| 00:02.35 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:06.47 | unpaidbill | what the heck... what does hasvoicemail = no have to do with the way a user is dialed |
| 00:07.14 | unpaidbill | if i have hasvoicemail = no in the user config, it dials without my Dial arguments (twk), if i comment that out it dials fine? |
| 00:08.26 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 00:08.31 | [TK]D-Fender | BBIAB |
| 00:09.51 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 00:17.22 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 00:17.34 | grandpapadot | Hi all. Have any of you seen this: copy_header: No field 'Call-ID' present to copy |
| 00:17.48 | grandpapadot | It just started on one of our systems and just repeats over and over. |
| 00:19.38 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 00:20.00 | SteveTotaro | i assume you are not getting a sipcallid |
| 00:20.27 | SteveTotaro | has anything changed? |
| 00:21.23 | *** join/#asterisk edibrac (n=edibrac3@netblock-66-245-252-161.dslextreme.com) |
| 00:21.54 | edibrac | after a Dial(), i can execute AGI() right? it doesn't seem to run, at least from my asterisk -rvvvvvvvvvv |
| 00:22.00 | edibrac | messages |
| 00:22.39 | edibrac | IOW, this is what I mean: http://pastebin.com/m4185fb62 |
| 00:25.37 | grandpapadot | SteveTotaro: Nope. Very odd. |
| 00:26.33 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 00:26.38 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-9531ea948f096576) |
| 00:26.41 | grandpapadot | hrm... hung channel maybe? |
| 00:27.02 | drmessano | ~wizard |
| 00:27.03 | jbot | i heard wizard is enchancement to howto's |
| 00:27.05 | drmessano | ~wizard! |
| 00:27.06 | jbot | i heard wizard is enchancement to howto's |
| 00:27.12 | drmessano | crap |
| 00:27.47 | unpaidbill | i agree |
| 00:27.56 | unpaidbill | this problem is driving me nuts |
| 00:28.05 | drmessano | WIZARD! |
| 00:28.40 | edibrac | grandpapadot: hung channel for what I'm talking about? |
| 00:29.55 | grandpapadot | No, my problem, sorry. |
| 00:31.51 | drmessano | Have you checked the channels? |
| 00:32.26 | grandpapadot | drmessano: Me? Yea, there's like 160 calls right now, three of them are over 90 minutes long. |
| 00:33.07 | drmessano | Hmm |
| 00:33.22 | *** join/#asterisk jyap (n=jyap@66.135.230.101) |
| 00:33.30 | grandpapadot | It's consistent, about every 10 seconds it pops up again. Never seen this before -- ever. |
| 00:34.26 | grandpapadot | It doesn't seem to be affecting anything - just odd. |
| 00:35.23 | jyap | i seem to be having issues with iax2 on the asterisk 1.4.19. core show locks is similar to this: http://www.pastebin.ca/995767 |
| 00:35.49 | fmueller | edibrac, can you paste the messages that are written to the cli please? |
| 00:36.20 | grandpapadot | Well, Hans Reiser is guilty.. He basically fracked himself ... |
| 00:36.31 | tzanger | I haven't read everything on it yet |
| 00:36.43 | tzanger | but there better be some damned good reaosns there, circumstancial evidence is not evidence |
| 00:36.49 | grandpapadot | Hit /. about an hour ago |
| 00:37.12 | jyap | this is the lock message: Tried and failed to get Lock #0 (channel.c): MUTEX 4591 ast_channel_trylock &chan->lock 0x883c6e0 (0) |
| 00:37.14 | grandpapadot | Apparently, he made an ass of himself and didn't give the jurors any reason to like him. |
| 00:37.30 | tzanger | agreed |
| 00:37.35 | tzanger | but it isn't a popularity contest |
| 00:37.54 | lmadsen | anyone have a website that you can go to, type in a SIP URI, and have it call you? |
| 00:37.58 | tzanger | he could be the biggest asshole in the world |
| 00:38.01 | lmadsen | I want to test from an external source my SIP URI |
| 00:38.07 | rob0 | Not liking him is one thing, but convicting without hard evidence and a viable if improbable story ... |
| 00:38.07 | tzanger | but if he's not guilty, then he's not guilty |
| 00:38.09 | lmadsen | tzanger: he couldn't be... you're still alive |
| 00:38.11 | tzanger | it's not illegal to be a dick |
| 00:38.16 | tzanger | lmadsen: damn |
| 00:38.21 | grandpapadot | I think it's a travesty. Yea, I agree it isn't a popularity contest -- but these are just people and innocent until guilty might look good on paper but in reality it's guilty until proven innocent in this country at least ... |
| 00:38.22 | tzanger | lmadsen: come here so I can kick your ass |
| 00:38.28 | lmadsen | tzanger: pfffffffft |
| 00:38.50 | tzanger | grandpapadot: yeah |
| 00:39.25 | jyap | does anyone know if my dead lock is a recent issue? or anything about it? |
| 00:40.09 | tzanger | asterisk? deadlocks? |
| 00:40.13 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584477.dsl.bell.ca) |
| 00:40.43 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 00:40.45 | rob0 | he may well be guilty, nobody but him really knows. But it's scary that the gov't can get a conviction without presenting a prima facie case. |
| 00:40.58 | tzanger | rob0: that's what I'm concerned about |
| 00:41.12 | tzanger | rob0: kind of ruins my plans with my wife... |
| 00:41.17 | rob0 | ;) |
| 00:41.52 | tzanger | I gotta say though.. the circumstantial evidence against him *is* pretty fucking weird |
| 00:41.59 | rob0 | indeed |
| 00:42.08 | jyap | yes, asterisk and deadlocks |
| 00:42.35 | rob0 | Reiser is guilty of the asterisk deadlocks, beyond a reasonable doubt. |
| 00:42.40 | tzanger | haha |
| 00:42.50 | tzanger | well there was definitely a nina deadlock |
| 00:42.55 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 00:42.55 | tzanger | she was quite pretty, too |
| 00:43.10 | rob0 | unless she ran off to Russia, which is quite possible |
| 00:43.16 | tzanger | I gotta wonder about the kids, though... I can't imagine what they're going through |
| 00:43.25 | jyap | or is there another channel i post my question? |
| 00:43.34 | rob0 | #reiserfs ? |
| 00:44.11 | jyap | anyone running reiserfs? i hear it's killer |
| 00:44.15 | grandpapadot | The entire situation is weird .. |
| 00:44.48 | rob0 | jyap, this is Linux? What distro? |
| 00:44.50 | unpaidbill | is macro-stdexten special in any way? is it automatically invoked if hasvoicemail = yes is specified i nthe users configuration or is this something asterisknow is doing on its own? |
| 00:45.18 | unpaidbill | i really want to punch myself in the face for using asterisknow :P but client wants pretty interface what can i do.. |
| 00:45.18 | jyap | yes, running linux on centos. |
| 00:45.37 | [hC] | asterisk-gui has a ways to go yet. |
| 00:46.01 | rob0 | aha, it could be SELinux then |
| 00:46.08 | jyap | selinux is disabled |
| 00:46.39 | unpaidbill | yea asterisk-gui should not be version 1.0.2, it's more like .1.0.2 :P |
| 00:48.38 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 00:48.41 | fmueller | when i call * via isdn and make asterisk to playback a speex encoded audio file does * translates speex to g.711 in that case? |
| 00:49.12 | [TK]D-Fender | fmueller, yes |
| 00:49.14 | *** join/#asterisk rcy` (n=rcy@S010600207810532a.vc.shawcable.net) |
| 00:50.42 | fmueller | i am always getting this when i try to playback the speex file: ast_streamfile: Unable to open quant (format 0x8 (alaw)): No such file or directory |
| 00:51.07 | fmueller | i have no problems playing back a wav file. |
| 00:51.42 | fmueller | this is how i encode the wav file to speex: speexenc -n --quality 8 quant.wav quant.spx |
| 00:52.10 | fmueller | does anybody know why asterisk complains with the message mentioned above? |
| 00:52.28 | grandpapadot | fmueller: file permssions? |
| 00:52.44 | fmueller | probably, i am going to check. |
| 00:53.52 | fmueller | no, permissions are rwxrwxrwx now. |
| 00:54.41 | Damin | fmueller: Looks like you are converting the .wav to a .spx file, but your channel is Alaw.. |
| 00:55.09 | fmueller | grandpapadot, what i also do not understand is that asterisk mentiones alaw |
| 00:55.12 | Damin | fmueller: Does your Asterisk have speex? |
| 00:55.30 | fmueller | yes, i think so. i have the compiled module in /var/lib/asterisk/modules. |
| 00:55.34 | Damin | fmueller: When you do a "show codecs" what does it show? |
| 00:56.05 | *** join/#asterisk RobH (n=RobH@dsl017-048-227.sfo4.dsl.speakeasy.net) |
| 00:56.57 | fmueller | Damin, Well, first of all that command does not seem to tell anything usefull about the current configuration since it outputs this: |
| 00:56.58 | fmueller | Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. |
| 00:57.20 | [TK]D-Fender | fmueller, If you have a choice, don't encode to speex |
| 00:57.30 | fmueller | Damin, but i am happy to post it to pastebin. |
| 00:57.55 | Damin | fmueller: 1.4? Or 1.2? |
| 00:58.03 | fmueller | http://pastebin.com/m4e5e307c |
| 00:58.04 | fmueller | 1.4 |
| 00:58.08 | Damin | fmueller: The commands got mucked around w/ in 1.4.. |
| 00:58.13 | fmueller | Asterisk 1.4.19 |
| 00:58.16 | Damin | fmueller: try "show translation" |
| 00:58.29 | Damin | Look to see if you have a translation path between speex and alaw. |
| 00:58.52 | fmueller | Damin, show translation: http://pastebin.com/m5c9288b6 |
| 00:59.06 | Damin | fmueller: Doesn't really do me any good. I'm on a VT-100 terminal right now.. |
| 00:59.15 | fmueller | if i get it right i have it. |
| 00:59.36 | fmueller | Damin, ah, ok |
| 00:59.46 | Damin | speex on X axis, alaw on Y.. where they intersect should be a number... |
| 00:59.54 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 01:00.03 | fmueller | Damin, yes. exactly that is the case in output. |
| 01:00.10 | fmueller | Damin, the number is 16. |
| 01:00.31 | Damin | ok.. so Asterisk knows about speex and knows about alaw and how to transcode between them.. |
| 01:00.48 | Damin | fmueller: Somewhere there is a command that shows you the mappings of the file extensions to the codec.. |
| 01:01.07 | Damin | fmueller: been a while since I looked at it, but perhaps you have the file extension misnamed for speex? |
| 01:01.11 | fmueller | core show config mappings |
| 01:01.42 | fmueller | Damin, if you mean that command i can tell you that i get no output from it. |
| 01:01.52 | vader-- | so is 1.2.28 the last build of the 1.2 line? |
| 01:02.00 | vader-- | they are now working on 1.4 and 1.6? |
| 01:02.00 | Damin | fmueller: Could be it.. |
| 01:02.14 | fmueller | Damin, more precisely i get three empty lines ... |
| 01:02.58 | Damin | fmueller: That is about all i can offer to you now.. |
| 01:03.27 | Damin | fmueller: For some reason, Asterisk wants a .alaw file for the channel. |
| 01:03.52 | fmueller | Damin, ok, thanx so far. i will continue to look at this mapping command and why i get no output from it. |
| 01:05.02 | fmueller | Damin, well, that * wants alaw does not surprise me since i am in germany and alaw is standard (in europe) for isdn network as far as i know. |
| 01:05.43 | Damin | fmueller: Sure.. try converting your .spx file to an alaw file. |
| 01:05.53 | fmueller | Damin, i also have no problem in playing back alaw. |
| 01:05.53 | fmueller | <PROTECTED> |
| 01:05.57 | Damin | fmueller: I generally transcode all of my stuff into all the different formats.. |
| 01:06.07 | Damin | fmueller: Avoids having to transcode on the fly. |
| 01:07.00 | fmueller | Damin, the reason for speex is this: i am not satisfied by the audio quality. if i get it right the sample rate of the file must be 8000Hz. |
| 01:07.24 | fmueller | Damin, I would like to have at least 16000Hz. |
| 01:08.07 | Damin | fmueller: Well something is messed up in your transcoding configuration I guess.. |
| 01:08.10 | fmueller | therefore i though: "ok, i convert my 16kHz wav file to speex and let asterisk playback the speex file." |
| 01:10.04 | lmadsen | vader--: correct -- the only changes to 1.2.x is for security issues |
| 01:10.05 | fmueller | Damin, ok, than i will investigate further in that direction. but i am right that asterisk cannot playback audiofiles with a sample rate more than 8kHz, am i not? |
| 01:10.46 | fmueller | Damin, (at least if i use this alaw thing ...) |
| 01:11.34 | Damin | fmueller: Alaw and Ulaw are both 8kHz yes.. I believe that if you play it back at 16khz it will play 2x as slow. :) |
| 01:12.51 | fmueller | Damin, hehe ... well, i guess that i have to try what´s better :-) the problem with 8kHz is that i cannot really hear a difference between "s" and "f". |
| 01:12.54 | Damin | fmueller: Well.. if you play a 16khz file at 8khz that is.. |
| 01:13.34 | Damin | fmueller: Interesting.. I don't have a problem running straight ulaw here in the states. are you using text to speech or something? |
| 01:14.14 | fmueller | Damin, that is exactly what i am doing! :-) |
| 01:14.45 | Damin | fmueller: Try a different speech engine? :) |
| 01:15.30 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 01:15.35 | fmueller | Damin, i have the speech from the speech engine in a wav file. in the wav file it sounds ok. but i am not able to get the same quality via the phone. |
| 01:15.39 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:15.54 | *** join/#asterisk Nasra (n=Nasra@CPE001839494bc9-CM00111ade9528.cpe.net.cable.rogers.com) |
| 01:16.32 | *** join/#asterisk gormux (n=julien@asuka.genux.info) |
| 01:16.32 | fmueller | In the mailinglists i found a posting from Steve Underwood (i guess you know him). in that posting he wrote: |
| 01:16.35 | fmueller | "There certainly are real benefits. With 16kHz sampling you can tell "s" from "f". That is almost impossible with 8kHz sampling." |
| 01:16.39 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 01:17.27 | fmueller | not so easy ... |
| 01:18.15 | *** join/#asterisk ftp3 (n=none@pool-71-117-212-7.ptldor.fios.verizon.net) |
| 01:18.15 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 01:18.15 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 01:19.15 | *** join/#asterisk ixx (i=foobar@cpe-70-112-149-89.austin.res.rr.com) |
| 01:19.43 | ftp3 | hello :-) is it possible to use asterisk with skype in so that someone can call my skype number and asterisk will take the call and do as I wish with it? |
| 01:21.33 | ftp3 | google showed me: Skype-to-Asterisk(SIP) which has no new info since 2006.. so I suppose that did not get finished |
| 01:21.50 | JayTee52 | not directly AFAIK |
| 01:23.49 | *** part/#asterisk tclineks (n=tclineks@adsl-75-39-141-211.dsl.tpkaks.sbcglobal.net) |
| 01:25.01 | ftp3 | ok :-) thank you |
| 01:26.21 | RobH | anyone have a good vendor for telecom stuff? I want to put a one button intercom at our front door that uses a standard rj11 connection to the FXS module in my * box |
| 01:26.32 | jjshoe | rob0 tons of those on ebay |
| 01:26.44 | jjshoe | rob0 and google in general, I mentioned a viking one in here the other day |
| 01:26.45 | jjshoe | about $150 |
| 01:27.13 | RobH | cool, just looking for some kinda company that folks have used, (not sure what I would call the intercom thing, heh.) |
| 01:27.14 | RobH | thank yas |
| 01:27.25 | jjshoe | i of course found it in the article on doorbell on voip-info |
| 01:27.37 | RobH | viking was the one that someone else told me about, i lost the bookmark |
| 01:27.40 | RobH | thank you =] |
| 01:27.56 | rob0 | And the Viking women wear those steel bras. |
| 01:28.04 | rob0 | Way cooler than pirates. |
| 01:28.38 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 01:30.02 | RobH | ha |
| 01:30.09 | RobH | well, damn viking doesnt have pricing =[ |
| 01:30.11 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 01:33.37 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 01:41.06 | *** join/#asterisk BeeBuu (n=beebuu@219.130.253.189) |
| 01:42.27 | *** join/#asterisk auslandr (n=auslande@20.ptr.cs.rogers.wave.ca) |
| 01:43.42 | auslandr | can anyone offer any advice as to why /dev/zap/pseudo would be painfully slow w/ ztdummy? |
| 01:44.00 | auslandr | like, results of -200% in zttest slow |
| 01:44.37 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 01:44.37 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 01:45.09 | [TK]D-Fender | auslandr, Check your kernel timing |
| 01:45.19 | [TK]D-Fender | auslandr, needs to be set for 1000hz |
| 01:47.22 | auslandr | Running a stock kernel w/ asterisknow |
| 01:47.27 | auslandr | i assume it would be correctly compiled. |
| 01:47.55 | auslandr | is there a place i can doublecheck in /proc or sysctl or something? |
| 01:48.07 | [TK]D-Fender | auslandr, pastebin dmesg |
| 01:49.16 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-245-162.balt.east.verizon.net) |
| 01:49.25 | auslandr | http://pastebin.com/da0ad8a9 |
| 01:51.53 | [TK]D-Fender | auslandr, ok, "cat /proc/interrupts |
| 01:52.41 | auslandr | http://pastebin.com/d5dac5d7 |
| 01:59.02 | [TK]D-Fender | auslandr, Ok, not sure what to make of it yet, rerun "ztcfg -vvvv" then re-run zttest |
| 01:59.42 | auslandr | 5 cycles of zttest ok? |
| 02:01.35 | auslandr | http://pastebin.com/d67d3daa9 did 10, just for good measure |
| 02:01.39 | auslandr | http://pastebin.com/d67d3daa9 |
| 02:01.43 | auslandr | whoops. |
| 02:12.24 | *** join/#asterisk golumn (n=golumn@190.6.194.118) |
| 02:14.29 | *** part/#asterisk ftp3 (n=none@pool-71-117-212-7.ptldor.fios.verizon.net) |
| 02:18.24 | golumn | in what format do I need to have the files to be reproduce by Background() and do I need special software for that? |
| 02:18.32 | *** part/#asterisk fmueller (n=user@p548F718D.dip.t-dialin.net) |
| 02:19.04 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 02:19.27 | BeeBuu | sox can be help |
| 02:20.04 | golumn | thanks |
| 02:30.09 | [hC] | huh. is it normal for network traffic to increase when enabling iax2 trunking, comparing the bandwidth usage for a single call against a single call going through an iax2 trunk? |
| 02:33.11 | *** join/#asterisk thx2000 (n=bob@netblock-208-127-150-56.dslextreme.com) |
| 02:34.24 | thx2000 | Having issues getting outgoing zaptel to work, incoming works great, but outgoing gets "Everyone is busy/congested at this time" ... any ideas? |
| 02:35.46 | jyap | thx2000, are you dialing the correct number of digits for your provider? |
| 02:35.57 | thx2000 | yea |
| 02:36.25 | jyap | is it every call or just certain numbers? has it ever worked? |
| 02:36.53 | thx2000 | every call, never...its a new install |
| 02:37.42 | jyap | try and run a debug with 'core set verbose 9' and 'agi debug' if you are using agi. might be clues there |
| 02:39.44 | thx2000 | Been doin a bit of that...its pretty ambiguous. The calls just instantly fail. If I choose a channel that doesn't have a line connected to it, the call won't fail instantly leaving me to believe it has something to do w/ the signaling, otherwise it's just an instant fail |
| 02:41.56 | JayTee52 | is it a PRI or FXO? |
| 02:42.32 | thx2000 | TDM2402 |
| 02:43.12 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 02:43.14 | [TK]D-Fender | thx2000, pastebin the complete CLI output of your failed call at verbose 10 |
| 02:43.15 | [TK]D-Fender | ~pb |
| 02:43.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:43.16 | [TK]D-Fender | ^^^^^^^^^ |
| 02:43.24 | thx2000 | kk |
| 02:45.07 | thx2000 | http://pastebin.ca/1001291 |
| 02:45.26 | thx2000 | want the .conf's too? |
| 02:50.02 | golumn | I am trying to build an ivr, but don't know which functions will be usefull. I want play a gsm file telling th options, 1 for this, 2 for that, etc. an wait for the user to push a button, what functions can I use for this? |
| 02:51.00 | *** join/#asterisk Yosam (n=No@S0106001c109e98db.no.shawcable.net) |
| 02:51.04 | Yosam | Hello guys |
| 02:51.14 | thx2000 | golumn: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Background |
| 02:51.21 | BeeBuu | ~book |
| 02:51.21 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 02:51.28 | Yosam | what is the best opensource speech recognition software that works with asterisk |
| 02:51.36 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 02:51.44 | *** join/#asterisk TJNII (n=TJNII@209.234.89.237) |
| 02:51.50 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 02:52.43 | golumn | so, I only need to name the extend with the digit I want extend=>1,.... for option 1 and so on |
| 02:52.50 | thx2000 | yup |
| 02:53.28 | golumn | didn't know that, does Background() has a timeout? |
| 02:54.54 | Yosam | no idea ?? |
| 02:55.36 | Yosam | i want to use Sphinx but i heard it does not work with * because it does not have 8khz files |
| 02:55.58 | thx2000 | Waitexten for timeout length, then exten => t,1,... for what to do on a timeout |
| 02:58.32 | thx2000 | [TK]D-Fender, any ideas? |
| 02:58.34 | JayTee52 | Yosam, only speech recognition software I know that works with * is LumenVox and it's not open source and not free |
| 02:59.50 | thx2000 | golumn: BeeBuu's right though, "The Future of Telephony" is a great read and a great way to get started |
| 03:00.31 | golumn | thanks, will start reading it right now |
| 03:01.38 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 03:02.10 | Yosam | I am creating a non-commercial application and sorry but i would not pay for lumenvox i know it is good i worked with it but |
| 03:02.11 | Yosam | no |
| 03:04.42 | JayTee52 | FYI on the book, the examples for an IAX2 trunk on pages 112, 113 and 114 have the contexts wrong. In the iax.conf they are incoming_osaka and incoming_toronto and in the extensions.conf examples they are osaka_incoming and toronto_incoming so they don't match properly. |
| 03:05.04 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 03:05.08 | JayTee52 | I already emailed errata@oreilly.com about it. |
| 03:08.18 | [hC] | Well.. I'm sufficiently confused as hell. Why would putting a line on hold increase bandwidth utilization by a whole lot? like 20x |
| 03:08.43 | JayTee52 | music on hold? |
| 03:08.48 | *** join/#asterisk gitguy (n=diego@adsl-145-202.click.com.py) |
| 03:08.56 | [hC] | yeah, but why would music on hold take up that much more bandwidth than the conversation itself? |
| 03:09.14 | [hC] | Im doing some tests where I call between two boxes, and loop an audio file |
| 03:09.15 | JayTee52 | seems a bit high |
| 03:09.19 | thx2000 | what codecs are available on the line? |
| 03:09.32 | [hC] | usage with iax/g729 are about 2.9kB/s each direction |
| 03:09.48 | [hC] | put the call on hold (hold music in g729) and its around 15-20kB each direction |
| 03:10.04 | [hC] | if there is no hold music, the bandwidth usage drops to dead zero. |
| 03:10.06 | [hC] | no rtp moves. |
| 03:11.49 | [hC] | ah. it looks like it depends on what i execute on the other side |
| 03:12.01 | [hC] | if i loop an audio file, it doesnt change. if i have the call put into milliwatt() it does. |
| 03:13.52 | *** join/#asterisk azidenth (n=aby_azid@227.215.50.60.cbj04-home.tm.net.my) |
| 03:14.18 | azidenth | hi, where can i read the latest asterisk TFOT online..thanks? |
| 03:14.27 | JayTee52 | ~book |
| 03:14.27 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:15.11 | azidenth | tq |
| 03:15.11 | JayTee52 | you can download the pdf too |
| 03:15.51 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 03:15.52 | JayTee52 | personally I use the pdf but bought the print version so I can read on the crapper and killing all those trees really pisses off the druids. |
| 03:19.00 | jblack | I prefer books. |
| 03:19.30 | *** join/#asterisk dssman (n=no@CPE001d7e602900-CM0011aec52a9c.cpe.net.cable.rogers.com) |
| 03:19.47 | jblack | Give me an aspect ratio and a resolution equivilant to books, and I would probably change my mind, though. |
| 03:20.34 | tzanger | not so much resolution for me as contrast ratio |
| 03:20.39 | Strom_C | imagine how wonderful your argument will be once you learn to spell "equivalent" |
| 03:22.54 | JayTee52 | lol |
| 03:22.56 | jackson__ | coughs #spellingbee if you're anal enough to correct someones spelling :) |
| 03:23.44 | JayTee52 | ok, I have to get some sleep. nite everyone |
| 03:24.19 | lmadsen | nite |
| 03:24.22 | lmadsen | :) |
| 03:25.37 | tzanger | how on earth would you have an irc spelling bee?? |
| 03:26.14 | tzanger | "spell punctuation" ... uh, okay, punctuation? |
| 03:28.07 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 03:28.07 | *** mode/#asterisk [+o jmhunter] by ChanServ |
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| 03:30.12 | lmadsen | you could have it on the phone in a conference room |
| 03:30.28 | lmadsen | you just invented a new phone game |
| 03:31.11 | BeeBuu | if i dial(zap/1/123&zap/2/234) ,how can i get which one is answered? |
| 03:33.58 | [hC] | man am i ever getting myself turned around in circles trying to do bandwidth measuring on calls |
| 03:34.04 | ZaVoid | anyone ever install the amr codec on asterisk? |
| 03:34.13 | [hC] | what im seeing on the wiki that john todd did just isnt adding up |
| 03:34.17 | lmadsen | BeeBuu: hrmm... looking at the channelvariables.tex file... I'm thinking you might be able to get something back with ${BRIDGEPEER} |
| 03:34.37 | BeeBuu | lmadsen: thanks |
| 03:34.51 | lmadsen | BeeBuu: can you try that out and show me the output from that variable? |
| 03:35.30 | lmadsen | exten => h,n,DumpChan() or exten => h,n,Verbose(1,BRIDGEPEER: ${BRIDGEPEER}) |
| 03:36.17 | BeeBuu | i just reading the pages about channel var.... |
| 03:37.27 | BeeBuu | i have no idea to fix my question,thanks for your point... |
| 03:38.39 | ZaVoid | anyone ever try the SigC5561 card? |
| 03:41.19 | [hC] | Is there any way besides measuring current throughput to check that iax2 trunking is infact on and working? |
| 03:41.57 | lmadsen | BeeBuu: what are you trying to do exactly? |
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| 03:49.54 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:50.26 | BeeBuu | lmadsen: i want to catch whois talking and do something when dial(one&two) |
| 03:50.56 | lmadsen | BeeBuu: please be more specific... you mean from the manager? |
| 03:51.06 | lmadsen | like from an external application? |
| 03:51.48 | BeeBuu | just a context,can i? |
| 03:52.23 | lmadsen | BeeBuu: I don't understand what you're asking, sorry |
| 03:53.34 | BeeBuu | anyway, you give me a good point to fix my problem,thanks again. |
| 03:53.47 | *** join/#asterisk ftp3 (n=none@pool-71-117-212-7.ptldor.fios.verizon.net) |
| 03:54.07 | ftp3 | is there a website that has good text to speech that i could use to make some menus? |
| 03:54.35 | russellb | why not just pay for cepstral? it's so cheap ... |
| 03:54.56 | ftp3 | didnt know about it |
| 03:55.00 | ftp3 | will google it :-) |
| 03:55.17 | Strom_C | russellb: shush! you can't tell people the easy, simple answer! :) |
| 04:01.08 | jameswf-home | we bought the cepstral as a toy we dont even use it for anything... had allison doing the minimall rap :) |
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| 04:05.40 | Yosam | cepstral voices suck |
| 04:05.52 | Yosam | the sound quality is poor |
| 04:06.00 | Yosam | Neospeech voices are better |
| 04:06.25 | jeffspeff | hi365, which is more preferred? Asterisk-Now, TrixBox or use FreePBX with a vanilla Asterisk install? |
| 04:06.59 | jeffspeff | *hi, |
| 04:07.20 | jeffspeff | (i don't know where the 365 came from. :P ) |
| 04:07.32 | *** join/#asterisk RoyK (n=roy@ip-206-23-149-91.dialup.ice.no) |
| 04:07.54 | lmadsen | Yosam: try out the latest version of cepstral with the allison voice -- I just heard a sample of it and it was really really good |
| 04:08.08 | lmadsen | I agree all the other voices I've heard were sub-par |
| 04:08.25 | lmadsen | I was actually shocked at how good the new version sounded |
| 04:09.09 | jeffspeff | any recommendations? |
| 04:09.56 | drmessano | ReiserFS kills the competition |
| 04:10.20 | JT | at least the significant others |
| 04:10.33 | Yosam | let me check the new version |
| 04:10.39 | Yosam | you mean the online sample? |
| 04:11.14 | jackson__ | lmadsen, you were listening to the 8khz version of allison on the demo page? |
| 04:11.30 | drmessano | lol |
| 04:11.47 | lmadsen | jackson__: nah, someone here at work had a copy of it and generated a file for me |
| 04:12.31 | Yosam | nah its the same quality |
| 04:12.32 | Yosam | i have |
| 04:12.33 | Yosam | on the web |
| 04:12.46 | lmadsen | shrugs |
| 04:13.14 | Yosam | any good speech recognition apps? |
| 04:13.51 | jameswf-home | lumenvox |
| 04:13.56 | *** join/#asterisk solar_ant (n=John@61.12.7.81) |
| 04:14.10 | Yosam | otherthen lumenvodx |
| 04:14.37 | Yosam | i am working on a non-commercial application for people with disabilities and i dont want to pay for it |
| 04:14.43 | jameswf-home | well if theres goin to be conditions.... |
| 04:14.50 | Yosam | i need more then 1 concurrent licenses |
| 04:14.59 | jameswf-home | microsoft sync |
| 04:15.01 | Yosam | i mean connections / channels sorry not licenses |
| 04:15.27 | *** join/#asterisk obnauticus (n=obnautic@c-24-22-79-67.hsd1.mn.comcast.net) |
| 04:15.28 | Yosam | and lumenvox costs 195 dollars for each channel |
| 04:17.31 | *** part/#asterisk RoyK (n=roy@ip-206-23-149-91.dialup.ice.no) |
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| 04:17.53 | Yosam | microsoft sycn |
| 04:17.56 | Yosam | sync |
| 04:18.03 | Yosam | you mean the new thing that will be on cars? |
| 04:19.45 | jameswf-home | I dunno I am not a MS person... |
| 04:20.18 | jameswf-home | as far as I know lumenvox os ot for asterisk and VR |
| 04:20.40 | Yosam | lumenvox is good i know |
| 04:21.15 | jameswf-home | If you cant afford lumenvox maybe VR isnt the proper route |
| 04:21.40 | syntxerr | stupid MS I want my SP3 already >:[ |
| 04:22.06 | jameswf-home | spe? win 2000? |
| 04:22.21 | jameswf-home | I havent used MS in so long I am way out of the loop |
| 04:23.01 | Yosam | jameswf-home |
| 04:23.07 | Yosam | i used lumenvox before |
| 04:23.25 | Yosam | for this project i am just a developer |
| 04:23.49 | jameswf-home | and this project cant afford it so it is out... |
| 04:23.58 | jameswf-home | and with it VR |
| 04:24.23 | Yosam | :) |
| 04:26.17 | [hC] | I cant express how much i hate continental airlines. |
| 04:28.14 | jameswf-home | airtran was OK but man they spam the hell out of you |
| 04:40.30 | *** join/#asterisk VoicePulse_ (n=contact@unaffiliated/voicepulse) |
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| 04:55.14 | ftp3 | i am trying to find a did provider here in the usa that allows unlimited incoming calls for a good price (skype is $24/yr for the number and unlimited outgoing to usa/canada for $2.95/mo.. everyone I find for asterisk is really expensive compared to that |
| 04:55.17 | ftp3 | any suggestions? |
| 04:57.45 | [TK]D-Fender | ftp3, everything is much more costly |
| 04:57.51 | [TK]D-Fender | ~itsplist-us |
| 04:57.53 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 04:59.31 | ftp3 | ah nice |
| 04:59.32 | ftp3 | thank you |
| 05:03.22 | syntxerr | I really like http://www.link2voip.com, they are based in canada but have servers in the states and overseas as well. Rates are 1.1 to canada and 1.5 to the US, unlim chans no features |
| 05:03.56 | syntxerr | billing is 6/6 and 6/30 domestic |
| 05:04.06 | *** join/#asterisk aut- (n=aut@modemcable086.237-80-70.mc.videotron.ca) |
| 05:04.36 | aut- | anyone used a nokia E70? or any other wireless voip phones? i know the older ones suck really bad, but just checking to see if anything is changed |
| 05:04.37 | syntxerr | skypes termination is cheap :( I want that! |
| 05:04.52 | syntxerr | Ive go no idea |
| 05:05.33 | ftp3 | syntxerr, but all my minutes are incoming, i will make no outgoing |
| 05:09.19 | ftp3 | i am very new to voip, so if this is really stupid, please bear with me... i loked at link2voip as was suggested, and I see this: Unlimited DID Pricing: |
| 05:09.19 | ftp3 | $9.99/month and 0¢/min |
| 05:09.32 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 05:09.35 | ftp3 | does that mean that for $9.99/mo I can have unlimited incoming calls? |
| 05:12.48 | [TK]D-Fender | ftp3, means you should read the fine print |
| 05:14.45 | ftp3 | ok, well, i have been reading sites for about 3hrs. and everyone seems to have their own "fine print".. i just want unlimited (which for skype anyway means 10,000min) or a large amount free.. I mean, I can pay the phone company $19.95/mo and have unlimited incoming calls.. why cant i do that with voip? voip is suppose to be cheaper |
| 05:15.24 | ftp3 | i suspect it is my own ignorance.. but I am not learning much from reading these sites ;-) |
| 05:15.37 | [TK]D-Fender | ftp3, no VoIP is supposed to be "not POTS". Everything else is fine print :) |
| 05:16.54 | ftp3 | ok, :-) do you have a company you suggest? |
| 05:27.32 | [TK]D-Fender | ftp3, nothing in particular |
| 05:28.01 | [TK]D-Fender | ftp3, VoicePulse Connect tends to have the lowest /$ for US 48 |
| 05:32.09 | [TK]D-Fender | ok, checkout time... |
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| 05:38.13 | Nasra | ~itsplist-ca |
| 05:38.14 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
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| 05:51.00 | salviadud | when doing a macro, should my voice channels be inside the macro context, or a context of their own? |
| 05:54.15 | stony | anyone seen the readme of chan_misdn ? |
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| 06:07.24 | stony | yeah that's what i thought... |
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| 06:26.41 | salviadud | exit |
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| 06:26.46 | salviadud | oops |
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| 06:32.45 | Silicium | ReiserFS: For when you need to partition your wife. :D |
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| 06:55.20 | mort_gib | Morning |
| 06:56.30 | mort_gib | When using Xtfer (*2) I can't use my "extensions" programmed on my Snom phones, because they are like sip:200@asterisk and * wants DTMF |
| 06:56.38 | mort_gib | Is there a way to do this?? |
| 07:02.00 | *** join/#asterisk mistergibson (n=mistergi@c-76-126-157-152.hsd1.ca.comcast.net) |
| 07:02.55 | mistergibson | anyone know a good how-to tutorial on a simple asterisk h323 pbx? |
| 07:03.12 | mistergibson | just looking to host h323 |
| 07:06.17 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
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| 07:08.38 | mistergibson | in fact, any how-tos on getting started (utter noob) |
| 07:08.49 | mort_gib | ~thebook |
| 07:08.49 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 07:08.54 | mistergibson | thanks |
| 07:09.02 | mort_gib | nopro |
| 07:11.41 | mort_gib | You want to start simple, and if you don't need telco stuff it's going to be ehm, easy?? |
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| 07:59.58 | akirahz | i was curious what sort sys requirements are needed to run asterisk, such as do i need the X platform installed? or just basic run of the mill shell account with zlib/bison? |
| 08:00.10 | akirahz | how much memory would be required? |
| 08:05.08 | Strom_C | akirahz: you defnitely DON'T want X installed |
| 08:11.15 | *** join/#asterisk stony (n=oloch@p57B3908D.dip0.t-ipconnect.de) |
| 08:11.17 | stony | h |
| 08:11.20 | stony | i |
| 08:11.57 | stony | the chan_misdn module i'm running is only using one channel per port - is there a workaround for that ? |
| 08:12.53 | JT | how so? |
| 08:13.33 | stony | jt? |
| 08:14.00 | mrdigital | akirahz: i can help you, |
| 08:14.19 | mrdigital | still here? |
| 08:15.06 | Silicium | http://blog.wired.com/27bstroke6/2008/04/reiser-guilty-o.html |
| 08:16.30 | akirahz | ah okay i did some more digging and found out my system can handle it, i was just curious how exactly you patch into a phone line with it if i have it in a x86 linux machine, is there a card or a device i need to purchase? |
| 08:16.31 | *** join/#asterisk shinao1 (n=shinao1@41.219.221.117) |
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| 08:23.48 | s0ck | stony: is that misdn? |
| 08:24.15 | stony | s0ck: yes |
| 08:25.05 | JT | stony: be more specific? |
| 08:34.16 | stony | JT: beronet BN8S0 card with 8 isdn ports and 16 b-channels - configured 4 ports internal, 4 ports external (nt,te), so this should make 8 channels on "everey side" |
| 08:34.26 | stony | but asterisk only uses the primary channel of each port |
| 08:34.46 | stony | so i only have 1 channel per port - the other is just not used |
| 08:35.12 | stony | so i only have 8 instead of the 16 expected channels ... |
| 08:35.48 | *** join/#asterisk adrin_ (n=adrin@chello089078212055.chello.pl) |
| 08:36.36 | adrin_ | Hello i have a question, how does data packets flow look like when creating a call through asterisk? do the data packets traverse through asterisk? |
| 08:43.10 | JT | stony: you still aren't giving the important information |
| 08:43.14 | JT | like inbound |
| 08:43.16 | JT | outbound |
| 08:43.36 | JT | how you are calling |
| 08:43.39 | stony | doesn't matter |
| 08:43.58 | stony | the calls are bridged between the outer and the inner network |
| 08:44.16 | JT | of course it bloody matters |
| 08:44.22 | stony | JT: the problem appears the whole time (calling out, or incoming calls) |
| 08:44.23 | JT | do you want the problem solved or not? |
| 08:44.43 | JT | pastebin your configuration |
| 08:45.13 | stony | JT: misdn.conf ? |
| 08:45.34 | *** join/#asterisk Jakobsen (n=ask@gandalf.dansupport.dk) |
| 08:46.11 | JT | and extensions.conf |
| 08:46.20 | *** part/#asterisk Jakobsen (n=ask@gandalf.dansupport.dk) |
| 08:46.46 | stony | JT: http://rafb.net/p/MWBBGG50.html |
| 08:48.05 | stony | JT: http://rafb.net/p/GJKKfC98.html |
| 08:50.58 | JT | ~freepbx |
| 08:51.00 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
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| 08:51.46 | stony | JT: this isn#t a freepbx problem - it's simply a misdn/chan_misdn problem |
| 08:51.59 | stony | JT: in my handmade contexts it's not working too |
| 08:52.24 | adrin_ | Hello i have a question, how does data packets flow look like when creating a call through asterisk? do the data packets traverse through asterisk? |
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| 08:57.17 | awk | hmm,cause 21 on hangup... does anyone have a cause table, explaing what each hangup cause is for? |
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| 09:34.58 | JT | stony: prove it isn't a freepbx issue, what are your dial strings? |
| 09:35.12 | saftsack | hi, is there no linear queues ring strategy before * 1.6 available? |
| 09:44.45 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
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| 09:49.09 | Uatec | Hello |
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| 10:05.30 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 10:08.48 | *** join/#asterisk R0land (n=Roland@193.227.191.91) |
| 10:08.50 | R0land | hello all |
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| 10:14.32 | *** part/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 10:14.42 | R0land | could any1 help out with extensions.conf plz! |
| 10:15.55 | R0land | i managed to dial out |
| 10:16.05 | R0land | but when i call in, it keeps ringing .. |
| 10:16.28 | R0land | i dont know how to point it to a specific extension.. or set a specific audio file to b played like "plz dial extension or 0 for assistance" |
| 10:19.35 | stony | R0land: isdn or pstn ? |
| 10:19.44 | R0land | pstn |
| 10:20.36 | R0land | stony pstn |
| 10:23.03 | tzafrir | R0land, so run in the CLI: core set verbose 3 |
| 10:23.11 | tzafrir | and pastebin what you see there |
| 10:23.20 | R0land | ok just a sec |
| 10:23.35 | tzafrir | And probably also the relevant parts of extensions.conf |
| 10:23.46 | stony | R0land: check the context incoming calls go to and add an s extension there |
| 10:23.47 | *** join/#asterisk Ubluzok (n=ubluzok@62.141.89.219) |
| 10:23.47 | R0land | core set verbose 3 |
| 10:23.48 | R0land | Verbosity is at least 3 |
| 10:24.07 | masus | R0land: http://pastebin.ca/1001621 maybe this help ? |
| 10:24.27 | R0land | stony the prob is im at a loss of wht to add in extensions.conf to do so |
| 10:24.45 | stony | R0land: ever thought of using freepbx webfrontend ? |
| 10:24.57 | R0land | stony i did.. but its giving so many errors |
| 10:25.09 | R0land | stony when i started doing things manualy.. it started working |
| 10:25.31 | R0land | masus thts a macro and since im a newbie with all of this im afraid id unno which part i should edit to fill mine with |
| 10:25.52 | stony | R0land: then you have todo it the hard way ... |
| 10:26.47 | R0land | stony ya i know.. but at least it would work |
| 10:27.42 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
| 10:27.50 | stony | R0land: when a call arrives, where should it go ? |
| 10:27.53 | stony | phone on another port ? sip ? |
| 10:28.03 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:28.27 | R0land | well when a call comes through.. i want an audio msg to b played and if an extension is dialed.. itll b transfered to a SIP extension |
| 10:28.53 | stony | R0land: ok - then you need to use the read() function |
| 10:28.53 | JT | ~thebook |
| 10:28.55 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 10:29.22 | stony | R0land: and after you read the digits then you need to dial the extension via Dial() |
| 10:29.31 | R0land | stony exactly |
| 10:29.55 | stony | R0land: and before you can read any digits you should answer() the channel |
| 10:30.59 | tzafrir | hmm... that's Background and then WaitExten, right? |
| 10:31.14 | tzafrir | Why use Read there? |
| 10:31.16 | R0land | tzafrir well i guess so, since in the background the audiop file would b played |
| 10:32.01 | R0land | [incoming] |
| 10:32.01 | R0land | exten => s,1,Answer() |
| 10:32.01 | R0land | exten => s,n,Playback(cc_welcome) |
| 10:32.01 | R0land | exten => s,n,Hangup() |
| 10:32.03 | R0land | exten => 2,1,Goto(internal,s,2) |
| 10:32.07 | R0land | this is wht i got so far |
| 10:32.28 | stony | R0land: the moment you Hangup() it jumps to the h extension |
| 10:33.03 | R0land | well im at a bit of a loss |
| 10:33.46 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 10:34.42 | stony | R0land: this is how i would do it (it's all freehand) |
| 10:34.42 | stony | exten => s,1,Answer() |
| 10:34.45 | R0land | that would make asterisk answer the call, but who would take it! |
| 10:35.13 | stony | exten => s,n,Read(digitsToDial,file_to_playback,5,,3,30) |
| 10:35.43 | stony | exten => s,n,Dial(technology/${digitsToDial}) |
| 10:35.47 | R0land | wht does 5,,3,30 <<== stand for |
| 10:36.05 | stony | R0land: you can get all the information you need on http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read |
| 10:36.50 | stony | it means that you read a maximum of 5 digits, no options, the caller has three attempts to dial the digits and it's a timeout of 30 secs per attempt |
| 10:37.07 | R0land | ah thts gr8 |
| 10:37.11 | R0land | tht would b perfect |
| 10:37.30 | R0land | so if i just set the exten u just gave me.. it would b enough ? |
| 10:37.32 | stony | yeah, but you should check if the digits match your available extensions before you Dial() them |
| 10:37.48 | stony | yes, the "s"-extension is always used on pstn connections |
| 10:38.19 | R0land | digitstodial should b substituted with wht! |
| 10:38.32 | stony | ?? |
| 10:39.24 | R0land | exten => s,n,Dial(SIP/${digitsToDial}) |
| 10:39.28 | R0land | so thts how its supposed tob e?! |
| 10:39.41 | JT | R0land: have you read the book? |
| 10:39.59 | R0land | JT part of it just enough to configure sip/outgoing calls |
| 10:40.19 | JT | read the bits on the dialplan |
| 10:40.23 | JT | it is the most important |
| 10:41.01 | R0land | well i agree.. but i want to make things work at a minimal level |
| 10:41.12 | R0land | then start tweaking/ makiing things professional enough |
| 10:42.13 | stony | R0land: yes SIP/${digitsToDial} is the right way (if you're dialing a sip extension) |
| 10:42.15 | JT | this is fundametnally important |
| 10:42.33 | R0land | stony thank u |
| 10:42.51 | R0land | JT i agree.. but as i said trying to make things work at a minimal level and then ill move up |
| 10:43.05 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 10:43.06 | stony | JT: so back to my misdn problem |
| 10:44.45 | stony | JT: i don't see the point where dial commands that only dial on channel groups without selecting any channel directly can cause this trouble |
| 10:45.38 | stony | JT: if there would be a direct selection of the channel it would be comprehensible if a channel is in use - but not in this way |
| 10:45.53 | R0land | one more question..! stony do i set them in [incoming] context in extensions.conf! or in the same context that i had my outgoing rules in..! |
| 10:46.01 | stony | jt: it also seems that misdn only export 8 channels |
| 10:46.30 | stony | R0land: in the incoming context (the context you defined in the config of the technology you are using for incoming calls) |
| 10:46.39 | R0land | stony ok thank u :) |
| 10:47.10 | JT | stony: misdn is utter rubbish really |
| 10:47.23 | JT | it doesn't support group dial on NT ports |
| 10:47.29 | JT | only TE ports |
| 10:49.20 | stony | JT: i'm thinking about two things: 1st: migrating from misdn to bristuff (would work with the card here ..) |
| 10:49.48 | stony | JT: 2nd: migrating from asterisk to callweaver (t.38/fax, and other things) |
| 10:49.57 | JT | i would choose bristuff where possible |
| 10:50.07 | JT | as misdn is quite pathetic |
| 10:50.19 | stony | but i need this "shiny" freepbx frontend for some users that want to manage the callgroups and IVR thingie .... |
| 10:50.21 | JT | i don't think i would choose callweaver, as bri support sucks |
| 10:50.41 | stony | yeah, that's the point ... |
| 10:51.03 | JT | really |
| 10:51.08 | JT | the reasiest option with bri |
| 10:51.10 | JT | for reliability |
| 10:51.11 | *** join/#asterisk Penggu (n=me@203.213.102.59) |
| 10:51.15 | JT | is to get a bri sip gateway |
| 10:51.39 | *** join/#asterisk EugenA (n=eugen@dialbs-213-023-178-155.static.arcor-ip.net) |
| 10:51.41 | JT | as each different bri card and bri driver package has its own battle you need to fight |
| 10:51.47 | stony | this asterisk pbx with the 8 port isdn card is the bri/sip gateway... |
| 10:51.59 | JT | you know what i mean. |
| 10:52.06 | JT | a HARDWARE BRI to SIP gateway |
| 10:52.08 | Penggu | hi all. i got snom phones..their initial registration always fails and the phones then retry anyway after 5 minutes. any idea why? tcpdump shows packets coming in from the phones, but * refuses to reply to them in time enough for the registration to work. i always get either authorization failed or network failure in the phones. |
| 10:52.26 | EugenA | hi! time in my cdr-csv.log file is -2 hours. How do i change this? |
| 10:52.35 | Penggu | if i manually say "re-register" in the snom phone before the 5 minute re-cycle, it usually works quite quickly |
| 10:52.48 | stony | JT: yeah - no money |
| 10:52.55 | JT | EugenA: are you in timezone UTC+2? |
| 10:52.59 | Penggu | they are all on the local LAN, nat=no, qualify=yes, host=dynamic |
| 10:53.07 | EugenA | JT, yes |
| 10:53.18 | JT | EugenA: it is normal for CDRs to be in UTC |
| 10:53.45 | stony | Penggu: you added type=friend ? |
| 10:54.01 | EugenA | JT, so i should not change this? |
| 10:54.20 | JT | EugenA: probably not, i've never looked into changing it to be honest |
| 10:54.33 | Penggu | stony: we got some company to do it for us initially, and we've just continueed that way. voip-info's page on snoms shows type=fried |
| 10:55.06 | tzafrir | JT, Asterisk is a BRI <-> SIP gateway |
| 10:55.16 | JT | tzafrir: just not a very good one |
| 10:55.31 | JT | if i was to start over again i'd only use hardware BRI gateways |
| 10:55.53 | yang | hello tzafrir |
| 10:55.59 | tzafrir | yang, hi |
| 10:58.47 | stony | Penggu: read the packets between the phones and the pbx with ngrep ? they are mostly in plain text and good to read (even the errors) |
| 10:59.38 | Penggu | tcpdump won't cut it? |
| 11:00.54 | Penggu | snom interface also has sip traces |
| 11:01.42 | *** part/#asterisk EugenA (n=eugen@dialbs-213-023-178-155.static.arcor-ip.net) |
| 11:01.48 | *** join/#asterisk EugenA (n=eugen@dialbs-213-023-178-155.static.arcor-ip.net) |
| 11:07.01 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:12.00 | mort_gib | Penggu: Check network mask, iptables on * |
| 11:13.08 | mort_gib | Penggu: If you have the Snoms on static IP address, you have to turn off DHCP |
| 11:13.14 | Penggu | mort_gib: server and phones have same n/w settings. iptables all blank (ACCEPT) |
| 11:13.25 | Penggu | all phones on DHCP |
| 11:13.38 | mort_gib | Ping gtom * |
| 11:13.46 | mort_gib | £$%^^& Ping from * |
| 11:13.59 | *** join/#asterisk smps (n=smps@217.71.50.228) |
| 11:14.20 | Penggu | they get ip addr, download their settings, they are ping replying and their web interfaces accessible during the time the registration fails |
| 11:15.43 | mort_gib | tcpdump on * don't gine you anything?? What about tail -f /var/log/asterisk/messages and asterisk -rvvvvvvv "sip set debug" |
| 11:15.56 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:16.08 | Penggu | ill do that now.. |
| 11:16.41 | Penggu | my tcpdump command: tcpdump -w sip223.dump -v -v -s 1500 -i eth0 port 5060 and host <phone ip addr> |
| 11:16.57 | smps | hi anyone made motorola modem (1057:3052) work with zaptel ? |
| 11:17.35 | mort_gib | Basically you need to have a god long look at the sip set debug if you can see the traffic using tcpdump |
| 11:17.53 | Penggu | im doing "sip debug ip <ip address of phone>" |
| 11:18.13 | Penggu | hmm |
| 11:18.16 | Penggu | packetdump on console |
| 11:18.24 | mort_gib | ?? |
| 11:18.25 | Penggu | makes things a little easier... |
| 11:20.43 | *** join/#asterisk muthah (n=muthah@41.207.106.34) |
| 11:21.49 | Penggu | Received from udp:10.0.0.10:5060 at 24/12/2001 11:01:00:789 (474 bytes): SIP/2.0 401 Unauthorized |
| 11:21.53 | Penggu | although i dont see that in * console |
| 11:21.57 | Penggu | that's from the phone's sip log |
| 11:22.35 | Penggu | * is just happily transmitting options messages, and getting back the replies |
| 11:22.52 | Penggu | probably reminiscent of the previous registration |
| 11:24.07 | Penggu | does this mean anything: Apr 29 21:23:38 WARNING[18081]: chan_sip.c:1229 retrans_pkt: Maximum retries exceeded on transmission 3c26701d1a9c-o5c6xie3gojy for seqno 103 (Non-critical Request) |
| 11:24.16 | Penggu | probably the failed end of the phone's requests |
| 11:24.53 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 11:26.07 | mikkel | I have a ISDN Digium B410 quad-bri card and have plugged a cable from port 1 to my telco (TDC Denmark) into the NT box where I would normally plugin a ISDN phone. Should the port be in TE or NT mode ? |
| 11:26.54 | mikkel | I presume it should be in TE mode and act like a normal phone ? |
| 11:26.56 | mort_gib | Ah, passwd/user |
| 11:26.56 | mort_gib | or domain |
| 11:26.58 | mort_gib | So you get any messages complaining about domain |
| 11:27.41 | Penggu | mort_gib: nope |
| 11:28.14 | gr0mit | mikkel, you should be running your astiersk in TE mode |
| 11:28.25 | JT | mort_gib: TE |
| 11:28.34 | JT | mikkel: even |
| 11:28.36 | mort_gib | JT:?? |
| 11:29.40 | mort_gib | Penggu: Looks like soem of the traffic is not received by * |
| 11:29.41 | mikkel | gr0mit: Okay, so I don't change the jumper on the card. It should only be in NT mode is I bypassed the telco box a plugged the cable directly to the card, is that true ? |
| 11:29.56 | mort_gib | Try IP address as registrar on the Snom hpones |
| 11:29.56 | gr0mit | yup |
| 11:30.11 | mikkel | gr0mit: Thanks. So that is not the problem. |
| 11:30.19 | gr0mit | if you want to connect an isdn phone to the card you need to put the card into NT mode |
| 11:30.27 | gr0mit | mikkel, what prob are you having? |
| 11:30.55 | JT | mikkel: no, you cannot plug a line from the telco directly into the card without an NT1 |
| 11:31.14 | JT | mikkel: NT mode on the card is for it to connect to a BRI TE device |
| 11:31.59 | mikkel | JT: gr0mit: So I have the change the jumper to NT to plugin a ISDN phone, really ? |
| 11:32.06 | JT | yes |
| 11:32.09 | gr0mit | yup |
| 11:32.15 | JT | and you need to feed power to connect an ISDN phone |
| 11:32.40 | mort_gib | mikkel: On some cards it's just turning the card around |
| 11:32.49 | mikkel | JT: And how do I feed power ? |
| 11:32.57 | JT | mikkel: depends on the card |
| 11:33.05 | mort_gib | mikkel: most cards have a connector on the back |
| 11:33.19 | JT | either an inline power feeder that goes between the card and the phone, or you need to get a suitable power supply for the card |
| 11:33.29 | JT | if the card supports ISDN phone power feed |
| 11:33.44 | mikkel | JT: I think the power is taken from the BUS, I have a Analog card where I plug a power cable, but this B410 don't have that. |
| 11:34.11 | mort_gib | mikkel: NO, the card would need separate power |
| 11:34.37 | JT | mikkel: no. |
| 11:34.47 | JT | mikkel: you CANNOT run an ISDN phone from bus power |
| 11:34.58 | JT | it can't pull enough current |
| 11:35.06 | mikkel | mort_gib: That card don't have a power plug, so mayby it is an extention. |
| 11:35.23 | tzafrir | mikkel, what card is it? |
| 11:35.36 | mikkel | tzafrir: Digium B410 BRI |
| 11:35.41 | mort_gib | mikkel: No I think the card dosen't support it |
| 11:35.42 | JT | :( |
| 11:36.02 | JT | you will probably need inline power injectors |
| 11:36.10 | mort_gib | I use Sangomas A500 |
| 11:36.20 | mort_gib | They have the connector on the back |
| 11:36.32 | JT | the digium card only supports misdn afaik |
| 11:36.38 | JT | so pretty much useless ;) |
| 11:36.54 | mikkel | JT: Why is that ? |
| 11:37.10 | mort_gib | And you can turn the modules around to support TE or NT |
| 11:37.25 | *** join/#asterisk jblack (n=jblack@pool-72-79-176-12.sctnpa.east.verizon.net) |
| 11:37.26 | mikkel | mort_gib: It is a jumper on this card. |
| 11:37.29 | *** join/#asterisk nixguy (n=matmoj@fw.packetfront.com) |
| 11:37.29 | JT | misdn is awful |
| 11:37.34 | gr0mit | has only ebver used junghanns.net |
| 11:37.39 | gr0mit | for BRI cards |
| 11:37.50 | JT | full of bugs, appauling NT support |
| 11:38.02 | JT | it's renamed ISDN4Linux |
| 11:38.06 | gr0mit | but there is no support for 1.4 in bristuff |
| 11:38.13 | nixguy | hi , if i want sip phones to be able to "handshake" over asterisk and then just talk "peer to peer" do i need to configure this specifically? |
| 11:38.18 | tzafrir | gr0mit, sure there is |
| 11:38.22 | nixguy | or is this the default behaviour? |
| 11:38.29 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-66-31.bstnma.east.verizon.net) |
| 11:38.34 | gr0mit | so tzafrir since when? |
| 11:38.55 | tzafrir | both from junghanns.net and from http://updates.xorcom.com/astribank/bristuff/1.4 |
| 11:39.01 | gr0mit | last i saw was a beta6 which compiled but had probs |
| 11:39.21 | mikkel | JT: I'm using asterisk 1.6 and mISDN, but have'nt got it working yet. |
| 11:39.35 | gr0mit | yes, test6 - |
| 11:39.43 | gr0mit | not a production stable version |
| 11:39.52 | tzafrir | mikkel, 1.6 has support for BRI in chan_zap |
| 11:39.55 | mikkel | gr0mit: 1.6beta7.1 |
| 11:41.11 | gr0mit | tzafrir, native zaptel support for junghanns cards then? |
| 11:41.47 | tzafrir | well, you still need zaptel from bristuff (qozap and a patch or two to zaptel) |
| 11:42.02 | tzafrir | but zaptel has always been the smaller part of bristuff |
| 11:42.05 | gr0mit | aah ok |
| 11:44.15 | gr0mit | tzafrir, is it also working well with the cheap cologne chipset cards? |
| 11:44.24 | gr0mit | i have 3 cheap cards in my home box |
| 11:44.29 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 11:44.35 | tzafrir | gr0mit, the zaphfc driver |
| 11:44.43 | gr0mit | yup that is what i use at the moment |
| 11:44.49 | tzafrir | as usual |
| 11:44.55 | gr0mit | ok cool. |
| 11:45.08 | tzafrir | (only now there's a certain florz patch included there) |
| 11:45.41 | gr0mit | yes, i noticed last time i tried 1.4 bristuff in about jan or feb that all the masses of interrupts had gone |
| 11:46.35 | *** join/#asterisk oej (n=olle@216.64.24.250) |
| 11:47.02 | tzafrir | though qozap in 1.2's bristuff seems to include some extra fixes |
| 11:47.45 | mikkel | Seems I need a PWR400M module to supply power to phones. |
| 11:48.06 | mort_gib | Mikkel: Why are you doing this to yourself?? |
| 11:48.18 | mort_gib | Why don't you just do SIp phones?? |
| 11:48.27 | mikkel | Thanks all for the info, will prob. ask some more another day. |
| 11:48.44 | *** join/#asterisk sw (n=sw@unaffiliated/sw) |
| 11:49.14 | mikkel | mort_gib: I would, be they are all ugly and the customer is only interesseted in trendy phones, like B&O design. |
| 11:49.26 | JT | sip phones do make an awful lot more sense |
| 11:49.36 | JT | i doubt that, there are heaps of nice sip phones out there |
| 11:49.40 | JT | the customer is on drugs |
| 11:50.10 | mikkel | I know, but that is what they want. |
| 11:50.25 | JT | a phone with way less usability, they are crazy |
| 11:50.34 | mort_gib | mikkel: B&O phones are analogue though |
| 11:50.47 | mort_gib | btw you CAN get nice looking SIP phones |
| 11:51.13 | mort_gib | And by now B&O is highly overrated, they do car stereos now.... |
| 11:51.59 | mikkel | mort_gib: Where are those nice SIP phones ? |
| 11:52.28 | JT | so B&O don't even do ISDN phones? |
| 11:52.37 | mort_gib | Some of the Polycom ones lokks, if not nice at least prof. |
| 11:52.50 | mort_gib | -Not as far as I know, they do home stuff, mostly |
| 11:54.06 | mikkel | JT: Not anymore, but the old base station is available in ISDN. |
| 11:54.23 | mort_gib | So they already HAVE the base station?? |
| 11:54.40 | mort_gib | Have you had a look at the Sangoma A500 card |
| 11:54.49 | *** part/#asterisk sw (n=sw@unaffiliated/sw) |
| 11:54.57 | mikkel | mort_gib: The phone have to be cordless. |
| 11:55.12 | mort_gib | Do they already have the phones?? |
| 11:55.16 | mikkel | mort_gib: Yes they have the base station. |
| 11:55.22 | mikkel | mort_gib: Yes |
| 11:55.52 | mort_gib | Ok, I would still have a look at the Sangoma cards, nice Hardware Echo cancelation |
| 11:56.18 | mort_gib | Pity though... They loose out big time on functionality |
| 11:56.42 | mort_gib | mikkel:Are you Danish by the way |
| 11:56.50 | mikkel | mort_gib: Yes |
| 11:57.01 | mort_gib | Heh, |
| 11:57.22 | mort_gib | I Danmark stadig?? |
| 11:57.54 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 11:58.32 | mikkel | mort_gib: Ja, det er du måske ikke ? |
| 11:59.02 | mort_gib | Nix, paa Gibraltar/Syd Spanien :-) |
| 11:59.51 | mikkel | mort_gib: Cool, så kan jeg ikke få hjælp fra dig, andet end her på chatten. |
| 12:00.04 | mort_gib | Joda! |
| 12:00.22 | mort_gib | Har du hoert om Internettet og SSH ?? |
| 12:00.43 | mort_gib | :-) |
| 12:04.13 | mikkel | mort_gib: Ja. Men jeg har nemlig et enmands firma og er begyndt at sælge et lægeprogram som jeg selv har udviklet under linux til klinikker og her skal jeg også integrere telefon system så de kan se patienten når de ringer (udfra nummer) og kunne derfor godt bruge en mand til at hjælpe mig. |
| 12:04.30 | mort_gib | Jeg er paa! |
| 12:04.31 | SteveTotaro | Ja. Men jeg har nemlig et enmands firma og er begyndt at s?lge et l?geprogram som jeg selv har udviklet under linux til klinikker og her skal jeg ogs? integrere telefon system s? de kan se patienten n?r de ringer (udfra nummer) og kunne derfor godt bruge en mand til at hj?lpe mig. |
| 12:05.12 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 12:05.15 | mort_gib | Mikkel: Men saa laener du dig faretruende meget op ad callerid |
| 12:05.54 | mikkel | mort_gib: Okay, men det er jo lidt besværligt når du ikke er i danmark. Ja callerid skal jeg bruge, håber det virker. |
| 12:05.59 | glaz | Sprechen Sie Englisch! |
| 12:06.01 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:06.07 | mort_gib | mikkel: Suk, ikke altid... |
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| 12:06.40 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
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| 12:06.49 | mort_gib | glaz: Ja, Ich sprecht Englisch, aber nich mit mine Danisher freunden.... |
| 12:07.23 | glaz | Sorry, I don't speak german. |
| 12:07.48 | mort_gib | glaz: That's okay, we all have our short comings |
| 12:07.54 | mikkel | mort_gib: grineren. |
| 12:08.02 | mort_gib | :-) |
| 12:08.09 | glaz | you said you speak english, but not with your friends |
| 12:08.12 | glaz | or something like that. |
| 12:08.20 | mort_gib | Yes, that is very true. |
| 12:08.56 | mort_gib | the thing is though that I do speak German, and Spanish :-) |
| 12:09.06 | dominic1 | He speaks english, but not with his danish friends |
| 12:09.36 | glaz | Ich denke, wenn Sie chatten möchten mit Ihren Freunden in Deutsch, sollten Sie sich einen anderen Kanal. |
| 12:09.51 | mikkel | mort_gib: Maybe we should talk in a separate room. If I could ask you a few more questions. |
| 12:10.02 | mort_gib | mikkel: Sure |
| 12:10.25 | dominic1 | they wasn't speaking german, they was speaking danish |
| 12:10.48 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:11.06 | mort_gib | glaz: Maybe, but all sorts of conversations seems to run in parallel in here anyway ;-) |
| 12:12.09 | muthah | is it possible to run Asterisk without any of the Digium/sangoma cards |
| 12:12.21 | muthah | I only want to use it for intrabranch VOIP |
| 12:12.22 | masus | muthah: yes |
| 12:12.34 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
| 12:12.43 | muthah | what about the requirement of a timing device on the cards |
| 12:13.01 | JT | only for certain things |
| 12:13.05 | dominic1 | yes, but you should use zaptel for things like IAX or meetme |
| 12:13.12 | JT | app_meetme, music on hold, iax trunking |
| 12:13.27 | nixguy | anyone who can answer what i need to do to allow sipphones to communicate directly between each other? |
| 12:13.35 | nixguy | or is this the default behaviour in asterisk? |
| 12:14.01 | [TK]D-Fender | nixguy: "canreinvite=yes" in the phone's entry |
| 12:14.44 | nixguy | mucho gracias |
| 12:14.53 | masus | nixguy: and Dial(SIP/${EXTEN}) |
| 12:14.58 | masus | :) |
| 12:15.20 | nixguy | masus: hmm not really what i was asking but i appreciate the effort! |
| 12:16.29 | masus | :) |
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| 12:56.47 | *** join/#asterisk Penggu (n=me@203.213.102.59) |
| 12:57.00 | Penggu | hi all. how would you make * forget the peers in "sip show peers" |
| 12:57.11 | Penggu | eg say im testing and i reboot a phone - i dont want * to keep poking it |
| 12:57.19 | Penggu | and checking. it should forget the IP add |
| 12:57.36 | *** join/#asterisk coppice (n=chatzill@218.0.193.139) |
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| 12:58.18 | stony | Penggu: sip prune realtime peer |
| 12:58.49 | Penggu | "Peer '223' is not a Realtime peer, cannot be pruned." |
| 12:59.21 | Penggu | in the list, i get: "223/223 10.0.0.223 D 2063 UNREACHABLE" |
| 12:59.38 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:59.38 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:00.23 | *** join/#asterisk Dovid (n=Dovid@bzq-79-177-124-172.red.bezeqint.net) |
| 13:02.44 | Penggu | where is that list stored anyway? an asterisk restart doesn't clear it |
| 13:03.32 | *** join/#asterisk mikeshank (n=sam@c-68-37-250-134.hsd1.pa.comcast.net) |
| 13:04.10 | [TK]D-Fender | stony: "database del SIP Register/[peernamewithoutbraces]" + "sip reload" |
| 13:04.43 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 13:05.05 | Dovid | Any Marshals here ? |
| 13:05.12 | Dovid | seems there is no http://svn.digium.com/svn/zaptel/trunk |
| 13:05.24 | lmadsen | Dovid: this is on purpose |
| 13:05.51 | Dovid | y ? |
| 13:05.52 | lmadsen | Dovid: zaptel/trunk was removed because it was stale -- zaptel/branches/1.4 is where the latest zaptel code is |
| 13:06.06 | lmadsen | it works for both 1.4 and 1.6.x beta's |
| 13:06.40 | lmadsen | version numbers between libpri, zaptel, and asterisk do not necessary match up anymore (they haven't for a while now) |
| 13:06.48 | Dovid | oh didnt know that |
| 13:06.52 | lmadsen | now you know |
| 13:06.55 | Dovid | i:) |
| 13:06.59 | Dovid | thats a lot more convenient |
| 13:07.02 | lmadsen | and knowing is power |
| 13:07.07 | Dovid | yup |
| 13:07.16 | Dovid | which is y i sit here all day |
| 13:07.36 | lmadsen | hrmm... you could sit in more productive arenas :) |
| 13:07.47 | Dovid | lol |
| 13:07.55 | Dovid | i sit in here and help when i can |
| 13:08.47 | mikeshank | Hi all. I have a small home office with 3 phones, I have an old G4 mac I want to use as the server, firstly can I install asterisk on the G4 and secondly what other hardware would I need to get started? |
| 13:08.57 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 13:09.05 | Dovid | mikeshank: it depends. what kind of lines will you be using ? |
| 13:09.11 | Dovid | PSTN or VOIP ? |
| 13:09.55 | Dovid | for testing purposes you cna use VOIP in the begining and if you want to use "regular" lines you can always add a card |
| 13:10.06 | mikeshank | im pretty new to this, PSTN is that just the plain old phone line? |
| 13:10.08 | lmadsen | or a breakout box that converts analog to SIP |
| 13:10.29 | Dovid | yup |
| 13:10.31 | lmadsen | then you don't need to stick any hardware directly in the box... since I"m not sure if it would work on a G4 Mac (it doesn't use standard PCI does it?) |
| 13:10.41 | mikeshank | ok pstn then |
| 13:10.41 | lmadsen | I know very very little about G4's... but it's not Intel based right? |
| 13:10.51 | lmadsen | ~pstn |
| 13:10.52 | jbot | extra, extra, read all about it, pstn is Public Switched Telephone Network, or "please stop the nonsense" |
| 13:10.57 | lmadsen | ~pots |
| 13:10.58 | jbot | [~pots] POTS (Plain Old Telephone Service) is the term for a common analog phone line service as is used world-wide. The "phone company" is called FXO (~fxo), and the user end-point (or phone) is called FXS (~FXS). POTS supports 1 channel, and possibly call-waiting, 3-way calling, CID, as signalled to the telco. |
| 13:11.09 | mikeshank | no not intel base |
| 13:11.50 | lmadsen | mikeshank: ya... you're going to be better off sticking with pure VoIP in that system (so using something like an analog->voip box will be a better bet than internal hardware) |
| 13:12.16 | lmadsen | try it out with pure voip before you buy any hardware to see how it goes -- you'll want to use a Linux distro ideally on there |
| 13:12.53 | [TK]D-Fender | mikeshank: What kind of access to the PSTN are you looking to use with your system? |
| 13:13.06 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 13:13.25 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
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| 13:14.02 | mikeshank | [TK]D-Fender: sorry, I'm really new to this, I'm not sure what that means |
| 13:14.32 | [TK]D-Fender | mikeshank: Looking to use something like your basic analog line with *? |
| 13:15.07 | [TK]D-Fender | mikeshank: Multiple lines? Perhaps Digital? (ISDN, PRI, T1, E1) |
| 13:15.27 | mikeshank | [TK]D-Fender: just 1 line |
| 13:15.35 | [TK]D-Fender | mikeshank: So basic analog line? |
| 13:15.41 | mikeshank | [TK]D-Fender: yes |
| 13:16.23 | [TK]D-Fender | mikeshank: http://www.telephonydepot.com/product_p/105-054-312.htm |
| 13:17.03 | [TK]D-Fender | mikeshank: there you go. That will let you use your analog line with *, and use 1 analog phone as well. Both will cpeak the SIP VoIP protocol to your * server and you won't have to worry about configuring hardware. |
| 13:17.13 | [TK]D-Fender | speak* |
| 13:18.16 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:18.55 | mikeshank | [TK]D-Fender: when you say 1 analog phone do you mean 1 number or actual physical phones? forgive my stupidity |
| 13:19.31 | mikeshank | [TK]D-Fender: i have 3 phones in the office |
| 13:19.37 | [TK]D-Fender | mikeshank: Actual phone line. Good 'ole Ma Bell POTS |
| 13:21.57 | mikeshank | [TK]D-Fender: so for 3 phones I would need an adapter with 3 FXS ports correct? |
| 13:22.17 | *** join/#asterisk ajohnstone (n=ajohnsto@host81-137-217-96.in-addr.btopenworld.com) |
| 13:22.18 | [TK]D-Fender | mikeshank: Everything depends how you want to run things. |
| 13:22.27 | [TK]D-Fender | mikeshank: How would you like your setup to operate? |
| 13:22.28 | Dovid | trying to run bootstrap.sh and I get this error: http://pastebin.ca/1001758 |
| 13:22.51 | *** join/#asterisk ReD-MaN (i=rox-ur-s@172-220.static.golden.net) |
| 13:23.06 | Dovid | i am using trunk |
| 13:23.32 | Maliuta | Dovid: you are elephant? ;) |
| 13:23.38 | Dovid | huh ? |
| 13:23.49 | Maliuta | makes bad telecoms joke |
| 13:24.05 | Maliuta | elephant ... trunk ... elephant :) |
| 13:24.10 | Dovid | ah ok |
| 13:24.12 | Dovid | tired |
| 13:24.36 | Maliuta | I could make others, but they'd be more culture specific |
| 13:25.36 | Maliuta | are you sure your version of m4 is up to the one required? |
| 13:25.41 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-245-162.balt.east.verizon.net) |
| 13:25.45 | Dovid | dont know y its not working. |
| 13:25.52 | Dovid | i think that may be the issue which i am upgradign now |
| 13:26.41 | Maliuta | the fact that m4 is saying it's not accepting the argument would point at m4 not being the correct version |
| 13:27.11 | Dovid | i would think that the script would complain about it |
| 13:27.17 | Maliuta | you could see if there is a nodebug option on that script though |
| 13:27.22 | Dovid | like it did about a diff package that i needed to upgrade |
| 13:27.30 | Maliuta | well it did, that is the error message |
| 13:27.52 | mikeshank | [TK]D-Fender: truth be known i'm not sure, whats typical? Basically have a small business along with a "business" phone account which costs me $150 a month and for that I basically get a listing in the phone book and voicemail. Im looking to drop the business account and use my home line for both my residence and business |
| 13:28.01 | Dovid | it did for another package not for m4 |
| 13:28.17 | Dovid | upgraded m4 manually will see now. looks like its working |
| 13:28.30 | Dovid | yup. m4 was the issue |
| 13:28.38 | Dovid | after bootstrap comes configure ? |
| 13:29.05 | [TK]D-Fender | mikeshank: Ok, what kind of internet connection do you have? Type and bandwidth up/down. |
| 13:29.25 | Maliuta | Dovid: are you sure you're up to compiling? |
| 13:30.11 | Dovid | yea |
| 13:30.14 | Dovid | doing that now ) |
| 13:31.14 | Maliuta | Dovid: what does the README say about what comes after bootstrap.sh? |
| 13:31.18 | *** join/#asterisk kombi (n=kombi@port-213-160-14-18.static.km-it.de) |
| 13:31.24 | *** join/#asterisk adjohn (n=adjohn@i220-221-4-188.s05.a013.ap.plala.or.jp) |
| 13:31.35 | mikeshank | [TK]D-Fender: cable, 768k i think |
| 13:32.25 | [TK]D-Fender | mikeshank: What do you want * to do for you? |
| 13:32.28 | mikeshank | [TK]D-Fender: comcast says 6 megabits per second download |
| 13:32.29 | stony | JT: i tested the whole thing through, put the freepbx away and simply did 5 calls to the speaking clock - the 5th call isn't coming through, it says there are no free channels on the ports i try (and i try all 4 one by another ...) |
| 13:33.19 | JT | stony: is it a TE or NT port? |
| 13:33.24 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 13:33.42 | mikeshank | [TK]D-Fender: sorry,what does that mean |
| 13:34.16 | stony | JT those are TE ports |
| 13:34.27 | [TK]D-Fender | mikeshank: Means what it sounds like. What functionality are you looking for Asterisk to provide to you in your setup? |
| 13:34.39 | JT | stony: you are using group dialling yes? |
| 13:34.52 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:35.03 | stony | JT: no, i simply dial through all the available ports until one is working |
| 13:35.17 | kombi | An extension with MusicOnHold plays instantly when called but not at all when connected via Originate. Why might this be? |
| 13:35.18 | JT | why don't you use group dial? |
| 13:35.28 | stony | JT: i put the whole freepbx thing away and simply tested it with a "normal" dial |
| 13:36.10 | [TK]D-Fender | kombi: ...... PASTEBIN |
| 13:36.17 | kombi | coming.. |
| 13:36.29 | stony | JT: just for testing, to see that there is ABSOLOUTLY nothing that affects dialing |
| 13:36.55 | stony | JT: sometimes i can bring up to channels on one port and sometimes not |
| 13:37.07 | stony | JT: but in the end i cannot do more than 4 channels |
| 13:37.35 | stony | s/to channels/two channels/ig |
| 13:37.35 | JT | sounds like a misdn issue |
| 13:37.42 | stony | JT: NO WAY |
| 13:37.44 | stony | ;) |
| 13:37.44 | JT | ? |
| 13:37.49 | JT | yeah |
| 13:37.50 | stony | lol |
| 13:37.52 | JT | misdn is pretty bad |
| 13:38.01 | JT | i abandoned it long ago |
| 13:38.05 | mikeshank | [TK]D-Fender: oh god Im dense, just realized * meant asterisk, really at this point just want to set up voicemail and play with it |
| 13:38.37 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
| 13:38.43 | kombi | Fender: http://pastebin.se/194382 |
| 13:39.43 | [TK]D-Fender | mikeshank: Ok here's a consideration : If you're on the phone your next caller will either busy-out, or hit telco voicemail. If it hits telco voicemail, * won't be able to tell you that there is one. Second that means 2 places you'll have to check for VM's (in * itself if it got answered by *), and your telco's box. |
| 13:40.06 | [TK]D-Fender | mikeshank: Just a business consideration. |
| 13:40.56 | *** join/#asterisk fmueller (n=user@p548F6ACA.dip.t-dialin.net) |
| 13:41.14 | [TK]D-Fender | kombi: Where's the call that failed? |
| 13:42.20 | rupa | or permanently forward your POTS line to a SIP provider like voicepulse connect that lets you have multiple channels. Then you can accept multiple incoming... |
| 13:42.26 | mikeshank | [TK]D-Fender: didnt think about that. is there anything you would suggest given what im trying to do? |
| 13:43.13 | [TK]D-Fender | mikeshank: If you're jsut looking for VM I'd say stick with the line you've got. You haven't listed any reasonable benifit that * would give you. |
| 13:43.46 | rupa | [TK]D-Fender, btw, upgraded my poly 320 to the 3.0 firmware. Now the line light follows the presense status of the associated extension. woohoo |
| 13:44.38 | [TK]D-Fender | mikeshank: Now if you wanted a separate phone # to use for your business I might suggest setting up with an ITSP, at which point * would become useful. |
| 13:44.44 | kombi | That's the thing, it does not fail, only no sound.. |
| 13:44.53 | kombi | ..Fender.. sorry |
| 13:45.00 | [TK]D-Fender | rupa: So 3.0 adds presence on the line-keys? Cool. |
| 13:45.05 | rupa | [TK]D-Fender, yup |
| 13:47.03 | kombi | Fender: http://pastebin.se/194384 |
| 13:47.36 | mikeshank | [TK]D-Fender: so with an ITSP i wont run into the busy issue you mentioned above |
| 13:47.36 | rupa | [TK]D-Fender, but I've run into an issue that I'm sure is a config issue in *. Incoming calls to the extension set the hint. But outgoing calls from that extension do not change the hint. Is there something I have to do in the outbound dialplan to get hints to work on that side? |
| 13:47.54 | rupa | mikeshank, up to the # of channels the ITSP lets you use. |
| 13:48.23 | [TK]D-Fender | mikeshank: well you can arrange to have them support multiple simultaneous calls. This would allow the overflow to still hit your * and then you could ahve them fall to VM on your sever as well. |
| 13:49.08 | [TK]D-Fender | rupa: basics require something like "call-limit=99", "type=peer" |
| 13:49.25 | [TK]D-Fender | rupa: If that doesn't do it then perhaps calls aren't being auth'd right in both directions. |
| 13:49.48 | rupa | [TK]D-Fender, ok, I'll play with both directions. The Hint only needs to be where the exten is defined, right? |
| 13:50.04 | mikeshank | [TK]D-Fender: that sounds like what i might be looking for, although what Im really after is not paying 150 for basically voicemail, is the ISTP costly? |
| 13:51.20 | rupa | [TK]D-Fender, type=peer can't happen since the Linky has 2 SIP registrations (one for PSTN and one for the TA) and peer can't distinguish between the two since it only uses IP to distinguish. |
| 13:51.39 | [TK]D-Fender | rupa: needs to be in the context the peer is using (or linked to it), or in "subscribecontext=" is you did that |
| 13:51.54 | [TK]D-Fender | mikeshank: cheaper than residential lines. |
| 13:51.59 | [TK]D-Fender | ~itsplist-us |
| 13:51.59 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 13:52.06 | [TK]D-Fender | mikeshank: ^^^ go do some research |
| 13:52.26 | mikeshank | [TK]D-Fender: thanks so much for your guidance |
| 13:53.57 | *** join/#asterisk adjohn (n=adjohn@i220-221-4-188.s05.a013.ap.plala.or.jp) |
| 13:55.28 | Katty | pouts |
| 13:55.33 | Katty | i hate access. and i hate sql. |
| 13:55.37 | Katty | just so everyone knows. |
| 13:57.32 | [TK]D-Fender | Katty: Yes, Access blows, but serves a few smaller needs. Nothing wrong with SQL, just depends what for, and which back-end |
| 13:57.36 | [TK]D-Fender | Katty: (Mew) |
| 13:57.39 | [TK]D-Fender | mikeshank: You're welcome |
| 13:59.12 | Katty | [TK]D-Fender: mew. |
| 13:59.20 | Katty | [TK]D-Fender: well i can't get access to nicely export via odbc into sql. |
| 13:59.24 | Katty | [TK]D-Fender: and this makes me MAD! |
| 13:59.40 | Katty | stabity access |
| 13:59.42 | [TK]D-Fender | Katty: Which SQL? |
| 13:59.54 | Katty | [TK]D-Fender: 2005 express... comes with sugarcrm |
| 14:00.03 | [TK]D-Fender | Katty: *shudder* |
| 14:00.15 | Katty | mhmm |
| 14:00.21 | Katty | now you know. |
| 14:00.26 | Katty | you KNOW |
| 14:00.36 | Katty | cries. |
| 14:02.56 | *** join/#asterisk af_ (n=getsmart@88-149-230-0.dynamic.ngi.it) |
| 14:12.07 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:13.58 | kombi | just so I get it right: To make an outbound call an play a message, I set up an extension with playback and originate to connect it to the outbound call. Correct? |
| 14:14.28 | *** join/#asterisk Skarmeth (n=Skarmeth@iris.aspec.com.br) |
| 14:20.10 | [TK]D-Fender | kombi: pretty much. |
| 14:21.18 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 14:21.26 | kombi | Fender: Thing is, the message just does not play. It does when you call that extension direct but not when connecting with originate |
| 14:22.33 | [TK]D-Fender | kombi: Thing is you still aren't showing me anything useful. |
| 14:22.50 | kombi | point taken, let me try again |
| 14:23.31 | [TK]D-Fender | kombi: SIP debug, etc... |
| 14:24.35 | s0ck | Sorry, your post tripped our spam filter - let us know if you think this could be improved |
| 14:24.39 | s0ck | anyone seen that on pb? |
| 14:24.46 | kombi | Fender: How do I sip debug again? |
| 14:25.02 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 14:25.35 | kombi | sip set debug, got it.. |
| 14:25.54 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:28.15 | s0ck | http://pastebin.ca/1001807 <--- spot anything fishy in there? |
| 14:28.38 | *** join/#asterisk CVirus (n=GoD@196.205.193.110) |
| 14:30.15 | [TK]D-Fender | s0ck: Yes, you're not stating your problem and expect us to go fishing for it. |
| 14:31.07 | kombi | Fender: I'm not sure this is very helpful..->http://pastebin.se/194385 |
| 14:31.41 | s0ck | [15:05:23] <s0ck> anyone have an idea why when dialling inbound i see the cli identifies the call and passes to vm |
| 14:31.42 | s0ck | [15:05:28] <s0ck> but the caller never actually gets picked up ;s |
| 14:31.59 | *** part/#asterisk stony (n=oloch@p57B3908D.dip0.t-ipconnect.de) |
| 14:32.06 | kombi | Maybe I should explain what I try to do: Call a few people and put them into a queue |
| 14:32.28 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 14:33.05 | *** join/#asterisk fish-bulb (n=cstewart@216.207.245.1) |
| 14:33.11 | *** join/#asterisk ldleworker (n=ldle@ip-007-165.its.niu.edu) |
| 14:33.39 | ldleworker | How do I pass information from an AMI Originate to the Context/Extension? |
| 14:33.59 | *** part/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 14:35.01 | seanbright | ldleworker: http://readlist.com/lists/lists.digium.com/asterisk-users/14/72296.html |
| 14:35.02 | *** join/#asterisk jfg (n=jfg@dyn-88-121-23-232.ppp.tiscali.fr) |
| 14:35.16 | acxty | hey guys, I have a small extensions.conf file that is not working how I want to. I want to make the call the tt-weasels to play and wait for the extension to be press. I press 1 and it should play honolulu sound, but instead WaitExten timeout and it hangup. http://dpaste.com/47404 |
| 14:35.29 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com) |
| 14:35.29 | [TK]D-Fender | s0ck: FreePBX is not supported here, please visit their channel for support on it. |
| 14:36.19 | *** join/#asterisk __freedom__lover (n=__freedo@200.178.163.129) |
| 14:37.01 | *** join/#asterisk xenonex (n=xenonex@92.47.14.6) |
| 14:37.01 | s0ck | ya... |
| 14:37.21 | s0ck | fxo ports, zaptel.conf should read fxsks=x, yeh? |
| 14:37.30 | tzafrir | right |
| 14:37.53 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:38.35 | [TK]D-Fender | s0ck: Typically. |
| 14:38.41 | *** join/#asterisk deneenm (n=fullstop@neptune.saucontech.com) |
| 14:38.49 | [TK]D-Fender | s0ck: This of course has nthing to do with your hitting VM. |
| 14:38.53 | hmmhesays | is there any way to catch alpha characters without _. in asterisk? |
| 14:39.03 | acxty | and it doesn't give me any error at all, seems the press key never arrive |
| 14:39.03 | s0ck | i have set it to vm on purpose |
| 14:39.12 | s0ck | cli shows it hitting vm |
| 14:39.17 | [TK]D-Fender | hmmhesays: as a pattern, not really. Fixed, yes |
| 14:39.19 | s0ck | actual incoming line keeps ringing |
| 14:39.23 | s0ck | stupid card does not pick up the line |
| 14:39.33 | s0ck | despite the first few lines of that pb showing it does |
| 14:39.48 | s0ck | event 18/event 2 |
| 14:40.43 | *** join/#asterisk killmel8tr (n=IceChat7@ip-198-22-65-131.quickconnectusa.com) |
| 14:40.50 | *** join/#asterisk [T]an3 (n=[T]ank@206.71.78.158) |
| 14:41.14 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 14:41.21 | [TK]D-Fender | kombi: Please pastebin all of the related dialplan. |
| 14:41.43 | deneenm | hi.. I'm fairly new to asterisk, especially the analog side. I have just installed a TDM422E card. From what I have read, the green modules are FXS and the red modules are FXO. According to the numbers under the modules, the green FXS should be ports 1 & 2, and the red FXO should be 3 & 4. When I run ztscan, they show up switched. Is this because the FXO uses FXS signalling, or do I really have it backwards? |
| 14:42.00 | deneenm | I don't want to blow the card up by plugging a PSTN line in. :-) |
| 14:42.13 | acxty | do you find something wrong with the file? |
| 14:42.28 | *** join/#asterisk harryv (n=fufufu@3e6b0130.rev.stofanet.dk) |
| 14:42.36 | harryv | hey -- any idea why this call isen't established ? http://pastie.org/pastes/187226 |
| 14:42.43 | s0ck | ztscan on my 805e shows ports 1-5 as fxo deneenm |
| 14:42.47 | s0ck | they are all pstn ports |
| 14:42.51 | [T]an3 | if asterisk is telling me: gbx-01*CLI> cdr mysql status |
| 14:42.51 | [T]an3 | Not currently connected to a MySQL server. |
| 14:42.51 | [T]an3 | gbx-01*CLI> |
| 14:42.57 | [T]an3 | and my config is set up |
| 14:43.04 | [T]an3 | what else could I be missing? |
| 14:43.36 | kombi | [TK]D-Fender: http://pastebin.se/194386 |
| 14:43.48 | s0ck | is that a pci-e card, by any chance, deneenm? |
| 14:43.50 | [TK]D-Fender | harryv: Perhaps they don't auch calls they send you. Try with "insecure=port,invite" |
| 14:44.11 | deneenm | s0ck: here's the output from ztscan http://www.pastie.org/188662 |
| 14:44.26 | s0ck | btw, im here trying to get help too :P |
| 14:44.26 | [TK]D-Fender | kombi: 61 is answering and hanging up. |
| 14:44.39 | deneenm | s0ck: I am fairly certain that it is a regular PCI-2.2 card. |
| 14:44.41 | [TK]D-Fender | kombi: and you clearly don't understand how app_queue works at all. |
| 14:44.59 | s0ck | from that and comparing it with my own |
| 14:45.03 | [TK]D-Fender | kombi: You do not call MoH MANUALLY, and you NEVER call extens that ANSWER the call focibly like that |
| 14:45.06 | [T]an3 | doing a load mysql_addon_mysql.so gives me a: Module 'cdr_addon_mysql.so' already exists. |
| 14:45.06 | s0ck | i see port 1/2 as your pstn ports |
| 14:45.13 | s0ck | 3/4 as the handsets? |
| 14:45.23 | *** join/#asterisk CVirus (n=GoD@196.205.193.110) |
| 14:45.30 | s0ck | bearing in mind, i can't get my card to work |
| 14:45.37 | s0ck | so probably not a great baseline :P |
| 14:45.55 | deneenm | s0ck: That's what I see as well. When I looked under the modules on the card itself, 1 & 2 were green and 3 & 4 were red. |
| 14:46.07 | deneenm | s0ck: Thanks for the heads up. ;-) |
| 14:46.13 | [T]an3 | is there a way to get mysql connected without restarting asterisk? |
| 14:46.28 | hunmonk | digium.com seems to be down -- i'm getting a godaddy page. is there a mirror that i can use to download asterisk/zaptel? |
| 14:46.30 | s0ck | does it dial in/out btw? |
| 14:46.48 | kombi | [TK]D-Fender: hmm, so how would you go about calling three people and storing them in a queue, just as an outline? |
| 14:47.00 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:48.27 | [TK]D-Fender | kombi: Go read the sample configs, and check the WIKI for more info. |
| 14:48.46 | harryv | [TK]D-Fender: yep! thanks! |
| 14:48.49 | harryv | spent hours on this. |
| 14:49.06 | deneenm | hunmonk: does this work? http://216.27.40.102/pub/telephony/zaptel/zaptel-1.4.10.tar.gz |
| 14:49.27 | [T]an3 | i can manually connect to the database from the mysql command. but asterisk is not connecting to it. |
| 14:49.34 | [TK]D-Fender | harryv: You're welcome. |
| 14:49.35 | [T]an3 | anyone able to assist? |
| 14:50.09 | [T]an3 | here is my cdr_mysql.conf: http://pastebin.ca/1001818 |
| 14:50.54 | *** join/#asterisk babarak (n=babar@ouest.adelux.fr) |
| 14:51.02 | babarak | hi |
| 14:51.17 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 14:51.43 | [T]an3 | the output of cdr mysql status is: Not currently connected to a MySQL server. |
| 14:52.38 | babarak | i have a problem with TC400B, i have compile zaptel and load the module, dmesg give me "Zaptel DTE (G.729a) Transcoder support LOADED (firm ver = 6.12)", but when i try to see the card in asterisk with "transcoder show", i have "No Zaptel transcoders found" |
| 14:52.49 | babarak | someone have an idea, please ? |
| 14:53.00 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 14:53.10 | [T]an3 | here is what happenes when i manually try to load mysql: gbx-01*CLI> load cdr_addon_mysql.so |
| 14:53.10 | [T]an3 | [Apr 29 08:53:06] WARNING[23591]: loader.c:627 load_resource: Module 'cdr_addon_mysql.so' already exists. |
| 14:53.19 | acxty | [TK]D-Fender, may you help me with this. http://dpaste.com/47406 . When I press the extension it ignores it and don't execute Playback(honolulu) |
| 14:53.35 | [T]an3 | anyone? |
| 14:53.39 | [TK]D-Fender | acxty: first guess, DTMF mode isn't set right |
| 14:54.09 | acxty | guess no |
| 14:54.56 | acxty | does it should be set on the extensions.conf file or on a different one? |
| 14:55.21 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 14:56.53 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 14:56.53 | [TK]D-Fender | actin the channel driver config of whatever device you're calling in from. |
| 14:57.29 | [T]an3 | ive tried doing a reload of cdr_custom.so cdr_manager.so and cdr. nothing seems to be working. |
| 14:57.35 | outtolunc | PLUTO/1 |
| 14:57.48 | [T]an3 | Module 'cdr_addon_mysql.so' does not support reload |
| 14:57.49 | [T]an3 | [Apr 29 08:57:57] ERROR[24334]: cdr_addon_mysql.c:23 |
| 14:58.12 | acxty | [TK]D-Fender, I test it from a normal telephone line and from a cellphone |
| 14:58.12 | glaz | unload it first, than load. |
| 14:58.41 | [T]an3 | still says i am not connected to a mysql server |
| 14:58.42 | [TK]D-Fender | acxty: Meaningless.... |
| 14:58.48 | [T]an3 | but I can connect to it manually |
| 14:58.57 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 14:59.05 | glaz | [T]an3: paste your cdr_mysql.conf to pastebin. |
| 14:59.15 | [T]an3 | already did... let me get the link again.... |
| 14:59.18 | glaz | ok |
| 14:59.19 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 14:59.25 | [T]an3 | (08:50:09 AM) [T]ank: here is my cdr_mysql.conf: http://pastebin.ca/1001818 |
| 14:59.36 | glaz | ok, comment the sock line |
| 14:59.44 | [T]an3 | k... hang on |
| 15:00.01 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:00.03 | babarak | i have a problem with TC400B, i have compile zaptel and load the module, dmesg give me "Zaptel DTE (G.729a) Transcoder support LOADED (firm ver = 6.12)", but when i try to see the card in asterisk with "transcoder show", i have "No Zaptel transcoders found" |
| 15:00.07 | glaz | and also the port line, you don't need it. |
| 15:00.24 | [T]an3 | then unload and load again? |
| 15:00.31 | glaz | yes. |
| 15:01.24 | ldleworker | Anyone know how to prevent ringing during a Call() ? |
| 15:01.25 | [T]an3 | same thing except now it complained of the sock file not being specified |
| 15:01.47 | glaz | what version of asterisk-addons did you install? |
| 15:02.18 | [T]an3 | asterisk 1.4.11 and asterisk addons 1.4.2 |
| 15:02.46 | [TK]D-Fender | ldleworker: and what is a "Call()"? |
| 15:02.53 | ldleworker | Dial() sorry |
| 15:03.07 | [TK]D-Fender | ldleworker: use "m" and an empty class. |
| 15:03.08 | glaz | isn't your mysql.sock located in /var/run/mysql/mysql.sock , you wrote /var/lib.. |
| 15:03.20 | [T]an3 | looking |
| 15:03.22 | ldleworker | TK, empty class? |
| 15:03.51 | ldleworker | m() |
| 15:03.54 | ldleworker | ? |
| 15:04.02 | [T]an3 | glaz: srwxrwxrwx 1 mysql mysql 0 Apr 17 16:27 /var/lib/mysql/mysql.sock |
| 15:04.07 | [TK]D-Fender | ldleworker: MoH. |
| 15:04.18 | acxty | [TK]D-Fender, I add dtmfmode=inband in sip.conf, it is working now, but I found different values for dtmfmode (inband, rfc2833, info, auto), which one do you recommend? |
| 15:04.45 | [TK]D-Fender | acxty: Whichever non-inband one your provider supports. |
| 15:05.03 | glaz | [T]an3: try mysql -uasterisk -h 10.110.1.181 -p asterisk_billing , and type scubapro1 as the password |
| 15:05.15 | acxty | [TK]D-Fender, ok, will change that line right now |
| 15:05.22 | glaz | sock and not localhost as the hostname shouldn't work. |
| 15:05.24 | [T]an3 | glaz: yeah, when I do that it works. |
| 15:05.40 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:05.40 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:05.56 | glaz | is the mysql server on the same box as the asterisk server? |
| 15:06.02 | [T]an3 | no |
| 15:06.07 | [T]an3 | on a db server |
| 15:06.43 | glaz | so you don't need sock= |
| 15:06.49 | glaz | sock is for local connections. |
| 15:06.53 | [T]an3 | ok |
| 15:06.59 | [T]an3 | so i still have it commented out |
| 15:07.16 | glaz | I had issue too with 1.4.2 , I installed 1.4.6 and it worked fine |
| 15:07.47 | [T]an3 | I have it working on another server. I had a failure of hardware last night and copied everything over to this server. |
| 15:07.57 | *** join/#asterisk nighty^ (n=nighty@p4069-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 15:07.57 | [T]an3 | had it working on another server I should say |
| 15:08.23 | babarak | someone can help me please ? |
| 15:08.36 | glaz | did you recompile it or just copied the backuped files over? |
| 15:08.52 | [TK]D-Fender | babarak: Virtually noone here even mentions the TC400 let alone has one. I highly suggest you contact Digium for support on it. |
| 15:09.20 | babarak | [TK]D-Fender : ok thanks |
| 15:09.34 | acxty | [TK]D-Fender, guess what, I try them all, and the onlyone that works is inband jeje |
| 15:09.51 | [TK]D-Fender | acxty: then your provider sucks, but at least its functional. |
| 15:10.20 | acxty | [TK]D-Fender, hehehe, what will be the disadvantages of using inband compare to the other ones |
| 15:10.52 | [TK]D-Fender | acxty: inband means its being detected by your * instead o ftheir equipment and that poor audio conditions means you'll miss DTMF's |
| 15:11.37 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 15:11.42 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 15:11.42 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:13.23 | ldleworker | [TK]D-Fender, How do you specify an empty class? |
| 15:13.43 | ldleworker | Dial(${CHAN},m()) |
| 15:13.44 | ldleworker | ? |
| 15:14.23 | [TK]D-Fender | ldleworker: No, specify a class that you create that doesn't have anything to actually play |
| 15:16.28 | *** join/#asterisk quaqo (n=quaqo@89-97-102-198.ip17.fastwebnet.it) |
| 15:35.11 | ftp3 | hello, |
| 15:35.27 | __freedom__lover | ftp3: hi friend |
| 15:35.34 | ftp3 | <PROTECTED> |
| 15:36.05 | [TK]D-Fender | ftp3: You tell us. |
| 15:36.25 | [TK]D-Fender | ftp3: try them and see. Jitter is worse than a little latency |
| 15:36.45 | ftp3 | lol, well, i am paying someone to setup a asterisk server for me. and i didnt want to pay to setup a whole service if the speed is to slow |
| 15:37.13 | errr | Im having some trouble getting a gotoif working: http://fluxbox.pastebin.ca/1001855 No matter what I have houstonoper set to it goes to exten => s,n(message),Dial(IAX2/toVMail/3333) |
| 15:37.16 | [TK]D-Fender | ftp3: You'll still want * regardless of the provider, no? |
| 15:37.50 | ftp3 | oh yes.. :-) but I am not sure how much $$ (time) it takes to swap provider info each time |
| 15:38.10 | Qwell | it shouldn't take but 30 seconds |
| 15:38.12 | [TK]D-Fender | errr: you might want to try NOT setting 2 extens to Priority 1... |
| 15:38.32 | errr | crap I didnt even notice that |
| 15:38.36 | errr | thanks |
| 15:38.52 | ftp3 | qwell: ahh.. thats good.. I didnt know if we were talking hundreds of dollars in time or something like that |
| 15:38.54 | [TK]D-Fender | errr: You're welcome. |
| 15:42.12 | acxty | hey guys, If I want to create my own sounds I need to have them in every format (alaw, g722, g729, gsm)? |
| 15:42.26 | [TK]D-Fender | acxty: No, any one that * supports |
| 15:42.27 | acxty | *ulaw, and wav |
| 15:43.01 | [TK]D-Fender | acxty: * will transcode if needed. You should probably ahve them in the most common codec you'll be using to avoid transcoding in the first place. |
| 15:43.08 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 15:43.13 | acxty | [TK]D-Fender, right now I make a test will only having a wav file, but it doesn't reproduce it, does asterisk have a specific frequency |
| 15:44.41 | [TK]D-Fender | acxty: 8khz mono |
| 15:44.55 | acxty | thanks |
| 15:45.00 | *** join/#asterisk Rico29 (n=Rico@ARennes-256-1-155-252.w86-195.abo.wanadoo.fr) |
| 15:46.17 | hunmonk | can anybody point me to an alternate location to checkout the asterisk GUI? svn.digium.com is down |
| 15:46.44 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 15:46.51 | file | what makes you think it is down? |
| 15:48.14 | hunmonk | file: visit the site, or try a checkout |
| 15:48.44 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 15:49.13 | file | perfectly fine |
| 15:49.31 | hunmonk | file: hrm |
| 15:49.33 | hunmonk | ok |
| 15:49.40 | file | what happens for you? |
| 15:49.41 | hunmonk | file: guess i'll try again |
| 15:50.11 | hunmonk | file: visiting svn.digium.com in a web browser takes me to a godaddy page |
| 15:51.08 | file | uh huh... |
| 15:51.40 | hunmonk | file: colossus:local root# svn co http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui |
| 15:51.40 | hunmonk | svn: PROPFIND request failed on '/svn/asterisk-gui/trunk' |
| 15:51.40 | hunmonk | svn: PROPFIND of '/svn/asterisk-gui/trunk': 501 Not Implemented (http://svn.digium.com) |
| 15:51.46 | hunmonk | that's the svn error |
| 15:51.56 | hunmonk | file: i have to run for a bit. be back on later |
| 15:53.20 | *** join/#asterisk CrummyGummy (n=dude@dsl-242-43-91.telkomadsl.co.za) |
| 15:53.53 | CrummyGummy | Hi all, is asterisk better packaged for ubuntu or redhat? |
| 15:54.23 | [TK]D-Fender | CrummyGummy: We advise against packages for * period. |
| 15:54.49 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com) |
| 15:55.24 | CrummyGummy | Oh, ok |
| 15:56.21 | rupa | is ilbc better than gsm if you want to use less bw than ulaw? |
| 15:58.34 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 15:59.49 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 16:01.08 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 16:02.39 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 16:03.01 | *** join/#asterisk l2cache (n=chatzill@117.178.101.97.cfl.res.rr.com) |
| 16:03.24 | *** join/#asterisk sohmestra (n=david@mx.hrcsb.org) |
| 16:03.49 | l2cache | Is there a way to use the features.conf transfer when you have an agent as a member to the queue? If I test it now, the only key that works as an agent is * and that hangs up. |
| 16:04.26 | sohmestra | is there any decent free/open source "receptionist" type of interface for asterisk? |
| 16:13.16 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 16:14.11 | *** part/#asterisk CrummyGummy (n=dude@dsl-242-43-91.telkomadsl.co.za) |
| 16:14.29 | *** join/#asterisk yakkop (n=yakkop@c-69-181-20-21.hsd1.ca.comcast.net) |
| 16:14.33 | *** join/#asterisk mintojoseph (n=mvaliyav@nat/redhat-in/x-345958954f7cd40a) |
| 16:16.29 | l2cache | Is there a way to use the features.conf transfer when you have an agent as a member to the queue? If I test it now, the only key that works as an agent is * and that hangs up. |
| 16:18.38 | *** join/#asterisk gitguy (n=diego@adsl-131-47.click.com.py) |
| 16:18.39 | gitguy | hi |
| 16:18.59 | gitguy | what kind of stuff do i have to install on centos or in any other distro to make meetme/zaptel/asterisk work |
| 16:19.10 | gitguy | in the kernel part |
| 16:19.35 | *** join/#asterisk Cyon (n=cyon@fxp1.dmz1.cro1.bestweb.net) |
| 16:21.09 | l2cache | are you using any digium cards? |
| 16:21.20 | l2cache | or just want the ztdummy timer? |
| 16:21.32 | gitguy | not using any card |
| 16:21.40 | gitguy | just want ztdummy timer |
| 16:21.42 | gitguy | for meetme |
| 16:21.46 | l2cache | do a "modprobe ztdummy" |
| 16:21.54 | l2cache | then make sure you start zaptel, then asterisk |
| 16:21.56 | l2cache | and it should work |
| 16:22.13 | l2cache | verify ztdummy is loaded with "lsmod | grep ztdummy" |
| 16:22.19 | gitguy | FATAL: Module ztdummy not found. |
| 16:22.21 | gitguy | i'm on debian |
| 16:22.22 | gitguy | now |
| 16:22.34 | l2cache | have you already compiled all the packages? |
| 16:22.35 | rob0 | "start zaptel" means what? |
| 16:22.40 | [TK]D-Fender | l2cache: "core show application queue" |
| 16:22.43 | l2cache | start the service |
| 16:22.58 | gitguy | l2cache: just asterisk |
| 16:23.10 | gitguy | l2cache: do i need zaptel for that kernel module? |
| 16:23.11 | gitguy | or? |
| 16:23.16 | l2cache | D-Fender, the queue application isnt the problem.. The agent that is logged in and connected cannot transfer calls |
| 16:23.19 | [TK]D-Fender | gitguy: Yes |
| 16:23.43 | [TK]D-Fender | l2cache: You talking about "app_agentlogin"? |
| 16:23.49 | l2cache | yes |
| 16:24.14 | [TK]D-Fender | l2cache: Think that it may still depend on how you call queue. |
| 16:24.16 | l2cache | How would an agent transfer a call? without using the phones transfer capability. I want to incorporate the features.conf transfer via "#" |
| 16:24.38 | rob0 | zaptel is a service? |
| 16:24.43 | [TK]D-Fender | l2cache: make sure they are allwed in your call to it. |
| 16:24.48 | l2cache | The only dtmf that the agent's phone picks up is "*" to soft hangup |
| 16:24.57 | [TK]D-Fender | l2cache: pastebin your dialplan. |
| 16:25.03 | l2cache | I have tested that, but it does not work for agents. |
| 16:25.34 | *** join/#asterisk kombi (n=kombi@port-213-160-14-18.static.km-it.de) |
| 16:25.37 | *** part/#asterisk [T]an3 (n=[T]ank@206.71.78.158) |
| 16:25.41 | l2cache | Normal extensions added to a queue works just fine, but agents bound to a queue via "member => Agent/1234" do not have features |
| 16:26.48 | kombi | I was under the impression that ${CALLERID(num)} contains the caller's phone number. However, NoOp(${CALLERID(num)} returns an empty string, why might that be? |
| 16:26.58 | [TK]D-Fender | l2cache: ... PASTEBIN |
| 16:27.17 | [TK]D-Fender | kombi: You too... |
| 16:27.34 | kombi | ;).. |
| 16:27.48 | gitguy | can i have notification sounds (know how many people are in a conference, etc) in app_conference as well as on meetme? |
| 16:27.59 | gitguy | that would be easier to do with meetme right? |
| 16:29.02 | l2cache | there is nothing to pastebin, i have the call entering the queue "exten => 1234,1,Queue(test||tT)" |
| 16:29.56 | [TK]D-Fender | l2cache: What ver of *? |
| 16:30.03 | l2cache | 1.4.18 |
| 16:30.12 | [TK]D-Fender | l2cache: Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI]) |
| 16:30.31 | [TK]D-Fender | l2cache: then you are clearly calling it wrong. |
| 16:30.31 | l2cache | well it works clearly |
| 16:30.40 | [TK]D-Fender | l2cache: not for your transfer its not. |
| 16:30.52 | [TK]D-Fender | l2cache: Your options are in the wrong field |
| 16:31.04 | l2cache | standby |
| 16:34.29 | l2cache | working now |
| 16:34.31 | l2cache | thanks Fender |
| 16:35.06 | *** join/#asterisk R0land (n=Roland@193.227.191.91) |
| 16:35.09 | R0land | hello all |
| 16:35.28 | R0land | could anyone help me out in recieving calls on my asterisk through pstn! |
| 16:35.52 | R0land | ive handled the outgoing calls! i could perfectly call an outside number through spa, or dial a specific spa extension on my lan and it will answer |
| 16:36.11 | R0land | but when some1 calls me on my pstn line, they just hear the ringing sound and it doesnt ring here! |
| 16:36.24 | R0land | this is my extensions.conf http://www.pastebin.ca/1001923 |
| 16:36.35 | [TK]D-Fender | l2cache: You're welcome. |
| 16:36.53 | [TK]D-Fender | R0land: That doesn't show us anything at all. |
| 16:37.11 | ldleworker | Can anyone explain why this doesn't merge the calls but drops the first upon answer? :: http://pastebin.com/m238ed4f0 |
| 16:37.13 | R0land | [TK]D-Fender thts the only thing ive set in my extensions.conf |
| 16:37.25 | R0land | as for sip, ive configured my extension tht i have on my softphone |
| 16:37.29 | [TK]D-Fender | R0land: Yes, but you aren't showing us what * sees upon this incoming call. |
| 16:37.30 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 16:37.32 | R0land | as well as the extension tht i need to dial to reach the pstn |
| 16:37.38 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 16:37.56 | R0land | [TK]D-Fender well lets say i gave u my pstn number and u tried calling me! the phone just rings and no one answers.. |
| 16:38.04 | R0land | and me! on the other end.. dont have a clue tht its ringing |
| 16:38.18 | [TK]D-Fender | ldleworker: "Call()" is not an application. |
| 16:38.27 | ldleworker | LOL |
| 16:38.33 | ldleworker | [TK]D-Fender, thanks |
| 16:38.36 | ldleworker | smacks himself |
| 16:38.38 | [TK]D-Fender | R0land: we need CLI output with SIP debug enabled. |
| 16:38.46 | R0land | [TK]D-Fender k coming just a sec |
| 16:39.13 | gitguy | i'm running linux-2.6.16-xenU on debian etch and there is only linux-2.6.18* on the repository, shit |
| 16:39.25 | gitguy | 2.6.18 headers |
| 16:39.25 | ldleworker | [TK]D-Fender, it still results in a hangup instead of a transfer |
| 16:39.39 | [TK]D-Fender | ldleworker: And you aren't showing me anything current. |
| 16:39.52 | ldleworker | What do you mean? |
| 16:39.58 | ldleworker | I just changed it to Dial() |
| 16:40.01 | ldleworker | I mean Dial() |
| 16:40.09 | ldleworker | meant* |
| 16:40.24 | [TK]D-Fender | ldleworker: If I don't see it, I don't care about your claims. Show me something real. |
| 16:40.33 | ldleworker | Like the AMI events? |
| 16:40.36 | R0land | [TK]D-Fender i cant seem to remember how to turn on debugging on! |
| 16:40.38 | R0land | im in cli.. |
| 16:40.48 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 16:40.48 | [TK]D-Fender | ldleworker: Like CLI output, like your new dialplan, like SOMETHING. |
| 16:40.56 | [TK]D-Fender | R0land: "sip debug" <- |
| 16:41.33 | R0land | k coming ight up |
| 16:42.08 | ldleworker | Okay here we go: http://pastebin.com/m7e132dbf |
| 16:42.24 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 16:42.58 | [TK]D-Fender | ldleworker: Whre's the CLI output? |
| 16:43.51 | ldleworker | I don't have access to it |
| 16:44.09 | ldleworker | Just the dialplan and AMI |
| 16:44.21 | ldleworker | I have a feeling its the way I'm doing the Dial command |
| 16:44.29 | ldleworker | Maybe the MusicOnHold flag |
| 16:44.32 | [TK]D-Fender | ldleworker: How do you not have access to CLI? |
| 16:44.34 | ldleworker | Or the way I'm passing a variable |
| 16:44.54 | ldleworker | [TK]D-Fender, I'm doing this work as a favor I owe. I'm in a remote location |
| 16:45.05 | [TK]D-Fender | ldleworker: SSH doesnt' care how far you are. |
| 16:45.18 | rob0 | ate his sshd |
| 16:45.26 | ldleworker | Ah well I wasn't aware you could login directly |
| 16:45.31 | R0land | [TK]D-Fender http://www.pastebin.ca/1001928 |
| 16:45.46 | ldleworker | [TK]D-Fender, no idea why it fails with that information? |
| 16:46.18 | [TK]D-Fender | ldleworker: Sorry, I'm not commenting blind... |
| 16:46.51 | [TK]D-Fender | R0land: Looking for s in spa (domain 192.168.0.60) SIP/2.0 404 Not Found |
| 16:46.54 | ldleworker | I wonder how to login to the SSH |
| 16:47.01 | [TK]D-Fender | R0land: Congratulations, go fix your dialplan. |
| 16:47.10 | R0land | hmm |
| 16:47.35 | [TK]D-Fender | ldleworker: Time to go learn Linux. |
| 16:47.43 | [TK]D-Fender | *nix |
| 16:48.55 | R0land | well i could set an incoming context.. bas to b honest! i dont know wht to set in it.. i know i wanna set a background command to have my audio file play.. as well as a waitforinput command to wait for the digits but honestly! i dunno how |
| 16:49.07 | R0land | i was wondering if you could help me out for a few seconds with this [TK]D-Fender |
| 16:49.13 | R0land | if you dont mind that is.. [TK]D-Fender |
| 16:49.43 | [TK]D-Fender | R0land: Its telling you right to your face exactly what exten in which context its looking for. GO MAKE IT. |
| 16:50.02 | R0land | mm good point k ill see wht to do |
| 16:50.02 | R0land | thanks :) |
| 16:50.12 | [TK]D-Fender | R0land: If you have no idea how to set up an IVR, go read the BOOK, and check out the WIKI |
| 16:50.17 | [TK]D-Fender | ~book |
| 16:50.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 16:50.19 | [TK]D-Fender | ~wikis |
| 16:50.20 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 16:50.24 | R0land | [TK]D-Fender thts wht im gonna do thank u :) |
| 16:50.27 | R0land | thnkas for hte help |
| 16:50.28 | ldleworker | [TK]D-Fender, wouldn't I have to stop my Asterisk server and restart it to gain CLI to it? |
| 16:50.41 | [TK]D-Fender | ldleworker: ..... |
| 16:50.42 | ldleworker | Or do you know of a way I can "take control" of the output of a process |
| 16:50.43 | *** join/#asterisk dmz (n=dmz@216.16.218.89.dyn-cm-pool46.pool.hargray.net) |
| 16:50.56 | [TK]D-Fender | ldleworker: Have you every really worked with * before? |
| 16:51.19 | ldleworker | [TK]D-Fender, hell no. I'm doing this as a favor. Everything I know about it I learned it yesterday |
| 16:51.24 | ldleworker | So |
| 16:51.30 | glaz | asterisk -r |
| 16:51.32 | [TK]D-Fender | ldleworker: You're in over your head it seems... |
| 16:51.33 | ldleworker | Sorry for that short comming |
| 16:51.34 | jbeez | So how's the recipient of this favor :P |
| 16:51.38 | jbeez | who's |
| 16:51.48 | glaz | pokes jbeez. |
| 16:51.48 | rob0 | "man asterisk" |
| 16:51.51 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 16:51.54 | jbeez | hey glaz |
| 16:51.57 | ldleworker | The thing is, I don't want to disrupt service |
| 16:52.22 | glaz | I just finished the php interface I made to listen to recorded calls. pretty sweet. |
| 16:52.35 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:54.29 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:54.30 | *** join/#asterisk Tako-san (n=jmkiffia@24.108.192.144) |
| 16:54.33 | *** join/#asterisk LCPodcasts (n=LCPodcas@adsl-158-159-221.msy.bellsouth.net) |
| 16:54.40 | kombi | how do I tell how many callers are in a queue? |
| 16:54.42 | *** join/#asterisk PepOSX (n=angeldav@200.90.123.15) |
| 16:54.49 | smash- | how do i reload asterisk again |
| 16:54.50 | smash- | ? |
| 16:54.54 | kombi | reload |
| 16:55.11 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:56.32 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:57.38 | *** join/#asterisk RobH (n=RobH@dsl017-048-227.sfo4.dsl.speakeasy.net) |
| 16:57.50 | smash- | please help me when i call the did pointed to this ext |
| 16:57.55 | smash- | Apr 29 09:57:27 WARNING[31139]: pbx.c:2491 __ast_pbx_run: Channel 'Zap/2-1' sent into invalid extension '500' in context 'external', but no invalid handler |
| 16:57.56 | smash- | <PROTECTED> |
| 16:58.35 | [TK]D-Fender | smash-: You tried dialing a # in an IVR that doesn't have a match and you have not "i" exten. |
| 16:58.45 | [TK]D-Fender | kombi: "show queues" |
| 16:58.57 | kombi | thanks fender! |
| 16:59.28 | RobH | Anyone have an issue where voicemail email notifications with attachments come through as multi-part messages rather than a single email with attachment? |
| 16:59.48 | gitguy | i found that my vps provider have a custom built kernel.... and they didn't included the linux-headers |
| 16:59.55 | gitguy | what can i do in this case? |
| 17:00.25 | ldleworker | [TK]D-Fender, http://pastebin.com/m43a6c72a Thanks for the help btw |
| 17:01.16 | [TK]D-Fender | ldleworker: - Executing Dial("IAX2/ics.iax.trunk-42", "IAX2/ics.iax.trunk/1|30| m(none)") in new stack <- # sure doesn't look legit to you does it? |
| 17:01.49 | ldleworker | It doesn't. Is it the way I'm passing the variable to the context? |
| 17:02.27 | [TK]D-Fender | ldleworker: Where did you you read your syntax formatting from? |
| 17:02.37 | ldleworker | voip.info |
| 17:03.05 | Katty | [TK]D-Fender: what's a good name for a fox. |
| 17:03.51 | ldleworker | Variable: <Variable 1>=<Value 1><CRLF> |
| 17:03.54 | [TK]D-Fender | Katty: "Fluffy" |
| 17:04.06 | Katty | [TK]D-Fender: cute, but thinking more.... serious. |
| 17:04.13 | ldleworker | obviously in my real code I've replaced mynumber and theirnumber with real phonenumbers |
| 17:04.47 | Katty | [TK]D-Fender: http://youtube.com/watch?v=gZziQsuy7N8 <- as soon as i have a permit.... |
| 17:06.08 | Katty | wonders what a fox in a doggie sweater would look like |
| 17:07.02 | [TK]D-Fender | ldleworker: I'd go verify... |
| 17:07.21 | ldleworker | Verify what? |
| 17:08.06 | [TK]D-Fender | ldleworker: Whats really getting passed around, and perhaps reconsider how you're dilaing out to the target #. |
| 17:08.36 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 17:10.11 | ldleworker | I think its a problem with the fact that doing Variable: PHONE=8155552896 |
| 17:10.25 | ldleworker | Doesn't create a ${PHONE} variable in the context the originate is going to |
| 17:11.50 | [TK]D-Fender | ldleworker: Then do it another way |
| 17:12.14 | ldleworker | Isn't that how you're supposed to pass variables into a context? |
| 17:12.27 | killmel8tr | [Apr 29 13:07:43] ERROR[2643]: chan_sip.c:15468 sipsock_read: We could NOT get the channel lock for SIP/5060-081de088! |
| 17:12.27 | killmel8tr | [Apr 29 13:07:43] ERROR[2643]: chan_sip.c:15469 sipsock_read: SIP transaction failed: 00C0.95CA.E4E0.4817.54BA.0398 |
| 17:12.27 | killmel8tr | [Apr 29 13:07:43] WARNING[2643]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission 00C0.95CA.E4E0.4817.54BA.0398 for seqno 7159 (Critical Response) |
| 17:12.30 | ldleworker | I don't want to change the dialplan and reload extensions each time I need to change the number |
| 17:12.31 | killmel8tr | yay |
| 17:13.00 | acxty | I want to redirect a call to a cellphone or telephone line that has nothing to do with the asterisk box, I have been reading about DIAL() but it seems I need to define the channels that I want to use. Can that be done with dial()? |
| 17:14.46 | [TK]D-Fender | acxty: If the call has nothing to do with *, how do you expect * do do anything about it? Telekinesis? Chan_psychic.so? |
| 17:14.49 | killmel8tr | just do dial(SIP/peer/1235551212) |
| 17:15.06 | killmel8tr | or out an analog port if thats your method |
| 17:16.35 | acxty | what I want to do, is that asterisk hangup that call, but send it to a cellphone |
| 17:16.38 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 17:17.00 | killmel8tr | ummm... dude.... well... |
| 17:17.38 | rupa | maybe buy callforward from your phone company and remove asterisk from the equation completely ? |
| 17:17.56 | killmel8tr | asterisk would generally stay in the media path unless there was a reinvite from your origination gateway to your termination gateway or something |
| 17:18.07 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:18.07 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:19.29 | [TK]D-Fender | acxty: You can't hangup a call AND send it out somewhere else./ |
| 17:19.37 | [TK]D-Fender | acxty: * sits in the middle of all calls. |
| 17:19.38 | R0land | [TK]D-Fender http://www.pastebin.ca/1001946 <<-- is this right! to recieve incoming calls! |
| 17:20.00 | [TK]D-Fender | R0land: You tell me... |
| 17:20.22 | R0land | [TK]D-Fender well i know theres something wrong.. bas cant figure out wht |
| 17:20.26 | kombi | How would one make the length of a queue available in php? |
| 17:20.44 | [TK]D-Fender | R0land: Go look at the debug. |
| 17:20.48 | acxty | [TK]D-Fender, Call to asterisk then asterisk call the cellphone, and acts as a bridge between them? can that be done? |
| 17:21.05 | [TK]D-Fender | acxty: if * has an interface it can dial out on, sure. |
| 17:21.27 | acxty | [TK]D-Fender, so cellphone won't work in this case |
| 17:22.00 | [TK]D-Fender | acxty: because? |
| 17:22.34 | [TK]D-Fender | kombi: AMI |
| 17:22.51 | acxty | [TK]D-Fender, done know, of what I have read, I need to add a channel, and the cellphone is from a different company, so I don't get how can I add the channel for it |
| 17:22.54 | *** join/#asterisk qdk_ (n=qdk@195.242.194.42) |
| 17:23.42 | [TK]D-Fender | acxty: What ways of readching the PSTN do you have on your server? |
| 17:25.25 | acxty | the only think I have is a network cable connected to the nic |
| 17:26.08 | R0land | [TK]D-Fender SIP/2.0 404 Not Found |
| 17:26.19 | [TK]D-Fender | acxty: well you need a way to be able to dial out to your cellphone. Go get a clue. |
| 17:26.21 | R0land | [TK]D-Fender umm wht is tht supposed to mean! |
| 17:26.36 | [TK]D-Fender | R0land: means its not finding what you're dialing. |
| 17:26.49 | ldleworker | Does anyone know how to, purely from AMI, without MeetMe, orignate two calls and bridge them? |
| 17:26.49 | R0land | [TK]D-Fender im not dialing anything, im recieving calls |
| 17:27.10 | [TK]D-Fender | R0land: Your SPA is dialing something into your * server. |
| 17:27.26 | R0land | hmm |
| 17:27.26 | [TK]D-Fender | ldleworker: The way you're already doing it. |
| 17:27.36 | [TK]D-Fender | ldleworker: Originate is it. |
| 17:28.09 | ldleworker | why doesn't Variable: PHONE=Anything result in ${PHONE} == "Anything" in the dialplan?! |
| 17:28.24 | ldleworker | Maybe Variable: isn't for dialplan variables? |
| 17:28.26 | ldleworker | Something else |
| 17:29.44 | *** join/#asterisk amstel92109_ (n=john@cpe-75-85-189-174.san.res.rr.com) |
| 17:30.00 | NovceGuru | Anybody happen to have a work around for pfsense/m0n0wall allowing nat'd clients access to an external tftp :( |
| 17:30.19 | amstel92109_ | i am having problems with my AsteriskGUI, i am getting a 404, nothing to see here message when logging in |
| 17:30.26 | [TK]D-Fender | NovceGuru: Try in #networking |
| 17:30.35 | jasonwoot | anyone get parity errors on TE410P with Dell PE servers? |
| 17:30.44 | [TK]D-Fender | amstel92109_: Please visit their support channel. |
| 17:30.51 | NovceGuru | well it had to do with a sip phone so I was stabbing in the dark, thanks though :) |
| 17:30.57 | amstel92109_ | what is the support channel? |
| 17:31.06 | amstel92109_ | i thought thats what this was |
| 17:31.08 | [TK]D-Fender | amstel92109_: Read the channel topic. |
| 17:31.12 | amstel92109_ | soryry |
| 17:33.08 | acxty | [TK]D-Fender, I was thinking, I have a peer from which I make the connection to the sip server, can that channel be use to make the call to the cellphone |
| 17:33.25 | [TK]D-Fender | acxty: Gee I don't know... can you dial out with it? |
| 17:33.32 | *** join/#asterisk doolph (n=doolph@201.218.103.170) |
| 17:33.45 | *** join/#asterisk autojack (n=owen@pdpc/supporter/active/autojack) |
| 17:33.47 | acxty | yes, but in asterisk don't know how |
| 17:35.09 | [TK]D-Fender | acxty: here's some inspiration : |
| 17:35.13 | [TK]D-Fender | ~jerjerguide |
| 17:35.13 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 17:38.43 | autojack | I've been using Asterisk with a VOIP termination provider and a DID to make international cell to cell calls; I dial my DID, it connects me to my * server, then I dial out from there to anyone. technically it works well, but the call quality is sometimes "muddy" when I get connected. when I dial direct, it's much clearer. |
| 17:38.59 | autojack | I know there are limitations with VOIP, but I'm wondering what simple steps I could take to try to improve things. |
| 17:39.10 | autojack | specific protocols/codecs to try? |
| 17:39.21 | autojack | I'm guessing most of the problem is with my provider. |
| 17:39.41 | *** join/#asterisk mort_gib (n=mjensen@77.208.43.208) |
| 17:40.17 | rupa | autojack, what codecs are you using? |
| 17:40.42 | autojack | rupa: er, I'm not even sure. I did a default Asterisk setup and it "just worked." |
| 17:40.47 | acxty | [TK]D-Fender, is not how to use DIAL(), instead how to use that channel to make the call for example. I did Dial(SIP/user) being user the line I have, so it call again to itself, but what I want to do is DIAL(SIP/cellphone) And that return the error that cannot create a sip channel and all that, The question is do, I need a card or modem to that kind of call? |
| 17:40.52 | autojack | I'm using IAX between myself and the VOIP provider. |
| 17:41.01 | autojack | but I know that's a protocol, not a codec. |
| 17:41.32 | rupa | autojack, when you have your call established, do a "aix show channels" -- it'll show the codec |
| 17:41.39 | [TK]D-Fender | acxty: No, you don't need a card. You don't need a card to GET a call from your provider do you? Of course not. Neither do you need one to PLACE A CALL. |
| 17:41.46 | autojack | OK |
| 17:41.50 | autojack | I see "format for the call is ulaw" |
| 17:42.26 | rupa | ok, so you are using what... 60kbps for each leg 120+. what is your net connection? |
| 17:42.42 | acxty | [TK]D-Fender, may you help me with the DIAL() syntaxis DIAL(SIP/cellphone) won't work, done have a clue on how to do it, all the documentation use Dial() only on there own channels |
| 17:42.52 | autojack | it's a server in a colo facility, so, some sort of high-bw link. |
| 17:43.00 | autojack | I have no set limit. |
| 17:43.14 | [TK]D-Fender | acxty: look at the sample and comple the style to the way you set up to receive calls from your provider. |
| 17:43.29 | [TK]D-Fender | autojack: switch from IAX2 to SIP. |
| 17:43.39 | rupa | autojack, is the quality any better if you use a softphone from your pc? |
| 17:43.52 | autojack | rupa: yes, oddly enough. considering that my PC is on a cable modem. |
| 17:44.10 | autojack | when I use the softphone, I am bypassing my * server. |
| 17:44.16 | autojack | I connect directly to my VOIP carrier and then dial. |
| 17:44.33 | rupa | softphone->asterisk->voip carrier -- how is that quality? |
| 17:44.47 | autojack | [TK]D-Fender: interesting, that's actually what my provider recommends also. not sure why I went with IAX initially. I can switch that easily. |
| 17:44.54 | autojack | rupa: I don't think I've tried that. |
| 17:45.18 | [TK]D-Fender | autojack: Many quality issues vanish along with IAX2... |
| 17:45.40 | autojack | interesting. some people have told me the oppostie. |
| 17:45.43 | autojack | opposite... |
| 17:45.43 | Qwell | [TK]D-Fender: welcome to 6 months ago |
| 17:45.53 | Qwell | 6? 9? something |
| 17:46.15 | ftp3 | what is a "DSO" ? |
| 17:46.20 | Qwell | ~ds0 |
| 17:46.21 | jbot | somebody said ds0 was Digital Signal 0 and is one 56kbps framed channel out of the 24 in a DS1 channel. |
| 17:46.25 | bkruse | ds0 |
| 17:46.38 | bkruse | jbot: Qwell++ |
| 17:46.53 | [TK]D-Fender | Qwell: Probably far newer than his install :) |
| 17:47.45 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 17:47.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:48.29 | acxty | [TK]D-Fender, do I have to add another channel on sip.conf and use that one to make the calls? |
| 17:48.57 | ftp3 | oh, thank you |
| 17:49.20 | Qwell | ftp3: depending on the context though, people often refer to "one channel" as a ds0 |
| 17:49.28 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 17:49.57 | [TK]D-Fender | acxty: You simply aren't getting it... go read some of these sample configs and actually pay attention to what theya re doing and how they access the resources they are showing oyi how to set up. |
| 17:50.37 | ftp3 | so, if someone charges $20 for a dso and $1 for a did, would the dso charge be a one time thing or monthly? (i know you would be guessing, but i am on my second day of learning about this stuff and I am still kinda confused by pricing since so many sites are so different) |
| 17:51.10 | Qwell | ftp3: monthly |
| 17:51.17 | autojack | so I'll switch over to SIP. any other suggestions for stuff to look at/test? |
| 17:51.37 | autojack | I can try softphone->asterisk->voip carrier tonight, but not right now. |
| 17:52.07 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 17:52.44 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 17:52.58 | ftp3 | qwell: so $21 for a phone number and one channel.. thats high :-).. Thanks! I am starting to learn this.. i wish companies would just make a simple price matrix instead of trying to trick you by hiding the total cost.. i mean, i will pay it or not.. hiding it isnt going to make it better |
| 17:53.04 | autojack | oh, I also have my DID routing over IAX. |
| 17:53.23 | [TK]D-Fender | autojack: well you've spelled out BW as not being an issue. So it either your provider at source, or mangling of the traffic in between |
| 17:53.40 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 17:53.49 | bkruse | ftp3: Actually, the bell charges ~$20 for a line to your house and DID |
| 17:54.06 | bkruse | just local calling, so depending on the plan, that is not bad. If that is the physical line rent, then maybe you should do some shopping :] |
| 17:54.20 | Qwell | hunmonk: interesting.. I like drupal's hostmasks |
| 17:54.53 | ftp3 | bkruse: yeah, but bell has perfect sound and no latency.. |
| 17:55.02 | ftp3 | bkruse: well.. mine anyway |
| 17:55.06 | Qwell | ftp3: and 2c/min |
| 17:55.11 | hunmonk | Qwell: we put that in awhile ago |
| 17:55.21 | bkruse | Qwell: Exactly. |
| 17:55.26 | Qwell | hunmonk: very nice addition |
| 17:56.08 | ftp3 | qwell, i live in Oregon.. its unlimited here.. i realize that in places on the east coast it is different, but its just one fee and unlimited incoming and local minutes and then their umlimited usa/canada is $15.00 mo |
| 17:56.22 | hunmonk | has anybody here compiled zaptel from source on an intel mac? i'm getting an error saying i need kernel headers, but i'm not having any luck finding where to download those :| |
| 17:56.23 | ftp3 | regardless, I want viop :-) |
| 17:56.37 | Qwell | hunmonk: linux, I assume? |
| 17:57.03 | hunmonk | Qwell: no, OSX |
| 17:57.04 | hunmonk | Qwell: all i really need is ztdummy for the timing stuff |
| 17:57.11 | Qwell | can't.. |
| 17:57.19 | hunmonk | bleh |
| 17:57.22 | hunmonk | alright |
| 17:57.39 | gitguy | hey i get this when i load ztdummy http://pastebin.com/m12931048 |
| 17:57.41 | unpaidbill | you can get an sbc for less than 200 bucks though |
| 17:57.41 | gitguy | anyone? |
| 17:57.50 | unpaidbill | or get a mac mini and put darwin on it |
| 17:58.24 | unpaidbill | fun little dedicated system! |
| 17:58.26 | acxty | [TK]D-Fender, my peer name is 5075010700, that is my telephone line. If I use DIAL(SIP/5075010700) It will dial itself, what I want to do is to use that peer to make a call outside, Been searching but done fine the way how to do it |
| 17:58.54 | *** join/#asterisk artz22 (n=artz@190.189.0.33) |
| 17:58.59 | artz22 | hi everybody |
| 17:59.08 | unpaidbill | hi dr nick |
| 17:59.12 | bkruse | gitguy: Was that version of zaptel compiled with that kernel? |
| 17:59.14 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:59.15 | bkruse | Does not look like it |
| 17:59.20 | artz22 | i'm having some issues with asterisk on xscale |
| 17:59.36 | unpaidbill | arcom board? |
| 17:59.37 | artz22 | when chan_sip.o gets loaded asterisk gets killed, how can I diagnose this? |
| 17:59.38 | gitguy | bkruse: zaptel 1.4.10 |
| 18:00.00 | bkruse | gitguy: I know, but did you compile that module with the kernel? Or just copy it in |
| 18:00.12 | gitguy | bkruse: yes i compiled zaptel 1.4.10 |
| 18:00.12 | bkruse | I would compile the module, the insmod ./ it |
| 18:00.26 | bkruse | insmod ./ztdummy.ko |
| 18:01.27 | gitguy | insmod: error inserting './ztdummy.ko': -1 Invalid module format |
| 18:02.14 | acxty | [TK]D-Fender, I make the call to the peer, but what I want to do is use the peer to make an outside call |
| 18:02.53 | gitguy | any ideas? |
| 18:03.21 | unpaidbill | yeah |
| 18:03.23 | bkruse | gitguy: It looks like you booted into another kernel, then tried to insmod it |
| 18:03.24 | unpaidbill | strip -S modulename |
| 18:03.28 | unpaidbill | try to reinsert |
| 18:03.46 | ldleworker | ARrrghghghghgh, can anyone else set channel variables from the AMI when originating a call!? |
| 18:03.47 | gitguy | bkruse: no, it's running the same kernel |
| 18:03.55 | unpaidbill | i think that's the command at least |
| 18:04.00 | bkruse | unpaidbill: It looks like (especially with the whole xen thing), that he compiled for one kernel, then tried to insmod it (already build modules) into the xen kernel |
| 18:04.34 | unpaidbill | ah |
| 18:04.43 | unpaidbill | that would do it too |
| 18:05.45 | gitguy | lrwxrwxrwx 1 root root 27 Apr 29 13:51 /lib/modules/2.6.16-xenU/build -> /usr/src/linux-2.6.16-xenU/ |
| 18:05.47 | unpaidbill | i did have an issue with a kernel module that required me to strip -S to get it to insmod though.. i think it was for the rt61 network drivers or something |
| 18:05.49 | gitguy | i made that symlink |
| 18:05.50 | gitguy | and hten compiled it |
| 18:05.52 | gitguy | then* |
| 18:06.04 | unpaidbill | horrible network drivers. |
| 18:06.46 | gitguy | unpaidbill: i did the strip -S ztdummy.ko but it doesn't work... |
| 18:07.41 | *** join/#asterisk Guza (n=Guza@84.41.116.162) |
| 18:07.47 | *** join/#asterisk _lando_ (n=a@pool-68-238-172-133.tampfl.fios.verizon.net) |
| 18:07.49 | Guza | hi |
| 18:07.58 | Guza | can someone help me with asterisk |
| 18:08.06 | hunmonk | Qwell: is there any way to get a timing source on OSX so that i can use music on hold? ztdummy is the only timing source i know of |
| 18:08.31 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 18:08.40 | Guza | i have sip client behind NAT |
| 18:08.47 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 18:08.50 | unpaidbill | yeah it just removes debugging stuff from the module, i guess it had an impact on the rt61 stuff because it's still in heavy development and probably had all sorts of crazy debugging stuff in it :/ |
| 18:09.12 | Guza | when i call someone i can hear them but thay can't hear me |
| 18:09.17 | unpaidbill | broken debugging stuff no doubt. like the damn driver. |
| 18:09.33 | _lando_ | Guza: is your asterisk server behind the NAT |
| 18:09.43 | Guza | yes |
| 18:10.02 | acxty | anyone may help me with this. My peer name is 5075040700 I want to dial 99921342 (This is a cellphone) how do I say to dial the cellphone using the peer DIAL(SIP/5075040700 ???????????? ) |
| 18:10.12 | Guza | but i nat all ports to asterisk |
| 18:10.17 | Guza | and all protocols |
| 18:10.21 | unpaidbill | nat=yes ? |
| 18:10.24 | Guza | yes |
| 18:10.31 | *** part/#asterisk artz22 (n=artz@190.189.0.33) |
| 18:10.36 | Guza | in my acc i was set nat=yes |
| 18:10.42 | gitguy | bkruse: should i get a system without xen? |
| 18:10.50 | gitguy | bkruse: to avoid this issues |
| 18:11.12 | bkruse | gitguy: Well, if you ever have to deal with zaptel, xen is NOT quite there for that |
| 18:17.39 | l2cache | What would you recommend for the best SIP provider with asterisk? I need the cheapest 30 concurrent channel plan I can get, unlimited minutes. |
| 18:18.35 | Qwell | l2cache: you aren't going to get multiple concurrent channels (especially 30) on an unlimited plan |
| 18:18.57 | gitguy | bipolar: so the answer is yes? |
| 18:19.25 | [TK]D-Fender | Guza: Read up : |
| 18:19.27 | [TK]D-Fender | ~sipnat |
| 18:19.28 | 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 |
| 18:19.39 | Guza | i fix it |
| 18:19.55 | Guza | tnx |
| 18:20.35 | Guza | i was add externip = my_public ip :D |
| 18:20.50 | Guza | and now works fine |
| 18:21.04 | killmel8tr | I have found that sometimes you need to set the localnet as well |
| 18:22.15 | gitguy | but ztdummy should work on a xen environment right? |
| 18:23.19 | killmel8tr | I thought I read something somewhere about not needing a timing souce with the 2.6 kernel |
| 18:23.25 | [TK]D-Fender | gitguy: Don't bet on it... |
| 18:23.33 | gitguy | [TK]D-Fender: ? |
| 18:23.38 | Qwell | killmel8tr: that would be incorrect |
| 18:23.40 | [TK]D-Fender | killmel8tr: No. |
| 18:24.01 | killmel8tr | well i dont use it for conferencing so I never messed with it and always have a card in anyhow |
| 18:24.14 | gitguy | should i get rid of this xen stuff? |
| 18:24.16 | killmel8tr | there was something new with the 2.6 kernel related to timing though |
| 18:24.52 | [TK]D-Fender | killmel8tr: Linux 2.6 kernels allow you to use th RTC timer as a timing source to use with Zaptel. |
| 18:25.05 | killmel8tr | oh ok, that was it then. ;) |
| 18:25.18 | [TK]D-Fender | killmel8tr: 2.4 needed a USB EHCI interface |
| 18:27.21 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-154-213.dsl.stlsmo.sbcglobal.net) |
| 18:27.49 | LemensTS | pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use my /etc/asterisk/dundi.conf file is editied out tho |
| 18:28.02 | ldleworker | YES OR NO: Variable: anyvariablename=anyvalue |
| 18:28.13 | ldleworker | ^^^ Correct syntax for originate via AMI?? ^^^ |
| 18:28.39 | gr0mit | l2cache, where do you want to call with these 30 channls? |
| 18:29.14 | *** join/#asterisk oej (n=olle@216.207.245.1) |
| 18:30.25 | Guza | is smart use extern ip ? |
| 18:30.40 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.142) |
| 18:31.04 | [TK]D-Fender | Guza: If you're behind NAT, its somewhat required... |
| 18:32.04 | Guza | what is consequents? |
| 18:33.26 | ftp3 | i need something so to allow me to receive voip phone calls thru my computer to a phone right? some box or something? |
| 18:34.06 | Qwell | what kind of phone? |
| 18:34.07 | gitguy | is zaptel 1.2.10 old? |
| 18:34.20 | gitguy | err |
| 18:34.25 | gitguy | 1.2.25 |
| 18:35.23 | LemensTS | as for my dundi problem, i removed dundi.con, then asterisk -vvvc and it went in...that is odd because the bind and port in dundi.conf were edited out anyways |
| 18:35.40 | Qwell | LemensTS: commenting it out just means to use the default |
| 18:36.31 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 18:37.27 | killmel8tr | you know I wonder.... if you are getting support for a digium card that is not working right using the free asterisk release, are you going to get the same level of support as if you were having problems with a card but has the business edition? |
| 18:37.56 | gitguy | can someone here confirm me that zaptel/ztdummy *wont* work on xen? |
| 18:37.59 | killmel8tr | i mean they seem to be working on it fairly good but I cant help but wonder. |
| 18:38.42 | LemensTS | Qwell: ahh ok that make sense. asterisk is still failing to start on a reboot. can start from asterisk -vvvc still....on the reboot it says Starting Asterisk PBX: asteriskUnable to open pid file `/var/run/asterisk/asterisk.pid`: No such file or directory |
| 18:38.53 | [TK]D-Fender | killmel8tr: Same support pretty much. BE offers access the the BE source (some locked stuff). All support is extra for that really. |
| 18:39.05 | killmel8tr | try it? |
| 18:39.14 | killmel8tr | .. |
| 18:39.21 | [TK]D-Fender | killmel8tr: I wouldn't. |
| 18:40.11 | killmel8tr | cool, thats what i was thinking... i mean you get in the same ticket system and seem to talk to the same engineers and stuff so.... |
| 18:40.25 | *** part/#asterisk Travo (n=travis@adsl-75-12-45-209.dsl.bumttx.sbcglobal.net) |
| 18:41.37 | killmel8tr | I had a funny thing today though.... I was talking to a front line guy and described my symptoms and asked if he had any ideas.... "well its usually a bug" <grins> |
| 18:41.55 | gitguy | [TK]D-Fender: ztdummy wont work on xen right? |
| 18:42.25 | [TK]D-Fender | gitguy: Might be a way but I wouldn't trust it... * needs a reliable timer source for things needing it. |
| 18:42.51 | *** join/#asterisk oej (n=olle@216.207.245.1) |
| 18:43.13 | gitguy | ok |
| 18:43.49 | *** join/#asterisk _gm (n=gmustafa@117.20.28.50) |
| 18:44.10 | *** join/#asterisk mocker (n=kyle@mocker.org) |
| 18:44.25 | ftp3 | qwell: well, i have a Philips VOIP3212G/37, its a "regular" and skype phone |
| 18:45.05 | ftp3 | qwell: so, i guess i need something to connect a regular phone to my computer.. right? my asterisk box (that is being installed today) is in a data center, not my office or house |
| 18:48.40 | *** join/#asterisk oej (n=olle@216.207.245.1) |
| 18:49.06 | gitguy | is there a server i could buy on internet running already without xen? |
| 18:49.19 | gitguy | to put asterisk on it asap |
| 18:49.33 | gitguy | wihout xen, without nat |
| 18:49.48 | gitguy | running on real hardware |
| 18:50.49 | gitguy | nat is not really a problem anymore |
| 18:50.53 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 18:51.21 | [TK]D-Fender | gitguy: go co-locate your own. |
| 18:51.32 | *** join/#asterisk luisjose (n=luisjose@nelug/coreteam/luisjose) |
| 18:51.38 | gitguy | [TK]D-Fender: i'm in paraguay, internet sucks here |
| 18:51.41 | [TK]D-Fender | gitguy: And what are you actually trying to accomplish? |
| 18:51.42 | luisjose | is there a way to replace flash for hold |
| 18:51.57 | [TK]D-Fender | luisjose: "flash for hold"? huh? |
| 18:52.16 | [TK]D-Fender | luisjose: on what? |
| 18:52.41 | gitguy | [TK]D-Fender: i'm trying to use meetme() and have all these fancy sound notifications, i'm using app_conference now |
| 18:52.52 | gitguy | [TK]D-Fender: i already have dial in/out working |
| 18:53.01 | [TK]D-Fender | gitguy: And why do you need some co-located server for this? |
| 18:53.13 | [TK]D-Fender | gitguy: Why not jsut run a server locally? |
| 18:53.49 | gitguy | [TK]D-Fender: this is for my boss... he lives in the US, I live in Paraguay |
| 18:54.06 | gitguy | if i could go there i would buy a local server there |
| 18:54.20 | gitguy | s/buy/build |
| 18:54.24 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:54.37 | [TK]D-Fender | gitguy: It'l likely cost a lot less then paying extra fees for hosting. |
| 18:54.49 | gitguy | yeah |
| 18:56.00 | ldleworker | Is there a way to list any linked asterisk boxes? |
| 18:57.25 | [TK]D-Fender | ldleworker: Linked to what? |
| 19:00.39 | *** join/#asterisk jetlagmk2 (i=jetlag@70.17.48.245) |
| 19:02.26 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:02.55 | M1s3ry | [TK]D-Fender, he's pretty idle... |
| 19:06.17 | *** join/#asterisk zacwolfe (n=zxwolfe@c-67-171-236-13.hsd1.or.comcast.net) |
| 19:07.55 | zacwolfe | Product Announcement: New Asterisk Visual IVR/Callflow Designer |
| 19:08.02 | zacwolfe | www.safisystems.com |
| 19:08.20 | zacwolfe | Check it out! free demo |
| 19:09.02 | *** join/#asterisk oej (n=olle@216.207.245.1) |
| 19:09.24 | LemensTS | hey TKD-Fender, remember you making fun of that one asterisk server i had with asterisk at home? Its debian now. Also that pid error was because the a@h config files were doing something, i restored the originals and am going to hand edit them back in |
| 19:11.57 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 19:12.52 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.229.111) |
| 19:12.59 | ftp3 | zacwolfe; no pricing... |
| 19:13.05 | shido6 | screenshots? |
| 19:13.08 | shido6 | no screenshots? |
| 19:13.08 | Cyorxamp | Lo folks, my manager.conf seems set correctly - but I still can't log in using it :S |
| 19:13.08 | zacwolfe | ah...sorry |
| 19:13.19 | zacwolfe | screenshots are there though |
| 19:13.25 | shido6 | Eugene? |
| 19:13.29 | zacwolfe | one sec |
| 19:13.38 | zacwolfe | Eugene OR! |
| 19:14.06 | zacwolfe | ok..the screenshots are hidden in a link at the bottom of the "products" page |
| 19:14.09 | gitguy | what about ztxen? |
| 19:14.14 | zacwolfe | http://www.safisystems.com/index.cfm?pageMode=screenshots |
| 19:14.20 | gitguy | would that work? |
| 19:14.37 | zacwolfe | haven't decided on pricing yet..still in alpha |
| 19:15.05 | Qwell | not open source? |
| 19:16.31 | Katty | [TK]D-Fender: you would NOT believe the hoops i'm having to jump through to obtain a wildlife permit. |
| 19:16.34 | [TK]D-Fender | gitguy: Keep trying to throw away money & cause problems... |
| 19:16.51 | [TK]D-Fender | Katty: Fox's are not domesticated... |
| 19:17.01 | Cyorxamp | Lo folks, my manager.conf seems set correctly - but I still can't login! (File: http://rafb.net/p/OHcvuH67.html ) (What I try to login... http://rafb.net/p/Sca1Rk33.html ) |
| 19:17.09 | Cyorxamp | Any ideas I'd be really happy :P |
| 19:17.14 | [TK]D-Fender | Katty: And I entirely believe it, my sister had 3 Servals |
| 19:17.33 | [TK]D-Fender | Katty: http://en.wikipedia.org/wiki/Serval |
| 19:17.42 | gitguy | [TK]D-Fender: heh, what else could i do? |
| 19:18.13 | [TK]D-Fender | gitguy: put a real server somewhere. You never stated your actual full NEEDS. |
| 19:19.05 | gitguy | [TK]D-Fender: my boss is a moron, he will say "but amazon ec2 is cool, but amazon ec2 is scalable, but amazon ec2 is blah blah blah" |
| 19:19.06 | Katty | [TK]D-Fender: i do not require a domesticated pet. |
| 19:19.12 | Katty | [TK]D-Fender: i am not the typical pet owner. |
| 19:19.13 | zacwolfe | Qwell: no sorry its a commercial product |
| 19:19.22 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 19:19.51 | [TK]D-Fender | gitguy: how does "but it doesn't &#@^%$ING work" dound to him? |
| 19:20.05 | gitguy | [TK]D-Fender: i will tell him that now |
| 19:20.37 | rupa | ec2 is great -- but wrong hammer for this problem |
| 19:20.50 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-59-180.lns10.syd7.internode.on.net) |
| 19:21.21 | gitguy | [TK]D-Fender: My boss: i heard about something about a 'timer' in the forums for AWS |
| 19:21.38 | gitguy | :S |
| 19:22.12 | [TK]D-Fender | gitguy: http://www.zazzle.com/i_hear_dumb_people_customized_mug-168670728950882139 |
| 19:22.49 | gitguy | haha |
| 19:23.47 | gitguy | i hear dumb people and i hate cheap asses |
| 19:24.07 | gitguy | well being cheap is not so bad |
| 19:24.11 | gitguy | but being dumb is |
| 19:24.51 | *** join/#asterisk stony (n=oloch@p57B3908D.dip0.t-ipconnect.de) |
| 19:24.53 | stony | re |
| 19:26.20 | rupa | i'm seeing some people talking about using asterisk on ec2 <shrug> |
| 19:26.21 | stony | i'm planning on writing a interface that uses hylafax to fax via app_[txfax|rxfax] - has this already been done or is this project still "open" ? |
| 19:27.08 | doolph | how do I restart samba? |
| 19:27.24 | doolph | because I made a modification to smb.conf |
| 19:27.27 | stony | /etc/init.d/samba restart |
| 19:27.30 | stony | rcsmb restart |
| 19:27.34 | stony | depends on the distro |
| 19:27.41 | stony | perhaps is best to do this via swat |
| 19:27.44 | stony | http://sambaserver:901 |
| 19:28.15 | doolph | oops wrong channel |
| 19:28.37 | [TK]D-Fender | stony: waste. IAXModem <---- |
| 19:28.48 | rupa | no zaptel in ec2 though -- so no timing which is what youa re seeing |
| 19:28.55 | stony | [TK]D-Fender: iaxmodem smells |
| 19:29.04 | stony | half of the time the faxes are cut off |
| 19:29.13 | stony | or it takes 4-5 trief to get the faxes through |
| 19:29.20 | stony | s/trief/tries/i |
| 19:29.47 | [TK]D-Fender | stony: Don't expect RXFax to be any better |
| 19:30.10 | stony | atm it works perfect |
| 19:30.47 | stony | best would be t38modem and gateway in asterisk |
| 19:30.56 | stony | but i think it'll take ages till this is in asterisk ... |
| 19:32.19 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 19:32.53 | *** join/#asterisk sbingner (n=john@pdpc/supporter/sustaining/sbingner) |
| 19:34.53 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 19:37.17 | gitguy | my boss suck |
| 19:37.45 | gitguy | i should rm -rf / his fucking server |
| 19:37.57 | b11d` | do it |
| 19:38.14 | b11d` | DO IT |
| 19:38.15 | b11d` | come on!!@! |
| 19:38.16 | b11d` | big man |
| 19:38.17 | b11d` | DO IT |
| 19:38.19 | b11d` | :) |
| 19:38.27 | lmadsen | rolls eyes |
| 19:38.29 | *** join/#asterisk edibrac (n=edibrac3@netblock-66-245-252-161.dslextreme.com) |
| 19:38.31 | b11d` | lol |
| 19:38.42 | stony | hey, don't become a reiser ;D |
| 19:39.19 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:40.31 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-70-59.lns10.syd6.internode.on.net) |
| 19:42.53 | _lando_ | lol |
| 19:43.16 | _lando_ | come on dude reiser that mf! |
| 19:43.22 | gitguy | reiser own |
| 19:43.37 | gitguy | i want to be like him when i grow up |
| 19:44.06 | *** part/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 19:44.36 | gitguy | jk xD |
| 19:46.38 | _lando_ | I have a linksys spa3102 (sipura) It's all working dandy. I use the FX0 as a trunk and the FXS connects to a couple analog stations. |
| 19:47.06 | _lando_ | I'm looking for a way to recognize the call waiting that comes over the FX0 |
| 19:47.49 | _lando_ | for example when I am on a call out that trunk on my cisco 7960 I can hear the call waiting beep, but I have not found a way to hook flash it. |
| 19:50.30 | _lando_ | on the same 7960 if I am on a call place via viatalk SIP trunk the call waiting works fine..infact awesome because I just put the first call on hold and pick up the second. |
| 19:51.36 | [TK]D-Fender | _lando_: I don't think there's a realistic way to signal the SPA to hook-flash... |
| 19:51.38 | *** join/#asterisk gitguy (n=diego@adsl-153-72.click.com.py) |
| 19:52.10 | _lando_ | I tend to aggree, just thought ide toss it out there for the guru's to chew on. |
| 19:53.08 | _lando_ | I did recently aqiured a TDM400 with FXO and FXS modules. It's installed and ready to go in my asterisk box, I'm just waiting on getting everything wired to my server closet. |
| 19:53.45 | _lando_ | I assume then that I can reliably hook flash with the digium tdm 400? |
| 19:54.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:58.30 | luisjose | [TK]D-Fender, when someone use "flash" i want it to acts as hold |
| 20:01.34 | [TK]D-Fender | luisjose: "flash" on what? |
| 20:03.51 | mocker | Who are the main GUI vendors for Asterisk right now? Switchvox, FreePBX, Fonality, ...? |
| 20:05.31 | zacwolfe | mocker there's also CogoBlue |
| 20:05.37 | LemensTS | can you install asterisk-addons after asterisk? |
| 20:05.39 | zacwolfe | never used it...looks cool |
| 20:06.27 | mocker | zacwolfe: Hmm, haven't heard of them. |
| 20:06.41 | [TK]D-Fender | LemensTS: Yes |
| 20:07.11 | mocker | Crap, switchvox requires you to buy their hardware? |
| 20:08.12 | zacwolfe | haha..so does Cogo I think |
| 20:08.40 | SwK | anyone have any hints for getting raw frames off a channel on a zaptel card... not raw voice frames, but really raw for the channel |
| 20:09.47 | [TK]D-Fender | _lando_: there are some PITA ways using features.conf |
| 20:12.13 | gitguy | [TK]D-Fender: what is my only choice if my boss don't want to get a dedicated server? |
| 20:12.16 | gitguy | [TK]D-Fender: app_conference? |
| 20:12.33 | M1s3ry | mocker, #switchvox , but it is not required |
| 20:13.04 | M1s3ry | mocker, if you want T1 or analog cards though, it must be digium |
| 20:18.28 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 20:23.23 | *** join/#asterisk nhuisman_work (n=nhuisman@dhcp36.IfA.Hawaii.Edu) |
| 20:24.03 | nhuisman_work | hey does anyone know of a good t1 failover type switch. I have a single t1 line but I want it to failover to whatever asterisk box is active. What do you folks use? |
| 20:24.59 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 20:26.26 | *** join/#asterisk oej (n=olle@216.207.245.1) |
| 20:27.21 | LemensTS | I have a tdm400p, channel 1 fxo went out. I took the module out, do i need to rename channe 2-3 to 1-3? Didnt know if it would work without a channel 1 or not] |
| 20:27.38 | LemensTS | i mean 2-4 to 1-3 |
| 20:28.28 | M1s3ry | shouldn't matter. I had a customer the other day who's provider had bchan set to 3-14, and dchan to 24 |
| 20:32.50 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-91-217.lns10.syd6.internode.on.net) |
| 20:33.31 | *** join/#asterisk xenonex (n=xenonex@89.218.235.34) |
| 20:38.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:02.23 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 22:02.23 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.2.28, 1.4.19.1, 1.6.0-beta8 (2008/04/22), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 22:02.23 | beighto | for example: Joe/Blow Phone:Email <---can I call the Phone and the Email variable separately in extensions.conf? |
| 22:05.01 | *** join/#asterisk ruied (n=ruied@bl7-223-94.dsl.telepac.pt) |
| 22:07.42 | acxty | where can I find good documentation on how to implement python on asterisk? |
| 22:08.00 | unpaidbill | for agi scripts? |
| 22:08.03 | mog | ac1djazz, pyst |
| 22:08.13 | mog | er acxty pyst |
| 22:09.00 | _ShrikE | anyone using the snom m3? |
| 22:09.57 | acxty | mog, I am only trying to send an receive some information by sockets, do you thing agi along can handle this, or will nee to use pyst |
| 22:10.16 | mog | pyst is just agi |
| 22:10.28 | mrbond82 | unpaidbill - suppose I have all the dialing rules in an #include file in the extensions.conf... how do I add to [globals]... is [globals](+) valid? |
| 22:10.57 | acxty | mog, so I don't need to install anything else only #!/usr/bin/python and start writing the code |
| 22:11.10 | acxty | and save the file as .agi with chmod 777 |
| 22:11.19 | unpaidbill | i'm not sure if you can specify a second globals section mrbond |
| 22:11.20 | unpaidbill | let me try |
| 22:11.57 | ruied | Hi have cdr being stored in a postgres table... I'm having a problem tracking the total time of a call! Example: An extension makes a call to the operator (A), the operator place the call on hold an makes a new call to the outside (B) than, the operator transfers (B) to(A). How can I get the total time of this call? I dont have anything that relates call (A), (B) and the transfer from (A) to (B)... is there any way that I can relate this calls |
| 22:11.58 | ruied | to make a total billsec related to the number dialed to the outside? |
| 22:13.26 | mog | agi is essentially just input output |
| 22:13.31 | mog | its fairly straight forward |
| 22:13.34 | mog | see voip-info |
| 22:14.36 | *** part/#asterisk OgRo (n=ogro@c95161d9.virtua.com.br) |
| 22:15.10 | unpaidbill | no you cant have two globals sections |
| 22:15.45 | mrbond82 | unpaidbill - but if you use [globals](+) in the #included file then it appends to [globals] I think |
| 22:16.01 | unpaidbill | hrm, maybe let me try |
| 22:16.44 | unpaidbill | not working |
| 22:16.48 | ruied | does my question makes sense? I'm trying to get a solution for days and I still don't know how to solve it...... |
| 22:17.10 | unpaidbill | i see on voip-info where it says that works |
| 22:17.17 | unpaidbill | but .. unless i spelled the variable wrong it's a no go |
| 22:17.36 | unpaidbill | haha i did spell it wrong |
| 22:18.12 | *** join/#asterisk metfan2007 (n=metfan20@201.103.142.225) |
| 22:19.26 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-66-31.bstnma.east.verizon.net) |
| 22:19.26 | unpaidbill | yeah it works mrbond..just make sure that the second instance of the [globals] section has the (+) next to it |
| 22:19.42 | unpaidbill | if you call your include before [globals] in extension.conf, then the extensions.conf section gets (+) |
| 22:19.58 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 22:20.01 | metfan2007 | Hi all!, I need to report the result of a call in Asterisk, like DIALSTATUS, hangup, noanswer, blah blah, but I'm dialing from a perl app that creates .call files, that first connect to a PSTN number and then to a internal Asterisk extension. Any idea how to monitor if PSTN call was answered? |
| 22:20.23 | *** join/#asterisk Horvath (n=Horvath@bas1-toronto26-1279484350.dsl.bell.ca) |
| 22:20.49 | unpaidbill | manager interface should be able to do that metfan, or if you dump your CDR info into mysql you can probably grab it near realtime |
| 22:21.02 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:23.28 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 22:41.14 | grandpapadot | Right, but the Spartan Gyro works better with my Stratus than the 611, regardless of the brand ... |
| 22:54.41 | acxty | hey guys, tell me if this is possible. I want to have several calls at the same time, but if I asterisk is handling a call a another person call, it doesn't answer and sound as busy. Can asterisk handle several calls using the same line? |
| 22:55.56 | RobH | acxty: you want a single POTs line to hold two independent, non-conferenced calls at the same time? |
| 22:56.08 | acxty | yes |
| 22:56.33 | acxty | if possible more than 2, like 20 - 25 |
| 22:56.37 | jackson__ | grandpapadot, I'm about headed out to fly the little trex now :) |
| 22:56.47 | RobH | I mean, there is call waiting, but my understanding is there is no way to split those two calls, over a standard plain old telephone line into different calls. |
| 22:57.04 | grandpapadot | Eh? lol, wrong channel, but cool that you got a TREX!!!!!!! |
| 22:57.06 | RobH | You can get into data lines and voip provididers to do it, but not on a single POTs line. |
| 22:57.22 | RobH | (however, i am often incorrect.) |
| 22:57.26 | jackson__ | grandpapadot, what channel were you intending on posting in so I joing the correct one? :) |
| 22:57.37 | acxty | RobH, I am using a SIP line |
| 22:57.48 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.135) |
| 22:58.14 | RobH | acxty: Ok, folks are calling you over a sip connection or over a plain old telephone line? |
| 22:58.45 | acxty | I have a sip connection on asterisk, I am calling from a cellphone and a normal telephone line |
| 22:59.21 | RobH | so sip is your * servers incoming connection and you are getting a busy tone on your phone trying to call it when anohter call is already setup on the sip connection to * |
| 22:59.23 | RobH | ? |
| 23:00.11 | RobH | If you are using a sip provider, you need to check and see if they limit your number of simultaneous channels. |
| 23:00.20 | ManxPower | Chances are it's a NAT issue |
| 23:00.44 | RobH | You have awoken ManxPower, yea.... he knows waaaayyyy more about this than me. |
| 23:00.53 | RobH | ;] |
| 23:01.18 | acxty | RobH, so it is not on asterisk configuration, nothing I can do if it only give me one simultaneous channel |
| 23:02.02 | RobH | acxty: IF your sip provider only allows a single simultaneous connection, then yer stuck. However, you should check with that before saying yer stuck. ManxPower suggests NAT issues, you will want to check that as well. |
| 23:02.38 | acxty | I don't have any NAT, I connected trough a public ip address directly to them |
| 23:03.10 | RobH | if there is no firewall/NAT between your server and your sip provider, then yea, check and see if they allow multiple channels at the same time |
| 23:03.17 | RobH | i know some of the cheaper sip providers do not. |
| 23:04.57 | acxty | will call them. Second issue, they don't allow me to call to cellphones also |
| 23:10.54 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:18.13 | *** join/#asterisk seanbright (n=sean@c-69-251-175-43.hsd1.md.comcast.net) |
| 23:18.27 | metfan2007 | can I retrieve the hangup cause in a "failed" standard extension? |
| 23:18.52 | gitguy | what do you usually do when the boss is a fucking moron that insists on doing things that you know that wont work? |
| 23:19.00 | ManxPower | Most flat rate monthly plans limit you to one or two channels (the 2nd channel is for three-way calling), most per/min plans allow you unlimited number of channels |
| 23:19.04 | Qwell | gitguy: quit |
| 23:19.29 | ManxPower | metfan2007: define "standard extension" |
| 23:19.38 | ManxPower | gitguy: let him |
| 23:21.08 | gitguy | i'll try to change his mind once again, otherwise i'll gtfo |
| 23:21.13 | ManxPower | gets bored waiting for metfan2007 to respond and goes back to work |
| 23:22.19 | metfan2007 | ManxPower: using the .call files to originate calls you can use the "failed" asterisk extension if the call does not complete, but I need to know the cause of that "failed", if it is busy, hangup, chanunavail, blah |
| 23:24.10 | ManxPower | Just got a message on the teliax web site " Rejected!! You may be an evil hacker. " |
| 23:24.11 | ManxPower | 8- |
| 23:24.53 | ManxPower | metfan2007: not familiar with the "failed" extension. I suggest you use chan_local and then handle it all in the dialplan the normal way |
| 23:25.22 | ManxPower | metfan2007: the .call file really runs exten => failed,1,whatever? |
| 23:25.26 | ManxPower | what context does it run in? |
| 23:25.43 | grandpapadot | jackson__: Sorry, was away, we were in #cx (carbon extreme) but we're pretty much done now. |
| 23:25.53 | metfan2007 | ManxPower: yes sr http://www.voip-info.org/wiki/index.php?page=Asterisk+standard+extensions |
| 23:26.14 | ManxPower | metfan2007: try looking in Asterisk official documentation |
| 23:27.03 | ManxPower | metfan2007: I wonder why that does not set HANGUPCAUSE or DIALSTATUS |
| 23:27.27 | ManxPower | In any case, I always use Local/1234@blah as the channel to dial, then handle it all in the dialplan |
| 23:28.33 | *** join/#asterisk [T]an2 (n=ckwall@86.sub-70-192-39.myvzw.com) |
| 23:29.33 | ManxPower | ~itsp |
| 23:29.34 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 23:29.34 | metfan2007 | ManxPower: do you know if I can use the variables that I use in the call file "Set" in the Local extension? |
| 23:29.53 | ManxPower | ~itsplist-us |
| 23:29.53 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 23:29.58 | ManxPower | metfan2007: yes they are |
| 23:30.26 | [T]an2 | I have my phones connecting to an asterisk server pbx, then the asterisk pbx connects to an asterisk "T1" gateway. I am running into an issue where if for whatever reason the pbx crashes (which is a whole different issue I am dealing with) the cli starts spouting out hdlc errors and eventually my d-channel will drop for about 30 seconds. |
| 23:30.31 | ManxPower | Teliax's new signup procedure just plain sucks |
| 23:30.42 | [T]an2 | I am looking for some assistance in narrowing down why this happens and getting it to stop. |
| 23:30.54 | [T]an2 | I have verified that the t1 card is on its own irq |
| 23:31.07 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:31.15 | ManxPower | [T]an2: HDLC errors mean "line errors", it's usually an IRQ latency problem, but can be a bad cable or bad T-1 |
| 23:31.36 | jblack | manxpower: I can't stand their authentication page. It doesn't work for me at all. I have to go to the old site via the ip |
| 23:31.48 | jblack | It's as if they don't undersatnd who their customers are |
| 23:32.14 | ManxPower | jblack: For THREE accounts I tried signing up (three different customer) not a single managed to bill the credit card. |
| 23:32.25 | [T]an2 | yeah, we talked about this last week a little bit. I verified the irq is unique. and the cable is brand new. i have swapped out the card to make sure that was not the issue. and the phone company maintains that it is not an issue on their end. |
| 23:32.31 | [T]an2 | are there any other possibilities? |
| 23:32.38 | jblack | broken driver? |
| 23:32.50 | jblack | Bad bios implementation? |
| 23:32.57 | ManxPower | [T]an2: no,. HDLC errors means Asterisk got corrupted data from the T-1 (on the D-Channel). |
| 23:33.24 | ManxPower | [T]an2: I did NOT say "IRQ CONFLICT", I said "IRQ LATENCY" |
| 23:33.33 | *** part/#asterisk beighto (n=chatzill@65.78.162.14) |
| 23:33.48 | [T]an2 | ok, understood on the latency. how can i check for that? |
| 23:33.55 | jblack | Try a building newer drivers, using a newer kernel, replacing the motherboard. in that order |
| 23:33.59 | ManxPower | You can do a basic IRQ latency test by using zttest and making sure your results are within normal ranges |
| 23:34.08 | *** join/#asterisk oej (n=olle@216.64.24.250) |
| 23:34.25 | ManxPower | jblack: I'd turn off the onboard ethernet to start out with, as well as disable all motherboard devices you don't need. |
| 23:34.38 | [T]an2 | ***google'ing for normal ranges*** |
| 23:35.08 | ManxPower | In EVERY instance of HDLC errors that I've experienced were fixed by either not using the onboard ethernet or replacing the motherboard with a different model |
| 23:35.13 | jblack | You think it possible he has pci bus contention? |
| 23:35.19 | [T]an2 | will zttest affect connected calls? |
| 23:35.31 | jblack | Yeah. I'm inclined to blame the bios implementation myself. |
| 23:35.34 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:35.38 | ManxPower | [T]an2: you can try asking what the normal range would be on #asterisk-dev, but don't try to get end user support there. |
| 23:35.51 | [T]an2 | nope... gone to google for that one |
| 23:36.15 | ManxPower | [T]an2: I don't think you can even run zttest if Asterisk is running. |
| 23:36.20 | [T]an2 | ok |
| 23:36.30 | [T]an2 | yeah, reading through the manpage on it. |
| 23:36.42 | ManxPower | [T]an2: what span are you getting T-1 sync from? The telco span or the span going to the other Asterisk |
| 23:37.07 | ManxPower | If you did something stupid like try to get sync from the other Asterisk system, that could explain all your issues |
| 23:37.31 | [T]an2 | the telco span. to be specific, the PRI goes into my T1 gateway then from there it is sip over ethernet to the pbx level |
| 23:38.33 | ManxPower | so exactly what happens? |
| 23:38.35 | [T]an2 | syncing is where i get confused... I know I am not syncing to the other server because it is eth to that server rather than t1.... gimme a sec to log into that server and I will paste my zap configs |
| 23:38.45 | ManxPower | You said when the gateway server goes down you get HDLC errors |
| 23:38.51 | [T]an2 | sorry... |
| 23:38.59 | ManxPower | sync means nothing unless you are on a T-1 |
| 23:39.10 | ManxPower | I assumed you has a T-1 between the two servers. |
| 23:39.29 | ManxPower | As far as I can tell you are getting HDLC errors when your server is crashing. In that case what the hell are you asking here for? |
| 23:39.59 | ManxPower | T-1/PRI <-> Asterisk <-> unknown <-> Asterisk <-> SIP phones |
| 23:40.02 | ManxPower | Is that correct? |
| 23:40.11 | [T]an2 | if the pbx level goes down, that is when I start to see this problem. it may be completely unrelated, but i have not seen this issue unless I have crashed at the pbx level. I see the hdlc errors then the d-channel errors on the gateway level. |
| 23:40.21 | [T]an2 | sorry was typing as you were asking. let me answer... |
| 23:41.10 | [T]an2 | pri<-> asterisk (gateway level)<->Point to Point ehternet<->asterisk (pbx level)<->sip phones |
| 23:41.24 | ManxPower | Now what system is crashing? |
| 23:41.34 | [T]an2 | pbx level |
| 23:41.43 | [T]an2 | hdlc errors happen on the gateway level |
| 23:42.03 | [T]an2 | i thought it may be related to calls that were dropped between the two systems when the pbx crashed |
| 23:42.06 | ManxPower | [T]an2: put your /etc/zaptel.conf on pastebin.ca |
| 23:42.26 | [T]an2 | yep... hang on. was working on that. gonna be a sec. on cellular modem right now. |
| 23:42.47 | ManxPower | then I'm going for a smoke |
| 23:43.13 | zacwolfe | in case you missed it: |
| 23:43.21 | zacwolfe | ANNOUNCE: NEW Visual callflow/IVR designer and server for Asterisk |
| 23:43.28 | zacwolfe | free non-time-restricted demo avail www.safisystem.com |
| 23:44.46 | [T]an2 | ManxPower: /etc/zaptel.conf: http://pastebin.ca/1002389 |
| 23:45.09 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 23:48.21 | *** join/#asterisk oej_ (n=olle@216.64.24.250) |
| 23:50.02 | jblack | um, anyone going to do anything about zacwolfe spamming the channel? |
| 23:50.28 | zacwolfe | umm...its relevant isnt it? |
| 23:50.41 | jblack | On freenode? Not in my opinion. |
| 23:50.46 | zacwolfe | only my second msg |
| 23:50.57 | zacwolfe | ok sry..will refrain from now |
| 23:51.37 | jeffspeff | hi365, I'm trying to get a forwarding button press function to work. I posted on the forums, and the code that somebody replied with didn't work. I have the code from the menu, and the error generated on the forum. http://forums.digium.com/viewtopic.php?t=22036 Could somebody please take a look at this and tell me what I'm doing wrong? Thanks. :) |
| 23:52.31 | jeffspeff | *hi |
| 23:52.36 | jblack | Oh, he just wants standard ivr stuff. |
| 23:52.45 | jblack | jeffspeff: Hold a sec. |
| 23:52.51 | jeffspeff | yes, but i can't get it to work. thanks. |
| 23:55.08 | jblack | http://pastebin.ca/1002400 |
| 23:55.11 | jblack | Should be reasonably close |
| 23:55.33 | jblack | You might want to add a 10 sec wait after telling them to press 9 if emergency |
| 23:55.53 | [T]an2 | ManxPower: I dont know if any of this info is relevant, but I pasted it anyway: http://pastebin.ca/1002401 |
| 23:55.57 | jeffspeff | true |
| 23:56.07 | jeffspeff | a waitexten |
| 23:56.13 | ManxPower | [T]an2: make the 2nd field a 0 for all the ethernet stuff |
| 23:56.39 | ManxPower | jeffspeff: We don't support GUIs here. |
| 23:56.48 | [T]an2 | actually its 4 pris going in. D chan is on span 1. Its nfas |
| 23:57.04 | ManxPower | [T]an2: Then I have no idea how your two servers are communicating |
| 23:57.06 | jeffspeff | i'm not asking anything about the gui. its the strings i'm trying use for a dail feature |
| 23:57.21 | jeffspeff | *dial |
| 23:57.24 | [T]an2 | ManxPower: sorry... I am really bad at explaingin.... |
| 23:57.41 | ManxPower | jeffspeff: fix your Dial line |
| 23:57.49 | [T]an2 | 4 pris into the t1 card. then over the ethernet port on the network i am delivering sip to the other server. just a sip peer |
| 23:57.57 | jeffspeff | what's wrong with it, that's what i can't figure out... thanks. |
| 23:58.01 | ManxPower | you apparently don't have a [1234567] section of sip.conf |
| 23:58.28 | ManxPower | [T]an2: So you are using SIP to communicated between the two systems? |
| 23:58.35 | [T]an2 | yes |
| 23:58.45 | jeffspeff | I'm wanting it to dial my cell phone number, not an extension... |
| 23:58.57 | jeffspeff | i'm still learning here, how do i need to go about that? |
| 23:58.57 | ManxPower | [T]an2: telling us that in the first place would have saved us much time |
| 23:59.09 | ManxPower | jeffspeff: I suggest you go read the book |
| 23:59.11 | ManxPower | ~book |
| 23:59.12 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:59.25 | [T]an2 | ManxPower: sorry |
| 23:59.27 | ManxPower | jeffspeff: You don't even know enough to ask the right questions at this point. |
| 23:59.34 | jeffspeff | i've read the dial sections of it... |
| 23:59.58 | ManxPower | jeffspeff: then you understand that how you use Dial depends on how you designed your system. |