeper | not even I'm that desperate :P |
| 19:10.27 | *** join/#asterisk saftsack (n=saftsack@pD9E079C3.dip.t-dialin.net) |
| 19:10.30 | [TK]D-Fender | Stupidity is LIMITLESS, why should rates be FIXED? :) |
| 19:10.58 | FFForever | because i am not rich and this biz does not make really any money |
| 19:11.25 | [TK]D-Fender | FFForever: Then just stick with what you've got. Apparently it got you this far. |
| 19:15.00 | FFForever | where can i find a local asterisk contractor?, in the los angeles area |
| 19:15.08 | datachomper | How do I turn OFF sip debug? |
| 19:16.08 | elixer | sip set no debug? |
| 19:16.42 | elixer | or 'sip no debug' for 1.2 |
| 19:17.05 | datachomper | sip set debug off, for 1.4, thanks |
| 19:17.10 | elixer | ah |
| 19:17.12 | elixer | i was close |
| 19:17.36 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 19:17.36 | elixer | next time you could try first and then ask |
| 19:17.41 | elixer | just a thought. |
| 19:18.43 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 19:20.35 | JerJer | so who else has beat their head against the wall with NAT on a OpenSER+Asterisk solution |
| 19:20.49 | JerJer | we do not want to use mediaproxy, since our calls are being dealt with by asterisk |
| 19:20.52 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 19:21.04 | JerJer | ...for this particular solution |
| 19:21.09 | datachomper | I did, but thanks |
| 19:21.14 | anonymouz666 | JerJer: just don't use force the rtp proxy then |
| 19:21.36 | JerJer | we aren't - and we get one way audio on some NAT configurations |
| 19:21.56 | JerJer | personally i run mediaproxy and deal with it, but these guys want no mediaproxy |
| 19:22.27 | *** join/#asterisk dharrigan (n=dharriga@82-71-62-76.dsl.in-addr.zen.co.uk) |
| 19:22.28 | blitzrage | JerJer: I tell the other end to get better NAT devices... mine works perfectly fine with a Linksys or iptables based solution |
| 19:22.36 | blitzrage | and I'm not using mediaproxy |
| 19:22.45 | blitzrage | Sonicwall's don't work though |
| 19:22.58 | JerJer | this is like vmware |
| 19:23.04 | JerJer | based crap |
| 19:23.16 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:23.55 | JerJer | unfortunately - they have s morbid desire to run vmware instances with some custom soft-client (for a strange call center type situation) |
| 19:24.11 | [TK]D-Fender | blitzrage: I run my * behind a SonicWALL just fine.... |
| 19:24.29 | blitzrage | [TK]D-Fender: I'm talking about in an OpenSER+Asterisk combination |
| 19:24.41 | blitzrage | [TK]D-Fender: if you'd read the scrollback :) |
| 19:25.09 | [TK]D-Fender | blitzrage: You and your silly sense of context! |
| 19:25.27 | blitzrage | and even without OpenSER I had absolutely no luck with SonicWall's |
| 19:25.40 | blitzrage | never have a problem with my network here running a simple linksys device though |
| 19:25.56 | blitzrage | people tell me they run sonicwall and want me to help them, I refuse to waste my time with them |
| 19:28.21 | *** join/#asterisk timtim (n=conor@91.186.21.88) |
| 19:28.39 | *** join/#asterisk MdeP (n=mdep@200.124.36.28) |
| 19:36.43 | JerJer | so like um - this sucks |
| 19:37.16 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 19:37.46 | Yourname` | Oh man, sipp is sooo what I need.. and I can't get to make it work :( |
| 19:40.02 | JerJer | blitzrage: so what if we can't just drop the nat solution ? |
| 19:40.25 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 19:41.40 | JerJer | i wonder what mediaproxy does that asterisk cannot do - since we have mediaproxy (the python script addon to openser) on different ip addresses and audio is totally solid |
| 19:43.21 | *** join/#asterisk jsaunders (n=super@70.70.0.33) |
| 19:43.46 | kombi | colleague went to the loo, I take her call <- What is that feature called? |
| 19:45.32 | kombi | anyone? |
| 19:46.41 | *** join/#asterisk minkus (n=minkus@pool-72-84-46-134.clrkwv.east.verizon.net) |
| 19:47.24 | [TK]D-Fender | kombi: "loo" ? |
| 19:47.27 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 19:47.37 | kombi | ..bathroom |
| 19:48.15 | kombi | I mean some phone rings and someone on another desk can take the call |
| 19:48.25 | *** join/#asterisk BitBandit (n=PX2@168-103-185-124.slkc.qwest.net) |
| 19:48.27 | [TK]D-Fender | kombi: looking for "call pickup" |
| 19:48.36 | [TK]D-Fender | kombi: "pickup groups" |
| 19:48.37 | kombi | THANKS! |
| 19:48.45 | [TK]D-Fender | kombi: Get wiki-ing :) |
| 19:49.04 | kombi | what do you press on the phone to do that? |
| 19:49.57 | [TK]D-Fender | kombi: *8 It hink. Its in "dynamic features" IIRC |
| 19:50.03 | [TK]D-Fender | kombi: Never used itpersonally. |
| 19:50.26 | kombi | they seem to want it (damn) |
| 19:52.37 | kombi | and my other question for today: In a setup that is all sip, how does one fax? |
| 19:53.07 | [TK]D-Fender | kombi: LOL.... |
| 19:53.43 | kombi | ..;) something I always wondered but never dared to ask.. |
| 19:54.18 | kombi | stick an analog card in and hook up the old fax machine to it? |
| 19:54.38 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:57.10 | luke-jr|work | grr, companies annoy me |
| 19:57.16 | luke-jr|work | always looking to make money instead of doing things rihgt |
| 19:57.33 | luke-jr|work | even when doing it right would work just as well |
| 19:58.58 | jsaunders | Any asterisk/networking guru's in the Vancouver, BC, Canada area willing to assist a 25 staff company w/ our issues? |
| 20:01.07 | *** join/#asterisk Ebola (n=Ebola@host86-138-7-229.range86-138.btcentralplus.com) |
| 20:01.54 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 20:02.13 | BitBandit | how many calls should i expect to have at one time with a 7meg down and a 896k up DSL connection ? |
| 20:03.17 | kombi | BitBandit: 896 / codec bandwidth |
| 20:04.16 | BitBandit | whats a good codec to use? |
| 20:04.29 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 20:04.53 | [TK]D-Fender | BitBandit: 896 / channel encapsulation protocol( # channels * codec bandwidth) |
| 20:04.57 | *** join/#asterisk guillote_GNU (n=bancaria@host222.201-253-44.telecom.net.ar) |
| 20:05.06 | [TK]D-Fender | BitBandit: Describe your NEEDS first |
| 20:05.20 | BitBandit | right now i have the G726 and if i get 4 calls going it starts breaking up and i need ot have more calls at a time than that |
| 20:05.49 | [TK]D-Fender | BitBandit: GSM is far lighter, but the real killer is RTP overhead if you're talking SIP. |
| 20:06.12 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 20:06.14 | *** join/#asterisk Strom_M (n=strom@24-240-128-76.dhcp.davl.vt.charter.com) |
| 20:06.36 | BitBandit | i have my phones are sip to the asterisk box then IAX2 from my box to the provider |
| 20:07.35 | [TK]D-Fender | BitBandit: try switching to GSM & SIP then. Sometimes IAX does "go to shit" so to speak. |
| 20:08.11 | BitBandit | k i will give that a shot and see how it goes from there, thanks |
| 20:08.20 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:09.03 | BitBandit | will i just need ot open port 5060 up on my router for sip or is there anything else i should open up |
| 20:09.23 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 20:11.31 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 20:13.08 | [TK]D-Fender | BitBandit: here |
| 20:13.10 | [TK]D-Fender | ~sipnat |
| 20:13.17 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:13.38 | BitBandit | thank you |
| 20:14.24 | Yourname` | Hi, if I see INVITEs in sip debug.. but nothing on CLI, but I want to.. like extension 20 is getting a cal.. and I don't, what could be wrong? |
| 20:17.02 | [TK]D-Fender | Yourname`: dialplan , sip peer setup or your client. |
| 20:17.21 | [TK]D-Fender | Yourname`: sip debug reveals all. get pastebin-ing |
| 20:17.45 | Yourname` | lol |
| 20:18.01 | Yourname` | [TK]D-Fender: One sec, I just found something.. thanks to the Defender mantra |
| 20:18.27 | *** join/#asterisk lbow (n=lbow@41-195-77-226.access.uunet.co.za) |
| 20:21.07 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 20:21.21 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 20:22.49 | Yourname` | [TK]D-Fender: You. Rock!! Got it, thank you so much! |
| 20:23.50 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 20:24.02 | coolbeans | Hey guys, what do you see wrong with this? I'm just trying to loop 3 times: exten => s,n,Set(LOOP=${MATH(${LOOP}+1)}) exten => s,n,GotoIf($[${LOOP}>3]?5) |
| 20:24.32 | Mercestes | Your gotoif logic appears wrong. |
| 20:25.09 | Mercestes | shouldn't it be GotoIf($[${LOOP}>3]?5:1) ? |
| 20:25.26 | Mercestes | and you should be using s,1 not s,n if you want to *goto* that priority |
| 20:26.03 | Mercestes | the way it's written it will pass yoru loop once, LOOP will = 1, gotoIF will find it less than three, so it will *not* go to five and proceed on to the next priority under your GotoIf because you have no false condition. |
| 20:26.09 | blitzrage | coolbeans: Set(LOOP=$[${LOOP} + 1]) is better |
| 20:26.21 | Mercestes | that too |
| 20:26.29 | blitzrage | and what Mercestes said :) |
| 20:26.33 | coolbeans | The goto is right, it's something with initially setting the variables ... |
| 20:26.44 | *** join/#asterisk Olobola (n=chatzill@74.95.13.57) |
| 20:26.45 | coolbeans | blitzrage: I'll give that a shot... tnx |
| 20:26.58 | blitzrage | oh wait, the GotoIf() is fine -- it's LOOP greater than 3, not less than |
| 20:27.05 | Yourname` | Hi, so what's that command that shows managers being logged in on the CLI? show managers doesnt work :S |
| 20:27.18 | blitzrage | although what he's trying to loop through is not being shown |
| 20:27.52 | Mercestes | Yourname`, try core show managers. =/ Just a case. |
| 20:27.59 | [TK]D-Fender | xten => s,n,Set(LOOP=$[${LOOP}+1]) |
| 20:28.19 | Yourname` | Mercestes: No workie. I just want to see which managers are logged in. thats all |
| 20:28.22 | coolbeans | Do I have to declare LOOP before I can evaluate it? |
| 20:28.30 | blitzrage | [TK]D-Fender: ya, thanks for keeping up :) |
| 20:28.37 | blitzrage | coolbeans: yes |
| 20:28.44 | blitzrage | Set(LOOP=1) |
| 20:28.45 | coolbeans | Ahh.. that's probably my problem, then. |
| 20:28.55 | blitzrage | might not be -- but it's best to in my experience |
| 20:29.06 | [TK]D-Fender | blitzrage: its time to bring on,.... the ILLOGICAL operators! |
| 20:29.09 | blitzrage | or else you get a null value, and asterisk doesn't like that too much :) |
| 20:29.14 | [TK]D-Fender | blitzrage: X = MAYBE Y |
| 20:29.15 | [TK]D-Fender | ! |
| 20:29.35 | blitzrage | and I prefer using priority labels over priority numbers (which are way outdated) |
| 20:29.45 | Olobola | this is odd: trying to grab a variable from lumenvox through agi/php... the var will appear ONLY after calling get_variable('SPEECH_TEXT(0)'); three times! $agi->exec('NoOp ' . $text['data']); |
| 20:31.08 | coolbeans | Got it! Thanks, all. |
| 20:31.27 | Yourname` | Mercestes: Found it! It's show manager connected |
| 20:31.35 | Mercestes | ah, cool |
| 20:31.39 | Mercestes | Yourname`, try <tab> sometime. |
| 20:31.43 | Mercestes | Yourname`, it's cool. |
| 20:31.52 | [TK]D-Fender | ok, heading home, later all |
| 20:31.57 | Mercestes | l8s |
| 20:31.58 | Yourname` | Mercestes: Oh it is!!! Thanks man! |
| 20:32.00 | Yourname` | bye [tk |
| 20:32.50 | *** join/#asterisk Blackthorn (i=blacktho@76.77.160.10) |
| 20:33.20 | Yourname` | Oh, man.. this <tab> is like a boon to mankind, lol.. especially the new man. |
| 20:34.31 | Blackthorn | Hello, setup a replacement * server a while ago and we are just getting around to moving the new pri to it. The zttool shows the line is ok, zttcfg shows good, and doing a lsmod | gre zaptel shows up fine too. but there is no zap menu in *. thoughts? |
| 20:35.08 | Sweeper | Blackthorn: check modules.conf to see if you're loading chan_zap |
| 20:36.19 | Blackthorn | the only thing st to noload is chan_alsa.so |
| 20:37.05 | Sweeper | so you type zap show channels, and it says it doesn't know about the command |
| 20:37.06 | Sweeper | ? |
| 20:37.06 | Blackthorn | were is the module directory? so i can just check to see if chan_zap exists in there |
| 20:37.20 | Sweeper | dunno |
| 20:37.28 | Blackthorn | yes sweeper, thats correct. thers no zap in the menu |
| 20:37.30 | JerJer | WTF - i feel like a jackhole |
| 20:37.44 | JerJer | a fucking nat=yes solved that audio problem - go finger |
| 20:37.48 | sheppard | does anyone have experience hooking a nortel PBX that uses a PRI up to a * box? |
| 20:37.58 | Mercestes | sheppard, lots of experienec. |
| 20:37.59 | Sweeper | sheppard: yes, actually |
| 20:38.04 | *** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 20:38.13 | sheppard | did your existing equipment migrate nicely? |
| 20:38.23 | Mercestes | sheppard, I also have experience with prison dates on death row... |
| 20:38.25 | JerJer | define nicely |
| 20:38.33 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:38.33 | Sweeper | for as long as it took me to rip it out and replace it, sure |
| 20:38.33 | Mercestes | sheppard, between the two......I think I prefered the prison dates |
| 20:38.35 | sheppard | because the PBX should still handle all the phone sets correct? and the * box only routes calls to the real world right? |
| 20:38.37 | Sweeper | about 4 weeks |
| 20:38.41 | sheppard | Mercestes: lol |
| 20:38.51 | Sweeper | sheppard: yea |
| 20:39.04 | Sweeper | I put * inline between bell's pri and the option11 |
| 20:39.11 | Mercestes | sheppard, in theory....yea....but in theory, democracy works too |
| 20:39.14 | Blackthorn | "/usr/lib/asterisk/modules" is were the modules reside btw :) |
| 20:39.17 | sheppard | hahahaha |
| 20:39.38 | sheppard | Hrm |
| 20:39.41 | Mercestes | Blackthorn, give modules.conf a nice load => chan_zap.so |
| 20:39.52 | sheppard | fight with our pbx, or spend the cash to move to a bunch of sip capable handsets |
| 20:39.55 | Mercestes | Blackthorn, maybe you have to be implicit. |
| 20:40.00 | Blackthorn | i don't see chan_zap.so in the directory |
| 20:40.14 | Blackthorn | so i proiblby need to do some make/isntall etc etc |
| 20:40.17 | Mercestes | Blackthorn, did you modprobe your card driver and zaptel? |
| 20:40.32 | Mercestes | Blackthorn, Also, install zaptel src first, then asterisk. |
| 20:40.41 | Blackthorn | yea the card is found fine. it shows up in zttool and is configured |
| 20:40.43 | Mercestes | sheppard, move to a bunch of sip capable handsets. |
| 20:40.55 | Mercestes | Blackthorn, try to give asterisk a recompile then. |
| 20:41.03 | sheppard | Mercestes: yeah but we have a fancy board for reception that I'm not sure * will be able to replicate |
| 20:41.06 | Mercestes | Blackthorn, maybe it didn't see zaptel so therefore didn't install it. |
| 20:41.12 | Mercestes | sheppard, FoP |
| 20:41.16 | sheppard | FoP? |
| 20:41.22 | Mercestes | sheppard, flash operator panel |
| 20:41.26 | sheppard | that's it! |
| 20:41.29 | Mercestes | sheppard, and a polycom 601 with a couple of expansion modules. |
| 20:41.29 | sheppard | s/it/them/ |
| 20:41.32 | Blackthorn | merc: thats what i'm thinking too since * been running but only just put the card in a few days ago |
| 20:41.34 | *** join/#asterisk _Krieger_ (n=warsword@91.102.176.47) |
| 20:41.40 | Mercestes | sheppard, works with asterisk then. |
| 20:41.58 | Mercestes | Blackthorn, yea, likely your problem then. |
| 20:42.10 | _Krieger_ | which frequencies of wav files cat asterisk play? |
| 20:42.34 | _Krieger_ | because i got format_wav.c:156 check_header: Unexpected freqency 22050 |
| 20:42.35 | Mercestes | the normal frequencies. |
| 20:42.50 | Mercestes | Um...google wiki asterisk ....one sec. |
| 20:42.50 | sheppard | Mercestes: like, they a) allow you to transfer calls to other people by just pushing the button that relates to their name, as well as showing you whos on the phone and who isn't |
| 20:43.08 | Mercestes | _Krieger_, http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 20:43.37 | Mercestes | sheppard, Sorta.. On the polycom yo uhave to hit transfer, then the button next to their name, then transfer again. |
| 20:43.57 | sheppard | ok that's not a huge issue, but the important thing is seeing whos on a call and who isn't |
| 20:44.07 | Mercestes | The lights *are* there though and show status and who is on a call. |
| 20:44.13 | sheppard | ah nice |
| 20:44.15 | Mercestes | It also interfaces with Jabber |
| 20:44.29 | sheppard | just need one that'll handle ~50 internal lines and 10 external |
| 20:44.41 | Mercestes | 48 limit if I recall. |
| 20:44.50 | Mercestes | Minus 1 for the secretary herself though. |
| 20:44.59 | Mercestes | so 49 is pretty close to 48. |
| 20:44.59 | Blackthorn | whats the * wiki address? |
| 20:45.13 | Mercestes | http://www.google.com wiki asterisk |
| 20:45.14 | Mercestes | ~wikis |
| 20:45.15 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 20:45.23 | tzanger | blitzrage: pong |
| 20:45.28 | Blackthorn | yea voip=info was what i was looking for |
| 20:46.39 | sheppard | Mercestes: lol |
| 20:46.48 | JerJer | tzanger: word |
| 20:46.53 | sheppard | Mercestes: thanks for your time |
| 20:47.49 | Mercestes | sheppard, your welcome |
| 20:49.10 | tzanger | JerJer: hey man |
| 20:49.16 | tzanger | you're the h323 guy I've been looking for |
| 20:49.28 | tzanger | what's the stack you use? |
| 20:50.08 | JerJer | i haven't touched h323 in years - but OpenH.323 |
| 20:50.31 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 20:51.42 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:54.03 | BitBandit | what are the most common causes for audio on incoming calls to only be one way, they can here me but i cant hear them |
| 20:54.17 | JerJer | fucking nat |
| 20:54.22 | elixer | heh |
| 20:54.32 | elixer | what he said |
| 20:54.36 | JerJer | put nat=yes in [general] and leave it there |
| 20:54.40 | anonymouz666 | haha |
| 20:54.43 | anonymouz666 | JerJer rlz |
| 20:54.46 | JerJer | grr |
| 20:54.56 | JerJer | that just totally slapped me in the face |
| 20:54.56 | elixer | JerJer: having a good day? |
| 20:55.36 | MindTheGap | hello all, im having a problem w chanspy. as soon as its started on a channel, asterisk drops the call and it beeps forever. I wonder if it requires canreinvite=no on clients... |
| 20:55.42 | [TK]D-Fender | BitBandit, Here : |
| 20:55.43 | JerJer | my own systems have always used mediaproxy, so i simply don't have one-way audio issues |
| 20:55.44 | [TK]D-Fender | ~sipnat |
| 20:55.45 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:56.57 | JerJer | at the expense of more CPU.....these guys did not want mediaproxy in their config.... which is what complicated things.... there was nat=yes on the peer to the openser proxy, but it needs to be in [general] to asterisk does the right thing for everyone else who isn't a peer but asterisk is sending audio to them |
| 20:57.06 | JerJer | i think this is a blog entry in the making actually :D |
| 20:59.22 | [TK]D-Fender | JerJer, only needs to be in general if * is itself behind NAT and need to know what externIP to send |
| 21:00.18 | JerJer | not correct |
| 21:00.50 | JerJer | situation: |
| 21:00.51 | Sweeper | he just doesn't want to type it in for every user :P |
| 21:00.56 | JerJer | phones registering to openser |
| 21:01.00 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 21:01.19 | JerJer | openser sends call to asterisk |
| 21:01.26 | JerJer | asterisk proxy's RTP (canreinvite=no) |
| 21:01.40 | JerJer | if that phone is behind nat, asterisk must have nat=yes in [general] |
| 21:01.54 | JerJer | if not we can duplicate one-way audio all day (ask about 4000 sales agents) |
| 21:02.29 | anonymouz666 | yes that is right. Since the call arrive in default [general] the nat=yes should be there. |
| 21:02.31 | JerJer | i cannot explain why, but it works |
| 21:02.40 | anonymouz666 | or am I wrong? |
| 21:02.46 | JerJer | anonymouz666: actually no - this is just media |
| 21:02.51 | JerJer | we have a peer for the openser proxy |
| 21:02.55 | anonymouz666 | hmm |
| 21:03.30 | JerJer | totally frustrating |
| 21:04.27 | JerJer | but hell if this works I may drop mediaproxy on my other configs |
| 21:04.44 | JerJer | would free up some damn CPU time |
| 21:04.46 | [TK]D-Fender | JerJer, Oh yeah... to the PROXY it'd be "no".... stupid NAT! |
| 21:05.13 | JerJer | correct - the proxy is one switch port away, same class C network |
| 21:05.19 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 21:05.34 | anonymouz666 | JerJer: well, you don't need two programas handling NAT |
| 21:06.21 | anonymouz666 | I use it rtpproxy on Openser and Asterisk as services provider |
| 21:06.34 | anonymouz666 | avpops is nice |
| 21:07.01 | JerJer | i liked mediaproxy for its proxydispatcher and ability to do accounting (broken, but ya know) |
| 21:07.57 | Mercestes | Can someone from digium send me some TDM400p brackets? |
| 21:08.04 | JerJer | oh yes - avpops is totally killer |
| 21:08.53 | *** join/#asterisk n00dle (n=ccraft@204.10.248.123) |
| 21:09.56 | n00dle | Anyone here use TDMoE and have trouble unloading/reloading the ztd_eth module? (Claims to be in use, use count is non-zero, but no modules are listed using it!) |
| 21:13.08 | De_Mon | I tried to setup a phone -> Nat Router -> NAT router -> internet -> asterisk. it wouldn't work... is that normal? |
| 21:13.23 | Mercestes | De_Mon, yes... |
| 21:13.38 | *** join/#asterisk Rahail (n=rahail@c-68-43-176-199.hsd1.mi.comcast.net) |
| 21:13.43 | Rahail | any one around |
| 21:13.51 | Mercestes | no |
| 21:14.03 | Dan0maN_Work | 263 people here, someone's bound to answer ;) |
| 21:14.13 | Mercestes | 273. |
| 21:14.17 | De_Mon | Mercestes ok, thats what I though ;) |
| 21:14.18 | Rahail | :) |
| 21:14.18 | Mercestes | 10 of them are ops |
| 21:14.21 | Dan0maN_Work | ah |
| 21:14.23 | Dan0maN_Work | my bad ;) |
| 21:14.24 | Rahail | ~ pastebing |
| 21:14.30 | Rahail | ? pastebin |
| 21:14.33 | Rahail | ~ pastebin |
| 21:14.33 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 21:14.34 | De_Mon | didn't realize it was double natted until this morning eh eh |
| 21:14.44 | *** join/#asterisk kkn088 (n=kikoun@88-138-6-52.adslgp.cegetel.net) |
| 21:15.20 | Rahail | I am trying to log in throw pap2 |
| 21:15.26 | Rahail | but i am getting fast buzzy |
| 21:15.27 | Rahail | http://pastebin.ca/660263 |
| 21:15.40 | Mercestes | how old are you? |
| 21:15.54 | Rahail | who |
| 21:15.56 | Rahail | me |
| 21:15.58 | De_Mon | 1050 |
| 21:15.59 | Mercestes | yes |
| 21:16.03 | Rahail | 21 |
| 21:16.05 | Mercestes | 1050? wow, that's old |
| 21:16.18 | De_Mon | Not really, I was reincarnated recently |
| 21:16.53 | Mercestes | your sip debug isn't real helpful, Rahail. Exactly what are you trying to do again? |
| 21:17.05 | n00dle | anyone else know how to get ztd_eth to unload? |
| 21:17.30 | De_Mon | what does the term 'split' mean on the polycom? it seems like it puts everyone on hold... |
| 21:17.33 | n00dle | ...besides rebooting. |
| 21:17.51 | Mercestes | De_Mon, it splits out a single call into two calls if you ahve them conferenced together. |
| 21:18.07 | Mercestes | De_Mon, in essence, it does put everyone on hold, but you can answer each one of them individually |
| 21:18.13 | De_Mon | hrm, the only choice I have for a split call is "resume" |
| 21:18.19 | tzanger | ugh, looks like hte head gaskets in my minivan have gone (again) |
| 21:18.24 | De_Mon | maybe I'm not looking hard enough |
| 21:18.26 | Mercestes | De_Mon, on *EACH* leg of the call |
| 21:18.35 | tzanger | there's no way I'm paying $1300 to have them fixed, I may have a go at it myself though... that'd be a bit of a chore :-) |
| 21:18.36 | [TK]D-Fender | De_Mon, split = AWESOME |
| 21:19.11 | [TK]D-Fender | De_Mon, after you SPLIT a 3-way call you can recombine into conference or treat them as seperate calls. |
| 21:19.25 | Mercestes | tzanger, It's easy.....you just remove the valve cover bolts....then the rocker arms and pins, and the springs, and the bolts, then you unhook all your air pressure hoses, and your air vaccume hoses, and your AC lines... |
| 21:19.40 | De_Mon | ahhh I see |
| 21:19.45 | Mercestes | tzanger, then you remove all of your head bolts.....and then you spend an hour and a half hammering at it trying to get it to break loose... |
| 21:19.47 | De_Mon | I didn't notice the 1/2 bit |
| 21:19.59 | Mercestes | tzanger, then you get mad...and crank it over a nice half a time and launch your head out of your hood. |
| 21:20.00 | De_Mon | I like the big clock on the front ;) |
| 21:20.07 | [TK]D-Fender | De_Mon, Split = feature I've only seen on Polycom. |
| 21:20.10 | tzanger | Mercestes: yeah ... while I feel I'm up for the challenge, I am not so sure I'll get it back together right :-) |
| 21:20.31 | tzanger | Mercestes: sounds like you speak from experience |
| 21:20.35 | Mercestes | >.> |
| 21:20.41 | De_Mon | don't like not being able to see CIDName and CIDNumber on missed calls without extra navigation tho. The linksys does a 2line view that's sexy |
| 21:20.43 | tzanger | turn the engine over when it's half disassembled?? I'd have never thought of htat |
| 21:21.02 | Mercestes | I couldn't get hte gasget to break loose. |
| 21:21.19 | Mercestes | compression works wonders |
| 21:21.19 | Rahail | Mercestes i am trying to make call from pap2 |
| 21:21.36 | Mercestes | Rahail, pastebin the cli output verbose 10 if your call attempt. |
| 21:21.43 | tzanger | heh I'd have expected it to launch clear out the engine compartment |
| 21:21.51 | Mercestes | it did. |
| 21:22.02 | Mercestes | well, mostly. |
| 21:22.03 | tzanger | hahaha |
| 21:22.10 | Mercestes | it kinda popped and then flopped over. |
| 21:22.30 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 21:22.39 | Mercestes | you over head your engine a few times and that gasget makes a *real* good seal. |
| 21:22.46 | Rahail | ok |
| 21:22.49 | Rahail | let me try and see |
| 21:23.12 | tzanger | there was a goodwill warranty on these things but I'm 260k km and 9 years old, I don't think it'll apply |
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| 21:26.19 | Rahail | Mercestes i got it work |
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| 21:26.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:26.22 | Rahail | but no Audio |
| 21:26.23 | tzanger | Mercestes: I'm looking at this now, as $1300+ isn't worth it on this vehicle: http://www.heal-a-seal.com/ |
| 21:26.25 | Rahail | :( |
| 21:26.40 | tzanger | this page though reads like the timecube dude |
| 21:26.42 | phoneproblems | I have a voicemail box with over 100 messages I need to clear out. Is there a simple command that will erase all messages in a voicemail box? |
| 21:27.58 | [TK]D-Fender | phoneproblems, "rm -rf /var/spool/asterisk/voicemail/[context]/[boxnumber]/INBOX/ |
| 21:28.00 | n00dle | phoneproblems, rm -f /var/spool/asterisk/voicemail/{VMCONTEXT}/{BOXNUMBER}/INBOX/* |
| 21:29.33 | phoneproblems | thanks! |
| 21:31.14 | n00dle | Anyone know how to get module ztd_eth to unload? I have tried to figure out what's holding it in use, but no luck. (rmmod, ifconfig ethX down, shutting down asterisk, ztcfg -s) |
| 21:31.31 | n00dle | ...in no particular order. |
| 21:31.36 | Mercestes | tzanger, I love how he quotes certain words like, "instantly" and "Small fortune" and "fix." |
| 21:31.48 | tzanger | yep |
| 21:32.07 | tzanger | I'll spend $100 to fix and get rid of the vehicle, but $1300... not so sure |
| 21:32.28 | tzanger | although |
| 21:32.42 | tzanger | since my head gaskets were already replaced once, I wonder if the bolts have just worked loose |
| 21:32.44 | tzanger | I should go check that |
| 21:36.56 | n00dle | Ok, if no one knows here, any referrals to linux lkm experts? |
| 21:36.58 | tzafrir_laptop | n00dle, as nobody picket that up yet, just my wild guess: ztcfg -s ? |
| 21:37.08 | n00dle | tzafrir_laptop, did that. |
| 21:37.31 | tzafrir_laptop | anybody using that span? |
| 21:37.39 | tzafrir_laptop | lsof /dev/zap |
| 21:37.40 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 21:37.41 | n00dle | Nope. I had shut * down. |
| 21:37.59 | tzafrir_laptop | e.g: ztmonitor or something |
| 21:38.14 | magic_hat | anyone run into problems upgrading from 1.2 to 1.4? or should I go ahead and compile away? |
| 21:40.24 | Mercestes | tzanger, http://www.complaints.com/2007/may/2/Sales_Heal-A-Seal.com__144265.htm |
| 21:47.49 | Mercestes | tzanger, http://www.complaints.com/directory/2005/july/25/32.htm |
| 21:49.52 | Mercestes | Never trust a salesman that uses quotation marks when he talks. |
| 21:52.23 | *** part/#asterisk anthm (n=anthm@adsl-76-229-157-60.dsl.milwwi.sbcglobal.net) |
| 21:53.09 | Qwell[] | Mercestes: more importantly - never trust a website that uses yellow |
| 21:56.09 | n00dle | Ach... maybe I'm not understanding TDMoE well enough yet, but I'm still getting RED in 'zap show status' |
| 22:01.19 | magic_hat | so i just upgraded from 1.2 to 1.4 and am now getting this: No application 'ResponseTimeout' for extension (greeter, s, 4). Help? |
| 22:01.25 | tzanger | I'm a tdmoe expert :-) |
| 22:01.38 | tzanger | you must have one side (clock 0) with a valid zap interface for timing |
| 22:02.20 | tzanger | the other side (clock > 0) does not need zap timing and takes it from the tdmoe span |
| 22:02.31 | tzanger | it really is that simple |
| 22:04.07 | tzanger | so, say you had a t1 you wanted to shoot over to another PC you could not put the t1 card in |
| 22:04.40 | tzanger | you have a computer with the t1 which you configure with a 24-span tdmoe interface with the mac address of the OTHER computer and a clock of 0 |
| 22:05.11 | tzanger | and the "destination" computer (the one with no hardware) you declare a 24-channel tdmoe span with the "source" comptuer's MAC and a clock of '1' |
| 22:07.37 | n00dle | tzanger, Thanks... it must be the timing parameter. Back after a reboot. (Damned module won't unload) |
| 22:08.18 | tzanger | ztcfg -s |
| 22:08.22 | tzanger | you must stop the span to get it to unload |
| 22:08.30 | n00dle | Tried that. |
| 22:08.34 | tzanger | and if asterisk is running you must stop it before you try ztcfg -s |
| 22:08.36 | tzanger | oh |
| 22:08.39 | n00dle | That too. |
| 22:08.45 | tzanger | if you change zaptel.conf BEFORE you ztcfg -s it won't stop it |
| 22:08.50 | tzanger | because you have a different span now, technically |
| 22:08.58 | Mercestes | Qwell[]: or bright orange. |
| 22:09.03 | n00dle | Heh. Ok. |
| 22:09.11 | Mercestes | Hrm? |
| 22:09.13 | Mercestes | ....oh yea... |
| 22:09.17 | tzanger | ztcfg -s looks at zaptel.conf to figure out which spans to stop, so if you alter the file before ztcfg -s it will say "oh, THAT span is already stopped" |
| 22:09.25 | Mercestes | s/orange/magenta/ |
| 22:09.26 | Mercestes | >.> |
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| 22:09.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:13.25 | fujin | sweet |
| 22:13.30 | fujin | got 128 concurrent calls on my * system |
| 22:13.35 | fujin | awesomeness. |
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| 22:16.21 | hmmhesays | bah ticketmasters phone contact is awful |
| 22:17.15 | n00dle | tzanger, thanks for the help, still have RED... I'll have to look at the patches to zaptel more closely to see if they broke anything. (Or try my config against unpatched first, then...) |
| 22:17.22 | n00dle | Ciao! |
| 22:21.04 | Rahail | is there way u can increase voice sound in pap2 |
| 22:21.07 | Rahail | or asterisk |
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| 22:32.45 | dlynes | Rahail: output gain |
| 22:33.29 | dlynes | Rahail: output gain is adjustable on the pap2, i believe |
| 22:33.54 | dlynes | Rahail: asterisk has no control over volume for sip channels; only zaptel channels...and again....there, it's output gain |
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| 22:41.27 | asterisknerds | <PROTECTED> |
| 22:47.26 | fujin | who maintains voip-info.org? |
| 22:47.35 | fujin | their register page is shite. Doesn't understand greylisting. |
| 22:47.42 | JT | incompetant fools |
| 22:47.46 | JT | i mean comm partners |
| 22:48.15 | fujin | Was just going to update some of the pages with stuff I've found to be useful |
| 22:48.24 | fujin | seems to be the only place to find publicly available info apart from the digium mailing list |
| 22:48.26 | fujin | and here |
| 22:48.39 | JT | and the book |
| 22:48.39 | dlynes | fujin: www.asteriskdocs.org, too |
| 22:48.52 | dlynes | fujin: if you want an asterisk-specific site, that is |
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| 22:51.15 | sopo2k4 | does anyone know how to start an extension via the Asterisk Manager? |
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| 22:51.42 | CoffeeKid | anyone ever have success using the PlayDTMF command using the manager? |
| 22:52.10 | CoffeeKid | sopo2k4, what do you mean by start an extension? |
| 22:53.48 | fujin | like, dial an extension? |
| 22:54.00 | sopo2k4 | nvm |
| 22:54.19 | sopo2k4 | any ideas on generating a random 5 digit number? |
| 22:54.27 | CoffeeKid | using what language? |
| 22:54.30 | sopo2k4 | english |
| 22:54.45 | JT | yes because computers love being programmed in english |
| 22:54.45 | CoffeeKid | in the dialplan? |
| 22:55.02 | sopo2k4 | yes |
| 22:55.06 | sopo2k4 | extensions.conf :/ |
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| 22:56.18 | tzanger | how about ${RAND()} |
| 22:56.24 | asterisknerds | <PROTECTED> |
| 22:56.48 | tzanger | or rahter RAND(10000,99999) |
| 22:57.24 | Rahail | dlynes |
| 22:59.37 | fujin | sdon't think rand() is a valid application |
| 22:59.47 | dlynes | Rahail: rahail! |
| 23:01.15 | minkus | RAND() is a function |
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| 23:02.30 | *** join/#asterisk VOiCi (n=o@132-199.sh.cgocable.ca) |
| 23:02.46 | VOiCi | hey, how do i ring a ringgroup ? |
| 23:04.33 | *** join/#asterisk sysreq (n=sysreq@modemcable238.182-131-66.mc.videotron.ca) |
| 23:10.42 | dlynes | VOiCi: Usually *8#, unless you've overridden it in features.conf |
| 23:11.03 | VOiCi | i mean, as a command |
| 23:11.09 | VOiCi | dlynes |
| 23:11.25 | VOiCi | i want to have a vocal menu, in which one option is to ring group 1 |
| 23:13.29 | dlynes | VOiCi: ah |
| 23:14.09 | dlynes | VOiCi: One method I believe is Dial(Local/*8#), but I might be off on that...one sec, and I'll get the proper way |
| 23:15.08 | dlynes | VOiCi: erm...nvm...that would pick up the ring group, not ring it |
| 23:15.19 | VOiCi | ow ok |
| 23:15.38 | VOiCi | thanks anyways for trying to help, much appreciated |
| 23:16.37 | dlynes | VOiCi: if you want to do that, what I would do, is define a local extension (option A) or a dialplan context (option B) to ring a group of phones, with option B being the preferred method |
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| 23:19.30 | VOiCi | ill do my best, mixing CLI and webUI aint the best |
| 23:19.59 | dlynes | VOiCi: your webui doesn't give you a way to do it? |
| 23:20.12 | fujin | VOiCi: |
| 23:20.16 | fujin | make a queue, in queues.conf |
| 23:20.20 | fujin | and then statically assign members to it |
| 23:20.49 | fujin | and then when you wan tthe quueue to ring, just put Queue(blah|tn||300); in your dialplan, where 300=number of seconds before going to next priority |
| 23:20.53 | *** part/#asterisk HockeyInJune (n=HockeyIn@pool-141-155-139-103.ny5030.east.verizon.net) |
| 23:20.58 | VOiCi | yeah it does dlynes, but generated content is far from my kind of scripting |
| 23:21.38 | VOiCi | wouldnt that ring them one by one? 300 seconds between each? |
| 23:29.44 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
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| 23:31.20 | Cyon | So anyone have any thoughts on having a cisco IAD interface with asterisk as SIP extensions for each of the pots lines? |
| 23:31.32 | Cyon | I can't seem to find anything good through google, though I'm obviously still searching |
| 23:33.36 | JT | Cyon: what does the device do? |
| 23:35.46 | *** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com) |
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| 23:43.49 | Cyon | JT: An IAD? It's Cisco's low-end voice specific router |
| 23:44.07 | Cyon | JT: http://www.cisco.com/warp/public/cc/pd/as/ia2400/index.shtml |
| 23:44.12 | JT | i doubt that's its full model number |
| 23:45.06 | MaliutaWrk | IAD??? is than Improvised Audio Device??? wouldn't they be bad ... kind like and "IED" |
| 23:45.08 | Cyon | JT: All the IADs have the exact same functionality, the only difference is the number of pots lines (FXOs/FXSs) it has, or if it's got an extra T1/E1/ISDN PRI interface. |
| 23:45.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:45.25 | JT | MaliutaWrk: Integrated Access Device |
| 23:45.54 | JT | Cyon: that's great, but i asked you what *your* device does |
| 23:46.11 | Cyon | JT: And the full model of the test unit I'm using is....Cisco IAD-2431-1T1E1 and IAD2431-16FXS |
| 23:46.37 | JT | 2 units? |
| 23:47.10 | x86 | what is TDMoE used for? |
| 23:47.20 | Cyon | JT: Ummm, it has an amphenol plug on the back of it that I could plug into phones; or it has a PRI interface that I can plug into anything that can take a PRI. And I'm trying to set it up so either the pots lines, or the pri channels can register as individual sip devices; I think I can see it with mgcp, but I prefer sip. |
| 23:47.25 | JT | nothing, usually, x86 |
| 23:47.54 | Cyon | JT: I have the two units for testing, yes. |
| 23:48.25 | x86 | JT: what can be done with it? |
| 23:48.26 | JT | so if it has sip firmware, you should be able to get it to work |
| 23:48.50 | JT | x86: it emulates running TDM over an Ethernet connection |
| 23:49.04 | JT | with a protocol digium invented |
| 23:50.08 | Cyon | JT: Yes, it does, but what I'm trying to figure out is how. I can have it make calls which are directed to asterisk with the session target; and have calls coming from asterisk to the IAD as sip as well -- but asterisk isn't able to check the status of the individual extensions; such as show hints obviously fails miserably, and I couldn't set it up to be dynamic registrations such as if the unit were to be brought to a customer location I couldn't just plug |
| 23:50.42 | Cyon | Those are the things I'm trying to figure out; and usually I can find sample configs either from cisco or from asterisk; but in this case there seems to be a surprising lack....I was thinking maybe someone here had done it already, or had a good reference I could look over. |
| 23:50.55 | fujin | get smartnet on your IAD's, log a fault with tacacs |
| 23:50.57 | MaliutaWrk | JT: I preffer my expansion of the TLA :) |
| 23:51.07 | fujin | I'm sure they'll be able to provide you with an explanation/configuration for sip |
| 23:51.34 | Cyon | fujin: Yeah, I could; but that's just lame. :-P |
| 23:52.01 | JT | Cyon: that was almost impossible to read |
| 23:52.06 | JT | run on sentences, yeah... |
| 23:52.14 | Cyon | fujin: And actually, I'm not positive they will -- I think they consider assistance in getting it to work on that level of detail with asterisk as outside tac's responsibility. |
| 23:53.23 | Cyon | JT: Sorry |
| 23:53.45 | fujin | uh, no |
| 23:53.52 | Cyon|Away | fujin: They will? |
| 23:53.56 | fujin | had a similar issue with our as5400, no one knew how to drive it |
| 23:53.57 | fujin | got smartnet |
| 23:53.59 | JT | Cyon|Away: yeah, and the ircd cuts off lines that are too long, without warning |
| 23:54.01 | fujin | and had them help us with the configs |
| 23:54.05 | Cyon|Away | fujin: Sweet |
| 23:54.10 | fujin | no issue whatsoever :) |
| 23:54.17 | fujin | I'm not sure about IAD's, though :) |
| 23:54.33 | Cyon|Away | fujin: I expect it's the same, nice, I have a last-resort then. :) |
| 23:55.18 | Cyon|Away | fujin: I just hate to give in, I've gotten everything else from 2600s, 2800s, 7200s, 7600s etc all interfacing exactly as I need with voice; but the IAD is the first time I'm putting a cisco device in on the client side. *sigh* |
| 23:55.22 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 23:55.28 | fujin | they've been very good about not asking us what we have connected to the as5400, at all. |
| 23:55.30 | Cyon|Away | fujin: Thanks for the advice; I'll take it if I have to. :) |
| 23:55.42 | fujin | they just assume that we're using some sort of SIP compliant hardware/software |
| 23:55.51 | Cyon|Away | fujin: hehe Nice. |
| 23:56.02 | fujin | although I hear from our account manager that TAC will (soon) be no longer supporting SIP products that aren't Cisco |
| 23:56.04 | fujin | although this may be heresy |
| 23:56.23 | Cyon|Away | fujin: Hum, might need to move fast then if I'm going to. I'll look into that too. |
| 23:56.32 | JT | if there's anyone you can bet on being arseholes, it's cisco :) |
| 23:56.51 | fujin | aint that the truth |
| 23:56.55 | Cyon|Away | Oh well, I'm off climbing, bbl. Thanks |
| 23:57.00 | fujin | toodles |
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