| 00:00.11 | aiksa[LV] | now, after correct initialization of those spans i see them as YEL alarmed |
| 00:00.12 | *** join/#asterisk Tuxguy (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 00:00.14 | Micc | hardwire, I don't think I need it. |
| 00:00.21 | Tuxguy | Is The win32 version supported here? |
| 00:00.22 | aiksa[LV] | which is ok because asterisk is not started yet |
| 00:00.24 | [TK]D-Fender | LOL |
| 00:00.28 | [TK]D-Fender | Tuxguy: No |
| 00:00.41 | hardwire | Micc: it's fun |
| 00:00.43 | hardwire | aktually |
| 00:00.43 | Tuxguy | Is there a channel? |
| 00:00.47 | Keizer | hardwire: Wow |
| 00:00.53 | hardwire | Keizer: ? |
| 00:01.03 | aiksa[LV] | now in zapata.conf i have signalling set as pri_net and the other as pri_cpe |
| 00:01.07 | hardwire | Micc: it's fun to register your numbers somewhere like e164.org and get calls in all over sip |
| 00:01.07 | [TK]D-Fender | Tuxguy: Nope |
| 00:01.09 | Keizer | hardwire: I just used your script |
| 00:01.12 | hardwire | form other providers |
| 00:01.12 | Keizer | It gives me errors |
| 00:01.14 | hardwire | Keizer: nice |
| 00:01.17 | hardwire | yer doing it wrong |
| 00:01.18 | Keizer | But it did have a result |
| 00:01.24 | Keizer | <PROTECTED> |
| 00:01.25 | Keizer | 208.50.100.0/26 dev eth0 proto kernel scope link src 208.50.100.28 metric 100 |
| 00:01.27 | Keizer | default via 208.50.100.62 dev eth0 |
| 00:01.27 | aiksa[LV] | when starting asterisk they keep staying with the YEL alarm |
| 00:01.29 | Keizer | default via 208.50.100.62 dev eth0 src 208.50.100.27 metric 10 y |
| 00:01.35 | aiksa[LV] | and i cant debug why |
| 00:01.51 | aiksa[LV] | [TK]D-Fender: perhaps you could shed a light on this? |
| 00:01.53 | Micc | hardwire, sounds interesting, you'll have to tell me more about that later. |
| 00:02.08 | Micc | I have to get a cheat sheet written up and the ivr menu finished for this customer. |
| 00:02.23 | Micc | you can hit exten 100 on default context now. |
| 00:03.07 | Keizer | Man I think Archlinux is sabatoging me |
| 00:03.34 | aiksa[LV] | what does " chan_dahdi.c:8413 dahdi_pri_error: !! Got a UA, but i'm in state 7" actually mean? |
| 00:04.51 | *** join/#asterisk tkbeat (n=tk@p54B95AA9.dip.t-dialin.net) |
| 00:07.04 | Keizer | hardwire: There has to be something else |
| 00:09.05 | hardwire | Keizer: practice makes perfect |
| 00:09.14 | hardwire | also.. check out vyatta and vrrpd and stuff |
| 00:09.54 | Keizer | hardwire: I don't think the IP metric did anything |
| 00:10.06 | Keizer | Something else is hindering |
| 00:10.15 | Keizer | I wish there was a log somewhere I could check |
| 00:10.47 | hardwire | Keizer: for.. one.. million.. dollars.. |
| 00:10.51 | hardwire | I'll just do it for you :P |
| 00:10.58 | hardwire | or sharks with lazer beams on their noggins |
| 00:11.57 | Keizer | Nov 6 11:43:13 Aster1 mgmtd: [2282]: ERROR: on_listen attach server socket failed |
| 00:11.59 | Keizer | Nov 6 17:21:40 Aster1 mgmtd: [2282]: ERROR: on_listen attach server socket failed |
| 00:13.00 | *** join/#asterisk kj5t (n=steve@65-120-138-35.dia.static.qwest.net) |
| 00:14.26 | hardwire | hah.. orkid is a Micheal.. so is Micc |
| 00:14.39 | hardwire | I got info emails from both of you.. and it was confusing |
| 00:15.19 | kj5t | I am having a strange problem with my PAP2T, not sure if this is an asterisk problem or a PAP2T problem. I just configured three new SIP accounts and three PAP2TS, two of them work fine. The third, when I call it from an extension on the local server (the PAP2T is connecting to a server in Dallas) the call connects, I can also call the telephone number assigned to it. When I call my local extension from the phone connecting to the Dall |
| 00:15.29 | Micc | I put my name in the subject. |
| 00:17.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:19.30 | kj5t | okay, just confirmed that it does the same thing when trying to call phones on the server it is connected to if I use the extension. However, when I dial the actual mumber it goes through |
| 00:24.02 | *** join/#asterisk jjshoe (i=jjshoe@cpe-76-175-157-237.socal.res.rr.com) |
| 00:31.30 | johnakabean | <PROTECTED> |
| 00:31.47 | Keizer | hardwire: I exhausted everything |
| 00:32.00 | Keizer | I don't think the problem is an IP metric |
| 00:32.09 | Keizer | I disabled all the port blocking |
| 00:32.12 | Keizer | So it's not that |
| 00:32.21 | Keizer | The service IP is in the route |
| 00:32.24 | Keizer | The metric is changed |
| 00:32.36 | Keizer | Asterisk is in the haresources |
| 00:32.45 | Keizer | So it's something completely different |
| 00:32.55 | hardwire | such is being an admin |
| 00:33.06 | Keizer | Yup |
| 00:33.14 | hardwire | do you have .27 assigned to .28 atm? |
| 00:33.39 | orkid | hardwire: reply eta? |
| 00:34.51 | Keizer | hardwire: Yep |
| 00:36.19 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 00:37.17 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.4) |
| 00:38.08 | *** join/#asterisk Ariel_Calzada (n=aricalso@adsl190-29-99-227.epm.net.co) |
| 00:38.42 | hardwire | Keizer: ip route show -> pastebin |
| 00:39.07 | hardwire | orkid: you know.. I dunno. I'm sorta wondering where the guy went that handles this stuff :) |
| 00:39.54 | Keizer | I'm using FreePBX and someone just told me there's an issue with default install and multiple networks |
| 00:40.03 | Keizer | But he stopped after saying that so I am searching google |
| 00:40.08 | Keizer | But google doesn't have the info on it |
| 00:42.28 | Keizer | ip route show |
| 00:42.30 | Keizer | 208.50.100.0/26 dev eth0 proto kernel scope link src 208.50.100.28 metric 100 |
| 00:42.32 | Keizer | default via 208.50.100.62 dev eth0 |
| 00:42.34 | Keizer | default via 208.50.100.62 dev eth0 src 208.50.100.27 metric 10 |
| 00:42.36 | Keizer | hardwire: There it is |
| 00:42.39 | Keizer | That looks correct to me |
| 00:42.41 | orkid | hardwire: lol |
| 00:42.46 | orkid | hook a brotha up! |
| 00:46.22 | hardwire | punches Keizer |
| 00:46.27 | *** part/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 00:46.30 | orkid | :O |
| 00:46.55 | Keizer | hardwire: Ouch |
| 00:47.00 | Keizer | Sup too many routes? |
| 00:47.35 | hardwire | http://pastebin.ca/1248319 |
| 00:47.38 | hardwire | make it look like that |
| 00:47.46 | hardwire | either way.. none of that may matter |
| 00:47.46 | orkid | what size should i use for packets when testing with iperf? hmm |
| 00:47.53 | hardwire | have you tried just setting .28 to be .27? |
| 00:48.08 | hardwire | maybe something is universally wrong with that IP? |
| 00:48.25 | hardwire | orkid: I usually use the iperf homepages examples |
| 00:48.27 | *** join/#asterisk RobH (n=RobH@dsl017-048-227.sfo4.dsl.speakeasy.net) |
| 00:48.44 | hardwire | but I thought there was a pretty little section on that in the voip-info wiki |
| 00:49.58 | orkid | yea, for iax2 size. i'm just wondering what -l size to set.. ie. what rtp uses |
| 00:50.05 | orkid | i guess i could sniff a call |
| 00:50.29 | Keizer | hardwire: How do I delete the default that doesn't specify IP src? |
| 00:50.49 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:52.34 | Keizer | lol I am stuck with one that won't remove |
| 00:52.44 | Keizer | The one before the src .27 line |
| 00:53.20 | Keizer | route del default |
| 00:54.12 | Keizer | hardwire: It works now G |
| 00:54.14 | Keizer | Thanks mang |
| 00:54.59 | hardwire | pastebin your routes again |
| 00:55.00 | hardwire | I wanna see |
| 00:55.04 | hardwire | goes to get the ruler |
| 00:55.14 | hardwire | if you did it weird.. I'm gonna get your knuckles |
| 00:55.36 | hardwire | orkid: use the NEB charts for some of that |
| 00:55.52 | hardwire | http://www.voip-info.org/wiki/view/Bandwidth+consumption |
| 00:56.16 | hardwire | I forgetz what IAX2's header size is |
| 00:57.10 | hardwire | it seems to be like 16kbps |
| 00:57.21 | hardwire | and the codecs are 20msec |
| 00:57.45 | hardwire | so maths.. yay! |
| 00:57.55 | orkid | i used -l 1280 ==>64*20 |
| 00:58.04 | hardwire | nice |
| 00:58.17 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 00:58.22 | *** join/#asterisk erikcw (n=erik@ip17.theneptune.sea.ygnition.net) |
| 00:58.41 | orkid | 64kbps * 20ms |
| 00:58.49 | hardwire | orkid: do you need IAX2 for termination? |
| 00:58.50 | erikcw | I'm trying to choose between PBX in a Flash and Elastix. Can anyone give me some pointers on which way to go? |
| 00:59.05 | hardwire | erikcw: install both.. play a lot.. |
| 00:59.05 | orkid | betamax is really killing me |
| 00:59.22 | orkid | i even get breaks in conversation sometimes |
| 00:59.30 | hardwire | who? |
| 00:59.36 | erikcw | thanks hardwire |
| 00:59.45 | hardwire | erikcw: sarcasm.. accepted. |
| 01:00.11 | hardwire | if theres one thing nobody needs to do is decide on a PBX in a hurry |
| 01:00.17 | hardwire | take your time and figure out what works best |
| 01:00.25 | hardwire | test druid out while you are at it. |
| 01:00.38 | erikcw | ha! seriously though. Where can I find a high level overview of where each excels? Google hasn't turned up much |
| 01:00.52 | hardwire | dunno.. |
| 01:00.53 | hardwire | I just download |
| 01:00.55 | hardwire | install |
| 01:00.58 | hardwire | say nope a lot |
| 01:01.04 | hardwire | then just end up using asterisk all by it's lonesome |
| 01:01.28 | Keizer | For the win |
| 01:01.39 | Keizer | I have the asterisk handbook right here |
| 01:02.15 | hardwire | Keizer: you're supposed to plug in dialplan with your left hand and have your right palm rest on the handbook. |
| 01:02.22 | Keizer | lol |
| 01:07.13 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 01:07.41 | *** part/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 01:17.05 | Keizer | hardwire: It's rather difficult to get the ip del to match on all cases |
| 01:17.25 | Keizer | I feel like the system will put something in the configuration to break the start/stop interface swap script |
| 01:17.32 | *** join/#asterisk Shamalamatech13 (n=chatzill@65.126.237.4) |
| 01:17.50 | Shamalamatech13 | Hello All |
| 01:18.10 | Keizer | How Shamalamatech13 |
| 01:18.40 | Shamalamatech13 | Does anyone in here know the program gastman? |
| 01:19.53 | Shamalamatech13 | We have it configured and working the only problem is that we have so many call queues we cant see them all |
| 01:20.08 | Shamalamatech13 | I cant seem to find a way to scroll in the program |
| 01:20.12 | Shamalamatech13 | any ideas? |
| 01:21.14 | Keizer | Never used it |
| 01:21.27 | Keizer | I'm amazed that route add default gw x.x.x.x isn't working |
| 01:21.44 | Keizer | hardwire: I'm modifying the interface script so it won't give errors for me |
| 01:29.57 | *** part/#asterisk Shamalamatech13 (n=chatzill@65.126.237.4) |
| 01:32.10 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 01:35.54 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 01:38.58 | *** join/#asterisk jpastore (n=jpastore@173.9.147.165) |
| 01:42.46 | *** join/#asterisk Ccomp5950 (n=Ccomp595@66.190.102.236) |
| 01:52.25 | *** join/#asterisk badcfe (i=christia@morra.di.er.kjip.no) |
| 01:57.52 | *** part/#asterisk badcfe (i=christia@morra.di.er.kjip.no) |
| 02:03.41 | orkid | hardwire: http://www.bce.ca/en/aboutbce/regulatoryinformation/tarrifs/index.php/ItemView.asp?Tariff=SFT%20%20&Part=%20B%20%20%20%20%20%20%20%20&Item=%20B%2076%20%20%20%20%20 |
| 02:05.55 | *** join/#asterisk drumkilla (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 02:05.55 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 02:11.15 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 02:11.39 | *** join/#asterisk drumkilla (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 02:11.39 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 02:12.32 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:21.01 | *** join/#asterisk RB2 (n=RB2@pool-71-127-212-121.nwrknj.east.verizon.net) |
| 02:21.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:26.25 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:27.56 | RB2 | evening jaytee |
| 02:28.12 | jaytee | RB2, evening |
| 02:33.26 | orkid | hello |
| 02:45.13 | *** join/#asterisk Yourname (i=Yourname@unaffiliated/yourname/x-837320) |
| 02:45.47 | *** join/#asterisk mindCrime (n=chatzill@cpe-075-177-141-190.nc.res.rr.com) |
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| 03:20.46 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 03:42.29 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-129-85.dsl.stlsmo.sbcglobal.net) |
| 03:42.38 | troy- | what syntax do i use to set background music while ringing?> |
| 03:43.09 | bkw_ | m or M on dial if I recall |
| 03:43.18 | troy- | thanks |
| 03:44.31 | jaytee | how long should I bake a chicken if the oven's set at 325F? |
| 03:45.18 | orkid | till it's done :P |
| 03:45.24 | orkid | u can't poke it to see? |
| 03:45.28 | LemensTS | do you have one of those thermometers |
| 03:45.54 | orkid | u can just poke it with a knoife, open it up all over the place and see if it looks done (white) |
| 03:45.57 | orkid | :) |
| 03:46.00 | jaytee | ah, screw it! I'll just get a bucket at KFC |
| 03:46.17 | orkid | really, the knife check is the way to go |
| 03:46.44 | LemensTS | i had one of those buffalo dipped spicy chicken sandwiches at wendys, i swear i had to get up and blow my nose 6 times because it was so spicy |
| 03:47.47 | jaytee | yeah, it's good. Glad they came out with that. Their other spicy chicken sandwich was pretty hot too |
| 03:48.06 | LemensTS | yea the old one only took 3 napkins |
| 03:48.08 | jaytee | I usually get their nuggets and drown them in Frank's Red Hot sauce |
| 04:00.04 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
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| 04:10.15 | orkid | does anyone have a sip uri where i can place an echo test call too? preferably in the usa or canada ? |
| 04:15.28 | *** join/#asterisk jameswf (n=james@ip68-3-60-66.ph.ph.cox.net) |
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| 05:18.49 | LemensTS | hal3 |
| 05:24.17 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
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| 05:50.14 | garywsmith | anyone around to help with trying to get asterisk-gui downloaded from svn? No one seems to be home in the asterisk-gui at night (or at least night my time) |
| 05:50.35 | mib_hdp882 | What do you all think is the best guide to get started with asterisk is? |
| 05:54.58 | [TK]D-Fender | ~book |
| 05:54.59 | jbot | [book] 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 |
| 05:57.30 | mib_hdp882 | thanks |
| 06:10.52 | *** join/#asterisk [gnubie] (n=bintut@cm61.omega118.maxonline.com.sg) |
| 06:11.01 | Micc | how do I setup E911 for my customers? I have a PRI that I think has 911, but it probably routes to the wrong place for some customers. |
| 06:18.22 | *** join/#asterisk newmember (n=chatzill@S010600036d1139bb.cg.shawcable.net) |
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| 06:29.14 | orkid | :O |
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| 07:22.09 | radgrills | needs some help with sip username mismatch problem |
| 07:23.08 | drmessano | I would suggest making them match |
| 07:23.19 | radgrills | lol, yeah, that would help |
| 07:23.27 | radgrills | hi there |
| 07:23.46 | drmessano | Are your grillz rad? |
| 07:24.04 | radgrills | on my local subnet, i have a 4xfxs grandstream gxw4004 |
| 07:24.18 | drmessano | Uh oh |
| 07:24.23 | drmessano | You said the G word |
| 07:24.32 | radgrills | lol ... is that a prb? |
| 07:24.35 | drmessano | You get to drink from the FIREHOSE |
| 07:24.45 | drmessano | ~grandstream |
| 07:24.46 | jbot | hmm... grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 07:24.46 | radgrills | oops |
| 07:25.29 | radgrills | man, it was working all along, but i 'tweaked' somethin, and now its giving me this prb |
| 07:25.55 | radgrills | of course, i didn't save my previous configs ... he he |
| 07:26.20 | radgrills | but this problem seems to be well known in asterisk circles ... |
| 07:26.38 | drmessano | oh? |
| 07:26.52 | radgrills | the 4004 uses a a single ip address and only 1 port |
| 07:27.09 | drmessano | .... |
| 07:27.33 | radgrills | so when the 2nd (and subsequent) extensions authenticate to asterisk, asterisk only identifies the first extension |
| 07:28.12 | drmessano | I remember someone coming in here a long time ago with one of those |
| 07:28.12 | radgrills | and it gives me the mismatch error |
| 07:28.16 | drmessano | Word week of my life |
| 07:28.21 | drmessano | Worst* |
| 07:28.44 | drmessano | smashes his brain a little |
| 07:28.46 | radgrills | yeah, the references i found were all dated 2007 more or less |
| 07:29.25 | radgrills | i'm using ver 1.6.1 of asterisk now, so i guess some work has been done by now |
| 07:29.38 | radgrills | let's get the exact message |
| 07:29.51 | drmessano | Why would the asterisk version have anything to do with it |
| 07:30.22 | radgrills | [Nov 8 09:15:35] WARNING[9712]: chan_sip.c:10772 check_auth: username mismatch, have <2009>, digest has <2011> |
| 07:30.23 | radgrills | [Nov 8 09:15:35] NOTICE[9712]: chan_sip.c:17509 handle_request_invite: Failed to authenticate device "2011" <sip:2011@sip.domain;user=phone>;tag=759968899 |
| 07:30.54 | radgrills | well, if there was a bug, a new version may have addressed it |
| 07:31.00 | drmessano | Dude |
| 07:31.10 | drmessano | This isnt a bug in asterisk, the grandstream is crap |
| 07:31.26 | drmessano | Most multiport devices present themselves as multiple sip clients |
| 07:31.32 | radgrills | lol ... its all i could afford |
| 07:31.35 | drmessano | Independent ports, etc |
| 07:31.50 | drmessano | This isn't an asterisk problem |
| 07:32.09 | radgrills | ok .... i'm pretty sure you're right, coz my softphone works perfectly |
| 07:32.30 | drmessano | You dont have to be "pretty sure" |
| 07:32.30 | drmessano | I am telling you |
| 07:32.52 | radgrills | ok, i'll keep tweakin away at the 4004 then, there are loads of options to try |
| 07:32.58 | radgrills | i believe you |
| 07:33.12 | radgrills | i did a factory reset .... that didn't help |
| 07:33.14 | garywsmith | is the RPATH version of asterisk dead at this point? |
| 07:33.34 | drmessano | that would be "asterisknow" and yes |
| 07:33.43 | garywsmith | bummer |
| 07:33.47 | drmessano | bummer? |
| 07:34.19 | drmessano | I think the new 1.5 kicks ass |
| 07:34.57 | garywsmith | well, I downloaded that and have it working pretty well with the x100p card I got for testing. things are going good now and I have it working with the asterisk-gui as well. So I pulled the drive and installed 1.5beta and the x100p cards don't seem to work. |
| 07:35.45 | drmessano | heh |
| 07:36.49 | drmessano | This is part of why I tried to talk you out of getting x100p's |
| 07:36.51 | garywsmith | I can live with the older version for now. I'm going to pickup a 4 port fxo card shortly and a new server for it, but I just wanted to start getting the configs down now. |
| 07:36.59 | drmessano | But you knew better |
| 07:37.55 | orkid | u also were saying something about sip doing flash hooks, which is doesnt, because it's not supported in chan yet, as per bug 12999 |
| 07:37.58 | orkid | :P |
| 07:38.00 | garywsmith | I'm going to keep the single cards and use them for play... |
| 07:38.43 | drmessano | orkid: SIP does do flash hooks.. check the spec |
| 07:39.42 | garywsmith | drmessano: If I setup everything I want on one box, is it as simple as copying over the /etc/asterisk/* to the new box when I'm ready to roll out the real thing? |
| 07:40.01 | drmessano | orkid: I find it fascinating you're trying to call me out on something when you can't get something as basic as an SPA-3102 working |
| 07:40.07 | drmessano | orkid: That's a fail |
| 07:40.25 | orkid | drmessano: http://bugs.digium.com/view.php?id=12999 |
| 07:40.27 | drmessano | garywsmith: More or less.. dialplan and whatnot will move right over |
| 07:40.40 | drmessano | orkid: Again, you're showing me nothing |
| 07:40.43 | orkid | sure |
| 07:40.50 | orkid | ;) |
| 07:41.03 | orkid | cya |
| 07:41.05 | drmessano | orkid: I told you exactly how the flash hooks needed to work |
| 07:41.18 | garywsmith | drmessano: what about custom voicemail prompt? I havent tried those yet (don't know where there stored) but we will have probably 100 of them before we are done. |
| 07:43.16 | drmessano | orkid: I think you're the one that insisted for days the SPA-3102 couldn't do flash hooks due to some forum post you read.. after a week or so of that crap, you finally moved on to doing it with asterisk like Fender and I told you it needed to be done, and you found a bug.. woopdeedoo.. I never said I made it work in Asterisk, no clue if it even worked.. I knew you were doing it WRONG. |
| 07:43.40 | radgrills | drmessano: thanks your wise comments inspired me ... i changed a few settings on the GS based on what you said, and it works now |
| 07:43.47 | drmessano | orkid: Don't you have some ITSP to go troll over charging a 2 cent fee for something? |
| 07:43.52 | drmessano | radgrills: Awesome |
| 07:44.09 | drmessano | garywsmith: You can pretty much do anything you can dream up |
| 07:44.13 | drmessano | (within limits) |
| 07:45.21 | garywsmith | drmessano: glad you qaulified that... I was already thinking about floating to the moon :) Actually, I think (I know) I stated my last question wrong. Where are the voicemail prompts stored so I can copy those over when the time is right? |
| 07:46.29 | drmessano | --> /var/spool/asterisk/voicemail |
| 07:46.30 | orkid | yeah, rate center not servicable, lnp rejected |
| 07:46.36 | orkid | so much for that idea |
| 07:47.09 | garywsmith | drmessano: thanks. Well, I'm going to play for a while. I have to think up how this voicemail will be setup. |
| 07:47.14 | orkid | hardwire suggested an rcf, but thats like 31/month here according to the bell tariff, and no forwarding to extended local exchanges |
| 07:47.24 | orkid | which is bullshit |
| 07:47.46 | orkid | so im stuck if i want to keep my number |
| 07:49.02 | orkid | and SendDTMF() flash to a sip channel didnt work |
| 07:49.41 | drmessano | So let me get this right |
| 07:51.10 | drmessano | You come in here bitching about the SPA-3102 and all these bugs and forum posts, and we go through the trouble of pointing you towards "What you're doing is internal in the SPA-3102, this needs to be done in Asterisk if it's gonna work at all" and you're gonna bitch about how it doesn't work, like there was some "You told me definitely this would work, you suck, waa waa waa" |
| 07:51.12 | drmessano | Please |
| 07:52.05 | orkid | u mean "puh-lease" |
| 07:52.11 | drmessano | No |
| 07:52.17 | drmessano | That's not what I mean |
| 07:53.19 | orkid | btw, 3102 hsa nothing to do with it. sendDTMF does not send the flash on the sip channel, asterisk gives a 'warning' that it doesn't know how to handle it. |
| 07:53.24 | drmessano | I'm sure it's my fault your LNP didn't go through because I somehow recommended Les.net under the pretense that "they are more awesome than google, and can port anything.. oh, and they give free cheese" |
| 07:53.34 | orkid | umm, no |
| 07:53.43 | orkid | dont beat yourself over the head for no reason |
| 07:53.47 | drmessano | lol |
| 07:53.51 | drmessano | You're the one being the ass |
| 07:54.02 | orkid | and you the hole? lol |
| 07:54.09 | drmessano | and for the record, the 3102 has everything to do with it |
| 07:54.46 | drmessano | because the way you started off was flash hooking between the SIP and PSTN, which was never gonna work |
| 07:55.11 | orkid | flash hooking on the spa-3102, between pstn and SIP actually work |
| 07:55.13 | orkid | works |
| 07:55.20 | drmessano | ..... |
| 07:55.31 | drmessano | But does FLASH the PSTN port |
| 07:55.35 | orkid | when i disable CW on the spa-3102, and the flash hook gets sent to asterisk (supposedly), it doesnt work |
| 07:55.37 | drmessano | The flips ports |
| 07:55.39 | drmessano | and I told you that |
| 07:56.08 | drmessano | wow |
| 07:56.15 | orkid | in any case, i didn't use the 'flash' to send a flashhook, i used a feature code i programmed (eg. #9) and when asterisk saw it during a call it would SendDTMF to callee |
| 07:56.42 | orkid | thats when it threw the error |
| 07:56.51 | orkid | s/error/warning |
| 07:57.07 | orkid | try to sendDTMF(F) |
| 07:57.24 | drmessano | ... |
| 07:57.40 | orkid | or however u think u send flash through sendDTMF |
| 07:58.03 | drmessano | man |
| 07:58.13 | drmessano | there is something wrong with you |
| 07:58.30 | drmessano | Maybe I am speaking a different english |
| 07:58.40 | orkid | ur a funny guy |
| 07:59.19 | drmessano | No, you're way funnier |
| 07:59.24 | drmessano | Almost hilarious |
| 07:59.54 | garywsmith | drmessano: Can we talk about something a little more useful? I have this rash... |
| 08:00.23 | orkid | if asterisk cant send the flash to the sip channel, there is nothing for the 3102 to even try to sent to the pstn |
| 08:00.39 | drmessano | orkid: THAT IS WHAT _WE_ told YOU |
| 08:01.00 | drmessano | Mr "Hook flash doesn't work on the SPA-3102, it's broke" |
| 08:01.20 | drmessano | Seriously, are you stupid? |
| 08:01.23 | orkid | .... so tell me, how do i send a dtmf flash tone to a sip channel |
| 08:04.11 | drmessano | I am so sorry there is a fucking bug in asterisk that you've pasted 10 times to channel that prevents it from working, but IIRFC, you wasted days of our lives bitching about double hook flashes, firmware bugs, etc.. I spent HOURS beating into your head "IF THIS IS GOING TO WORK, IT NEEDS TO BE DONE THROUGH ASTERISK", not internally in the SPA-3102 "That only switches the PSTN and SIP ports" |
| 08:04.27 | drmessano | Go bitch at the devs that asterisk doesnt honor it |
| 08:04.34 | drmessano | I told you how it needed to work on the device |
| 08:04.34 | garywsmith | in the lab only situation (or should I say for the purposes of my kids at this point), can asterisk interface with a stardard skype account. (I ask this knowing I'd never use this for anything other than leasure. |
| 08:04.52 | drmessano | garywsmith: Not without an addon, which is in beta |
| 08:04.57 | drmessano | (closed beta) |
| 08:05.32 | garywsmith | ah. no biggy. are there any other free ones out there (or very inexpensive ones) that do? |
| 08:06.19 | drmessano | garywsmith: Not inexpensive.. few buggy ones that require a full client to be running on a windows box |
| 08:06.30 | garywsmith | nah. O |
| 08:07.09 | garywsmith | I'd rather go with something somewhat stable. My kids have skype on their computer so I figured I'd play with that. I know networking, not voip is the problem. |
| 08:07.29 | drmessano | Closed source application is the real problem |
| 08:07.35 | drmessano | Closed protocol, etc |
| 08:07.39 | garywsmith | true |
| 08:08.26 | radgrills | wrt Skype ... why isn't there a public sip/skype gateway? i'm sure every asterisk user would be interested |
| 08:08.30 | drmessano | Existing solutions create virtual sound devices that skype uses as the default device.. and I think the rest is manipulated through API |
| 08:08.48 | drmessano | radgrills: because skype isn't going to open their network so freely |
| 08:10.41 | drmessano | Real answer is, much like XMPP, that more networks need to embrace open protocols |
| 08:11.13 | drmessano | If Yahoo, AIM, MSN, and Google had SIP access, the point would be moot |
| 08:11.51 | drmessano | I could SIP dial joe@gmail.com from joe@aim.com and they wouldn't even need gateways |
| 08:12.00 | drmessano | Would be all open road |
| 08:12.01 | radgrills | yeah i understand that skype *could* but choose not to support sip directly ... but someone "out there" - even one of the voip providers, should set up a sip box, with a skype client or that closed beta driver, and allow people to access it. |
| 08:12.29 | drmessano | Yes, except each gateway represents ONE SKYPE user |
| 08:12.39 | radgrills | oh no |
| 08:12.42 | drmessano | SO 1000 people using one Skype account would be a problem |
| 08:12.58 | radgrills | i understand |
| 08:13.27 | radgrills | bang goes that theory |
| 08:14.23 | drmessano | Thats the real problem.. the existing solutions represent one skype user.. So if a company wanted 10 skype accounts, they would need 10 PCs or Virtual PC's set up, each as a gateway |
| 08:14.42 | drmessano | The Asterisk beta driver does it differently, but with that, will come some cost |
| 08:14.48 | garywsmith | Is there a recommended economical IP phone you guys would recommend? |
| 08:14.57 | drmessano | SPA-941 |
| 08:15.02 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 08:16.19 | garywsmith | Thanks. |
| 08:16.21 | radgrills | i'd pay a once-off fee for a gateway, but i don't want to be charged per call, and i don't want closed source |
| 08:16.25 | drmessano | Right now Asterisk will interface with Googletalk in 1.6 |
| 08:16.35 | drmessano | actually, 1.4+ |
| 08:17.14 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-4ccade14f11c436c) |
| 08:18.04 | drmessano | radgrills: Some part of it will have to be closed.. Asterisk is going to have to hook into a closed source binary to do what they're doing, and as I understand it, the asterisk part is only going to be a conduit into Skype.. with the client presentation and communication taking place inside of the cloud. |
| 08:18.36 | garywsmith | drmessano: how well does it interface with googletalk? |
| 08:18.46 | drmessano | Works real well... |
| 08:18.58 | radgrills | well, if it must be closed source, and paid for, it might as well be an external hardware device (like a SIP gateway device) |
| 08:20.37 | drmessano | radgrills: Yeah, and it would be nice if there was a Skype box, whether it be a SIP gateway or (gulp) an ATA that you could at least loop an FXO port into, but they seem to have a good reason for requiring the tether into a windows box |
| 08:23.36 | radgrills | think i'll have my skype account deleted ... i don't want to support arrogance |
| 08:24.23 | drmessano | You can be a hippie freeloader like me that preaches SIP URI dialing for all! |
| 08:24.33 | radgrills | is there a windows live |
| 08:24.37 | radgrills | interface? |
| 08:24.44 | drmessano | gets out the acoustic guitar and sings songs of SIP freedom |
| 08:24.47 | drmessano | No |
| 08:24.48 | radgrills | yeah ... i'm a freeloader |
| 08:25.48 | radgrills | joins in the chorus |
| 08:26.25 | radgrills | i also have one of those (spa3102) ... would it be possible to do this?: I want to keep an incoming PSTN call in the PSTN domain, but I'd like to log a CDR for the call, and if the call goes unanswered, then I'd like the call to be transferred to asterisk voicemail |
| 08:26.26 | garywsmith | drmessano: is googletalk similar to skype for the puposes of making calls? is it something I can just put into asterisk and be able to make calls from a softphone? |
| 08:26.27 | drmessano | encourages everyone to buy his new single "SIP, SIP, ain't it hip!", available on LP and cassette |
| 08:27.00 | garywsmith | drmessano: sorry, I only have 8track |
| 08:27.08 | drmessano | radgrills: YES! That's a good application for one |
| 08:27.47 | radgrills | thanks ... but how? |
| 08:27.58 | drmessano | garywsmith: Not sure where the softphone comes in.. You can use Softphone ---> Asterisk ---> GTALK |
| 08:28.47 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 08:28.58 | drmessano | radgrills: you register the PSTN port of the SPA-3102 to asterisk, and direct incoming calls into your dialplan... you enable the PSTN passthru on the SPA-3102 and it effectively gets "spliced" inline with your PSTN |
| 08:29.12 | garywsmith | well, we have no long distance on our home phone and the kids use skype. I was thinking about how I can integrate the incoming line through the FXO and use something like skype/googletalk/whatever for outgoing. |
| 08:29.19 | drmessano | Actually |
| 08:29.38 | garywsmith | currently we just use our cell phones for all of the long distance but Can you hear me now usually ends with NO more often than less. |
| 08:30.07 | drmessano | radgrills: Minus the CDR bit, you can just add the SPA-3102 to a jack somewhere and have it act almost like an answering machine |
| 08:30.44 | drmessano | garywsmith: That's easy enough to do.. the Googletalk just becomes another "trunk" |
| 08:31.22 | garywsmith | Do I just need to tweak gtalk.conf and jabber.conf? or are there other files as well? |
| 08:32.00 | radgrills | drmessano: Yes, but the CDR is the cool bit ... I have the passthrough enabled (without going off-hook) ... but will asterisk detect if the call is answered and terminated in the PSTN domain? |
| 08:32.16 | garywsmith | better question. Is there any real documentation of this other than someones blog? |
| 08:32.22 | drmessano | garywsmith: Thats it for the config, then you need to work the gtalk stuff into yout extensions.conf |
| 08:32.53 | drmessano | garywsmith: Theres a very good blog post somewhere.. beyond that, HA.. Which goes for most things asterisk |
| 08:33.20 | garywsmith | The example for that has several predefined entries for extensions. Do I just need to add the s,n.Dial one? |
| 08:34.02 | drmessano | radgrills: If the PSTN on the SPA-3102 terminates to asterisk, minus the passthru, and you register the FXS on the SPA-3102 to asterisk and direct all calls to it, you have your CDR |
| 08:34.16 | drmessano | FXO ---> Asterisk ----> back to FXS |
| 08:34.40 | drmessano | garywsmith: Link me what you're looking at |
| 08:34.49 | garywsmith | http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk |
| 08:35.15 | drmessano | oh god |
| 08:35.18 | drmessano | no, hang on |
| 08:35.20 | garywsmith | my extensions.conf is full of whatever. I don't know where to begin to dechiper this. |
| 08:35.24 | drmessano | Wiki = outdated crap |
| 08:35.27 | garywsmith | ok... thanks.. :) |
| 08:35.36 | garywsmith | yes, that's why I hate blogs... |
| 08:36.09 | garywsmith | they get stayle more often than not. |
| 08:39.50 | radgrills | hmmm this jabber/gtalk stuff looks interesing |
| 08:40.23 | drmessano | http://www.nslu2-linux.org/wiki/Optware/AsteriskGoogleTalk |
| 08:40.51 | drmessano | Thats not the O.P., but what I worked off of stolen and posted somewhere else |
| 08:42.32 | garywsmith | is the external IP required? I have cable so I don't have a static. |
| 08:43.15 | drmessano | Yes, same as for SIP.. you can use externhost there just the same |
| 08:43.47 | garywsmith | ah, so dynamip dns is fine? |
| 08:43.53 | drmessano | yeah |
| 08:44.09 | garywsmith | and bind address, that appears internally. Is that correct? |
| 08:44.46 | drmessano | Mine is 0.0.0.0 here |
| 08:45.05 | drmessano | I think it can be omitted and defaults to that |
| 08:45.19 | garywsmith | so, just to clarify, instead of externip I can use externhost=dyniphost.whatever.com |
| 08:45.36 | drmessano | Just as you do for SIP |
| 08:46.10 | drmessano | hmm |
| 08:46.17 | garywsmith | you lost me on sip, but I'm still working through this :) (the first time I touched voice was last week). |
| 08:46.39 | drmessano | ok |
| 08:46.55 | drmessano | In order for SIP to work on the outside, you need similar settings |
| 08:47.06 | drmessano | Which is common, hence the reference |
| 08:47.32 | garywsmith | ah. okay. I'll catch on to this pretty quick. |
| 08:49.19 | garywsmith | I just have to do it a couple times and I'll have the basics down. |
| 08:49.37 | garywsmith | do I really need all of the default crap in extensions.conf? |
| 08:50.11 | drmessano | No, you would probably do well to read the book |
| 08:50.55 | garywsmith | true |
| 08:52.25 | garywsmith | okay, I did everything on that page, and I still don't see the new trunk available in the asterisk-gui. Do I need to restart asterisk or is reload enough? |
| 08:52.46 | tzafrir_laptop | drmessano, fix the wiki, or at least write "outdated crap" in outdated pages |
| 08:53.55 | drmessano | tzafrir_laptop: No time |
| 08:53.55 | tzafrir_laptop | specifically, make a link from that page to a page called "ThisPageIsOutdatedCrap" |
| 08:54.07 | tzafrir_laptop | This only taked a minute or so |
| 08:54.42 | tzafrir_laptop | Later on, someone could take a look at the back-references from the page ThisPageIsOutdateCrap |
| 08:55.27 | drmessano | tzafrir_laptop: Can I do that for most of the wiki, but say minus 10 or 12 pages? :) |
| 08:56.06 | tzafrir_laptop | you have to start somewhere |
| 08:56.22 | tzafrir_laptop | the wiki badly lacks something like wikimedia's chat page |
| 08:57.11 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:58.13 | drmessano | Yeah, well, everyone is so down on Mediawiki because it's "so corporate, like, yanno" that they put up Wikis with inferior software and then the wiki dies or suffers due to lack of the features needed to make a wiki thrive |
| 09:03.22 | garywsmith | should the google-out show up as one of the available trunks in asterisk-gui for calling rules or is there more to do? |
| 09:23.50 | radgrills | is there still such a thing as a free public dahdi hierarchy? |
| 09:28.03 | garywsmith | possbile dumb question. If I do a reload inside of asterisk, does this impact existing calls? |
| 09:38.00 | Mark_Logan | not really. |
| 09:38.14 | Mark_Logan | I mean, I speak for sip chans that are open, I don't really know for zap. |
| 09:38.32 | Mark_Logan | just makes them have a little jitter for about 250ms |
| 09:39.01 | garywsmith | thanks. |
| 09:40.03 | Mark_Logan | np |
| 09:40.06 | Mark_Logan | Gnight all. |
| 09:40.43 | garywsmith | is 1.6.1 stable? |
| 09:41.58 | *** join/#asterisk CRCinAU (n=CRCinAU@203.56.246.91.crc.id.au) |
| 09:42.12 | CRCinAU | just wondering if anyone knows of a way to place a call on hold via AGI? |
| 09:42.20 | CRCinAU | ie while I do some processing in the background... |
| 09:42.33 | CRCinAU | of course, I need to be able to pick the call up again after that processing has ended.... |
| 09:42.51 | CRCinAU | so far, my googling hasn't found much... |
| 09:50.22 | radgrills | looks like the current stable release is 1.6.0.1 |
| 09:50.59 | radgrills | garywsmith:is your gtalk working yet? |
| 09:52.08 | kerx | damn, can someone help |
| 09:54.44 | kerx | having some difficulties figuring out what logic to use |
| 09:54.58 | kerx | i'm trying to build a City to State database, and I have a file that looks like: |
| 09:55.08 | kerx | AL,HELENA |
| 09:55.08 | kerx | AL,HOLLY POND |
| 09:55.24 | kerx | Should I use hash or array for this? |
| 09:55.29 | kerx | oh, damn |
| 09:55.33 | kerx | wrong channel! |
| 09:55.35 | kerx | slaps himself |
| 09:56.03 | *** join/#asterisk feeds (n=feeds@85-135-228-40.adsl.slovanet.sk) |
| 09:56.37 | *** join/#asterisk feeds (n=feeds@85-135-228-40.adsl.slovanet.sk) |
| 09:57.21 | garywsmith | radgrills: no. But I |
| 09:57.47 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 09:57.49 | garywsmith | but I'm doing some destructive stuff right now to the machine. I think I will have to rebuilt it in the morning... |
| 09:58.17 | radgrills | oh no ... yeah, i had to compile in iksemel support and rebuild asterisk. |
| 09:58.25 | radgrills | I think i'm ready to test a call now |
| 09:58.45 | radgrills | But g2g, so the testings gonna have to wait |
| 09:59.09 | radgrills | good luck |
| 10:02.00 | *** part/#asterisk garywsmith (n=garywsmi@c-67-164-10-244.hsd1.ca.comcast.net) |
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| 13:06.51 | tzafrir_laptop | anybody else having problems? Or does it work for anybody else? |
| 13:07.13 | tzafrir_laptop | svn ls http://svn.digium.com/svn/asterisk |
| 13:08.14 | tzafrir_laptop | suspects that there should be at least somebody else here who either has or doesn't have that problem |
| 13:08.29 | tzafrir_laptop | but then again, there's one flaw in that logic: |
| 13:08.42 | tzafrir_laptop | it assumes that there is actually someone else here |
| 13:09.51 | [gnubie] | waves.. gtg now.. |
| 13:11.11 | riddlebox | tzafrir_laptop, I dont use the svn version |
| 13:12.10 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 13:12.38 | tzafrir_laptop | 'svn ls http://svn.digium.com/svn/asterisk' does not require that. It doesn't even download anything to your local copy |
| 13:12.45 | tzafrir_laptop | (or even need a local copy) |
| 13:13.53 | riddlebox | ok I can check it for you |
| 13:15.26 | Maliuta | looks like the host is down |
| 13:15.49 | Maliuta | can't get trace or icmp response from svn.digium.com |
| 13:15.57 | Maliuta | or open a port |
| 13:16.20 | Maliuta | no route to host |
| 13:19.04 | riddlebox | tzafrir_laptop, it just sits there |
| 13:19.09 | *** join/#asterisk sasargen (n=chatzill@174-146-130-228.pools.spcsdns.net) |
| 13:23.38 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 13:30.12 | riddlebox | tzafrir_laptop, james@eeejames:~$ svn ls http://svn.digium.com/svn/asterisk |
| 13:30.12 | riddlebox | svn: OPTIONS of 'http://svn.digium.com/svn/asterisk': could not connect to server (http://svn.digium.com) |
| 13:36.10 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-186-176-190.bflony.east.verizon.net) |
| 13:36.58 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
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| 13:48.43 | Cybertoy | uhm.. sry for noob question. I upgraded to 1.6.0.1 from 1.4.22 and now I get these: -- ast_get_srv: SRV lookup for '_sip._UDP.sip.stanaphone.com' mapped to host sip.stanaphone.com, port 5060 |
| 13:48.53 | Cybertoy | but everything works... can I ignore these messages? |
| 13:50.30 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 13:50.49 | *** part/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 13:54.03 | tzafrir_laptop | Cybertoy, aren't those debug messages? |
| 13:54.12 | Cybertoy | I think they are |
| 13:54.24 | Cybertoy | ah... maybe they now show up |
| 13:54.45 | Cybertoy | I always had verbosity 3 ... |
| 13:54.49 | Cybertoy | on the console |
| 14:00.17 | *** join/#asterisk d-Vybes (n=david@82.196.214.14) |
| 14:00.48 | d-Vybes | is it possible for a SIP ALG to mess up an adapter contacting the PBX through a TCP tunnel? |
| 14:00.59 | d-Vybes | s/adapter/softphone |
| 14:02.24 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:15.18 | tzafrir_laptop | d-Vybes, set up openvpn over that :-( |
| 14:15.32 | tzafrir_laptop | iax is single port, but UDP |
| 14:16.52 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 14:16.56 | d-Vybes | i don't have the access. i am working for a company using an implementation of sip with TCP tunnels which should supposedly get the packets through the NAT, but still having problems with one way sound, etc... |
| 14:17.02 | d-Vybes | tzafrir_laptop: ^ |
| 14:17.03 | x86 | IAXy's are disco now eh? |
| 14:17.18 | tzafrir_laptop | x86, they rock? |
| 14:17.31 | x86 | tzafrir_laptop: discontinued |
| 14:18.22 | tzafrir_laptop | there are at least two chineese companies that sell IAX-copable ATAs/phones |
| 14:18.27 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 14:18.44 | x86 | wtf means this "copable"? ;) |
| 14:19.02 | tzafrir_laptop | err... capable |
| 14:19.16 | x86 | yeah i knows, was just being a dick :P |
| 14:19.31 | x86 | why doesn't digium make the iaxy's anymore? |
| 14:22.11 | *** join/#asterisk stoffell (n=stoffell@d51A4D7C1.access.telenet.be) |
| 14:22.22 | *** part/#asterisk d-Vybes (n=david@82.196.214.14) |
| 14:23.14 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 14:25.44 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 14:29.39 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 14:42.58 | *** join/#asterisk Tili (n=tili@58-27-162-28.wateen.net) |
| 14:43.17 | Tili | how can I log the out of pri intense debug span x to a file? |
| 14:45.34 | x86 | log the out? |
| 14:48.49 | stoffell | the output i suppose ;) |
| 14:56.18 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 14:58.18 | tzafrir_laptop | pri set debug file |
| 14:58.38 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 14:58.42 | tzafrir_laptop | (or use bristuff where pri debug/errors are logged using standard asterisk functions) |
| 15:03.37 | stoffell | tzafrir_laptop, good news, I read that there's a native b410p driver in dahdi trunk, ... looks promising don't you think ? |
| 15:03.58 | tzafrir_laptop | yup |
| 15:08.16 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 15:08.56 | Tili | thanks |
| 15:12.49 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 15:13.46 | drmessano | Still no mention of Oslec with Dahdi |
| 15:13.50 | drmessano | Lame |
| 15:14.32 | stoffell | it's in dahdi svn, no ? |
| 15:15.51 | stoffell | or easily installable with.. |
| 15:16.26 | drmessano | I've seen some mention of it, was expecting a dahdi variation of the install procedure on the main oslec site |
| 15:17.25 | stoffell | it is in the readme i think but svn.digium.com is down for the moment ( http://svn.digium.com/view/dahdi/linux/trunk/README ) |
| 15:17.57 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 15:38.10 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
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| 16:14.14 | *** join/#asterisk brummel444 (n=chatzill@dslb-088-067-199-247.pools.arcor-ip.net) |
| 16:18.27 | brummel444 | hi, when i try to call my mobile number from sip -> asterisk -> isdn -> mobilephone i get this error: WARNING[10119]: app_dial.c:1210 dial_exec_full: Unable to create channel of type 'CAPI' (cause 44 - Requested channel not available), does sb know what that means ? |
| 16:18.59 | ManxPower | It means that Asterisk was not compiled with CAPI support. |
| 16:20.12 | brummel444 | this works, when i restart asterisk, when i then call an internal sip client from external phone, it doesnt work anymore.. , i have unbuntu and asterisk chan capi installed |
| 16:20.57 | brummel444 | i think asterisk crashes on external calls.. |
| 16:21.09 | jaytee | ah, there's the problem! he has Unbuntu instead of Ubuntu. It's been Unbunted. |
| 16:22.19 | brummel444 | its ubuntu 8.04 server edition |
| 16:22.54 | brummel444 | and my isdn card is avm b1 (active) |
| 16:24.07 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 16:27.34 | ManxPower | Asterisk does not magically know and then forget about channel types. |
| 16:27.45 | ManxPower | Maybe your CAPI drivers are not loading BEFORE Asterisk loads. |
| 16:28.04 | brummel444 | modules show like capi shows the driver |
| 16:29.23 | brummel444 | when i restart asterisk, internal -> external works, but external -> internal i get this in console: http://pastebin.com/m568fe8bc |
| 16:29.25 | *** join/#asterisk Segnale007 (n=Pietro@host17-251-dynamic.32-79-r.retail.telecomitalia.it) |
| 16:29.34 | *** join/#asterisk bshiva (n=alex@193.136.29.2) |
| 16:29.36 | ManxPower | brummel444: I'm just telling what I know after using Asterisk for 7 years. |
| 16:30.02 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.218) |
| 16:30.13 | ManxPower | it might help if you spell "general" correct in your config files. |
| 16:30.39 | ManxPower | Also why are you using AEL? |
| 16:31.27 | bshiva | Hi all. I want to setup a 'homemade' ivr for a small business i have.i have a internet connection and a server running linux in the moment. what i have to do? install and configure asterisk or i have to talk to third parties in order to have a valid VoIP number? |
| 16:31.48 | ManxPower | bshiva: bOTH. |
| 16:31.59 | ManxPower | Now go run along and read The Asterisk Book |
| 16:32.01 | ManxPower | ~book |
| 16:32.02 | jbot | from memory, book is 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 |
| 16:32.33 | ManxPower | You don't need a 3rd party if you just want a "VoIP number" but you do if you need a telephone number. |
| 16:33.27 | jaytee | holy cow! there's a book on Asterisk? awesome! runs off to read |
| 16:33.45 | brummel444 | ManxPower: whops, wrong spelled.. yes. now 2 lines left in the console output: http://pastebin.com/m7ab29447 , dont know what ael is, i just used the default settings, and added my configuration |
| 16:34.20 | bshiva | ManxPower: hmm the idea was to receive calls from my cell and do some things with ivr. its more cheap to call from cell to phonenumber or to voip number? |
| 16:34.25 | ManxPower | You understand that the "default" config files are not designed to actually work, right? They are to show you as many examples of stuff as they can. |
| 16:35.01 | ManxPower | bshiva: you can't call "voip numbers" from a cell or other telephone. I'm not even sure what you mean by "voIP" number, I assume you mean a SIP URI. |
| 16:35.31 | ManxPower | But if you want a telephone number you either need a phone line or a service provider (ITSP) |
| 16:35.41 | ManxPower | Now, go read the book. all this info is in the book. |
| 16:36.01 | bshiva | ManxPower: yap i mean SIP. to have a IVR all i need is a normal phone line connected to my modem? |
| 16:36.03 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:36.12 | ManxPower | brummel444: looks like a codec issue |
| 16:36.14 | Daejeo | anyone have nokia cell n-series(S60- 5th edition)? i want to test the GPS service with asterisk. i tested with emulator. it is ok , but i want to test it in real time. |
| 16:36.15 | bshiva | ok i go read the book ^^ |
| 16:36.23 | ManxPower | bshiva: GO READ THAT DAMN BOOK!!!!!!!!!!!!!!!!!!!1 |
| 16:36.57 | ManxPower | I am not here to tutor people on Asterisk, telecom and networking. I am here, volunteering -- not paid, to help people. I am not here to read for people. |
| 16:37.35 | codefreeze-lap | brummel444: here's a hint for you: It's not finding a common codec to use between parties. Look at your sip.conf allow, disallow stuff (see the book, examp configs, etc) |
| 16:38.03 | bshiva | ManxPower: sorry. im not here to bother u. thanks a lot with your help and thanks for the book link :) |
| 16:38.40 | brummel444 | yes i already googled.. and found that i have to put this to each sip client configuration: disallow=all |
| 16:38.41 | brummel444 | allow=alaw |
| 16:38.56 | brummel444 | so it should use the codec alaw |
| 16:39.13 | drmessano | Ha |
| 16:39.20 | drmessano | Building a new server for home |
| 16:39.28 | ManxPower | brummel444: your SIP device must also have at least alaw enabled. |
| 16:39.34 | drmessano | Trying to find 2 SATA drives at a reasonable, but cheap price |
| 16:39.36 | ManxPower | brummel444: also check for typoes. |
| 16:39.52 | brummel444 | its phoner app on winxp.. |
| 16:39.58 | drmessano | The most economical value for drives is actually 320GB SATA |
| 16:40.08 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 16:40.09 | *** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
| 16:40.20 | ManxPower | drmessano: Don't use SATA! Use the new SATA-Network (aka SATAN) |
| 16:40.21 | ManxPower | 8-) |
| 16:40.26 | drmessano | lol |
| 16:40.34 | brummel444 | oh yes.. phoner doesn have alaw |
| 16:40.49 | drmessano | I am not against using last gen drives, other than them having sat on a shelf for a year or so now.. but damn |
| 16:40.55 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 16:40.57 | coppice | actually some 1T drives are less per byte than 320G now |
| 16:41.05 | drmessano | Per byte, yes |
| 16:41.43 | coppice | well, there are still 40G drives a lot less per drive than 320G :-) |
| 16:41.45 | drmessano | But I am talking about $44 + $16 ship for 160GB, $55 + $10 ship for a 250GB, $60 + Free ship for 320GB |
| 16:42.19 | brummel444 | is it G.711 A-law ? its marked in my sip client.. now i check for this type thing, do i have to install the codec to asterisk or is it already enabled there ? |
| 16:42.21 | ManxPower | I have a bunch of 40GB drives you can buy! |
| 16:42.22 | drmessano | It would cost MORE to get a smaller drive |
| 16:42.26 | drmessano | lol |
| 16:42.58 | ManxPower | brummel444: There is G.711 alaw and G.711 ulaw |
| 16:42.58 | drmessano | I love the WD 40GB drives for project boxes.. Build a little PBX for someone with 40GB mirror = win |
| 16:43.09 | drmessano | Then you have "old drive redundancy" |
| 16:43.10 | jaytee | brummel444, the book explains codecs pretty clearly. |
| 16:43.14 | jaytee | ~book |
| 16:43.14 | jbot | [book] 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 |
| 16:43.16 | ManxPower | drmessano: I am not a big fan of WD drives. |
| 16:43.35 | drmessano | You have to know which ones were particularly crap |
| 16:43.40 | jaytee | I used to like Maxtor drives. Now Seagate owns them. |
| 16:43.41 | ManxPower | at least of the vintage of 30GB - 60GB. |
| 16:43.57 | drmessano | Like the 20GB drives were horrid |
| 16:44.01 | ManxPower | I like the brand with the longest warrenty. 8-) |
| 16:44.03 | drmessano | the 40GB were solid |
| 16:44.05 | drmessano | lol |
| 16:44.18 | drmessano | WD had horrible 60GB drives as well |
| 16:44.30 | drmessano | the 160GB's have been rockin.. havent had a bad one |
| 16:45.04 | drmessano | Ah and WD30s Another sour vintage |
| 16:45.30 | jaytee | had one server with 5 80GB 15K rpm drives in it. Drives sounded like being in a huge cloud of mosquitos. |
| 16:47.04 | drmessano | Seems like there would be a HUGE market for a UM gateway for Windows |
| 16:47.25 | drmessano | Like a SIP VM server that pushes wav's to IMAP boxes |
| 16:48.34 | coppice | UM has been around for >10 years with very poor sales. now the term is suddenly sexy, but are the real sales getting any better |
| 16:48.46 | drmessano | Not really |
| 16:48.52 | drmessano | The users havent gotten any better |
| 16:49.14 | ManxPower | The problem with UM is that the "M" is usually gotten right, it's the "U" that never seems to work as advertized. |
| 16:49.23 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 16:49.23 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:49.40 | *** join/#asterisk anthm][ (n=anthm@72.59.201.204) |
| 16:49.51 | coppice | actually, a number of systems worked pretty well 10 years ago. the snag is people love the idea, but never want to sign up |
| 16:49.55 | drmessano | Honestly, pushing VM to users via SMTP ala Asterisk, with users being able to listen to VM on their smartphones, is about as unified as people seem to get |
| 16:50.17 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 16:51.27 | coppice | we used to offer two related numbers for voice and fax, so we could completely avoid the quirks in differentiating call types. the thing people asked for most was for the two number to be unrelated. that's how unified people like things :-) |
| 16:51.27 | drmessano | when it comes to the IM part, the market is so fragmented it's absolutely nuts |
| 16:51.34 | jaytee | I'm using * with Exchange 2007 UM and I love it. Only feature I don't have working, because MS left it out is MWI lamp working on the Polycom phones. |
| 16:52.03 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:54.05 | drmessano | I was never a big Blackberry fan until recently.. and now mainly due to the incredible stability of BES over ActiveSync and Versamail, but what IM is offered by default and is well known for the BB? The completely isolated island of BB Messenger |
| 16:54.16 | drmessano | Which is as unified as the former soviet union |
| 16:54.31 | jaytee | lol |
| 16:55.14 | *** join/#asterisk SteelSide (n=SteelSid@217.76.87.68) |
| 16:55.28 | coppice | for MS, unified messaging == whatever you message, one company provides the bill |
| 16:56.08 | Bad_Robot- | MS just came out with Reponse Point as their new pbx have you guys seen it |
| 16:56.48 | drmessano | OLCS + Exchange is far better than anything else out there.. and at least there's desktop and mobile clients for the IM, that actually notify of VM and Email |
| 16:57.14 | drmessano | Thats what is most frustrating.. The integration bits are not that hard |
| 16:57.53 | SteelSide | Hi. Anyone got a list of compatible voice modems that work with asterisk? Or some general rule for if it works or not (found aopen on a mailinglist, but i can't find ANY information about such a standard) |
| 16:58.08 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 17:00.02 | jaytee | SteelSide, forget voice modems. |
| 17:00.08 | drmessano | Asterisk + IMAP to any mail server = win.. Now we need it to be more stable... Now for the IM.. going on the whole "free as in beer", you need an XMPP server that will allow config of your IMAP account <> IM account to give you VM notifications and email notification + a strong mobile app for the instant messaging |
| 17:00.21 | SteelSide | jaytee, how so :(? |
| 17:00.42 | jaytee | because they're not designed for VOIP |
| 17:00.49 | drmessano | SteelSide: If using a $10 modem were that great of an option, people would have spent time+beer on it |
| 17:00.57 | *** join/#asterisk getsql08 (n=getsql05@70.5.226.189) |
| 17:01.09 | SteelSide | well it's not voip im interrested in, but having asterisk pickup all incoming normal analog calls, and give out a selection menu |
| 17:01.25 | getsql08 | Looking for professional asterisk developer with 2+ years experience for consultation services | $75 / hour |
| 17:01.26 | drmessano | .... |
| 17:01.35 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 17:01.36 | drmessano | Yeah, jaytee |
| 17:01.57 | jaytee | ah, you want an IVR with an analog interface. Check out FXO cards for the asterisk server. Digium makes them, Sangoma does too. |
| 17:02.53 | SteelSide | So a normal data/fax/voice modem won't work well with asterisk out of the box? I'd need something special for XX dollars ? |
| 17:03.03 | *** join/#asterisk RB2 (n=RB2@pool-71-127-212-121.nwrknj.east.verizon.net) |
| 17:03.04 | jaytee | getsql08, does it involve travel? |
| 17:03.10 | getsql08 | jaytee: no |
| 17:03.25 | jaytee | ManxPower, you paying attention? |
| 17:03.34 | coppice | couch potatoes need apply :-) |
| 17:03.42 | jaytee | lol |
| 17:04.21 | getsql08 | I can up the fee to $150 |
| 17:04.27 | getsql08 | I just need some general information for a few hours |
| 17:05.25 | Superbartt | o_0 |
| 17:07.00 | Superbartt | anyone has experience using call deflection with bristuff? after 10 seconds it just seems to disconnect the caller |
| 17:09.47 | Superbartt | and i can't seem to get the capi stuff compiled :x |
| 17:09.53 | Daejeo | anyone have nokia cell n-series(S60- 5th edition)? i want to test the GPS service with my box. i tested with emulator. it is ok , but i want to test it in real time. |
| 17:10.55 | *** join/#asterisk [gnubie] (n=bintut@cm61.omega118.maxonline.com.sg) |
| 17:11.29 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:11.37 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 17:11.56 | brummel444 | i cant solve my problem.. phoner (sip client) and asterisk have the alaw (and ulaw) modules loaded, internal -> external call works, but not external -> internal. console output: http://pastebin.com/m42e2f564 |
| 17:17.39 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 17:24.17 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 17:25.16 | *** join/#asterisk _coredump_ (n=duke@dslb-084-061-222-159.pools.arcor-ip.net) |
| 17:25.57 | *** part/#asterisk _coredump_ (n=duke@dslb-084-061-222-159.pools.arcor-ip.net) |
| 17:27.14 | ManxPower | brummel444: you're not disabling the loading of some modules, are you? |
| 17:29.39 | ManxPower | brummel444: pastebin /etc/asterisk/modules.conf |
| 17:31.06 | ManxPower | Codec 65535 is a video codec. |
| 17:39.36 | beek | Hello all -- I'm having the quintessential "No CallerID Name on my PRI". chan_dahdi.conf has "immediate=no" and I have a Wait(2) prior to answer() in my context. |
| 17:40.22 | beek | I've googled this and those were the two things that were discussed to fix the problem. (Including some posts by ManxPower). |
| 17:40.46 | ManxPower | beek: then you need to contact your telco to turn on the option. you have done all the required config on the asterisk side. |
| 17:40.47 | beek | Using pri intense debug I can see where the callerid name packet is sent after I'm in context. |
| 17:41.01 | ManxPower | then make the Wait(1) be the FIRST priority |
| 17:41.11 | beek | ManxPower: It is. |
| 17:41.16 | ManxPower | also there are TWO ways of sending PRI CLID. |
| 17:41.24 | ManxPower | have your telco change it. |
| 17:41.57 | beek | ManxPower: This * box is going to sit between the telco and our legacy PBX. |
| 17:42.05 | beek | (which gets callerid name properly) |
| 17:42.23 | beek | I am getting the CallerID number, just not the name. |
| 17:42.26 | ManxPower | I'm just giving you advice based on 7 years of using Aserisk |
| 17:42.40 | ManxPower | 6 of those years using Asterisk with PRIs |
| 17:42.54 | beek | ManxPower: That's cool. I'll call them on Monday. |
| 17:42.58 | beek | ManxPower: Thanks |
| 17:43.09 | ManxPower | What specific variable are you using to access the CID Name info? |
| 17:43.25 | beek | CALLERID(name) |
| 17:43.58 | ManxPower | paste the line in the CLI that has that variable on it. |
| 17:44.26 | beek | exten => _XXXX,n,NoOp(Callerid: ${CALLERID(all)}); |
| 17:44.43 | ManxPower | BTW, it is possible, in theory, to have the wrong switch type and still have calls work. |
| 17:44.47 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 17:44.47 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:45.02 | beek | I used NI2 |
| 17:45.14 | beek | :q |
| 17:45.27 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 17:49.19 | beek | ManxPower: Yes, I feel stupid... TWO priority 1s in my dialplan. |
| 17:50.21 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 17:58.12 | *** join/#asterisk pyite (n=pyite@c-76-21-105-195.hsd1.ca.comcast.net) |
| 18:01.36 | *** join/#asterisk bmoraca (n=bmoraca@64-195-81-243.mod.clearwire-dns.net) |
| 18:04.41 | beek | ManxPower: Now that I have that problem solved I'm not getting the callerid(name) to the PBX. The callerid(num) does get there. Any sage advice on how to make that happen? |
| 18:11.13 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:12.47 | *** join/#asterisk km2 (n=x@mobile-166-217-134-109.mycingular.net) |
| 18:13.49 | *** join/#asterisk Eppos (n=alpin@i59F5FB42.versanet.de) |
| 18:27.20 | ManxPower | beek: no idea. |
| 18:28.05 | ManxPower | make sure you are not modifying the CID info (name or number) |
| 18:31.21 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 18:33.20 | jaytee | ~itsp-us |
| 18:33.26 | getsql08 | hello |
| 18:33.37 | getsql08 | ~itsp-us |
| 18:33.43 | jaytee | ~itsplist-us |
| 18:33.43 | 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 |
| 18:33.59 | beek | ManxPower: Will do. I'm having a blast doing this -- it's my first PRI connection so I'm learning a lot. I appreciate your help. |
| 18:36.01 | bmoraca | has anyone here had a chance to set Asterisk up in an ESX setting? |
| 18:38.46 | inv_arp | bmoraca: dont see why it wouldnt work.. |
| 18:42.59 | *** join/#asterisk RobertLaptop (n=rmiddle@m9f0736d0.tmodns.net) |
| 18:43.49 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 18:48.31 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 18:49.07 | bmoraca | well, my concern is zaptel virtualization |
| 18:49.20 | bmoraca | although I suppose I could use a separate asterisk box as my PSTN gateway |
| 18:49.56 | bmoraca | the idea is to create a multi-tenant hosted solution, optimally with my PRI cards in the same box |
| 18:52.31 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:56.33 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:56.33 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 18:58.03 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 19:07.31 | *** part/#asterisk pyite (n=pyite@c-76-21-105-195.hsd1.ca.comcast.net) |
| 19:14.37 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:19.07 | ManxPower | do not expect to get zaptel working well in a VM |
| 19:22.30 | ManxPower | And that also means no IAX2 trunking and no MeetMe |
| 19:37.48 | *** join/#asterisk oh207 (n=oh207@nylug/member/oh207) |
| 19:43.54 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 19:46.13 | hardwire | ManxPower: even though it's not ideal.. I do love asterisk + openvz |
| 19:46.39 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 19:46.39 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:47.28 | *** join/#asterisk garywsmith (n=garywsmi@c-67-164-10-244.hsd1.ca.comcast.net) |
| 19:48.52 | ManxPower | I love the bare hardware more. |
| 19:50.19 | garywsmith | drmessano: I had the x100p cards working fine under zap and so I'm taking a shot at getting them to work under dahdi. dahdi_hardware shows them just fine so I add fxsks=1-2 in the /etc/dahdi/system.conf but it's then removed when I run dahdi_genconf. what am I doing wrong? |
| 19:50.36 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 19:51.31 | garywsmith | when I run dahdi_genconf it complains: /usr/sbin/dahdi_genconf: Cannot read '/etc/dahdi/genconf_parameters': No such file or directory |
| 19:52.05 | garywsmith | when i create an empty /etc/dahdi/genconf_parameters file, it works but doesn't do anything real. What goes in the genconf_parameters file? |
| 19:54.05 | ManxPower | dahdi_genconf overwrites the file. |
| 19:54.16 | ManxPower | don't run it on a file that is already set up. |
| 19:54.48 | garywsmith | makes sense but shouldn't it be creating the file with the entries for the appropriate hardware? |
| 19:56.54 | garywsmith | so I add fxsks=1-2 back in and when I run the dahdi_cfg -t -vv it says everything is fine but when I run it without test mode it complains "DAHDI_CHANCONFIG failed on channel 1: No such device or address (6)" |
| 19:57.51 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 19:59.26 | tzafrir_laptop | garywsmith, you don't have those channels . See /proc/dahdi/1 |
| 20:00.27 | garywsmith | yep, nothing there. So if the hardware is detected by dahdi_hardware, how to i get them to initialize? |
| 20:00.44 | ManxPower | garywsmith: load the kernel modules. |
| 20:00.59 | garywsmith | modprobe dahdi. did that already. Am I missing one? |
| 20:01.23 | *** join/#asterisk mankash (n=rom10@CPE00062575886a-CM00186832000a.cpe.net.cable.rogers.com) |
| 20:01.35 | ManxPower | or do a "make config" and "chkconfig dahdi on" and a "service dahdi restart". You should have seen this info in the INSTALL or README for DAHDI. |
| 20:01.38 | *** join/#asterisk sasargen (n=chatzill@72.58.119.227) |
| 20:01.42 | ManxPower | Maybe someone forgot to put it in there. |
| 20:02.27 | ManxPower | garywsmith: you never modprobe dahdi. You modprobe the module for the card you have, then that module loads zaptel/dahdi kernel module automagically. |
| 20:02.36 | mankash | hi |
| 20:02.41 | garywsmith | dude, start dahdi now shows that two channels... I'll have to go back to the readme. |
| 20:02.48 | ManxPower | garywsmith: You did read the install doc, right? |
| 20:02.57 | mankash | I have a simple internal setup of asterisk with 2 soft phones resgitered |
| 20:03.17 | mankash | <PROTECTED> |
| 20:03.36 | mankash | <PROTECTED> |
| 20:03.44 | ManxPower | mankash: Does "sip show peers" show the correct IP address for each of the two peers? |
| 20:03.49 | mankash | on the asterisk log |
| 20:04.14 | mankash | wait pls |
| 20:06.16 | garywsmith | 1.5 is going with the freepbx. Does that mean that asterisk-gui is going away? |
| 20:07.28 | [TK]D-Fender | GaryNo, it means they want to placate whiners who can't handle the fact that *GUI doesn't do EVERYTHING for you |
| 20:07.34 | [TK]D-Fender | :) |
| 20:07.40 | mankash | does this command need to be used in asterisk cli |
| 20:08.19 | mankash | it is not accepting this command |
| 20:09.51 | garywsmith | [TK]D-Fender: it is easy to get overwhelmed with the config options if you're a newbie to pbx like myself though. But I usually manage. |
| 20:09.56 | [TK]D-Fender | mankash: What "command"? |
| 20:10.11 | mankash | sip show peers |
| 20:10.12 | [TK]D-Fender | garywsmith: the GUI is a pretty crappy way to learn. |
| 20:10.32 | garywsmith | [TK]D-Fender: what's the recommended gui at this point? |
| 20:10.42 | [TK]D-Fender | mankash: what does it tell you? |
| 20:10.47 | ManxPower | mankash: pretty much any command we tell you to use will be in the Astrisk CLI |
| 20:11.09 | garywsmith | actually, I'm looking through config files. But I will need semi to non techies to be able to do some limited stuff. |
| 20:11.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:11.41 | ManxPower | garywsmith: almost nobody here will help with GUI isssues. There are other channels for that. |
| 20:11.45 | [TK]D-Fender | Garywell some much MORE techie guy is going to have to learn the basics for *GUI |
| 20:11.59 | ManxPower | ~zeeek |
| 20:11.59 | jbot | zeeek is, like, someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 20:12.07 | mankash | Its is saying no such command |
| 20:12.24 | ManxPower | mankash: pastebin what is on your screen. |
| 20:12.26 | ManxPower | !pb |
| 20:12.29 | ManxPower | ~pb |
| 20:12.29 | 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 |
| 20:12.38 | [TK]D-Fender | garywsmith: FreePBX once installed runs the entire show and a barely trained monkey can do it. We see the completely untrained kind in here all too often |
| 20:13.14 | garywsmith | luckily for me I'm a trainable monkey. |
| 20:14.17 | ManxPower | It's not so wrong to order internet service when you are in a location you KNOW it's not available, just because their web site says it's available? |
| 20:17.02 | mankash | ManxPower here is the pastebin http://pastebin.com/m25d4c213 |
| 20:18.16 | tzanger | finally |
| 20:18.32 | tzanger | I have been tearing my garage apart for ... about a week now, trying to find an FXO board for my Adit600 |
| 20:18.36 | tzanger | finally found it today |
| 20:19.44 | [TK]D-Fender | tzaYou have a garage? *sniff* |
| 20:20.35 | [TK]D-Fender | mankash: do "module load chan_sip.co" |
| 20:21.33 | mankash | i did that |
| 20:21.42 | [TK]D-Fender | mankash: and,,,? |
| 20:22.40 | mankash | = Manager 'asterisk' logged off from 127.0.0.1 |
| 20:22.40 | mankash | <PROTECTED> |
| 20:22.40 | mankash | <PROTECTED> |
| 20:22.40 | mankash | <PROTECTED> |
| 20:23.21 | mankash | sorry it was just 4lin e so I pate it here |
| 20:23.30 | *** join/#asterisk jplank (n=gbove@reports.nyigc.net) |
| 20:24.03 | jplank | could someone tell me why in http://pastebin.com/m6df037bf the calls goes to n(eleven) and not to the next step? |
| 20:24.40 | ManxPower | mankash: in the shell can you "ping www.fnords.org" ? |
| 20:25.23 | [TK]D-Fender | mankash: pastebin that stuff next time, do not pam the channel |
| 20:25.37 | [TK]D-Fender | mankash: And I meant "sip show peers" |
| 20:25.50 | mankash | Please note that I am trying asterisk internally with 2 soft phones |
| 20:26.01 | ManxPower | [TK]D-Fender: chan_sip does not appear to be loaded as seen in his pastebin. |
| 20:26.03 | jplank | the cli shows -- Executing [s@custom-test:3] GotoIf("SIP/209.220.119.5-b7c6fc30", "9=11?eleven") in new stack and just jumps to n(eleve) |
| 20:26.24 | mankash | it is able to ping |
| 20:26.31 | [TK]D-Fender | jplank: exten => s,n,Gotoif(${clen}=10?ten:none) <-- because there is not EXPRESSION in this GotoIf |
| 20:26.47 | ManxPower | jplank: Hint: You never see the expression in the CLI, you only see what the expression evaluates as. since you can see the expression in the CLI you have a syntax error |
| 20:27.02 | [TK]D-Fender | ManxPower: his last paste does not show this |
| 20:27.12 | ManxPower | expressions normally evaluate as "1" or "0" |
| 20:27.16 | [TK]D-Fender | ManxPower: it would say "channel not implemented". It does not. Its there |
| 20:27.37 | [TK]D-Fender | jplank: Go read up on "Asterisk Expressions" on the WIKI and in the book again |
| 20:27.45 | jplank | oops |
| 20:27.49 | jplank | sorry |
| 20:27.51 | jplank | I see |
| 20:27.53 | ManxPower | sip show peers no such command "sip show" |
| 20:28.03 | jplank | thanks |
| 20:29.07 | ManxPower | [TK]D-Fender: starting to smell like a GUI. I don't think he'd be ABLE to edit the modules.conf. |
| 20:29.37 | [TK]D-Fender | ManxPower: I asked to see the result of trying to manually load the module. I smell incompetance. |
| 20:29.52 | [TK]D-Fender | ManxPower: 1 line. All I asked for. Was I too demanding? |
| 20:31.25 | ManxPower | [TK]D-Fender: people think we just sprinkle magical faerie dust and fix problems. They don't realize how important it is to do each step one at a time or it won't be fixed. |
| 20:31.48 | mankash | I can understand |
| 20:32.05 | [TK]D-Fender | mankash: So... whathappened when you tried to load the module? |
| 20:32.24 | mankash | it did not gane any message |
| 20:32.26 | mankash | gave |
| 20:32.31 | [TK]D-Fender | mankash: And next since it sooms OK at first glance you shoul look at your SIP PEER to confirm they are registered at all |
| 20:32.37 | ManxPower | mankash: You should go re-read The Book |
| 20:32.40 | ManxPower | ~book |
| 20:32.41 | jbot | extra, extra, read all about it, book is 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 |
| 20:32.45 | [TK]D-Fender | mankash: make sure you are at verbose 10 |
| 20:32.48 | `Sean | hey |
| 20:33.14 | mankash | it was 7 now is set it to 10 |
| 20:34.16 | ManxPower | [TK]D-Fender: Did you notice this "Manager 'asterisk' logged on from 127.0.0.1" |
| 20:35.31 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 20:35.31 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:35.46 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 20:35.48 | ManxPower | mankash: Try re-running the sip show peers command |
| 20:35.55 | mankash | ok |
| 20:36.11 | mankash | same result |
| 20:36.27 | mankash | is it possible to see what modules are loaded |
| 20:36.35 | ManxPower | "core show modules" |
| 20:37.04 | ManxPower | Did you install from a package. |
| 20:37.22 | mankash | I downloaded and then install it |
| 20:37.54 | mankash | no such command |
| 20:37.54 | ManxPower | Then I am out of ideas. |
| 20:38.08 | ManxPower | there is no reason for it to not work for you. |
| 20:38.09 | garywsmith | is svn down? |
| 20:39.00 | mankash | ok anyway thx for trying |
| 20:40.41 | mankash | but why manager is keep loggin on and logging off |
| 20:40.47 | tzanger | hmm |
| 20:40.54 | tzanger | any carrier-access familliar people here? |
| 20:41.03 | tzanger | trying to understand the difference between an FXS/DPT card and an FXS 5G card |
| 20:41.17 | garywsmith | looks like bugs.digium,com is offline as well. |
| 20:42.28 | mankash | ManxPower it is working now |
| 20:42.40 | mankash | I reloaded asterisk and it is working finr |
| 20:43.47 | mankash | thanks guys |
| 20:45.33 | mankash | anybody has any experience with voicetronix openline4 card |
| 20:45.35 | beek | Crap! ManxPower -- I figured out how to get CallerID name sent to the pbx. If I set facilityenable=yes in chan_dahdi.conf, then the name gets sent. Unfortunately, as soon as I do that I can no longer dial out because the telco has a fit. |
| 20:46.19 | beek | I thought that I had the callerid name problem fixed... |
| 20:47.34 | [TK]D-Fender | beek: thats set by channel.... set it to NO for the ones going to the telco |
| 20:52.24 | beek | [TK]D-Fender: I would have guessed that but I'm not sure where I need to put it. I have it now at line 29 (http://www.pastebin.ca/1248992), which sets it globally. I've tried at 44.5 but no go. |
| 20:52.33 | beek | [TK]D-Fender: would you mind taking a look please? |
| 20:53.49 | [TK]D-Fender | beek: 44.5 should be where it goes. you restart * and it still freaks? |
| 20:54.14 | beek | Yes. |
| 20:55.18 | *** join/#asterisk KP7 (n=chatzill@dsl001-145-071.phl1.dsl.speakeasy.net) |
| 20:57.17 | KP7 | does anyone know if there are problem with sip notify messages for mwi v1.6? |
| 20:59.40 | *** join/#asterisk kb3ien (n=kb3ien@isl177-max1.accesshighway.net) |
| 21:01.09 | kb3ien | any work on using the DND button on the polycoms to affect the presence on another phone? |
| 21:04.30 | *** join/#asterisk feeds (n=feeds@85-135-228-40.adsl.slovanet.sk) |
| 21:05.43 | [TK]D-Fender | kb3ien: No. |
| 21:07.16 | *** join/#asterisk loconut (n=jessica@webtrotter.com) |
| 21:07.20 | loconut | hello :) |
| 21:07.41 | loconut | anyone alive today? |
| 21:07.46 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 21:08.08 | loconut | I have a cisco 7960 with 3 NAT'd connections and one server I'm setting up on the lan. My local server shows with an X but everything works fine. Any idea what its showing an X for? Does it have to do with the NAT somehow? |
| 21:10.17 | beek | [TK]D-Fender: I was too quick on the draw to try and connect. I just noticed that it takes a bit for asterisk to bring up the B-channels after a restart. That took care of it -- thanks! |
| 21:10.18 | [TK]D-Fender | loconut: Where do yous ee this "X"? |
| 21:10.33 | [TK]D-Fender | beek: You're welcome. |
| 21:11.50 | loconut | TKID on the phone icon |
| 21:11.56 | loconut | (on the phone itself) |
| 21:12.13 | loconut | on the line for my server on the lan |
| 21:12.28 | loconut | i can pick up, dial numbers, calls come in, but it still shows the x |
| 21:13.21 | loconut | phone is of course registered |
| 21:13.34 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 21:13.57 | kb3ien | ‰on a similar toic is there a way to disapear the MyStat button on the olycom 650? |
| 21:14.03 | troy- | how can i receive sms over SIP? |
| 21:14.47 | [TK]D-Fender | troy-: * does not support receiving SMS |
| 21:15.01 | troy- | [TK]D-Fender, but there are modules for it that do? |
| 21:19.25 | [TK]D-Fender | troy-: Nothing I've seen |
| 21:19.57 | [TK]D-Fender | troyand it wouldn't just be a module. As this is SIP signalling, it'd have to be in the CHANNEL DRIVER. |
| 21:20.14 | [TK]D-Fender | troy-: So thats a no-go for most thoughts of a bolt-on solution. |
| 21:20.17 | troy- | gotcha - are there any other protocols such as IAX that do support? |
| 21:21.34 | [TK]D-Fender | troy-: No. |
| 21:22.13 | troy- | are there any other software PBXs that do? |
| 21:23.05 | [TK]D-Fender | troy-: this is the point where you stop fishing for a "yes", get off your ass, and go look. |
| 21:23.33 | troy- | haha well thanks for being patient ;) |
| 21:38.33 | loconut | TKID, any ideas on my x icon? |
| 21:38.43 | loconut | (if you have a moment) |
| 21:52.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:59.58 | jplank | how can I compare one string against multiple strings inside a dialplan? |
| 22:00.49 | jplank | something along the lines of Gotoif($["${careac}"="718" or "212" or "917" or "347 or "646"]?nyc) |
| 22:01.37 | *** join/#asterisk patkoscsaba (n=patkoscs@79.114.73.246) |
| 22:01.47 | patkoscsaba | hi everybody |
| 22:02.22 | [TK]D-Fender | jplank: Go read up on "Asterisk Expressions" on the WIKI and in the book again <---------- |
| 22:03.07 | patkoscsaba | I just installed Asterisk on Mandriva 2009 from the provided packages of my Linux distribution, I installed asterisknow too ... I have absolutly no ideea how to configure it, anyone can point me to a starting point, or to a GUI based configuration version/option |
| 22:03.09 | patkoscsaba | ?? |
| 22:04.48 | [TK]D-Fender | patkoscsaba: There ar no good GUI guides that I've ever heard of |
| 22:05.02 | [TK]D-Fender | patkoscsaba: for learning how * work underneath it : |
| 22:05.03 | [TK]D-Fender | ~book |
| 22:05.04 | jbot | book is, like, 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 |
| 22:05.22 | patkoscsaba | [TK]D-Fender: GUI or starting guide |
| 22:05.32 | [TK]D-Fender | patkoscsaba: the GUI will of course make learning the basic more painful. |
| 22:05.38 | [TK]D-Fender | ~jerjerguide |
| 22:05.38 | 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/ |
| 22:05.40 | patkoscsaba | [TK]D-Fender: if I have a GUI I can configure stuff |
| 22:05.49 | [TK]D-Fender | ^^ for some inspiration on a minimal system |
| 22:06.17 | patkoscsaba | [TK]D-Fender: and again, I just want it up quickly right now, the rest can be learned later |
| 22:07.59 | [TK]D-Fender | patkoscsaba: Don't get your hopes up to ohigh on "up quickly" either. * has a somewhat steep learning curve |
| 22:09.24 | ManxPower | patkoscsaba: your biggest problem with a GUI is finding other people using that GUI. |
| 22:09.26 | patkoscsaba | [TK]D-Fender: ok, so I restate my question, Can I get it up to make a test call (the rest will be configured later) betwenn 2 PCs in less than 30 minutes? |
| 22:09.29 | carrar | patkoscsaba, if you need to learn quick there are Asterisk classes out there |
| 22:09.42 | ManxPower | patkoscsaba: not a chance in hell. |
| 22:09.53 | carrar | hahha |
| 22:09.58 | ManxPower | Heck, it takes 3 days to teach that to people in a class. |
| 22:10.00 | [TK]D-Fender | patkoscsaba: 30 mins from scratch? Low odds. |
| 22:10.11 | [TK]D-Fender | ManxPower: Shitty class... |
| 22:10.15 | patkoscsaba | [TK]D-Fender: from zero-conf |
| 22:10.27 | ManxPower | [TK]D-Fender: these people didn't even know Linux |
| 22:10.28 | carrar | How Can I get a new outlet in my room in 5 mins or less!!! |
| 22:10.34 | patkoscsaba | [TK]D-Fender: it is already installed and has the default settings |
| 22:10.37 | [TK]D-Fender | patkoscsaba: best of luck to you. |
| 22:10.46 | ManxPower | Yup, best of luck to you! |
| 22:10.50 | [TK]D-Fender | patkoscsaba: Default means absolutely nothing in * |
| 22:11.02 | patkoscsaba | [TK]D-Fender: just put 2 users and make a test call ... if that takes 3 days ... I'll have to look for something more managable |
| 22:11.18 | [TK]D-Fender | patkoscsaba: Go try. I've linked you to the book... |
| 22:11.27 | ManxPower | patkoscsaba: *nod* If you don't want to do the work you should look at a different system. |
| 22:11.36 | [TK]D-Fender | patkoscsaba: there are 4 files you'll have to master as well as learning how to use *'s CLI |
| 22:12.05 | patkoscsaba | [TK]D-Fender: yes, oreilly's book has 576 pages, i've been already there ... |
| 22:12.16 | [TK]D-Fender | patkoscsaba: Good, then get cracking... |
| 22:12.30 | carrar | everything you need to know to make it work is in that book |
| 22:12.33 | carrar | start reading! |
| 22:13.09 | patkoscsaba | ManxPower: I may be a LInux System Administrator longer than your ... so please drop these presumptions like "these people didn't even know Linux" |
| 22:13.18 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 22:13.28 | ManxPower | patkoscsaba: Were you at the class? I was refering to the Asterisk Fast Start class. |
| 22:13.28 | patkoscsaba | ok ... thanks |
| 22:13.48 | patkoscsaba | that was very unhelpfull ... |
| 22:13.52 | carrar | or if you want it to do just do what you want without doing anything just type: Make Asterisk --complete-working-system |
| 22:13.55 | ManxPower | Now if you know Linux, it AND you have already read the book, then chances are you can get up what you want in a day or so. |
| 22:15.07 | ManxPower | Asterisk is NOT really a PBX. Asterisk is a TOOLKIT that lets you build/write/program a PBX. |
| 22:15.07 | patkoscsaba | I DON'T want to read 576 pages right now, I don't want conference calles right now ... I just want ot be able to define 2 USERS AND MAKE A TEST CALL! |
| 22:15.07 | [TK]D-Fender | ManxPower: I wish they'd drop that tagline, but it'd hurt their sales... |
| 22:15.07 | ManxPower | patkoscsaba: As we have said, best of luck with that. |
| 22:15.14 | [TK]D-Fender | patkoscsaba: I linked you something earlier..... |
| 22:15.29 | carrar | start microsoft netmeeting then |
| 22:16.06 | carrar | that should take less then 30 mins |
| 22:16.11 | ManxPower | Lets see, "define users" You don't define users in Asterisk. You define devices (sip.conf/zapata.conf/iax.conf/mgcp.conf/sccp.conf/skinny.conf) and extensions (extensions.conf) |
| 22:16.21 | patkoscsaba | and as I sayd ... very selfish and unhelpy from you ... lucky me, there are other VOIP/PBX servers for free ... |
| 22:16.24 | patkoscsaba | bye bye |
| 22:16.34 | carrar | hahah wtf |
| 22:16.35 | ManxPower | Extensions.conf is what ties a dialed number into a destination device. |
| 22:16.38 | ManxPower | Poor sod. |
| 22:16.48 | carrar | i think |
| 22:16.50 | [TK]D-Fender | ManxPower: You really did blast him... |
| 22:16.50 | carrar | Lucky for us |
| 22:17.31 | ManxPower | I was hoping he'd take is sorry ass to a different platform. |
| 22:18.01 | *** join/#asterisk N3o (n=N3o@79.114.65.151) |
| 22:18.15 | [TK]D-Fender | that was actually a very sad series of attacks... |
| 22:18.31 | ManxPower | At this point I expect that he might go thru the same experience with other platforms (not liking people that don't want to do the work is pretty universal) or he'll wise up and buckle down and start learning. |
| 22:19.58 | N3o | who the fuck do you think you are? if someone asks for help then help him! Remember, you asked yourself for help and someone helped you! damn idiots.... |
| 22:20.01 | *** part/#asterisk N3o (n=N3o@79.114.65.151) |
| 22:20.16 | carrar | there was no channel when I started |
| 22:20.32 | carrar | no book |
| 22:20.37 | ManxPower | I did not see an actual question. |
| 22:20.40 | carrar | not much of any web pages |
| 22:20.44 | *** join/#asterisk loproc (i=d40a3735@gateway/web/ajax/mibbit.com/x-847d7241e57bcb17) |
| 22:20.53 | carrar | had to actually read the readmes |
| 22:20.55 | carrar | heh |
| 22:20.56 | ManxPower | I saw someone expecting to have a working system in a couple of hours. |
| 22:21.45 | carrar | oh |
| 22:21.51 | carrar | I see, comeplete and quickly leave |
| 22:21.55 | carrar | err complain |
| 22:21.59 | carrar | that rocks |
| 22:23.06 | [TK]D-Fender | I learned how to program looking at the MS BASIC book which was basiaclly just syntax for each command. My imagination told me how to structure them to do what I wanted. Show me a rock and I'll make a statue. |
| 22:23.18 | loproc | Hey. I've installed asterisk-gui on top of my asterisk and I'm having a bit of trouble understandig how to use it - My setup includes 4 sip accounts at an external provider (for pstn connection) and a quite simple dialplan (call comes in on account A and a group of phones ring simultaineusly, or account B and another group, etc.). Anyone who can help me getting started? |
| 22:25.02 | ManxPower | loproc: best of luck almost nobody on this channel uses a GUI -- perhaps you'll find more people with experience with that GUI on a channel for that GUI? See the /topic. |
| 22:25.04 | [TK]D-Fender | loproc: What have you succeeded in doing so far? |
| 22:26.15 | loproc | [TK]D-Fender: I've got four trunks created and they register fine, however incoming calls aren't routed to the respective calling groups... |
| 22:26.45 | [TK]D-Fender | loproc: Go look in CLI how the call is coming in. Enable SIP debug, etc as appropriate. |
| 22:26.52 | [TK]D-Fender | loproc: PASTEBIN is your friend. |
| 22:26.54 | [TK]D-Fender | ~pb |
| 22:26.55 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:29.52 | jaytee | wow, just got back from shopping and I was reading the scrollback. boy the fun just never stops in this channel :-) |
| 22:30.09 | carrar | NEVER! |
| 22:30.20 | loproc | [TK]D-Fender: Just to make sure I've got the right setup - I'm supposed to create a trunk, and an Incoming call rule with Pattern "_X.", right? |
| 22:30.24 | [TK]D-Fender | jaytee: Why stop when we can stop it from ever starting ;) |
| 22:30.46 | carrar | It's consistant as the ever rising deficit |
| 22:30.52 | jaytee | "HEY!!! TELL ME HOW TO SETUP ASTERISK NOW WITH TWO PHONES IN LESS THAN 30 MINUTES AND DON'T GIVE ME ANY SHIT!!! I'VE BEEN A LINUX SYSADMIN RETARD WAY LONGER THAN YOU!!!" |
| 22:30.58 | [TK]D-Fender | loproc: that makes all sorts of assumptions. that its NUMERIC, 2 digits or longer, and even landing in that CONTEXT |
| 22:31.11 | ManxPower | jaytee: We need that on a t-shirt |
| 22:31.12 | [TK]D-Fender | loproc: Hece why you need to open your eyes and look at the CALL. |
| 22:31.22 | carrar | hahah |
| 22:31.26 | jaytee | ManxPower, that'd have to be a real small font |
| 22:31.44 | [TK]D-Fender | ManxPower: You bit his head off. Hard. This could have been handled much better... |
| 22:31.46 | ManxPower | jaytee: OK, maybe a dress then. 8-| |
| 22:31.57 | loproc | [TK]D-Fender: Maybe i should just stick to the plain old asterisk without any GUI crap... |
| 22:31.59 | ManxPower | [TK]D-Fender: I disagree other than near the end. |
| 22:32.26 | ManxPower | The whole "do it for me" mentality was getting annoying. |
| 22:32.30 | jaytee | "VOIP? Sure! Would you like that in Asterisk Digium Orange or Trixbox Puke Green? |
| 22:32.41 | [TK]D-Fender | ManxPower: He didn't ask us to, he asked for docs, etc. |
| 22:32.49 | [TK]D-Fender | ManxPower: Point him there and be done with it. |
| 22:32.50 | ManxPower | jaytee: I want it in sangoma blue! |
| 22:33.09 | [TK]D-Fender | ja/me delcares a War On Pastels |
| 22:33.13 | [TK]D-Fender | delcares a War On Pastels |
| 22:33.19 | ManxPower | [TK]D-Fender: we did, he said "(4:15:06 PM) patkoscsaba: I DON'T want to read 576 pages right now, I don't want conference calles right now ... I just want ot be able to define 2 USERS AND MAKE A TEST CALL!" |
| 22:33.25 | loproc | [TK]D-Fender: I've got it running quite well without the GUI - only it's not as stable as i would like it to be (often only one-way audio etc.) |
| 22:34.03 | [TK]D-Fender | loproc: 1-way audio is almost always because the system was not configured to work behind NAT |
| 22:35.28 | garywsmith | is digi doing some server maitanence today or something . Now I can't do a yum install of anything because it's reporting that packages.digium.com is refusing connections. |
| 22:35.33 | loproc | [TK]D-Fender: Yes, but the I would suppose it would be every time? This is "sometimes" and dependig on which party initiates the call... |
| 22:35.47 | [TK]D-Fender | loproc: no thats exactly WHY it happens |
| 22:36.14 | jaytee | garywsmith, not sure about the packaging site but their SVN was down earlier |
| 22:36.26 | loproc | [TK]D-Fender: Okay - I'll just try going through the configuration... |
| 22:36.27 | garywsmith | svn is up now |
| 22:36.36 | [TK]D-Fender | loproc: If * is behind NAT and calls OUT, then RTP keeps the NAT mapping alive for the call. ABout 2 mins (give or take) after, it closes up and that same endpoint you called can;'t call back because the router has no clue where to send the call to |
| 22:36.45 | *** join/#asterisk adorah (n=Administ@80.250.159.240) |
| 22:38.04 | carrar | Manx |
| 22:38.04 | carrar | http://www.osburn.com/asterisk-shirt.png |
| 22:38.06 | carrar | bwahaha |
| 22:38.55 | loproc | [TK]D-Fender: I've enabled NAT in sip.conf (nat = yes) and set externip and localnet. I've set fromdomain and at each peer (connected phone) I've set qualify = yes and nat = yes... What am I missing? |
| 22:39.29 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 22:40.18 | ManxPower | loproc: the portforwarding |
| 22:40.21 | ManxPower | ~sipnat |
| 22:40.22 | 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 |
| 22:40.23 | [TK]D-Fender | ^^^ |
| 22:40.41 | loproc | [TK]D-Fender: Actually I recently discovered that when removing tT from the Dial commands (transferring) it works much better (maybe because of reinviting) |
| 22:40.45 | [TK]D-Fender | carrar: You ARE a bastard... |
| 22:40.45 | ManxPower | loproc: nat=yes is only required if the REMOTE side is behind NAT. |
| 22:40.50 | carrar | hahah |
| 22:40.55 | loproc | ManxPower: I've forwarded the ports ;o) |
| 22:41.08 | [TK]D-Fender | loproc: Then again, PASTEBIN is your friend. |
| 22:41.33 | ManxPower | loproc: you forwarded all ports listed in rtp.conf (or 10,000 - 20,000 UDP not TCP) |
| 22:41.57 | carrar | err that was jaytee, not Manx that pasted that |
| 22:42.06 | loproc | ManxPower: I'll just check again... |
| 22:42.36 | ManxPower | loproc: With the exception of really brain dead routers the instructions in ~sipnat should work. |
| 22:42.37 | jaytee | what? pasted what? |
| 22:42.48 | carrar | jaytee, http://www.osburn.com/asterisk-shirt.png |
| 22:42.59 | ManxPower | I repasted it. It was written to the channel |
| 22:43.12 | jaytee | hahaha |
| 22:43.21 | ManxPower | I don't go around pasting private conversations to a public channel no matter how much someone deserves it. |
| 22:43.35 | loproc | ManxPower: The router is a pfSense |
| 22:43.46 | ManxPower | carrar: maybe a "wrap around" style? |
| 22:44.01 | ManxPower | loproc: Is there any SIP or RTP specific stuff enabled on the pfSense? |
| 22:44.06 | carrar | with beltbuckle locks at the end of each arm |
| 22:44.35 | loproc | ManxPower: No, I don't believe so... |
| 22:45.06 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1279505477.dsl.bell.ca) |
| 22:45.21 | loproc | ManxPower: RTP ports from rtp.conf is forwarded (UDP only) |
| 22:45.26 | [TK]D-Fender | sighs... |
| 22:45.28 | ManxPower | loproc: If there is it will undo the localnet/externip stuff Asterisk does. Also did you ever pastebin your sip.conf masking only passwords? I think [TK]D-Fender requested that. |
| 22:45.47 | [TK]D-Fender | ManxPower: No exactly... |
| 22:45.49 | loproc | ManxPower: Will do... |
| 22:45.57 | [TK]D-Fender | ManxPower: Could well indicate if its wrong... |
| 22:46.02 | [TK]D-Fender | Not* |
| 22:46.13 | *** join/#asterisk jov4n (n=jovan@87.19.106.7) |
| 22:46.22 | carrar | hrmm cafepress doesn't have any straightjackets |
| 22:46.38 | jov4n | Hi |
| 22:46.52 | jaytee | http://atom.smasher.org/highway/?l1=Hey%21+Tell+me+right+now&l2=howto+setup+2+phones+&l3=with+Asterisk+in+30mins&l4=%26+don%27t+give+me+shit%21%21 |
| 22:47.08 | carrar | hahah |
| 22:47.19 | jaytee | I love that site! |
| 22:47.30 | carrar | the cult has started! |
| 22:47.35 | [TK]D-Fender | jaytee: Not to validate the conversation that spawned it.... but thats fucking HILLARIOUS ;) |
| 22:47.36 | jaytee | hehee |
| 22:48.50 | jaytee | I think it clearly displays the guy's attitude. |
| 22:49.15 | loproc | [TK]D-Fender and ManxPower: http://pastebin.com/d47272764 |
| 22:49.51 | [TK]D-Fender | loproc: Common error |
| 22:50.07 | loproc | [TK]D-Fender: ? |
| 22:50.11 | [TK]D-Fender | loproc: REGISTERs must come AFTER everything else under [general] |
| 22:50.27 | [TK]D-Fender | loproc: they terminate the section |
| 22:50.32 | loproc | [TK]D-Fender: Aha, I'll just correct that |
| 22:50.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:51.12 | ManxPower | loproc: I'm sorry, I don't help with users.conf configs. |
| 22:51.12 | ManxPower | Perhaps [TK]D-Fender can help you. |
| 22:51.42 | [TK]D-Fender | ManxPower: thats sip.conf... |
| 22:52.00 | loproc | [TK]D-Fender: So when the * registers, it hasn't parsed the SIP settings? |
| 22:52.03 | ManxPower | [TK]D-Fender: this is a sip.conf? [unoteliktp](unotel) |
| 22:52.30 | [TK]D-Fender | loproc: * doesn't get to parse them. the Register statemenst say "[general] is FINISHED" |
| 22:52.35 | [TK]D-Fender | ManxPower: Yes, it is |
| 22:52.53 | ManxPower | [TK]D-Fender: OK then. I'm sorry, I don't help with templates. Maybe [TK]D-Fender can help you? |
| 22:53.13 | [TK]D-Fender | ManxPower: not the issue anyway.... I ID'd in about 3 seconds flat |
| 22:53.43 | ManxPower | [TK]D-Fender: enlighten us |
| 22:54.25 | [TK]D-Fender | ManxPower: I did... read up. |
| 22:54.38 | [TK]D-Fender | ManxPower: and go caffienate, you must be running on empty |
| 22:54.40 | ManxPower | [TK]D-Fender: I shall wait for the results. |
| 22:55.00 | ManxPower | I never believed that urban myth about register lines, but I have been proved wrong in the past. |
| 22:55.30 | [TK]D-Fender | ManxPower: I've proved it here countless times, and never ever doubt how silly much of *'s design is. |
| 22:55.43 | [TK]D-Fender | ManxPower: "register" shouldn't even be a directive like that. |
| 22:55.52 | ManxPower | I'm just waiting for loproc to report that it fixed the problem. |
| 22:56.11 | loproc | ManxPower: I'm testing... |
| 22:56.18 | [TK]D-Fender | ManxPower: the way variables and app parameters are handled is BLEH. I did better from scratch myself in the language I wrote. And I took no theory on it |
| 22:57.08 | [TK]D-Fender | ManxPower: Lack of type, raw parsed chars. funky nesting. * dialplan is a FUBAR design. |
| 22:57.20 | carrar | jaytee, and just to complete: http://www.osburn.com/asterisk-churchsign.png |
| 22:57.32 | [TK]D-Fender | ManxPower: So we do our best and simply bookmark the oddities |
| 22:57.48 | jaytee | <PROTECTED> |
| 22:58.00 | [TK]D-Fender | jaytee: Wanna see a program written in it? :) |
| 22:58.02 | ManxPower | jaytee: It's called ADA! |
| 22:58.31 | jaytee | carrar, the pentecostals will be gunning for you! :-) |
| 22:58.36 | carrar | heh |
| 22:58.38 | jaytee | [TK]D-Fender, sure! |
| 22:59.22 | [TK]D-Fender | jaytee: http://pastebin.com/m71cab482 |
| 22:59.34 | [TK]D-Fender | jaytee: behold the worlds first... MID level programming language! |
| 22:59.49 | [TK]D-Fender | jaytee: You can see where I was starting to templete coding a loop structure. |
| 22:59.55 | ManxPower | So I take it Pascal was the first language you learned? |
| 23:00.15 | [TK]D-Fender | ManxPower: BASIC, then Pascal. I wrote this language IN TP7 |
| 23:00.20 | loproc | ManxPower: When dialing out to a PSTN phone from a SIP phone, i get oneway audio... that is audio from the sip phone dosn't get to the PSTN |
| 23:00.51 | ManxPower | loproc: sounds like you might not have a canreinvite=no somehwere you need it. |
| 23:01.11 | [TK]D-Fender | loproc: Good. Step one down. Next re-read the guide. There is an EXTREMELY important thing you left out. |
| 23:01.11 | jaytee | kinda makes me think of what code would look like if Cobol II and Delphi were in a head-on collision driving at max speed in two Geo Metros. |
| 23:01.16 | ManxPower | BASIC has ruined to many potentially good programmers |
| 23:01.29 | ManxPower | Hey! I like COBOL! |
| 23:01.56 | jaytee | Completely Obsolete But Obstinately Lingering |
| 23:02.03 | ManxPower | The good old days when all you had to do was get totals and subtotals right. |
| 23:02.38 | loproc | ManxPower: There's a canreinvite = no in sip.conf in the [unotel](!) |
| 23:02.46 | [TK]D-Fender | jaytee: My language ran on 3 parsing engines. 1 was the "pre-scan" that read each line in. It "numbered" the labels, etc. then there are the directive "#SOMETHING" commands. that was handled in 1 subroutine. Then there is the App engin which did all of the applications. |
| 23:03.01 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:03.35 | [TK]D-Fender | jaytee: Here's another sample : http://pastebin.com/m63d455f4 |
| 23:03.38 | ManxPower | loproc: and for your PHONES? |
| 23:03.38 | jaytee | or try doing a rewrite of a cobol app written in 1981 by a guy no longer with the company and of course there's no documentation and he didn't bother to comment hardly at all and used actors and actresses names for subroutines. GOTO SINATRA |
| 23:04.46 | [TK]D-Fender | jaytee: WriteLn(' ', dircount, ' file(s) ', dirsize, ' bytes'); , FindFirst(s, $3F, filename, size, date, time, attr) <- See? I can mix vars & constant values, numic, text, hex, boolean, etc without the raw-text mashup that is extensions.conf |
| 23:05.00 | loproc | ManxPower: Nope - I'll try putting it there... |
| 23:05.13 | [TK]D-Fender | loproc: under [general] and EACH section. |
| 23:05.24 | lesouvage | Does any of you have experience with astercrm (http://astercc.org/about-astercrm). It looks like a proper lighweight crm solution on top of a normal non gui Asterisk. |
| 23:05.49 | [TK]D-Fender | jaytee: I'd love to adapt my parser to *... |
| 23:05.54 | garywsmith | jaytee: Been there, done that. They used names of wines though... Trefethen was misspelled in a report and they complained that I misspelled it... Not that it was BROKEN. |
| 23:05.58 | loproc | ManxPower: That did the trick! |
| 23:06.44 | loproc | I thought reinvites was i nice and pleasant thing... |
| 23:06.52 | loproc | i = a |
| 23:07.29 | [TK]D-Fender | jaytee: Here is the actual parser code : http://pastebin.com/m5157f201 |
| 23:07.44 | [TK]D-Fender | jaytee: down & dirty text line parsing. |
| 23:07.52 | [TK]D-Fender | jaytee: char by char building it up. |
| 23:08.52 | [TK]D-Fender | jaytee: Fun starts around 436 |
| 23:09.21 | ManxPower | loproc: It is a nice thing---when NAT is not involved |
| 23:09.39 | jaytee | [TK]D-Fender, maybe modify what's in Asterisk.NET using Visual C# 2005 to include a mod of your parser as a front end to Asterisk with AMI and SQL. Maybe even mix in the AsteriskCSTA project code too so it would support CTI apps :-) |
| 23:09.53 | [TK]D-Fender | jaytee: As we use "check levels" to keep count of where we are |
| 23:10.01 | [TK]D-Fender | jaytee: Now you're just being bitter :p |
| 23:10.25 | [TK]D-Fender | jaytee: It was really hard for me not to ManxPower that guy last week... |
| 23:10.43 | [TK]D-Fender | ManxPower: You've been verb'd. Blog it! |
| 23:10.56 | loproc | ManxPower: Oh - I'll remember that :o) Anyhow, thank you all so much for helping me on this... Asterisk is FUN! |
| 23:11.26 | jaytee | [TK]D-Fender, ok I just took a quick scan of lines 430 to 564 of that last pastebin and all that nesting's giving me a brain hemorrage. |
| 23:11.43 | garywsmith | loproc: like hot poker in the eye fun... ;) |
| 23:11.59 | [TK]D-Fender | jaytee: its pretty readable I find... |
| 23:12.32 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 23:12.43 | jaytee | [TK]D-Fender, it's very Pascalish |
| 23:13.13 | garywsmith | [TK]D-Fender: is that ADA? |
| 23:13.16 | [TK]D-Fender | jaytee: Yup... but not too different from C really. just more "wordy" |
| 23:13.28 | loproc | garywsmith: No - more like "Hahahaaaaa, I've got my own telephony system and YOU don't! In you face!" |
| 23:13.47 | [TK]D-Fender | garywsmith: that last past was Pascal code for part of the parser for the other code samples I PB'd |
| 23:14.09 | loproc | Anyhow, I'll go wash the cinnamon out of my hair (just turned 25) |
| 23:14.20 | garywsmith | It's been like, well, 18 years since I've seen pascal. |
| 23:14.36 | garywsmith | actually, 22 years. |
| 23:14.37 | [TK]D-Fender | jaytee: the App parser was actually pretty clean... the PoundParser would make your head pull a "Scanners" |
| 23:14.42 | jaytee | [TK]D-Fender, I bet you could cut that code down to 1/4 of the size porting it to C# |
| 23:14.52 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 23:15.02 | jaytee | lol, Scanners |
| 23:15.07 | [TK]D-Fender | jaytee: Nope... no chance. only thing it'd change are the {} |
| 23:15.51 | [TK]D-Fender | jatthe part that lets me define vars, do ASM-grade jumps instead of real "logic" |
| 23:16.16 | [TK]D-Fender | jaytee: Notice the use of the Compare() function. That sets a result flag that #JE #JNE, etc use |
| 23:17.06 | jaytee | [TK]D-Fender, ever heard of Anders Hejlsberg? |
| 23:17.19 | [TK]D-Fender | jaytee: funny part is this started as a single-line parser for a telecom app I wrote so I could send command strings w/ var data, etc :) |
| 23:17.29 | [TK]D-Fender | jaytee: Name sounds familiar.. |
| 23:18.28 | [TK]D-Fender | jaytee: reading up on him now |
| 23:19.13 | jaytee | [TK]D-Fender, I'm not giving you static. The concept is sound. I'm just saying, it has advantages over a plain text parser but 1) is it as fast and 2) how many people would want to code in it? |
| 23:19.30 | [TK]D-Fender | jaytee: I did it because I COULD |
| 23:19.54 | [TK]D-Fender | jaytee: it started as a macro-parser for a stupid termial app I wrote :) |
| 23:20.07 | [TK]D-Fender | jaytee: I just figured I'd add one more little thing... |
| 23:20.20 | [TK]D-Fender | jaytee: and then another.. next you know, its a full language. |
| 23:20.21 | jaytee | met him at a Boston Computer Society meeting back in 91. He handed out free copies of Borland's latest Pascal. Now he works for Microsoft. |
| 23:20.31 | [TK]D-Fender | jaytee: and yes, an interpreted one, but it works. |
| 23:21.03 | garywsmith | Windows 1.0 started really as an Ascii menuing system. They just wanted to add one more little thing... LOL |
| 23:21.04 | [TK]D-Fender | jaytee: Just saying that *'s retard Dialplan app parser sucks, does it my way, only WORSE, and I could do better with it :) |
| 23:21.17 | jaytee | garywsmith, hahahaha |
| 23:21.31 | [TK]D-Fender | Garywyou forgot "and needed to remain ABI compatible with everything right back to that ASCII release) |
| 23:21.52 | [TK]D-Fender | load chan_bloatware.so |
| 23:21.52 | jaytee | garywsmith, I'd always thought that Windows 1.0 was a complete rewrite of IBM's Topdesk |
| 23:22.04 | [TK]D-Fender | misses DesqView |
| 23:22.11 | jaytee | rofl |
| 23:23.24 | jaytee | or Quarterdeck's QEMM 386? trying to load drivers in the 384K upper memory region like that guy at Contadina trying to get 8 great tomatoes in that little bitty can. |
| 23:23.29 | [TK]D-Fender | jaytee: Seriously though... I'd love to rewrite the * App parser... |
| 23:23.42 | LoRez | hey now, QEMM kicked ass |
| 23:24.05 | garywsmith | I have an original Windows 1.0 disk here in one of my drawers. I don't recall windows 2.0 though. I think the first one I really used was 3.0 which I got for free. |
| 23:24.05 | ManxPower | jaytee: I used to be at the Quarterdeck telephone tech support, for both QEMM and DESQview. |
| 23:24.21 | [TK]D-Fender | jaytee: I 1-up'd that... I ran VIDRAM to recover VGA BIOS memory to free up 640K range. I always had over 640K FREE after loading my super-floppy :) |
| 23:24.22 | jaytee | [TK]D-Fender, I was being serious about Asterisk.NET with a full parser as front end with * AMI as a backend. |
| 23:24.59 | ManxPower | res_perl might be a good starting point for a new dialplan |
| 23:25.05 | [TK]D-Fender | jaytee: then I would load up DesqView for TP programming, etc/ |
| 23:25.09 | garywsmith | jaytee: download the source, split off and write it. I mean, C#'s claim to fame is rapid code release... |
| 23:25.12 | jaytee | ManxPower, before DOS 6 made QEMM "unneccessary" in most cases it was an awesome product |
| 23:25.41 | garywsmith | we used desqview 386 for multitasking a bbs |
| 23:25.46 | ManxPower | jaytee: Quarterdeck got the FIRST software patent. |
| 23:26.15 | garywsmith | ManxPower: and things haven't been the same since. |
| 23:26.19 | jblack | on an unrelated note, I wonder who was first to kill a cute ittle kitten. |
| 23:26.21 | jaytee | especially when you needed to load the Netware shell and IPX protocol stack to free up conventional mem for WordPigPerfect 5.1 |
| 23:26.22 | [TK]D-Fender | garywsmith: Been there, done that :) Also that language I wrrote was being adapted to be used in building a BBS |
| 23:26.31 | ManxPower | garywsmith: An example of not being able to see the consequences. |
| 23:26.52 | jaytee | jblack, I'm probably the oldest one here so I was probably the first guy in here to "rub on off". |
| 23:26.54 | garywsmith | true |
| 23:27.55 | ManxPower | It was only really a symbolic victory. Microsoft had used the technology in their own memory manager before the patent was granted, but after it was filed. I don't think they got a penny from it. |
| 23:28.53 | jaytee | ManxPower, I never once had to call Quarterdeck for tech support. Back in those days companies actually wrote documentation. |
| 23:29.28 | ManxPower | jaytee: there were many hardware incompatibilities, all special cases |
| 23:29.58 | jaytee | even after DOS 6.22 came out there were alot of instances where QEMM was a better solution because it was more flexible. |
| 23:31.06 | ManxPower | in that era all the magic was done by the "optimize" program. |
| 23:32.18 | KP7 | has anyone else had problem compiling dahdi-linux-2.0.0? |
| 23:32.20 | jaytee | with the exception of the ability to multitask and run concurrent programs and excluding linux from the mix, I've seen computers go from single megahertz speeds with 1MB of RAM to multi gigahertz speeds with 32GB of RAM and STILL the subjective response time almost "feels" about the same. |
| 23:32.47 | garywsmith | optimizer! what's that. We're writing software today that consumes 100mb's just to load a simple graphical form and do something simple. |
| 23:33.48 | jaytee | I remember the disaster that Lotus encountered when they took the codebase for Lotus 123 2.1 written in assembler and ported it to C. What a meltdown. |
| 23:34.07 | jaytee | Not as embarrasing as Ashton-Tate's DBaseIV debacle. |
| 23:34.40 | outtolunc | long live dbaseIII+ |
| 23:34.47 | outtolunc | giggles |
| 23:34.49 | jaytee | "World's First Shrink Wrapped Virus and Relational Database In One!!!" |
| 23:35.06 | [TK]D-Fender | jaytee: Lotus 1-2-3 was good until Excel came around. Excel was always better really, and lets not forget Lotus NOTES. Craptastic. Word was patently inferior to WordPerfect, but thats bundling for you... |
| 23:35.24 | [TK]D-Fender | outtolunc: I still work with DBase files... |
| 23:35.39 | [TK]D-Fender | outtolunc quite sad |
| 23:35.50 | jaytee | back in 91 Notes in beta showed alot of promise. Once IBM took over Lotus it was all downhill from there. |
| 23:35.54 | adorah | Hi Does any one knows the limit of recorded calls either in storage and/or in no. of files in *? |
| 23:36.10 | [TK]D-Fender | jaytee: I despise.. no wait, LOATH Lotus Notes. |
| 23:36.23 | [TK]D-Fender | adorah: Limit = your disk space |
| 23:36.38 | jaytee | [TK]D-Fender, that's just because you haven't used Sharepoint yet :-) |
| 23:36.43 | garywsmith | Lotus... Can't tell you how many times that program was bought and sold. Didn't Novell finally end up with it? |
| 23:36.52 | [TK]D-Fender | jaytee: We dodged that bullet! |
| 23:37.04 | [TK]D-Fender | jaytee: Shot down for budget reasons! w00t |
| 23:37.26 | [TK]D-Fender | jaytee: Now I'm looking for a good groupware package (OSS) to meet our needs |
| 23:37.32 | jaytee | garywsmith, no Novell came out with it's own alternative to Notes called Groupwise which was actually a decent product in comparison |
| 23:37.40 | ManxPower | [TK]D-Fender: consider http://atmail.comf/ |
| 23:37.51 | ManxPower | and atmail.con too. |
| 23:37.53 | [TK]D-Fender | jaytee: the calendaring on the ones I've seen has been only a single calendar and not "multi-project" oriented |
| 23:37.55 | ManxPower | argh. |
| 23:38.01 | adorah | <[TK]D-Fender>Not correct, I used about 18000 recirded files on a 73GB disk that used less than 10 Gigs and got a messege that the folder is full |
| 23:38.29 | [TK]D-Fender | adorah: that sounds more like an OS limit |
| 23:38.31 | garywsmith | but lotus was owned by corel which was absorbed by Novel at one point. |
| 23:38.34 | adorah | It stopped recording |
| 23:38.38 | garywsmith | in the end that is. |
| 23:38.49 | adorah | Maybe the OS is Centos 5.2 |
| 23:38.52 | ManxPower | adorah: that should have nothing to do with Asterisk. That is likely to be an OS or filesystem limit. |
| 23:39.39 | jameswf | the future of piracy? http://www.pinoytux.com/linux/tip-hiding-files-inside-an-image-in-linux |
| 23:39.43 | ManxPower | on the other hand I suspect just trying to read directory entries for 180,00 files would take a long time. |
| 23:40.11 | ManxPower | Most of the time apps use a tree to split the files across multiple directories. |
| 23:40.12 | jaytee | probably using LVM and the volume the mailboxes are in is maxed out |
| 23:40.21 | adorah | <ManxPower>That is not the point. the point is that asterisk stopped recording calls |
| 23:40.40 | ManxPower | adorah: I'm sorry to hear that. Fix your OS and Asterisk should start recording calls agian. |
| 23:40.50 | jblack | adorah: Yeah. You'll start seeing operating system problems once you get past a few thousand directory entries. |
| 23:41.37 | jblack | Though asterisk itself shouldn't have the problem. Probably you're going past what the filesystem itself can handle (the shell fails much sooner) |
| 23:42.13 | adorah | Any advise? I thought Centos 5.2 can handle much more entries |
| 23:42.33 | adorah | I had plenty of space there |
| 23:42.48 | jblack | Just set up a cron job each night that moves each day's recordings into a subdirectory.. say mkdir -p "+%Y/%m/%d", then move all the files into that dir. |
| 23:42.50 | garywsmith | adorah: well, inodes != size. Check the inodes... |
| 23:43.41 | adorah | Thx everyone will try it - need it now for a call center coz it is election time here..hehe |
| 23:43.44 | garywsmith | if you tweaked the format for large files you get less inodes. |
| 23:44.05 | garywsmith | adorah: "df -i" |
| 23:44.51 | jblack | Changing filesystems can help too, but putting 40K files in one unnested dir is a nightmare to navigate anyways. |
| 23:44.56 | ManxPower | The generally recognized way to handle large numbers of files is using a tree directory design. Mail servers do this so do proxy servers. |
| 23:45.13 | garywsmith | true. |
| 23:45.27 | ManxPower | Frequently based on the first few chars of the file name, but not always. |
| 23:45.39 | jaytee | jblack, especially since ext3 only handles 32000 links per inode |
| 23:45.49 | ManxPower | jaytee: I just saw that on a Wikki |
| 23:46.04 | jblack | Yeah. I couldn't remember the ext3 limit. |
| 23:46.05 | jaytee | if only Hans hadn't killed Nina |
| 23:46.23 | garywsmith | LOL |
| 23:46.33 | jblack | I've bumped into it a few times in various software I've worked on. |
| 23:47.02 | garywsmith | well, that's the opensource world for you. One day great software, next day, deprecated because someone killed someone... |
| 23:47.06 | jblack | Hans didn't have proper respect for the value of data. I think it's good he's... unavailable. |
| 23:47.22 | ManxPower | naw, that's just a function of poor management and planning. |
| 23:47.40 | garywsmith | just glad OJ wasn't an open source developer... |
| 23:47.50 | jblack | Did you ever see him use the word "redundant"? |
| 23:47.56 | jaytee | Ian Murdock used to live about 10 miles from me. Not sure if he's moved to the west coast to be closer to Sun's HQ now that he's workin on OpenSolaris |
| 23:48.23 | rob0 | oh no! Ian didn't murder Deb, did he? |
| 23:48.33 | jaytee | no, but they divorced a few years back |
| 23:49.03 | jblack | How is that release coming along, btw? |
| 23:49.15 | jaytee | but the user community refused his request to rename the distro Bitchian |
| 23:49.33 | jaytee | OpenSolaris? |
| 23:49.59 | jblack | Debian. |
| 23:50.03 | garywsmith | Actually didn't he want to name it Open Slo Ars? |
| 23:50.07 | jaytee | still alive and well |
| 23:50.16 | jblack | I guess lenny is the next. |
| 23:50.40 | jaytee | they call it Project Indiana since he's from here or at least lived here in Fishers, IN just north of Indy. |
| 23:51.11 | jblack | I thought they were going to go for quicker releases. I count 19 months since etch. |
| 23:51.13 | garywsmith | I just don't why everything has to have a name. |
| 23:51.14 | jaytee | I tried the beta but it's a pig and good luck getting linux code to run on it without a major hack |
| 23:51.51 | garywsmith | I always say that Linux is as free as your time to fix it... |
| 23:52.58 | jblack | Do people fall for that line often? |
| 23:53.51 | ManxPower | jblack: more often than "What's your sign?", I imagine. |
| 23:54.46 | garywsmith | no, the get stuck with that line |
| 23:57.05 | jaytee | jblack, if you like being on the bleeding edge with Debian why not try lenny or sid? |
| 23:57.25 | gcbirzan | Which are almost the same thing nowadays :-P |
| 23:57.37 | garywsmith | I had to do a setup of the old Veritas agent for a client on their linux box. They run CentOS for VMWare and Ubuntu for everything else. There are a couple issues getting the veritas agent installed as they don't support it. But they can't easily switch to another distro because their software is customized to ubuntu at this point and they don't have anyone to change it. |
| 23:57.50 | garywsmith | I have a similar problem with a lot of stuff in RPath at this point. |
| 23:58.31 | jaytee | makes sign of the cross at the mention of "that other distro" {shudders} |
