| 00:06.08 | *** join/#asterisk spiny (n=spiny@69-196-137-127.dsl.teksavvy.com) |
| 00:08.13 | *** join/#asterisk blinky42 (n=steveb@c-71-230-47-244.hsd1.pa.comcast.net) |
| 00:12.39 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 00:20.49 | *** join/#asterisk Zuchmir (n=dddddd@ool-18bd3bfc.dyn.optonline.net) |
| 00:21.00 | Zuchmir | is there a way to "busy" out a phone line |
| 00:21.13 | Strom_L | of course |
| 00:21.44 | Strom_L | what are you trying to accomplish by busying it out? |
| 00:22.20 | Zuchmir | i have 2 systems mirroring each other, and i want to do maintenance on one |
| 00:22.42 | Strom_L | analog circuits? |
| 00:22.47 | Zuchmir | i have 2 PRI's, one comming in to each system |
| 00:22.59 | Strom_L | uh, you don't "busy out" a PRI |
| 00:23.18 | Zuchmir | they "roll over" from one to the next |
| 00:23.22 | Strom_L | is the one set to roll over to the other if it's completly congested? |
| 00:23.23 | Strom_L | ok |
| 00:23.36 | Zuchmir | so if i can busy out 23 lines, automatically the 2nd system picks up |
| 00:23.39 | Strom_L | have you tried just unplugging it from the T1 card? |
| 00:24.00 | Zuchmir | i want to do it remotely |
| 00:24.27 | Strom_L | ok |
| 00:24.27 | Zuchmir | ... and while callers are on the system, i want to slowly move callers off the 1st system |
| 00:24.37 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 00:24.40 | christophocles | i am having trouble figuring out what this error message means, I am trying to make a call to my asterisk box from an outside line and I get this: [Jun 5 19:23:38] NOTICE[21194]: chan_sip.c:13865 handle_request_invite: Call from 'ipkall' to extension 'cr0d' rejected because extension not found. |
| 00:24.49 | Strom_L | you could just tell asterisk to automatically hangup with a "circuit congestion" cause |
| 00:25.12 | Strom_L | christophocles: you're calling a nonexistent extension. read the message. |
| 00:25.22 | Zuchmir | how do i do that |
| 00:25.24 | christophocles | ok, which extension is nonexistent, ipkall or cr0d |
| 00:25.34 | Strom_L | christophocles: READ THE MESSAGE |
| 00:25.45 | delparnel | "to extension 'cr0d' " |
| 00:26.04 | Strom_L | Zuchmir: use Hangup() and pass it whatever Q.850 says is for circuit congestion |
| 00:26.07 | christophocles | so cr0d is not defined correctly, ok i will try to fix that... but excuse me if I am new to this, jeez |
| 00:26.19 | Strom_L | christophocles: being new is OK -- not reading is not OK |
| 00:26.31 | christophocles | i DID read it, i did not UNDERSTAND it |
| 00:26.39 | Strom_L | sigh |
| 00:27.05 | delparnel | Haha |
| 00:27.21 | christophocles | ive been READING it for the past several hours and trying different things to fix it.. believe me i am tired of reading it |
| 00:27.24 | delparnel | It was just english |
| 00:29.15 | Zuchmir | so Hangup(34) would do? |
| 00:29.36 | Strom_L | is that what Q.850 says is the cause code for congested circuit? |
| 00:29.54 | ManxPower | Q.931 |
| 00:30.30 | Strom_L | ManxPower: also Q.850 |
| 00:30.41 | *** join/#asterisk erojasv (n=erojasv@190.40.83.177) |
| 00:30.51 | ManxPower | christophocles: exten => cr0d,1,Noop(What is a cr0d) |
| 00:31.13 | Strom_L | less non-cause-code related crap to wade through in Q.850 than in Q.931 ;) |
| 00:31.52 | ManxPower | 34 is Congestion/No Circuit available |
| 00:32.19 | Strom_L | Zuchmir: try 21 |
| 00:32.29 | ManxPower | 16 is normal clearing, 17 is busy, 1 is unassigned number |
| 00:32.43 | Strom_L | or 23 |
| 00:32.51 | Zuchmir | it looks like 34 is the code, my q. is, is the syntax hangup(34) |
| 00:33.09 | Strom_L | Zuchmir: that's the syntax, but I don't think you should use 34 |
| 00:33.09 | ManxPower | Zuchmir: "core show application hangup" |
| 00:33.23 | Strom_L | I want you to also try 21 and 23 |
| 00:34.45 | Zuchmir | i guok, thanks, will try... |
| 00:34.51 | ManxPower | also try 3 |
| 00:35.49 | ManxPower | Zuchmir: ask your carrier what they want. |
| 00:36.21 | christophocles | ManxPower, thanks I am making progress now |
| 00:36.24 | *** join/#asterisk fish-bulb (n=cstewart@216.207.245.1) |
| 00:36.53 | ManxPower | Zuchmir: honesty, "shutdown when convenient" is best. That will refuse new calls, not terminate existing calls |
| 00:37.10 | ManxPower | christophocles: now go read The Good Book |
| 00:37.12 | ManxPower | ~book |
| 00:37.13 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 00:37.49 | christophocles | manxpower, thanks im about 85 pages into that |
| 00:38.55 | Zuchmir | manxPower: my q. is, will that cause a rollover to the next PRI |
| 00:43.14 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-add7aeafe28f4f1d) |
| 00:43.45 | Zuchmir | manxPower: looks like "stop when convenient" keeps taking calls |
| 00:51.14 | drmessano | Maybe it's not "convenient" yet |
| 00:51.36 | Zuchmir | the hangup(34) / hangup(23) cause it to rollover to the next "line" on the same PRI, and after 3 "rollovers" it stops trying again! |
| 00:52.35 | Zuchmir | drmessano: i want to stop callers getting through to box A, i want them to get onto box B, untill i finish mainenance on box A |
| 00:59.20 | *** join/#asterisk dwclarkNU (n=dclark@24-151-134-55.dhcp.oxfr.ma.charter.com) |
| 01:00.37 | dwclarkNU | i just built a asterisk pbx with a digium TE210P for our T1s...some users are having their calls dropped after 10-15 minutes, any ideas why or how to troubleshoot? |
| 01:03.29 | JT | dwclarkNU: check zttest |
| 01:03.43 | dwclarkNU | ran that, everything is 99.99xxxx% |
| 01:03.51 | dwclarkNU | i'm assuming that's close enough to 100% |
| 01:04.18 | JT | how long do you run it for/ |
| 01:04.28 | JT | maybe there's some event occuring that makes it drop down |
| 01:04.38 | JT | make sure the card isn't irq sharing |
| 01:04.48 | dwclarkNU | it's been running since this AM |
| 01:06.27 | dwclarkNU | i also have a tdm410P for our fax modems |
| 01:06.30 | dwclarkNU | it's not configured at this time |
| 01:06.38 | dwclarkNU | but it's showing 10 IRQ misses |
| 01:06.45 | dwclarkNU | in zttool |
| 01:07.29 | dwclarkNU | but all calls have been going out over the te210p on our 2 T1 lines...we are using d4/ami em wink start |
| 01:11.19 | JT | yikes |
| 01:11.23 | JT | old school |
| 01:12.20 | dwclarkNU | yea, i know...it was the only thing our old PBX supported and i have a change order in to update it but it's 30 days |
| 01:12.32 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 01:13.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:14.20 | dwclarkNU | JT: any ideas why the unconfigured/unused TDM410P would be experiencing IRQ misses? |
| 01:14.45 | dwclarkNU | i have hyperthreading enabled, should it be disabled? |
| 01:30.01 | *** join/#asterisk asdx (n=diego@adsl-141-8.click.com.py) |
| 01:31.00 | dwclarkNU | i went into my irqs and set it so the t1 card and analog card had unique IRQs and they wernt shared with any other system device...i also disabled anything i could in the bios besides what i needed and disabled hyperthreading |
| 01:32.06 | asdx | why does asterisk opens more than 1 channel for 1 call? |
| 01:32.46 | dwclarkNU | 1 inbound channel to asterisk and one channel to the phone |
| 01:33.04 | dwclarkNU | if it doesnt do a re-invite, it will use 2 channels |
| 01:34.11 | asdx | ok |
| 01:34.17 | asdx | thx |
| 01:34.51 | dwclarkNU | np |
| 01:38.59 | JT | pretty pointless to just talk to yourself for most calls :] |
| 01:39.42 | Zuchmir | is there a way of setting the channel to unavailble using ztcfg? |
| 01:41.03 | Zuchmir | would unused=25 work? |
| 01:41.42 | Zuchmir | (in /etc/zaptel.conf) |
| 01:41.54 | ManxPower | there should be an examle of it in zaptel.conf.sample or something like that in the zaptel source dir |
| 01:53.01 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 01:53.28 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-54a2b553684c9e10) |
| 01:59.37 | hsv-al | JFC |
| 01:59.40 | *** part/#asterisk mprebello (n=marcel@189.78.166.231) |
| 01:59.43 | hsv-al | 5 days and counting, and this finally shows up: |
| 01:59.43 | hsv-al | 8:33 PM |
| 01:59.43 | hsv-al | Arrived at FedEx location |
| 01:59.43 | hsv-al | NASHVILLE, TN |
| 02:07.49 | DrukenLPY | hsv-al: what's the final dest? |
| 02:08.17 | *** join/#asterisk Sky-Knight (n=Sky-Knig@ip68-2-137-221.ph.ph.cox.net) |
| 02:08.27 | *** part/#asterisk Sky-Knight (n=Sky-Knig@ip68-2-137-221.ph.ph.cox.net) |
| 02:08.41 | hsv-al | bout 1.5 hours away |
| 02:08.44 | hsv-al | so should be here 2morow |
| 02:14.09 | nick125 | hsv-al: Whatcha ordering? |
| 02:16.00 | hsv-al | 411P |
| 02:16.07 | nick125 | mmm |
| 02:16.14 | hsv-al | $77 - 2 port |
| 02:16.16 | hsv-al | xo/xs |
| 02:16.21 | hsv-al | $239 total |
| 02:18.20 | *** join/#asterisk christoz (n=someones@ipa116.51.91.tellas.gr) |
| 02:25.25 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 02:36.02 | *** join/#asterisk ix33 (n=ix@206.222.13.162) |
| 02:44.27 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 02:44.41 | asdx | was chan_sip rewritten in ast 1.6? |
| 02:46.28 | *** join/#asterisk sack (n=sack@33.Red-81-32-160.dynamicIP.rima-tde.net) |
| 02:46.30 | ix33 | bleh support dragging their feet it seems |
| 02:48.49 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 02:53.44 | delparnel | Can I set a DISA to dial out through a different trunk? |
| 02:54.39 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:54.47 | ManxPower | delparnel: We don't really support GUI installs of Asterisk. |
| 02:57.38 | delparnel | Can I do it in the config files? |
| 02:58.05 | ManxPower | sure, but if you were using config files it would be totally obvious that your question makes no sense. |
| 02:59.03 | *** part/#asterisk xieliwei (n=vz1c1kPp@bb121-6-139-177.singnet.com.sg) |
| 03:01.29 | ManxPower | You tell DISA what context to jump to and you dial out however you want to dial out. |
| 03:01.37 | *** part/#asterisk spiny (n=spiny@69-196-137-127.dsl.teksavvy.com) |
| 03:03.21 | delparnel | So I'd create a custom context to dial out on a different trunk? |
| 03:04.13 | TJNII | ~trunk |
| 03:04.13 | jbot | trunk is probably a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 03:05.47 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 03:15.16 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:19.36 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-42.hsd1.ky.comcast.net) |
| 03:27.20 | ManxPower | YAY! Verizon is buying Alltell |
| 03:30.42 | jaytee | why is that a yay? |
| 03:32.13 | drmessano | WIIIIZARD! |
| 03:32.54 | asdx | will 1.6 introduce architectural changes? |
| 03:33.03 | drmessano | Yes |
| 03:33.53 | drmessano | That's a bit obvious, I would think.. |
| 03:34.00 | jaytee | SIP 2.0 |
| 03:35.34 | drmessano | S/1.4/1.6/ |
| 03:35.36 | ManxPower | jaytee: I roam from verizon to alltel between Tuscaloosa AL and Picayune MS |
| 03:36.04 | jaytee | that's a definite yay! then for sure. |
| 03:36.29 | drmessano | Alltel has a very extensive network in sparsely populated areas.. and they make a killing on roaming |
| 03:36.47 | drmessano | This is one of those buys that will benefit verizon customers greatly |
| 03:36.51 | jaytee | my old company was like that. they dominated in rural areas, still do |
| 03:37.23 | asdx | jaytee: sip 2.0? |
| 03:37.58 | jaytee | SIP tcp and supposedly tls support |
| 03:38.31 | asdx | does that makes it compliant? |
| 03:39.29 | jaytee | though even if that works out I'm still going to have to come up with some ungodly bastardization to get a MWI lamp indicator on a phone when there's a voicemail waiting in Exchange even though the idiot could just look in their inbox. |
| 03:39.57 | asdx | lmao |
| 03:40.09 | nick125 | ... |
| 03:41.49 | drmessano | That's one thing I can't stand |
| 03:42.23 | drmessano | asdx: What would it matter exactly if Asterisk with 100% SIP compliant? |
| 03:42.29 | drmessano | Would it work better than it does now? |
| 03:43.07 | asdx | it matters for me, maybe not for you |
| 03:43.12 | jaytee | as a B2BUA there isn't another product as flexible enough to do what I need it to do. |
| 03:43.21 | asdx | and that's it |
| 03:43.25 | drmessano | I just asked you how? |
| 03:43.34 | jaytee | which is an ugly kludge an a whole nuther story |
| 03:43.51 | drmessano | How would it work better if it was 100% SIP compliant? |
| 03:44.26 | asdx | who knows |
| 03:44.26 | drmessano | As opposed to where it is now, or it being 95% compliant.. or 98%... or 89% |
| 03:44.47 | asdx | i only want it to work well, if it's 100% compliant i guess it will work well |
| 03:44.52 | asdx | but who knows |
| 03:45.06 | drmessano | SO in other words, you read a blog somewhere that said it wasn't 100% SIP compliant, and you seem to think it matters |
| 03:45.16 | jaytee | if you don't have "exotic" needs then I'd say * is 99% there. |
| 03:45.29 | asdx | that's great hten |
| 03:45.31 | asdx | then* |
| 03:45.43 | asdx | good enough for everyone right? |
| 03:45.55 | drmessano | You must be from the same group of idiots that complaint that Firefox isn't 100% HTML compliant.. like, it's ruining their web surfing |
| 03:46.01 | jaytee | and even if you do have more defined needs you can still make * do just about anything. |
| 03:46.01 | *** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 03:47.36 | asdx | drmessano: i didn't call you an idiot, so don't call me an idiot |
| 03:47.41 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 03:48.04 | asdx | drmessano: i don't care if it's 100% as long as it works ok |
| 03:48.24 | drmessano | asdx: Well, does it now? |
| 03:48.45 | drmessano | asdx: So why does it matter if it conforms to some standard? |
| 03:48.54 | asdx | drmessano: it doesn't matter... forget it |
| 03:49.26 | asdx | drmessano: i'm just saying that it's nice to keep improving things, even if they work fine now |
| 03:50.36 | deeperror | ~c |
| 03:50.37 | jbot | c is, like, for maniacs C code. C code run. Run, code, run. Please? |
| 03:50.44 | asdx | i guess that's what they are doing now... with tls/tcp, etc |
| 03:50.46 | TJNII | If it ain't broke, don't fix it. |
| 04:01.42 | JT | TJNII: that would be a terrible philosophy for software development |
| 04:01.57 | JT | hell, it's a terrible philosophy for development of anything |
| 04:02.46 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 04:06.05 | lanning | lets port asterisk to Java! |
| 04:06.07 | lanning | :P |
| 04:06.30 | asdx | why would you want to do that |
| 04:06.48 | drmessano | Because it would be 100% java complaint, asdx |
| 04:06.54 | drmessano | compliant |
| 04:06.59 | asdx | it doesn't matter... |
| 04:07.13 | drmessano | Being 100% SIP compliant seems to matter |
| 04:07.15 | TJNII | Java is better because it runs on everything, duh. |
| 04:07.35 | florz | erm, if anyone really wanted to know: * is really far away from being SIP RFC compliant - actually, it's majorly broken in many ways. It just happens to work acceptably in the most common cases. |
| 04:07.39 | lanning | I wonder how good timing is in a JVM... |
| 04:07.56 | florz | (1.4, that is, haven't had a look at 1.6 yet) |
| 04:07.58 | jblack | Oh, did I miss the start of The Nighly Troll/ |
| 04:09.43 | drmessano | Internet Explorer is majorly not 100% HTML compliant, Exchange 2003 has a flawed IMAP implementation, Openfire doesn't completely implement the somewhat shaky XMPP spec.. Still can't figure out how any of these are really unusable and why it's worth trolling about. |
| 04:10.32 | drmessano | jblack: It's important that Asterisk 1.6 be 100% SIP complaint, because, you know, it works so lousy right now.. has for years. |
| 04:10.41 | asdx | drmessano: IE sucks |
| 04:10.52 | florz | (and the fun part really is that pedantic mode in parts makes the parser far worse at correctly understanding SIP ...) |
| 04:11.11 | lanning | actually, I bet if we make it 100% SIP compliant, it will stop working with everything else. :) |
| 04:11.11 | drmessano | asdx: You're really just here to troll, aren't you? |
| 04:11.20 | drmessano | lanning: Indeed LOL |
| 04:11.42 | asdx | drmessano: nope, i just asked about the sip implementation... is not a good reason to call someone a troll right? |
| 04:12.01 | asdx | drmessano: and have you ever did web development, and tried to make your css work fine in ie/firefox? |
| 04:12.07 | drmessano | asdx: I'm just going to put you on ignore. It's not really worth it.. |
| 04:12.35 | asdx | drmessano: do whatever you want |
| 04:14.59 | florz | And the current situation really is that you often have to work around deficiencies in asterisk if you want to support it, much like with IE, at least until not long ago. |
| 04:31.18 | asdx | compliance is good... it saves your ass of not having to deal with stupid workarounds |
| 04:32.14 | asdx | the only problem is that no one follows them |
| 04:33.03 | florz | well, the actual problem is that everyone doesn't follow the standards in a different way ;-) |
| 04:33.34 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 04:34.21 | asdx | yeah |
| 04:34.59 | jblack | I figure the biggest misbehaver, microsoft, is finally starting to play ball. |
| 04:35.21 | asdx | everyone seems to like to do the thing their "own way" and then we developers have to pay... |
| 04:35.31 | jblack | They just don't have that feeling of invulnerability/inevitability anymore |
| 04:37.03 | asdx | yeah |
| 04:37.55 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 04:38.02 | *** join/#asterisk MoreAllLess (n=justo@cpe-76-169-252-172.socal.res.rr.com) |
| 04:38.15 | drmessano | Worrying about standards compliance is a waste of time.. This isn't a flat planet, and the moon isn't made of cheese.. Being able to code for common implementations will get you a lot farther than getting worthlessly hung up on standards.. |
| 04:39.32 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-0062d882035f7388) |
| 04:40.16 | drmessano | If every product out there became SIP compliant tomorrow, in 6 months half of them would be left behind on "Only being compliant with the standard" and not supporting the 6 months worth of extensions that 5 or 6 apps have implemented since.. |
| 04:40.22 | jblack | I disagree, but I understand where you're coming from. |
| 04:40.41 | drmessano | Being compliant is an easy way to become programmatically obsolete |
| 04:41.05 | florz | drmessano: that's a false dichotomy |
| 04:41.27 | drmessano | florz: You don't have to tell me i'm wrong, I am well aware of it |
| 04:42.29 | florz | everyone else: what drmessano is saying is a false dichotomy |
| 04:42.31 | florz | drmessano: better? |
| 04:42.59 | jblack | I think standards become hidden when followed well. |
| 04:43.27 | drmessano | In any case.. technology is moving too fast to get hung up on some RFCs that are older than most kids writing the code powering the apps making the wheels turn right now |
| 04:43.51 | jblack | When they work, nobody thinks about them. There's no big argument about boot sectors, or whether this memory fits in that machine, or whether my TV is going to work with that cablebox. |
| 04:45.18 | drmessano | jblack: that's also why some tech takes forever to come to common use.. take digital TV.. had digital TV worked alongside analog, it would have been in place 5 years ago |
| 04:45.20 | jblack | If I buy a radio at walmart, I expect it to be able to tune into NPR just as I would expect some high end system. |
| 04:45.50 | drmessano | jblack: that actually furthers my point I just made (and you didnt even know it) |
| 04:46.15 | alrs | drmessano: digital TV has been around for 10 years |
| 04:46.18 | jblack | I agree standards slow things. I don't agree that standards complaince is a waste of time. |
| 04:47.00 | drmessano | alrs: Oh really, because I was under the mistaken impression that 80% of the digital TV installs took place in the last 3 years |
| 04:47.04 | florz | drmessano: well, "getting hung up" probably isn't quite the sensible thing to do - but what asterisk does with sip probably isn't, either - exactly for the reason that it hinders progress when you limit yourself to a few, very specializes use cases, when your code is a mess because it was written without much thought being put into it, when you get security problems for mostly the same reasons, ... |
| 04:47.12 | drmessano | But I guess I am wrong, even though I have worked in broadcast for years |
| 04:47.19 | alrs | drmessano: that's two different things |
| 04:47.28 | drmessano | alrs: No, it's not |
| 04:47.35 | drmessano | "in place" |
| 04:47.38 | jblack | I don't see how those two statements are mutually incompatible. |
| 04:47.58 | jblack | He says 10 years, you say 20% three years ago. |
| 04:48.14 | alrs | "had digital TV worked |
| 04:48.14 | alrs | <PROTECTED> |
| 04:48.15 | alrs | <PROTECTED> |
| 04:48.23 | drmessano | Yes, in place |
| 04:48.31 | alrs | it was in place 10 years ago |
| 04:48.32 | drmessano | As in |
| 04:48.32 | drmessano | DONE |
| 04:48.34 | drmessano | FINISHED |
| 04:48.34 | jblack | Doesn't digitial TV work alongside analog TV today? |
| 04:48.37 | alrs | http://www.ultimateavmag.com/news/10264/ |
| 04:48.37 | drmessano | EVERYONE GOT IT |
| 04:48.41 | drmessano | ALL OVER DA PLACE |
| 04:48.45 | drmessano | IMPLEMENTED FULLY |
| 04:48.49 | drmessano | 2009, OVER AND DONE |
| 04:49.38 | jblack | And now that I think about it, hasn't directv been a digital signal for at least a decade? |
| 04:50.05 | MoreAllLess | So is that with 802.11n is in "Draft" but already being implemented? |
| 04:50.24 | drmessano | alrs: I am well aware of how long HDTV has been getting rolled out.. I was there for the install of the third HDTV transmitter in the country back in early 1999 |
| 04:50.39 | jblack | Yeah. I think 802.11n is a good example of something that has slowed _because_ it hasn't been standardized yet. |
| 04:50.53 | *** join/#asterisk danchris (n=danichri@adsl-69-109-158-119.dsl.sndg02.pacbell.net) |
| 04:50.55 | drmessano | But HDTV is not IN PLACE yet |
| 04:51.07 | jblack | What do you define as in place? |
| 04:51.10 | drmessano | It wont be fully implemented until the analog in switched off in 2009 |
| 04:51.25 | drmessano | Right now BOTH are on the air |
| 04:51.25 | jblack | hurh? |
| 04:51.29 | drmessano | You still have a choice |
| 04:51.36 | drmessano | In 2009, you will not |
| 04:51.39 | MoreAllLess | The way I see it, HDTV and Digital TV are different things. |
| 04:51.53 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 04:52.01 | MoreAllLess | Just because the signal is digital, doesn't mean it has 1000+ lines of resolution. |
| 04:52.14 | jblack | Wait, so, HDTV isn't "in place" (I don't know what that means), because there's choice. |
| 04:52.35 | drmessano | Oh forget it |
| 04:52.41 | jblack | So, using your logic, shouldn't TV be going great guns because there's more than one way to watch it? |
| 04:53.53 | MoreAllLess | Getting back to the point: standards will slow development. Everyone agrees? |
| 04:54.21 | jblack | I agree it slows development, and speeds adoption |
| 04:55.21 | MoreAllLess | What if you have no standards? Does that mean there is no adoption? |
| 04:55.22 | florz | MoreAllLess: nope, for adoption can speed development |
| 04:55.53 | jblack | nah. I wouldn't say say. |
| 04:56.04 | jblack | That no standards means no adoption. |
| 04:56.16 | MoreAllLess | Now I see why the discussion. |
| 04:57.41 | jblack | I also think no standards, and I mean no in an extreme fashion, means a fully vertical stack. |
| 04:58.15 | jblack | From keyboard down to waht voltage and freq is supplied from the power line. |
| 04:58.18 | MoreAllLess | To this I say: standards or no standards, implementation and development will occur at it's pace. |
| 04:58.33 | jblack | Today, anyways. |
| 04:58.55 | jblack | I also think that 80 years ago, kids were digging in coal mines, and eighty years from now, kids will again be digging in coal mines. |
| 04:58.56 | MoreAllLess | It will depend on the technology being development, who decides to work on it, and the pace they decide to work on it. |
| 04:59.04 | jblack | I'd say that would be less development, and more regression. =D |
| 04:59.55 | MoreAllLess | Standards may help adoption will in turn will help development. |
| 04:59.56 | *** join/#asterisk UngaMan (n=jvannini@dynamic55-88.MASAYA.cablenet.com.ni) |
| 05:00.18 | MoreAllLess | But, standards may also hinder development and hinder adoption. |
| 05:00.20 | jblack | I'd say development without adoption is limited in use. ;) |
| 05:00.22 | UngaMan | hello |
| 05:00.36 | jblack | I ague that standards help adoption, not hinder it. |
| 05:00.39 | drmessano | I would rather have documented implementations than standards |
| 05:00.46 | drmessano | Standards are too binding |
| 05:00.50 | jblack | Standards provide commodity benefits. |
| 05:01.13 | UngaMan | commands |
| 05:01.22 | drmessano | Let implmentations work themselves into far more usable products that binding to standards created by old men who will never use them |
| 05:01.26 | MoreAllLess | I argue that it depends on the technology and the willness of development and implementation by the parties involved. |
| 05:01.30 | hardwire | Standards by standard are standardized by standardizers. |
| 05:01.41 | jblack | MoreAllLess: what depends? |
| 05:01.56 | MoreAllLess | The parties involved. |
| 05:01.57 | *** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled) |
| 05:02.07 | jblack | You completely lost me |
| 05:02.17 | MoreAllLess | Look at who developed asterisk. |
| 05:03.14 | drmessano | HA |
| 05:03.24 | drmessano | Firefox 3 RC2.. Gone again |
| 05:03.48 | jblack | Ok. If they didn't take advantage of TCP, Sip, UDP, the various PRI standards, Posix, etc etc... What would you have |
| 05:04.38 | jblack | You'd have some black box that only worked with other black boxes that you could only buy from one source. Standards is what makes asterisk such a powerful argument. |
| 05:05.12 | drmessano | I disagree.. being compliant with other implmentations is more important than being tied to a standard |
| 05:05.18 | MoreAllLess | jblack: I understand standards are needed. I agree with you. |
| 05:05.19 | jblack | I don't need a special nortel box with special nortel phones and special nortel headsets, that only work with cable bought just from nortel. |
| 05:05.48 | jblack | drmessano: What do you think a stardard is? |
| 05:05.59 | lanning | not a rule, more of a guideline |
| 05:06.02 | lanning | :) |
| 05:06.27 | MoreAllLess | But, there may be some circumstances, such as 802.11n, where those developing it and implemented do not agree and therefore hinder the development of such technology. |
| 05:06.31 | florz | drmessano: in a way, yes - and documenting how other implementations behave is exactly the purpose of a standard |
| 05:06.38 | jblack | I think of a standard as the definition of an implementation that, when you comply with it, you will interoperate with other things that comply with it too. |
| 05:07.06 | drmessano | Well, if my product was SIP compliant with the RFC, and FreeSWITCH, callweaver, SipX and whoever had implemented 10 or 15 additions to the RFC that added some neat shit, being complaint with the RFC wouldnt amount to a hill of damn beans |
| 05:07.13 | drmessano | I would rather be compliant with everyone else |
| 05:07.14 | MoreAllLess | Yes, jblack, I agree on your definition of a standard. |
| 05:08.15 | jblack | Ok, so, complying with an implementation doesn't guarantee that you won't work with things that don't try and follow the same implementation. |
| 05:08.17 | florz | drmessano: Well, obviuosly being standards compliant alone doesnt make you a good product for every purpose ... if that's what you want to say |
| 05:08.41 | jblack | doesn't guarantee you will or won't. I'd say it would be undefined |
| 05:08.45 | drmessano | Generally, the second that some body creates some hard standard on something it immediately becomes obsolote.. so whats the use in being bound to it? |
| 05:08.50 | florz | drmessano: so, a standard-compliant screw doesn't make for a good PBX - I think |
| 05:09.00 | jblack | drmessano: Interoperability. |
| 05:09.20 | drmessano | Dont make me paste |
| 05:09.27 | drmessano | <drmessano> Well, if my product was SIP compliant with the RFC, and FreeSWITCH, callweaver, SipX and whoever had implemented 10 or 15 additions to the RFC that added some neat shit, being complaint with the RFC wouldnt amount to a hill of damn beans |
| 05:09.45 | drmessano | I guess I wouldnt be INTEROPERABLE with them, would I? |
| 05:09.46 | jblack | Ok. if they add things, they're not longer following the standard, now are they? |
| 05:10.04 | jblack | So you fall into undefined land. THey may interoperate, they may not. |
| 05:10.17 | jblack | If everyone follows the standard, everyone interoprates. |
| 05:10.23 | drmessano | If they all implement the same "good idea", then the RFC becomes useless |
| 05:10.32 | florz | drmessano: yeah, exactly "a product supporting standard x isn't necessarily a good stand-in for a product supporting standard y" - now, how is that an argument? |
| 05:10.35 | jblack | If everyone just goes off in different competing directions, chances are good that no one interoperates at all. |
| 05:10.36 | drmessano | and increasingly useless as times go on |
| 05:10.52 | MoreAllLess | drmessano, but who is to say that two standards have to interoperate? |
| 05:11.07 | jblack | Wait. I think I misundersatnd you. You think that any product that interoperates with others is useless? |
| 05:11.07 | *** join/#asterisk rvhi (n=chatzill@udp255518uds.hawaiiantel.net) |
| 05:11.12 | alrs | drmessano: what is a good example of a debased standard that has left the rfc pathetic and irrelevant? |
| 05:11.33 | jblack | or do you think that any two indistinguishable products are useless? |
| 05:11.44 | drmessano | MoreAllLess: No one is.. the point was that "You must be compliant with some crap document someone calls a standard" doesn't make you interoperable |
| 05:11.48 | jblack | In order to be useful, one has to differentiate? |
| 05:11.59 | asdx | standards usually advance as technology goes |
| 05:11.59 | rvhi | hi, is there any program integrate asterisk with crm so that when a call comes in, CRM brings up that account based on caller id? |
| 05:12.20 | drmessano | If I don't support the 15 additions that 8 other products are using, I am behind the curve, am I not? |
| 05:12.21 | alrs | drmessano: an example of a "crap" document? |
| 05:12.23 | florz | drmessano: no, what makes you interoperable is that _everyone_ is compliant with some crap document all these everyones call a standard |
| 05:12.23 | asdx | usually, not always |
| 05:12.26 | drmessano | I am lacking feature |
| 05:12.29 | FreedomBI | alrs: I seem to recall that the tcp standard doesn't have all that much correlation with the rfc. |
| 05:12.40 | drmessano | alrs: Any RFC is already dated.. |
| 05:12.48 | jblack | drmessano: Maybe, maybe not. Certainly, you're not ahead of the curve. |
| 05:13.08 | jblack | Today, I bought CDRs. |
| 05:13.11 | florz | drmessano: plus, a "standard" doesn |
| 05:13.26 | drmessano | jblack: I am 15 features behind the other guys, right out of the box |
| 05:13.28 | jblack | They are not useless to me, even though they follow a standard. |
| 05:13.42 | florz | drmessano: plus, a "standard" doesn't necessarily have to be a document produced by a "standards body" - it's just a publicly/well known way of how to do things |
| 05:14.02 | jblack | I don't want Neat New-Feature Cdroms that burn 742 megabytes and don't work in my cd player. I want new stuff, that works with the stuff I have! |
| 05:14.16 | drmessano | florz: Well, when someone complains about Asterisk not being SIP compliant, what the hell do you think they're referring to? The almighty RFC.. |
| 05:14.59 | florz | drmessano: yes - I didn't say that a "standard" can not be produced by a "standards body", did I? |
| 05:14.59 | jblack | I guess "being compliant" is technically a boolean op, but I see it more as a scale, myself |
| 05:15.13 | *** join/#asterisk Swabby (n=dp@74-137-61-44.dhcp.insightbb.com) |
| 05:15.25 | MoreAllLess | You can't generalize standards and implementation and whether or not they are dated. |
| 05:15.27 | florz | drmessano: the SIP RFC simply happens to be a document that explains relatively well how everyone's implementations work |
| 05:15.29 | alrs | What part of the Asterisk SIP implementation is cooler than the RFC and should be forgiven? |
| 05:15.48 | MoreAllLess | Look at SMTP, how many implementation are there? Has there been a new standard for SMTP? |
| 05:16.05 | Swabby | I have a routing question...i'm using trixbox and i have my trunk setup (pots) basically i want every call just processed as stated. nothing added nothing taken off |
| 05:16.06 | alrs | My impression was that Asterisk SIP is incomplete and something of a moving target, having been rewritten for both 1.2 and 1.4 |
| 05:16.12 | MoreAllLess | Perhaps there has, but realistically is it implemented? |
| 05:16.15 | drmessano | MoreAllLess: No, exactly.. People just say "hey, that looks cool.. we need to add that too" and it's DONE.. |
| 05:16.19 | Swabby | will NXXXXXX do this? |
| 05:16.28 | jblack | SMTP isn't very well followed, and it's a pretty big mess. |
| 05:17.04 | [TK]D-Fender | Swabby, that is a pattern. That does not imply anything about what you are actually DOING once you match it. |
| 05:17.27 | florz | drmessano: and as far as I am concerned, the problem with asterisk's SIP implemenation really goes far beyond "not literally implementing the RFC (even where it describes things differently than done by everyone else)" |
| 05:17.32 | jblack | and yes, there's multiple close, competing standards. Some are 822, some are 2822, most are a mismash between the two, with stuff that's not part of eitehr standard thrown in. |
| 05:18.11 | jblack | And standards themselves are not always SHALL and MUST. There's a good helping of MAY and CAN thrown in too |
| 05:18.24 | Swabby | tkdfender: when i do my 9+# i can't get any calls through..is there a way to trace to see what asterisk is sending out? |
| 05:18.31 | *** part/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 05:18.38 | Swabby | i get stuff like..can't complete call as dialed, etc |
| 05:19.24 | MoreAllLess | Yes, jblack. That is the part of the point I am trying to make. |
| 05:19.26 | [TK]D-Fender | Swabby, What do you see in CLI? |
| 05:19.43 | [TK]D-Fender | Swabby, And you aren't showing us what you actually are having it DO. |
| 05:21.10 | Swabby | it looks like it's dialing 9+the number |
| 05:21.11 | Swabby | according to cli |
| 05:21.12 | [TK]D-Fender | Swabby, PASTEBIN is your friend. |
| 05:21.17 | [TK]D-Fender | ~pb |
| 05:21.18 | 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 |
| 05:21.19 | [TK]D-Fender | ^^^^^^^^^ |
| 05:21.23 | [TK]D-Fender | Swabby, use it. |
| 05:21.39 | Swabby | let me see if i can ssh in |
| 05:21.41 | drmessano | jblack: Problem is, a good majority do not speak on those terms.. "Asterisk isn't RFC compliant, so it sucks".. is a weak argument.. not supporting some features implemented in other products is a good argument.. whether or not those are even in the RFC. |
| 05:22.05 | florz | drmessano: now, if you are saying that it doesn't make sense to implement stuff in an obviously brain-damaged way when nobody else does, even if some "standards document" says you should - sure. But that, after all, is pretty much independent from the standards document, plus you in a way would still be implementing the standard: if you know everyone else does it the way you do, that obviously is the defacto standard. |
| 05:22.45 | [TK]D-Fender | Swabby, what have you been looking at all this time? |
| 05:23.28 | [TK]D-Fender | drmessano, RFC... Request For Comment. Isn't non-compliance a pretty obvious comment about how we feel about their ideas? ;) |
| 05:23.32 | Swabby | http://pastebin.com/d21133edc |
| 05:23.42 | drmessano | florz: As I just stated to jblack, the problem is that when the trolls come out of the woodwork, it's about RFCs and not about interoperability.. those two things aren't necessarily the same thing, although I do agree that having something documented is better than "Oh, what the hell" |
| 05:23.44 | Swabby | tk: i'm at the console |
| 05:23.50 | [TK]D-Fender | Swabby, Going to make this nice an clear : |
| 05:23.53 | [TK]D-Fender | ~freepbx |
| 05:23.54 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 05:23.57 | [TK]D-Fender | ^^^^^^^^^ |
| 05:24.03 | drmessano | [TK]D-Fender: HA! |
| 05:24.20 | Swabby | Isn't the architecture very similar though? |
| 05:24.36 | jblack | I never said I was with the majority. |
| 05:24.49 | drmessano | I never said you were either |
| 05:24.58 | [TK]D-Fender | Swabby, No. That GUI owns your ass. Through and through |
| 05:25.26 | [TK]D-Fender | Swabby, Your problem is you don't know how to configure it and you have to go through their craptastic web interface to do it. |
| 05:25.28 | florz | drmessano: Well, actually, I think that "doesn't comply with the RFC" often really is meant to say "deviates in problematic ways" - even though I'd certainly agree that, literally, it's not a strong argument against anything ... |
| 05:25.38 | [TK]D-Fender | Swabby, this is not the right channel for you to be in. |
| 05:26.21 | jblack | Poorly worded statements is a common problem. |
| 05:26.39 | [TK]D-Fender | florz, more like.... Yeah sure, we support SIP.... err.. KINDA.... |
| 05:26.49 | [TK]D-Fender | ..ISH? |
| 05:27.34 | drmessano | I would have some sympathy for those that complain about things they would like to see in Asterisk if they didn't start off with the bullshit about the RFC.. that's almost as bad as a girl I once dated who told me she was in anger management. |
| 05:28.19 | drmessano | "Hang on, I think.. I left my windows open.... at my beach house in .... pakistan.. be back in a few years..." |
| 05:29.20 | jblack | Huh. |
| 05:29.27 | drmessano | That's as weak as telling me Firefox doesn't properly implement some overly obscure CSS property.. "Uh, were you just born and this is your first week using a browser? |
| 05:30.05 | nick125 | drmessano: But it makes your computer output root beer! |
| 05:30.24 | jblack | You'd have more sympathy for people that say "it doesn't work" than you would for people that say "Asterisk is not working with device X, which follows the SIP rfc" |
| 05:30.25 | drmessano | Newsflash.... The internet browser _____ doesn't fully support ________. |
| 05:30.59 | [TK]D-Fender | Tomorrow its time for my PC upgrade. From AMD XP 3800+ (single core) to Celeraon E1200 (dual core) 16.ghz at 2.66ghz min, 3.2ghz max OC. |
| 05:31.03 | jblack | I guess I can see someone having sympathy for what I see as ignorance. |
| 05:31.09 | [TK]D-Fender | 1.6* |
| 05:32.32 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 05:32.38 | Swabby | It's Celeron |
| 05:33.26 | jblack | all the same, I'd respect the latter more than I would the former. The latter is a stronger argument with me than the former. |
| 05:33.50 | nick125 | jblack: I guess the only time I could see bringing up the RFC would be if the lack of compliance was causing the issue. |
| 05:34.30 | *** join/#asterisk techie (n=techie@adsl-76-214-30-11.dsl.lsan03.sbcglobal.net) |
| 05:34.31 | jblack | I agree. Pointing out hypothetical problems is a much lower priority in my book too. |
| 05:35.01 | drmessano | Pointing out specific implementation problems is more important that throwing the RFC out there, which everyone may or may not adhere too |
| 05:35.05 | drmessano | It's pretty simple |
| 05:35.21 | drmessano | I guess if you disagree, then that's fine |
| 05:35.41 | jblack | I can see where you're coming from, but man, I just can't get on board with it. |
| 05:36.23 | jblack | The alternative to having everyone chase one standard is having everyone try and make X seperate implementations for X different products. |
| 05:36.51 | jblack | I'd rather code to one documented standard, then try and code to 6 undocumented ones. |
| 05:37.47 | jblack | but that's my philosophy. Could I sum yours up as 'survival of the fittest' ? |
| 05:37.50 | [TK]D-Fender | Swabby, Its a typo... and its late. Anyways, go visit their channel and they can walk you through fixing your dial settings to it doesn't add a 9 or whatever. This isn't an * problem and FreePBX is not supported in here. It is responsible for you creating your configs and it is what you need to learn as long as you continue using it. |
| 05:39.21 | [TK]D-Fender | Checkout time. Later all. |
| 05:39.50 | drmessano | That's probably pretty accurate.. It's really about genetic mutations as part of evolution to remain a thriving product |
| 05:40.02 | drmessano | Darwinism at it's finest |
| 05:40.26 | drmessano | "Nice fins, everyone else has legs.. the food is over there on land.. Sorry charlie" |
| 05:40.34 | jblack | darwinism isn't efficient, but it does scale well. |
| 05:42.33 | *** part/#asterisk Swabby (n=dp@74-137-61-44.dhcp.insightbb.com) |
| 05:43.48 | drmessano | Probably the difference in reasoning has to do with my general outlook on things.. I try to be 110% flexible.. Application notes are great, but if every guy you talk to says "Cut the end off that panduit.. it makes it too hard to install, everyone does it".. then damnit, I am cutting the end off and making it work.. I never take instructions or hardset rules as being penultimate |
| 05:44.34 | drmessano | Rules tend to get in the way when proper application for the benefit of progress is far more imporatant |
| 05:44.50 | jblack | Yeah. I'd rather fix the problem then chop the end off, so that the next guy understands what he's looking at. |
| 05:45.43 | drmessano | It's like mounting an HP LH3000 server in a rack with a back door |
| 05:45.48 | jblack | I have pity for the SOB that'll take my place, so I like patterns, documentation, predictability, etc |
| 05:46.03 | drmessano | Sure, the door is supposed to be on, theres fans to suck the air out.. |
| 05:46.06 | florz | drmessano: well, there is a major difference between computers and humans, though - computers won't usually understand anything "similar" to the standard just because of the similarity |
| 05:46.21 | drmessano | But its hard to suck air out when the wire managment doesnt allow the door to close to begin with |
| 05:46.24 | jblack | florz: Heh. we moved past that. |
| 05:46.56 | florz | jblack: we did? =:-) |
| 05:46.57 | jblack | I bet you're big on TDD. |
| 05:47.45 | drmessano | TDD? |
| 05:47.46 | jblack | If not, you should look at it, as you'd love it. |
| 05:47.50 | jblack | Test driven development. |
| 05:48.19 | jblack | It's a series of successive 'just make it fucking work' hacks that are driven by adding test cases. |
| 05:48.56 | jblack | I can't stand it. Though it gets one to the immediate goal quickly, I think it ends up being really messy and not thought out |
| 05:49.02 | drmessano | You mean all software isn't just a bunch of miraculously working hacks? |
| 05:49.17 | asdx | i can't stand TDD/BDD that much too |
| 05:49.47 | drmessano | I probably wouldn't care much for that |
| 05:49.52 | jblack | I mean exactly that. Some software is miraculous "just patch it and go", and some is "think it through". |
| 05:50.18 | jblack | I think you've love it. HOnestly. It's very... evolutionary, which is right up the alley of a darwinist. |
| 05:50.38 | jblack | asdx: I have decided to invent the successor to TDD... TPD. |
| 05:50.59 | asdx | heh |
| 05:51.00 | drmessano | I prefer planning things out and attempting to follow my plans.. not to just gluing shit together as I go.. I have pride in my work |
| 05:51.31 | jblack | Test Proven Development. Write things in small, well thought out steps, then make assertions via unit tests that Prove your code is correct. |
| 05:51.45 | asdx | "if it compiles, it is good, if it boots up it is perfect" |
| 05:52.01 | drmessano | Sounds pretty lousy |
| 05:52.24 | jblack | chapter 1 of various books start off with a simple math parser. |
| 05:52.40 | asdx | drmessano: one of the best programmers said it |
| 05:53.06 | jblack | The first step is to make a unit test for an add function that adds numbers. the first test pases in "7 + 3" and does a return(10);. |
| 05:53.19 | asdx | "Regression testing"? What's that? If it compiles, it is good, if it boots up it is perfect. -- Linus Torvalds |
| 05:53.22 | asdx | :p |
| 05:53.40 | jblack | So then you make a new test... say 4+4, and your test fails. So you fix the function so that it can actually add. |
| 05:53.53 | jblack | The idea is that each time you add a new test, you do the absolute minimum to make the test work. |
| 05:54.02 | asdx | yeah |
| 05:54.11 | asdx | then if you want to add something, you modify the test |
| 05:54.20 | asdx | and make it pass with code again, etc |
| 05:54.38 | jblack | You don't modify the tests. You add new tests (new assertions) and modify the code to make the new assertions work. |
| 05:55.00 | asdx | yeah, that |
| 05:55.34 | jblack | And Linus is a frigging joke. |
| 05:55.38 | *** join/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 05:55.44 | asdx | Linus is cool |
| 05:56.25 | drmessano | My biggest pet peeve is people being unprepared.. which is where the ability to be flexible comes in.. If you know you're gonna need to implement something in a very non-standard way to make something work, and it's not just a half ass solution, you should know so beforehand.. Knowing beforehard also allows you to create a proper Plan B without having to make an on the fly hack decision in process |
| 05:56.30 | kraptv | why would I still get an american ring indication when calling from one extension to another when I get a UK indication when calling in from my sip-registered DID provider? |
| 05:56.42 | kraptv | I have country=uk listed in my indications.conf. |
| 05:56.58 | jblack | Then I think your peeve is with inexperience... |
| 05:57.08 | jblack | as it's with experience one can make predictions of what to expect. |
| 05:58.08 | drmessano | That's not it entirely.. it also has to do with how the problem is handled.. even with experience you can do a much better job of handling an "oh shit" that you may run into.. |
| 05:58.13 | jblack | bleh. I sound like a bleeping know-it-all tonight. I'm sorry. |
| 05:58.42 | drmessano | nah, that's every night.. but I enjoy arguing with you anyway.. ;) |
| 05:59.07 | jblack | Ok. That's true. I know everything every night. But it's still rude to show it off. =D |
| 05:59.26 | drmessano | Umm |
| 05:59.29 | jblack | anyways... I don't quite get ya |
| 05:59.33 | drmessano | I didn't say you had proven it |
| 05:59.42 | drmessano | I just said you ACT it :P |
| 05:59.46 | drmessano | anyways... |
| 05:59.52 | jblack | awww. I thought I'd get away with that |
| 05:59.55 | drmessano | lol |
| 06:00.15 | drmessano | I guess what I am saying is.. |
| 06:01.14 | *** join/#asterisk mosty (n=mosty@203.171.192.245) |
| 06:01.40 | jblack | hmm mostly.. moisty... musty... |
| 06:03.36 | drmessano | If you lack flexibility, you will be blind to those bumps that may come up.. and experience doesn't help that person any.. they're ready to apply everything 100% compliant with the application notes, and that's all there is to it.. until they hit a snag.. |
| 06:04.28 | jblack | Oh, ok. So, a combination of experience with to adapt to new facts? |
| 06:04.42 | jblack | with the ability to, that is |
| 06:04.56 | drmessano | Yes |
| 06:05.10 | jblack | Yeah. I agree. |
| 06:05.24 | mosty | is it possible to add additional columns to cdr's? |
| 06:05.29 | jblack | mosty: yes. |
| 06:05.34 | drmessano | There is nothing worse than someone who says "I've installed 100 of these.. they all go in the same.. " |
| 06:05.43 | jblack | mosty: You mean in an sql table? |
| 06:05.49 | drmessano | Until you tell them "Well, it needs to be done this way" |
| 06:05.52 | drmessano | "Oh, shit" |
| 06:05.53 | kraptv | never mind! I realize it was the application that was registering the american ring sound. |
| 06:05.54 | jblack | They do if they follow the standard! |
| 06:05.57 | mosty | jblack, yeah i want to record the ip address of sip clients for each call |
| 06:06.24 | jblack | mosty: Then yes, * will happily ignore the extra fields. |
| 06:06.33 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 06:06.55 | mosty | jblack, i need to set the extra fields from asterisk though |
| 06:07.21 | jblack | That may be tough, as I think * may insert the new row as the last thing it does. |
| 06:07.58 | jblack | But... you could make a separate table, and into it, insert things useful to you, like the call, the dst, the ip, etc etc. |
| 06:08.12 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:08.21 | mosty | jblack, oh i see the CDR function, it looks like you can use a custom name option |
| 06:08.32 | jblack | Oh, great! |
| 06:08.46 | jblack | tbh, I didn't check. My daughter moved my * book across the room |
| 06:09.04 | mosty | CDR(ip)=blah |
| 06:09.22 | jblack | Is it just me that finds it ironic that the west-call.com site is in russian? |
| 06:09.34 | jblack | mosty: That sounds perect for you |
| 06:10.02 | mosty | now i just need to find out how to extract the ip address of a sip channel |
| 06:10.21 | jblack | heh |
| 06:10.30 | asdx | why asterisk doesn't just use the sofia library? |
| 06:11.22 | jblack | asdx: Why don't you speak portuguese? |
| 06:11.28 | asdx | jblack: i do |
| 06:11.38 | asdx | jblack: eu falo |
| 06:11.40 | asdx | jblack: e voce? |
| 06:11.41 | jblack | Fine. why don't you speak french? |
| 06:11.59 | asdx | jblack: because i don't |
| 06:12.02 | jblack | mosty: http://www.voip-info.org/wiki-Asterisk+variables |
| 06:12.10 | jblack | That's why asterisk doesn't use the sofia library. ;) |
| 06:12.29 | asdx | ok |
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| 06:51.48 | mosty | i'm looking at the SIPCHANINFO function, what is the difference between peerip and recvip? |
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| 07:07.18 | HaMYaI | Hi, can we have multiple externip tags in sip.conf? |
| 07:08.19 | HaMYaI | I have incoming connections in both eth0 and eth1, and probably need to specify more than 1 external ips |
| 07:09.12 | HaMYaI | or just put none of them |
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| 07:41.08 | yang | Where can I set CDR log output to the local time, currently it logs in UTC |
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| 08:18.47 | stunsch | I'm having trouble with PICKUPMARK. When 2 persons are picking up a call on different extensions, the calls get mixed sometimes. How can this be solved? |
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| 09:20.57 | stunsch | I'm trying to check the crc setting of a primary. Where can I find this and how can I set it to ON? |
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| 09:33.16 | synthetiq | is it possible to keep a user in a state of ringing, if they are in a meetme room alone? |
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| 09:52.26 | Strom_L | synthetiq: ringing is going to annoy the user -- better to play music |
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| 09:59.40 | synthetiq | strom yea thats cute but ringing the suer isnt charged |
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| 10:20.09 | tinne | Hi @all! We have the problem that our asterisk pri gateway (2x pri-channels from/to our telco, 2x pri internally to an old pbx) doesn't seem to send any busy-signal to an outside caller if the number on the old pbx is busy. i can see that asterisk recognizes the zap-channel as busy/congested but the caller doesn't hear the busy tone... |
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| 11:51.06 | deeperror | http://sourceforge.net/awards/cca/?project_name=Asterisk&project_url=www.asterisk.org |
| 11:52.55 | *** join/#asterisk EE_BB (n=chatzill@212.58.24.138) |
| 11:55.44 | EE_BB | hii all I have configured my asterisk that connected with meridian 1C via E1 card all sip clients and meridian pstn client calls each other but I need some sip client call 0 and take signal an call another number that belog city telecom company |
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| 12:03.21 | BrokenNoze | Hi All, I have an issue with synced recordings. is there a patch for 1.4.18 to fix the mix monitor problem, and can i patch it without stopping asterisk at all? |
| 12:04.58 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 12:05.25 | dwclarkNU | i had a digium tdm410p and want to use them as fxs ports, they are in a dell 2650...where do I plug the 12V power into inside my server so I can use the card w/ asterisk |
| 12:06.18 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:06.18 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 12:08.39 | Whoopie | Hi, I compiled asterisk 1.4.20.1 for mipsel, and now I get "/usr/sbin/asterisk: can't resolve symbol 'ast_register_file_version'". any ideas? |
| 12:10.05 | BrokenNoze | can i recompile one single app and then reload it into asterisk? need to introduce a fix. any resources as to how i might do this anywhere? |
| 12:10.11 | *** join/#asterisk ManxPower (n=manxpowe@93.sub-75-200-144.myvzw.com) |
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| 12:14.08 | russellb | BrokenNoze: there isn't a really easy way to do that. You can just recompile as usual, and then move the one module into /usr/lib/asterisk/modules |
| 12:14.21 | russellb | then "module unload my_mod.so" ... "module load my_mod.so" |
| 12:14.22 | *** join/#asterisk blinky42 (n=steveb@69.7.229.5) |
| 12:15.28 | BrokenNoze | russellb: OK sounds OK, so I'd just do a standard "make" without a "make install" and copy to so file over? |
| 12:16.00 | russellb | yes |
| 12:16.05 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 12:16.05 | BrokenNoze | *the so file over once i've unloaded the module in the CLI, then reload it in the CLI |
| 12:16.17 | russellb | correct |
| 12:16.24 | BrokenNoze | russellb : awesome. that sounds fine. thanks mate |
| 12:16.29 | russellb | but by reload, we're talking about unload/load |
| 12:16.32 | russellb | not "module reload" |
| 12:16.35 | russellb | that's something different :) |
| 12:16.38 | russellb | reloads configuration ... |
| 12:16.45 | *** join/#asterisk artz22 (n=artz@190.189.0.33) |
| 12:16.47 | artz22 | hi everybody |
| 12:16.59 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 12:18.11 | EE_BB | anybody help me ? |
| 12:18.13 | the_5th_wheel | hello. Im in dire need of assistance |
| 12:18.25 | ManxPower | ~ask |
| 12:18.26 | jbot | from memory, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:18.26 | the_5th_wheel | Im getting horrible call qualtity over SIP |
| 12:18.34 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 12:18.48 | the_5th_wheel | (i accidentally pressed enter without thinking) |
| 12:18.55 | ManxPower | the_5th_wheel: using SIPura ATAs? |
| 12:19.06 | the_5th_wheel | I have debugged my internet connection and it seems fine. |
| 12:19.10 | the_5th_wheel | No, two asterisk |
| 12:19.24 | ManxPower | the_5th_wheel: define "horrible call quality" |
| 12:19.35 | the_5th_wheel | if i do ping floods, it gets thru at decent rates |
| 12:19.39 | the_5th_wheel | Very choppy |
| 12:19.49 | EE_BB | l I have configured my asterisk that connected with meridian 1C via E1 card all sip clients and meridian pstn client calls each other but I need some sip client call 0 and take signal an call another number that belog city telecom company |
| 12:19.52 | russellb | that's what jitterbuffers are fo |
| 12:19.53 | russellb | for* |
| 12:19.56 | EE_BB | ?? |
| 12:20.17 | the_5th_wheel | russellb: But it seemed to work well, nothing changed. |
| 12:20.17 | ManxPower | the_5th_wheel: do you have any telecom cards in the system? |
| 12:20.24 | artz22 | i'm trying sip phones with asterisk. I have two 729/ulaw phones so I added this to sip.conf in [general]: disallow=all; allow=g729; allow=g723; allow=gsm; allow=ulaw; I didn't add any disallow or allow in each user configuration in sip.conf. Apart from the 729/ulaw phones I have a gsm/ulaw softphone, but now if I try to call the softphone to the 729/ulaw one asterisk won't get down to ulaw to connect both phones, but will tell me that it can't |
| 12:20.24 | artz22 | convert from, for instance, ulaw to 729. What else should I try? |
| 12:20.28 | the_5th_wheel | Yes, ISDN cards |
| 12:20.43 | the_5th_wheel | But the voice promts that get generated by asterisk is cool. |
| 12:20.55 | ManxPower | artz22: Gow many G729 licences do you have? |
| 12:21.05 | ManxPower | the_5th_wheel: BRI ISDN or PRI ISDN? |
| 12:21.15 | EE_BB | I need a mode ? |
| 12:21.17 | artz22 | ManxPower: I have no 729 licenses, I want asterisk to get down to ulaw when there is no other choice |
| 12:21.21 | EE_BB | anybody hear me ? |
| 12:21.29 | ManxPower | artz22: you cannot do that. |
| 12:21.46 | artz22 | ManxPower: but wouldn't it be logically possible? |
| 12:21.48 | the_5th_wheel | and if i forward the audio to another sip server wich i own, 45ms away, i get decent quality |
| 12:21.48 | ManxPower | EE_BB: your question does not make sense |
| 12:22.09 | ManxPower | artz22: no. In my experience you only want to allow one codec per sip peer/friend/user. |
| 12:22.11 | EE_BB | then I can explain |
| 12:22.18 | ManxPower | Asterisk will always try G729 last. |
| 12:22.27 | synthetiq | using asterisk manager API, is it possible to set someones caller id? |
| 12:22.36 | artz22 | ManxPower: since both phones support ulaw, they are able to talk to each other. In fact, they are if I remove the allow=g729 and g723 |
| 12:23.00 | ManxPower | artz22: Correct. Don't allow g729 or g723 if you don't have the licenses |
| 12:23.42 | artz22 | ManxPower: :( well that's bad for me.. thanks |
| 12:24.06 | EE_BB | I need that Sip client call local telco number via using my meridian telco lines sip client call a number and take voice signal and dial local telco numbers |
| 12:24.20 | the_5th_wheel | ManxPower: ISDN BRI |
| 12:24.33 | EE_BB | ManxPower: clear ? |
| 12:24.48 | ManxPower | EE_BB: Not even close. What specific thing is not working? |
| 12:25.07 | ManxPower | the_5th_wheel: if it works fine with a different server then I would point to your ITSP as the problem |
| 12:25.37 | the_5th_wheel | ManxPower: Yes, i would think so aswell, but im trying to find a way to proove them that its their fault |
| 12:25.49 | ManxPower | the_5th_wheel: I can't think of a away |
| 12:26.19 | ManxPower | I can't think of a way, that is. |
| 12:26.46 | ManxPower | EE_BB: What specific error message are you getting? |
| 12:26.53 | EE_BB | I configured dial plan -- exten => 000,1,Dial,Zap/g1/8700 |
| 12:27.20 | EE_BB | 8701 is give telecom signal I mean fxs |
| 12:27.22 | ManxPower | you should not use the comma syntax any more, use exten => 000,1,Dial(Zap/g1/8700) |
| 12:28.08 | EE_BB | I am trying now |
| 12:28.28 | ManxPower | 8701 days dial out using lines in group 1, it says nothing about FXS, FXO, etc. |
| 12:28.39 | the_5th_wheel | Since pings show decent times, even when i flood it |
| 12:28.43 | ManxPower | So, what is g1 connected to? Your meridian |
| 12:28.54 | ManxPower | the_5th_wheel: ping flood tests nothing meaningful. |
| 12:29.06 | yang | A simple analog PCI modem card can act as an asterisk connection to the analog phone ? |
| 12:29.14 | ManxPower | yang: No! |
| 12:29.25 | yang | how not? |
| 12:29.26 | the_5th_wheel | Im wondering if they arentperhaps QOSing me |
| 12:29.35 | ManxPower | yang: Because you cannot |
| 12:30.08 | yang | ManxPower: but its the same thing as the ISDN PCI modem card |
| 12:30.08 | ManxPower | for one thing a modem expects to connect to a phone line, not a phone. |
| 12:30.08 | ManxPower | ~fxsfxo |
| 12:30.09 | jbot | [~fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 12:30.17 | ManxPower | yang: no it is not! |
| 12:30.18 | yang | and that one works if you plug it onto |
| 12:30.24 | yang | onto NT |
| 12:30.29 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:30.29 | ManxPower | ISDN and analog are totally different. |
| 12:30.48 | yang | ManxPower: you see i have line input and phone input....dual |
| 12:30.52 | ManxPower | Also modems don't even have the hardware capable of generating a ring. |
| 12:31.04 | ManxPower | yang: no, the two ports on the modem are wired togather. |
| 12:31.10 | ManxPower | it's like putting a splitter on the line |
| 12:31.22 | yang | geez |
| 12:31.34 | ManxPower | you can easily plug a phone into the 2nd port, but the modem is not capable to using the 2nd port. |
| 12:31.48 | yang | so i can throw this old modem PCI card away, i was allready getting happy that i will be able to plug on the phone |
| 12:31.52 | ManxPower | yang: the fact you don't LIKE the answer has nothing to do with the accuracy of the answer |
| 12:32.09 | ManxPower | yang: sure you can plug a phone into it, you just can't do anything with asterisk with that port. |
| 12:32.16 | ManxPower | in fact asterisk can't even talk to the modem |
| 12:32.27 | *** join/#asterisk LeBowlingAlley (n=derek@71.16.158.170) |
| 12:32.50 | ManxPower | unless it is a modem that uses a Ambien Intel IA92 chipset, for that exact make/model you can use the wcfxo driver. |
| 12:33.24 | lmadsen | but its not recommended for anything but light hobby use |
| 12:33.35 | yang | ManxPower: but for example on windows, you were able to dial with that thing to the internet (modem feature) and have also a phone plugged in at the same time |
| 12:34.06 | ManxPower | yang: I cannot help you further. Best of luck flapping your arms. |
| 12:34.07 | yang | but yes it got the signal from the analogue line....I am talking about sharing a SIP connection through asterisk later |
| 12:34.25 | ManxPower | I gave you the answer to your question |
| 12:34.27 | yang | well if it can't go it can't I am out of luck |
| 12:34.39 | yang | yes thanks, you saved me the trouble |
| 12:35.33 | LeBowlingAlley | i had a Digium TE110P that went bad and will be getting a TE122P today. Does anyone know if this should be as simple as dropping the new card in, or do I have to do some configuration? |
| 12:36.00 | russellb | LeBowlingAlley: pretty close. |
| 12:36.08 | russellb | LeBowlingAlley: it uses a different driver, that is the only difference |
| 12:36.14 | ManxPower | LeBowlingAlley: the zaptel readme tells you what card uses what driver |
| 12:37.09 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
| 12:37.30 | WildPikachu | guys ... how to get this inuse thing to stop bitching when an inbound call hits a sip phone? i'm sure i saw a config variable for it |
| 12:38.13 | ManxPower | WildPikachu: try removing call-limit and hints |
| 12:38.25 | ManxPower | and what is the ACTUAL error message? |
| 12:38.43 | WildPikachu | well ... i saw that error earlier, but in flash operator panel the user is showing as not busy |
| 12:38.52 | ManxPower | saw WHAT error? |
| 12:39.04 | ManxPower | PASTE the error to pastebin.ca |
| 12:39.13 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 12:39.29 | WildPikachu | waits till it pops up again |
| 12:39.48 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 12:39.59 | yang | ManxPower: how do the actual analog PCI cards look like , what is the difference with the modem card? |
| 12:40.03 | *** join/#asterisk RoyK (n=roy@ip-58-10-149-91.dialup.ice.no) |
| 12:40.13 | ManxPower | yang: I cannot help you further. |
| 12:42.31 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:42.43 | *** part/#asterisk Whoopie (i=Whoopie@unaffiliated/whoopie) |
| 12:42.54 | EE_BB | ManxPower: I received message "PROGRESS with cause code 28 received" |
| 12:42.54 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 12:43.07 | hsv-al | hello fellow irc addicts - are we all looking forward to another long & glorious day of injecting our veins with e-cocaine |
| 12:43.21 | hsv-al | :) |
| 12:43.31 | ManxPower | 28 is a code meaning INVALID NUMBER FORMAT |
| 12:43.39 | ManxPower | The destination does not like the number you are sending it. |
| 12:44.11 | ManxPower | hsv-al: I'd be happy with just getting some interesting questions |
| 12:44.19 | hsv-al | can you do me a favor |
| 12:44.24 | hsv-al | msg me my cable modem ip if you keep logs? |
| 12:44.28 | hsv-al | its a knology address |
| 12:44.34 | hsv-al | don't have it written down |
| 12:44.58 | ManxPower | whatismyip.com |
| 12:45.04 | hsv-al | no at home |
| 12:45.07 | hsv-al | Im at work now |
| 12:45.15 | hsv-al | google oddly doesnt have squat |
| 12:45.19 | ManxPower | and that nick might be? |
| 12:45.23 | hsv-al | hsv-al |
| 12:45.29 | LeBowlingAlley | russellb: ManxPower: is genzaptelconf all i need to do to change the driver? |
| 12:46.09 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 12:46.09 | ManxPower | It would take me 10 mins to find the logs |
| 12:46.09 | russellb | zaptel.conf will not change |
| 12:46.09 | ManxPower | LeBowlingAlley: I have never used genzaptelconf |
| 12:46.09 | hsv-al | n/p |
| 12:46.11 | russellb | whatever method you use to load drivers may need to change |
| 12:46.37 | LeBowlingAlley | ManxPower: what do you do to change what drivers are used? |
| 12:46.57 | ManxPower | LeBowlingAlley: I change the driver |
| 12:47.05 | ManxPower | if I need to edit the config files then I edit the config files. |
| 12:47.48 | ManxPower | I edit /etc/sysconfig/zaptel, uncomment out the required driver, comment out the old driver, stop asterisk stop and start zaptel, start asterisk |
| 12:48.07 | LeBowlingAlley | ManxPower: thank you very much. I'm sure that will help immensely! |
| 12:48.11 | ManxPower | assuming there are no other config changes required (like if I went from a 1-port card to a 4-port card. |
| 12:48.15 | hsv-al | heh |
| 12:48.32 | hsv-al | oh joy, on the 6th day, it arrives today |
| 12:48.35 | hsv-al | On FedEx vehicle for delivery |
| 12:48.35 | hsv-al | |
| 12:48.36 | hsv-al | MADISON, AL |
| 12:48.46 | hsv-al | my shipping was only $1.98 heh |
| 12:48.50 | BrokenNoze | Help!! I just unloaded and loaded my app_mixmonitor.so file and it it's working with the new app_mixmonitor app. have I done something wrong? |
| 12:48.54 | ManxPower | joy, I have to install a subfloor today |
| 12:49.30 | ManxPower | BrokenNoze: so it's working and you are complaining? |
| 12:49.37 | [TK]D-Fender | BrokenNoze: Yes, you're panicking over something you just said was WORKING. |
| 12:49.39 | yang | ManxPower: i am getting very different answers on the net - http://www.velocityreviews.com/forums/t233590-asterisk-with-modems-instead-of-phonecards.html , isn't analogue modem acting the same way as Sipura ? |
| 12:50.03 | ManxPower | yang: WHAT PART OF I CANNOT HELP YOU FURTHER DO YOU NOT UNDERSTAND????? |
| 12:50.49 | *** join/#asterisk jamuse (n=josh@bzq-219-135-48.static.bezeqint.net) |
| 12:51.01 | BrokenNoze | it loaded but the new module doesn't seem to be recording the calls |
| 12:51.13 | [TK]D-Fender | yang: generic modem don't work with *, only a very specifiy chiset |
| 12:51.38 | [TK]D-Fender | BrokenNoze: pastebin the CLI output of a failed call with a recording attemtp in it at verbose 10 |
| 12:51.39 | [TK]D-Fender | ~pb |
| 12:51.40 | 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 |
| 12:51.41 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 12:51.47 | [TK]D-Fender | specific* |
| 12:51.49 | ManxPower | [TK]D-Fender: he wants to use a modem as FXS |
| 12:52.11 | ManxPower | [TK]D-Fender: he thinks he can because the "modem has 2 ports" |
| 12:52.15 | [TK]D-Fender | yang: You can't use some craptastic modem as FXS, it wasn't built for that |
| 12:52.23 | BrokenNoze | ManxPower: i was trying to patch the app_mixmonitor module and it caused the system to crash. it then restarted (safe_asterisk) and the new module is creating 684 byte files |
| 12:52.30 | BrokenNoze | OK i'll give it a go |
| 12:52.37 | [TK]D-Fender | yang: taht is a PASSTHROUGH port, the card doesn't DO anything with it. |
| 12:52.56 | yang | [TK]D-Fender: yeah ok, i am gonna google some more |
| 12:53.13 | [TK]D-Fender | yang: What is there to Google? it is simply not possible. |
| 12:53.34 | yang | [TK]D-Fender: people also told me, that I cannot use ISDN PCI modem card, but these with HFC chips work just fine |
| 12:53.51 | [TK]D-Fender | yang: All the Google-ing in the world won't make my car be able to fly like a plane, nor will it make your shitty modem work as an FXS. |
| 12:54.02 | *** join/#asterisk remycouture (n=spid3r@229.87.modemcable.oricom.ca) |
| 12:54.06 | WildPikachu | how would I include parkedcalls context in ael? |
| 12:54.08 | [TK]D-Fender | yang: What do you have exactly? |
| 12:54.09 | yang | maybe i got a very special modem card |
| 12:54.11 | WildPikachu | thinks |
| 12:54.38 | yang | [TK]D-Fender: will tell you in a few hours |
| 12:54.47 | [TK]D-Fender | yang: uh huh.... |
| 12:54.57 | yang | [TK]D-Fender: I am not at home yet |
| 12:55.07 | [TK]D-Fender | \And can't access your server from where you are? |
| 12:55.19 | yang | yes, but i need to install that pci modem |
| 12:55.20 | [TK]D-Fender | SSH doesn't care about distance |
| 12:55.42 | [TK]D-Fender | yang: Its not installed yet, and you have no idea what it is? |
| 12:56.00 | yang | no idea, its some older modem but no idea about chipsets |
| 12:56.10 | [TK]D-Fender | yang: Basica analog modem? |
| 12:56.12 | ManxPower | [TK]D-Fender: you're right, he's wrong -- no need to convince him. |
| 12:56.48 | yang | [TK]D-Fender: yep the ones people had for connecting to internet 2x analogue ports one for line the other for a phone |
| 12:57.47 | yang | ManxPower: take a look here - http://24ur.com/ekskluziv/zanimivosti/video-ko-delavec-znori.html |
| 12:58.10 | ManxPower | yang: I must now put you on /ignore. |
| 12:58.22 | ManxPower | There. Much better. |
| 12:58.41 | [TK]D-Fender | yang: Lets try this one last time. No modem on this planet has got a FXS PORT on it. It does not supply power and act as a control device for a phone it is a 100% dumb PASS-THROUGH. When the "modem" is using the "line", it disconnects the "phone" port so it doesn't interfere. Thats all it does, and odds are you can't even use the "modem" part as an FXO either. |
| 12:59.13 | [TK]D-Fender | yang: Though I'm sure if you Google enough you can magically turn a motorcycle in your driveway into a CAR. |
| 12:59.19 | ManxPower | [TK]D-Fender: he's not going to leave you alone - EVER |
| 12:59.20 | yang | lol |
| 13:00.12 | [TK]D-Fender | yang: Feel free to keep wasting your time, because make no mistake thats exactly what you're doing. |
| 13:00.12 | yang | well then i am really out of luck |
| 13:00.29 | [TK]D-Fender | yang: No.... luck is when you HAVE a chance. You don't. |
| 13:01.14 | russellb | [TK]D-Fender: such kind words :-p |
| 13:02.02 | *** join/#asterisk blinky42 (n=steveb@64.9.15.189) |
| 13:02.07 | [TK]D-Fender | russellb: I refrained from attacking him personally or swearing. That puts me well in the clear :) |
| 13:02.34 | [TK]D-Fender | russellb: But few have ever accused me of being "unclear" :p |
| 13:03.56 | [TK]D-Fender | russellb: Now in this case if yang can will solder and silicon into existance and have his "modem" do what it is we all said he can't, I'm willing to bow to this miracle. |
| 13:04.25 | russellb | starts chanting and spinning in circles |
| 13:04.39 | *** join/#asterisk danchris (n=danichri@adsl-69-109-158-119.dsl.sndg02.pacbell.net) |
| 13:04.54 | lmadsen | .dance |
| 13:05.09 | remycouture | hi, is there a way to register to an iax2 server in the dialplan ? or in the cli ? |
| 13:05.27 | ManxPower | remycouture: not that I know of. |
| 13:05.52 | remycouture | =/ |
| 13:05.52 | lmadsen | no, that is a function of the channel configuration file |
| 13:05.52 | yang | [TK]D-Fender: my knowledge is limited |
| 13:05.54 | russellb | lmadsen: wrong channel :-p |
| 13:06.06 | lmadsen | russellb: I know :) but you knew what I wanted to happen, hehehehe |
| 13:06.14 | [TK]D-Fender | remycouture: No. |
| 13:06.15 | remycouture | so i'll have to have asterisk reload his iax2 conf |
| 13:06.37 | russellb | <PROTECTED> |
| 13:06.50 | lmadsen | yes! |
| 13:06.58 | lmadsen | dances with Kirby |
| 13:07.15 | russellb | jbot_: dance is <reply> <("<) <("^) ^("^) (^")^ (^")> (>")> |
| 13:07.17 | [TK]D-Fender | remycouture: Whats scarier is I could actually read that... |
| 13:07.18 | BrokenNoze | Hey. I've rebuild app_mixmonitor for version 1.4.20.1 and tried to unload it from 1.4.18.1 and replace the module. can i do this? the recordings seem to be only a few bytes long now |
| 13:07.22 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:07.29 | [TK]D-Fender | remycouture: Sorry, not directed to you. |
| 13:07.44 | remycouture | d-fender: np |
| 13:07.52 | hsv-al | man, freenx is snappy |
| 13:07.53 | [TK]D-Fender | BrokenNoze: recompile the proper oversion of the app. |
| 13:07.57 | hsv-al | over a long distance |
| 13:07.59 | BrokenNoze | but i can |
| 13:08.07 | lmadsen | ~dance |
| 13:08.08 | jbot | Sir! Too much booty in the pants, sir! |
| 13:08.20 | russellb | jbot_: forget dance |
| 13:08.29 | russellb | :( |
| 13:08.51 | lmadsen | jbot_: no, dance is <reply> <("<) <("^) ^("^) (^")^ (^")> (>")> |
| 13:08.52 | BrokenNoze | D-Fender: But i can't find the right .c file with the mixmonitor fix in for 1.4.18?? |
| 13:08.59 | lmadsen | ~dance |
| 13:09.22 | lmadsen | I'm totally waiting for jbot to go offline because we crashed him |
| 13:09.37 | BrokenNoze | D-Fender: is there somewhere I can go for patches? can't find anything on the asterisk.org site |
| 13:09.46 | [TK]D-Fender | BrokenNoze: Just downlaod the lastest full release and completely rebuild *. |
| 13:10.13 | [TK]D-Fender | latest* |
| 13:10.14 | BrokenNoze | I can't down the system, it's a 24 hour call centre. no way of doing it on a live system? |
| 13:10.23 | LeBowlingAlley | is there a potential problem with using newer zaptel drivers on an older version of asterisk? |
| 13:10.41 | russellb | LeBowlingAlley: shouldn't be |
| 13:10.45 | *** join/#asterisk chrisM8 (n=chatzill@78-105-1-131.zone3.bethere.co.uk) |
| 13:11.52 | LeBowlingAlley | russellb: my current installation does include the zaptel drivers to support the TE122P card. installing the new zaptel package should be okay? |
| 13:11.57 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:12.49 | russellb | LeBowlingAlley: yes |
| 13:12.54 | LeBowlingAlley | thanks |
| 13:13.55 | [TK]D-Fender | BrokenNoze: then jsut do a complete rebuild of the versio you hade and reload monitor apps. |
| 13:14.34 | [TK]D-Fender | LeBowlingAlley: What ver of zaptel and * are you looking at? |
| 13:15.53 | *** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk) |
| 13:16.06 | chrisM8 | hi, I am trying to use "realtime" MySQL extensions for IVR application. Everything works OK so far, but I need to send DTFM twice during 'Backgroud' in order to force asterisk to execute my app. Any ideas? |
| 13:16.38 | hsv-al | so how many here use freenx |
| 13:16.41 | hsv-al | for remote access |
| 13:18.32 | BrokenNoze | D-Fender. but that didn |
| 13:18.52 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:20.03 | BrokenNoze | t resove the issue. are you saying that 1.4.18, if i download the tar file from the website it will likely contain the fix? |
| 13:20.49 | *** join/#asterisk hsv-al` (n=asterix@user-24-214-126-81.knology.net) |
| 13:21.41 | *** join/#asterisk BeerSerc (n=dries@unaffiliated/beerserc) |
| 13:22.14 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 13:22.54 | *** join/#asterisk anthm (n=anthm@66.72.230.179) |
| 13:23.15 | BeerSerc | Hi there. I have been using asterisk 1.2 for managing 2 sip accounts with one sip phone for 2 years now. It stopped working because of some speex incompatibilities, and as 1.2 is the latest version in gentoo, I want to manually install it now. should I choose 1.4 or 1.6? |
| 13:24.00 | [TK]D-Fender | BeerSerc: 1.4 |
| 13:24.17 | BeerSerc | OK thx |
| 13:25.41 | hsv-al | d-fender |
| 13:25.44 | hsv-al | output from your req yesterday |
| 13:25.47 | hsv-al | Notice: Configuration file is /etc/zaptel.conf |
| 13:25.47 | hsv-al | line 0: Unable to open master device '/dev/zap/ctl' |
| 13:25.51 | hsv-al | 1 error(s) detected |
| 13:26.03 | hsv-al | ztcfg -vvvv |
| 13:26.52 | yang | [TK]D-Fender: don't be too happy, i am returning in a few hours |
| 13:27.46 | [TK]D-Fender | hsv-al: Yeah... but you haven't even RECEIVED your card yet... |
| 13:27.47 | tzafrir_laptop | yang, talking's cheap |
| 13:28.02 | [TK]D-Fender | yang: I'll try not to be... |
| 13:28.18 | tzafrir_laptop | hsv-al, that file is normally created by udev once zaptel is loaded |
| 13:28.32 | tzafrir_laptop | lsmod | grep zaptel |
| 13:28.42 | *** join/#asterisk adr3nalin3_ (n=afink@asa.redglaze.com) |
| 13:29.11 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.108.161) |
| 13:29.43 | florz | Erm, just looking for opinions: would you consider it a bug if Set(X=${X}) resulted in X changing its value? |
| 13:29.51 | ManxPower | BeerSerc: Be SURE to read the both upgrade.txt files when you download the 1.4 source |
| 13:30.01 | ManxPower | florz: it should not |
| 13:30.08 | [TK]D-Fender | FlyboySR22: As shown.... I'd say yes |
| 13:30.14 | florz | ManxPower: not be a bug or not happen? =:-) |
| 13:30.15 | ManxPower | what did it change from and to? |
| 13:30.22 | florz | ManxPower: \\ -> \ |
| 13:30.34 | [TK]D-Fender | ManxPower: watch the inversions... |
| 13:30.37 | ManxPower | \ is SPECIAL |
| 13:30.40 | BeerSerc | ManxPower: I will. Theres not much configuration anyway. If I encounter problems I will ask... |
| 13:30.48 | [TK]D-Fender | florz: Yes, that looks wrong. Now show us it HAPPENING. |
| 13:31.20 | florz | ManxPower: so you say that it's not a bug? |
| 13:31.35 | [TK]D-Fender | florz: Get out of the hypothetical and show us. |
| 13:31.52 | ManxPower | florz: I believe it is not a bug if you are having \\ converted to \ |
| 13:32.05 | ManxPower | it is a bug if you have FRED converted to BOB |
| 13:32.06 | [TK]D-Fender | ManxPower: I'd class that as a "quirk" |
| 13:32.24 | ManxPower | [TK]D-Fender: *nod* |
| 13:32.26 | hsv-al | d-fender |
| 13:32.26 | florz | [TK]D-Fender: well, it's a 1.4.18.1, so I don't know whether it has changed since then |
| 13:32.28 | *** join/#asterisk DrAkOSX (n=ljd@nelug/coreteam/luisjose) |
| 13:32.29 | hsv-al | its just a general error |
| 13:32.32 | DrAkOSX | morning |
| 13:32.36 | hsv-al | that maybe the settings are right, but bugging since no hardware in system? |
| 13:32.38 | DrAkOSX | im having a weird lock up |
| 13:32.43 | ManxPower | hsv-al: it's an error indicating you have no hardware |
| 13:32.48 | DrAkOSX | it just top give tone and then core show channels shows nothing |
| 13:32.51 | [TK]D-Fender | hsv-al: clearly. |
| 13:32.54 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 13:32.57 | DrAkOSX | and it happen all in the sudden, nothing on the logs |
| 13:32.57 | florz | [TK]D-Fender: and I discovered it by reading the source - and yes, I did try it, and it did "work" |
| 13:33.00 | DrAkOSX | nothing in the cli |
| 13:33.13 | hsv-al | when i pop it in tonight, ill just reformat the whole system i guess |
| 13:33.13 | Zeeek | and GM @ the rest of you |
| 13:33.15 | hsv-al | and start clean again |
| 13:33.17 | ManxPower | DrAkOSX: are you using the latest version? |
| 13:33.18 | [TK]D-Fender | hsv-al: Holy shit my Ferrari was stolen! Oh wait... I don't have a Ferrari.... nevermind. |
| 13:33.23 | DrAkOSX | ManxPower, yes |
| 13:33.24 | hsv-al | heh |
| 13:33.26 | DrAkOSX | 20.1 |
| 13:33.47 | ManxPower | DrAkOSX: try telling us the version again and do it right this time. |
| 13:34.00 | ManxPower | would that be 1.2.20.1 or 1.4.20.1? |
| 13:34.13 | florz | Set(X=\\\\\\\\);Set(X=${X});Set(X=${X});Set(X=${X}) and ${X} is empty |
| 13:34.22 | *** part/#asterisk danchris (n=danichri@adsl-69-109-158-119.dsl.sndg02.pacbell.net) |
| 13:34.30 | ManxPower | florz: that is expected |
| 13:34.35 | DrAkOSX | Asterisk 1.4.20.1 built by root @ margarita on a x86_64 running Linux on 2008-06-06 01:31:31 UTC |
| 13:34.59 | ManxPower | DrAkOSX: I guess it is time to read backtrace.txt and submit a bug report |
| 13:35.02 | florz | ManxPower: as in "intended design" or as in "given the source"? |
| 13:35.27 | ManxPower | florz: every time you evaluate a variable ( as in ${X}) you will lose one \ |
| 13:35.40 | ManxPower | actually, I think you will lose 1/2 of the \s |
| 13:36.10 | ManxPower | florz: it works the same way in most languages |
| 13:36.11 | florz | ManxPower: exactly - now, do I understand it right that asterisk doesn't support copying variables? |
| 13:36.33 | florz | ManxPower: hu? tell me one (other than asterisk diaplan) =:-) |
| 13:36.42 | ManxPower | florz: no, Asterisk does not, as far as I know, allow you to not evaluate escape characters. |
| 13:36.50 | [TK]D-Fender | florz: That's *'s shitty dialplan parser |
| 13:36.53 | ManxPower | florz: C, Perl, PHP |
| 13:37.06 | florz | ManxPower: none of those, really |
| 13:37.10 | ManxPower | *I* think the issue is that quotes are literals, not quoting chars |
| 13:37.30 | florz | [TK]D-Fender: well, yeah, that's pretty much what I think ;-) |
| 13:37.45 | [TK]D-Fender | *I* think the issue is that * doesn't have typed variables and real parser syntax |
| 13:38.43 | florz | ManxPower: now, C doesn't have string assignment - but strcpy certainly doesn't remove any chracters |
| 13:39.16 | florz | ManxPower: $x="\\";$x=$x;$x=$x; and $x still contains a single \ |
| 13:39.30 | Zeeek | RAGE against the latrine |
| 13:39.38 | florz | ManxPower: I don't know PHP very well, but IIRC same behaviour there |
| 13:39.57 | ManxPower | http://pastebin.ca/1040380 |
| 13:39.57 | [TK]D-Fender | florz: Yes, like we've said, thats jsut how *'s extensions parser works (or in this case FAILS) |
| 13:40.04 | ManxPower | you get 5 \ instead of 10 |
| 13:40.25 | florz | ManxPower: how ya mean? |
| 13:40.35 | florz | [TK]D-Fender: well, yeah, obviously, it is ;-) |
| 13:40.38 | ManxPower | florz: RUN THE PERL SCRIPT I JUST GAVE YOU |
| 13:41.01 | florz | ManxPower: oh, sorry, overlooked that URI |
| 13:41.22 | ManxPower | florz: do this: Set(X=\\\\\); Noop(Xis ${X}) |
| 13:41.52 | florz | ManxPower: erm, yeag, in so far it's all fine at actually what I'd expect |
| 13:42.17 | florz | ManxPower: or rather - no, not in the asterisk case, I think |
| 13:42.32 | florz | *test* |
| 13:43.24 | florz | ManxPower: Well, yeah, Perl behaves as I'd expect, and asterisk behaves differently from that |
| 13:43.34 | ManxPower | florz: but perl is eating \s |
| 13:43.48 | florz | ManxPower: that perl script indeed gives you 5 \s |
| 13:43.49 | [TK]D-Fender | NOM NOM NOM NOM NOM NOM NOM NOM NOM NOM NOM NOM NOM NOM |
| 13:43.52 | ManxPower | the fact that Asterisk eats all of them is something you cannot change. This is the way Asterisk works. |
| 13:44.00 | *** join/#asterisk didz_ (n=hoje@201.19.108.161) |
| 13:44.05 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:44.14 | [TK]D-Fender | <PROTECTED> |
| 13:44.17 | florz | ManxPower: Well, I doubt that can't be changed ;-) |
| 13:44.17 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 13:44.29 | ManxPower | florz: then go change it |
| 13:44.35 | [TK]D-Fender | florz: it can. You have the source... get to work :p |
| 13:44.48 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 13:45.56 | hsv-al | hello eXistenZ |
| 13:46.02 | eXistenZ | hai |
| 13:46.10 | hsv-al | plug your long rod into my back |
| 13:46.13 | hsv-al | and upload your data |
| 13:46.21 | florz | ManxPower: Nah, for various reasons - but after all, I was just asking for an opinion as to whether it is considered a bug that you can't copy variables in asterisk ... |
| 13:47.25 | *** join/#asterisk mmartinn (n=martins@n128-227-41-215.xlate.ufl.edu) |
| 13:47.53 | mmartinn | hello glorious asterisk developers, users, and friends :) |
| 13:48.10 | ManxPower | florz: you should be accurate. Asterisk eats escape chars when assigning variables. |
| 13:48.14 | [TK]D-Fender | florz: No.. its a FEATURE |
| 13:48.25 | hsv-al | mmartinn, summer classes at ufl? |
| 13:48.29 | ManxPower | I "copy" (aka assign) variables all the time in my dialplans. |
| 13:48.31 | hsv-al | irc'ing from the comp sci lab? :) |
| 13:48.58 | ManxPower | My variables just don't contain the \ escape character. |
| 13:49.39 | florz | ManxPower: well, yeah, that's the mechanism that leads to this behaviour - and that you can copy some variables doesn't imply that you can copy all variables - which would be required to fulfill the general case |
| 13:50.14 | mmartinn | hsv-al: Oh, no, I work here :) |
| 13:50.28 | ManxPower | florz: if you REALLY must, you can try something like Set(X=\\\\); Set(Y=X); Noop(X is ${${Y}})) |
| 13:50.34 | mmartinn | hsv-al: I just didn't feel like writing the -user list, and I wanted to ping some folks on an idea ;) |
| 13:50.36 | *** join/#asterisk exvito (n=exvito@195.23.54.153) |
| 13:51.09 | ManxPower | sorry, I hve an extra ) at the end of that example |
| 13:51.36 | exvito | hi all, can anyone point to a document explaining the 'core show locks' output to someone experienced with asterisk but not its internals ? (I'm experiencing 1.4.20.1 hangs...) :( |
| 13:51.38 | florz | ManxPower: by the way, didn't test that, but nested expressions probably are broken, too |
| 13:52.11 | mmartinn | Does anyone have a creative solution (that doesn't have me modifying app_chanspy.c) for giving people a way to chanspy only on calls that have just started? |
| 13:52.25 | ManxPower | exvito: there should be info in the doc directory |
| 13:53.21 | [TK]D-Fender | mmartinn: make a scrip that looks at the open channels and picks the last one and chanspy's it directly |
| 13:53.28 | ManxPower | florz: I use things like ${DIALOPTS[${INDEX}]} |
| 13:53.32 | exvito | ManxPower: ...did not find anything related... ..google did not help. Am i missing something particular or were you pointing to that as a possible source ? |
| 13:53.48 | ManxPower | exvito: as a possible source |
| 13:54.19 | mmartinn | [TK]D-Fender: What happens when the call ends? My users like chanspy because they can jump around... |
| 13:54.27 | florz | ManxPower: I mean stuff like $[${x$[${x}+100]}+5] |
| 13:54.39 | mmartinn | [TK]D-Fender: Do you know if app_chanspy will boot you off a channel when its spygroup changes? Maybe I can do a very brief spygroup |
| 13:54.44 | [TK]D-Fender | mmartinn: No jumping around. You don't jsut hop. |
| 13:54.47 | exvito | ManxPower: ...unfortunately not. (it could really gain from having a trobleshooting.txt) Thanks |
| 13:54.52 | florz | ManxPower: no clue what one could need that for, but the code looks like it doesn't work |
| 13:55.02 | ManxPower | florz: I never referenced $[] |
| 13:55.22 | mmartinn | hummm |
| 13:55.27 | ManxPower | exvito: ask on #asterisk-dev |
| 13:55.46 | ManxPower | florz: I cannot help you further |
| 13:55.51 | Zeeek | If you haven't heard of the VoIP Users Conference it's live to day in ~2 hours. Info is found at http://x2z.eu |
| 13:56.16 | exvito | ManxPower: again, thanks... I already tried it to no avail, I guess I'll have to politely insist. :) |
| 13:57.12 | ManxPower | exvito: ask further into the USA business day |
| 13:57.57 | exvito | ManxPower: ...sure. I'll certainly do that. Thanks. |
| 14:00.57 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 14:02.06 | *** join/#asterisk feqma (n=warriner@24.75.56.66) |
| 14:04.27 | kannan | the wiki says that fir meetme, X (option for user to quit with a 1 digit DTMF) doesn't work with the s option (user can adjust volume with * menu), ? but looks like this doc id for 1.2 version, does it apply for 1.4 latest also? |
| 14:07.05 | *** part/#asterisk exvito (n=exvito@195.23.54.153) |
| 14:10.25 | *** part/#asterisk Aurs (n=Ove_Aurs@ap39pb.ip.ssc.net) |
| 14:10.44 | ManxPower | kannan: the wiki is frequently wrong. The info was not listed in upgrade.txt? |
| 14:10.57 | kannan | ManxPower, i will see |
| 14:14.43 | *** join/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 14:16.43 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 14:17.12 | *** join/#asterisk hsv-al (n=asterix@user-24-214-126-81.knology.net) |
| 14:19.58 | rootlogin | is there a software available for "simply" converting an ISDN-channel into a SIP-channel ? maybe even for windows ? (shame on me i know:) |
| 14:20.33 | [TK]D-Fender | rootlogin: Highly doubt it. |
| 14:20.37 | [TK]D-Fender | rootlogin: More than. |
| 14:20.46 | rootlogin | acting like a proxy .. its that i cant connect my asterisk box directly to the ISDN lines directly :/ |
| 14:20.56 | rootlogin | hmm thats a pitty then :) |
| 14:21.26 | [TK]D-Fender | rootlogin: and why is it you can't plug the line into *? |
| 14:21.55 | rootlogin | distance and cablig effort :) there in ethernet but no phone lines |
| 14:22.07 | rootlogin | in -> is |
| 14:22.15 | [TK]D-Fender | rootlogin: drop another server at the entry point then. |
| 14:22.33 | rootlogin | yeah :) guess there is no other way |
| 14:22.58 | *** join/#asterisk af_ (n=getsmart@88-149-241-109.dynamic.ngi.it) |
| 14:23.00 | [TK]D-Fender | rootlogin: Or buy a BRI>SIP gateway. |
| 14:24.05 | rootlogin | [TK]D-Fender: what do you mean by a BRI>SIP gateway .. something like sipgate ? |
| 14:25.41 | [TK]D-Fender | rootlogin: nota device name I'm familiar with, got a link? |
| 14:26.59 | rootlogin | eh sipgate is a provider .. sec |
| 14:27.13 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
| 14:27.33 | rootlogin | www.sipgate.de hehe german .. well |
| 14:28.06 | [TK]D-Fender | rootlogin: A service provider is NOT a hardware line converter. |
| 14:28.16 | rootlogin | so there are devices .. BRI-SIP gateways .. guess they are in the >100$ class ? |
| 14:28.26 | [TK]D-Fender | rootlogin: Got some more apples & oranges to compare? |
| 14:28.43 | [TK]D-Fender | rootlogin: much more than $100 for sure |
| 14:28.49 | rootlogin | eh no :) sry .. that was why i asked |
| 14:29.17 | [TK]D-Fender | rootlogin: There might be an option. how long an extension cable can you have for BRI? |
| 14:29.21 | rootlogin | kk then i will set up another * machine at the ISDN-end |
| 14:30.00 | rootlogin | [TK]D-Fender: i dont know |
| 14:30.44 | rootlogin | but its also that i would have to drill a bunch of new holes :( so the cable option is tedious |
| 14:31.29 | [TK]D-Fender | rootlogin: if ther is no switch between your BRI demarc & the remote server you might be able to splice it onto the network wiring. |
| 14:31.43 | [TK]D-Fender | rootlogin: BRI requires 4 pins as does 10/100 ethernet. |
| 14:32.22 | rootlogin | hehe yes :) .. right that would be cool .. but the network segment is still coax |
| 14:32.22 | [TK]D-Fender | rootlogin: I wasn't implying running cable, but sharing the existing wire. You'd have to break it out at both ends, but if the cable length doesn't impede BRI, that might work. |
| 14:32.32 | *** part/#asterisk BrokenNoze (n=vircuser@host86-150-237-169.range86-150.btcentralplus.com) |
| 14:32.34 | [TK]D-Fender | rootlogin: Coax.... NETWROK? |
| 14:32.38 | [TK]D-Fender | network* |
| 14:33.02 | rootlogin | sure i see the point .. we did that at the company :) |
| 14:33.14 | [TK]D-Fender | rootlogin: Holy.... friggen.. hell.... |
| 14:33.22 | rootlogin | yes .. 50ohms coaxial ethernet hehe :) |
| 14:33.29 | [TK]D-Fender | rootlogin: Ok, you asked for pain, and you got it by name... |
| 14:33.39 | [TK]D-Fender | ~wglwat |
| 14:33.40 | jbot | wglwat is probably well, good luck with all that |
| 14:33.41 | [TK]D-Fender | :p |
| 14:33.55 | rootlogin | hehe :) |
| 14:34.18 | rootlogin | thanks for your time :) |
| 14:34.36 | coppice | rootlogin: the good old thick coax, I hope. none of this mamby, pamby RG58 stuff |
| 14:34.52 | rootlogin | coppice: exactly that stuff |
| 14:35.30 | coppice | that stuff's about as flexible as a woman |
| 14:35.33 | EE_BB | hii all I am looking for solution that like sip client dial 9 and take second ring tone then dial real number I hope I am clear |
| 14:35.34 | *** part/#asterisk benindy (n=chatzill@99-163-253-79.lightspeed.iplsin.sbcglobal.net) |
| 14:35.50 | rootlogin | there is lots of it here .. i did hamradio .. so :) |
| 14:35.52 | EE_BB | need like access code |
| 14:35.55 | EE_BB | ? |
| 14:36.01 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 14:36.46 | *** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3) |
| 14:36.59 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 14:37.08 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:37.09 | rootlogin | coppice: well it works fine .. except for that shitty collection detection crap .. so when busy the throughput is low |
| 14:37.36 | [TK]D-Fender | EE_BB: What SIP phone will they be using? |
| 14:37.43 | *** join/#asterisk mkl1525 (n=some@89.246.219.252) |
| 14:38.10 | EE_BB | linksys spa2102 |
| 14:38.19 | coppice | rootlogin: if you have the think silver plated coax, the bandwidth is soooo much better |
| 14:39.08 | rootlogin | coppice: but i guess i would have to switch eth-cards then as well ? |
| 14:39.21 | MRH2 | hi anyone know if there is a telco-speak word for a loud 'rushing water like' noise followed by disconnect? |
| 14:39.29 | [TK]D-Fender | EE_BB: Take a look at its dialplan, you can probably do it in there. |
| 14:39.42 | coppice | its a stinker trying to get gig-E over coax cards :-) |
| 14:40.00 | coppice | MRH2: leakage? :-\ |
| 14:40.03 | hsv-al | why is there such a low google response |
| 14:40.06 | hsv-al | when googling: TDM411P |
| 14:40.20 | [TK]D-Fender | hsv-al: Because that isn't an actual model. |
| 14:40.31 | hsv-al | what is it based off then? |
| 14:40.43 | [TK]D-Fender | hsv-al: its a "fake" # to imply 1 FXO/1FXS on a TDM400P |
| 14:40.51 | hsv-al | thx |
| 14:40.57 | [TK]D-Fender | hsv-al: But the combination doesn't have a real name. |
| 14:41.40 | hsv-al | Is it not based off TDM410? |
| 14:41.46 | hsv-al | 400P=end of life |
| 14:42.03 | MRH2 | something like call connects -> "Shhhhh"->disconnect - and no that's not a person saying 'shhhh' ;) |
| 14:42.33 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:42.36 | *** join/#asterisk iamhrh (n=iamhrh@74.7.128.170) |
| 14:42.42 | *** part/#asterisk simond (n=simon@76.74.139.38) |
| 14:42.44 | hsv-al | soon |
| 14:43.07 | iamhrh | is jabber implemented in 1.4? |
| 14:43.33 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:43.59 | [TK]D-Fender | hsv-al: It could be, but like I said, thats jsut someones "fake" name. What it really means is up to the description they tack on the end |
| 14:45.14 | *** join/#asterisk anebi (n=anebi@87-126-80-161.btc-net.bg) |
| 14:45.20 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 14:46.40 | anebi | hi |
| 14:46.43 | DavidR2008 | I've got a question about connecting Asterisk to an existing PBX using a Wink Start T1 |
| 14:47.15 | anebi | i'm beginner with asterisk and i wnat to ask |
| 14:47.24 | anebi | how i need to configure asterisk (trixbox CE) the way to use already an existing SIP account |
| 14:47.45 | anebi | we have a sip account, but i don't know how to configure asterisk to use this account. |
| 14:48.10 | spokra | read /etc/asterisk/sip.conf |
| 14:48.33 | spokra | there are examples that should get you going |
| 14:48.51 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 14:49.17 | hsv-al | aneb |
| 14:49.19 | hsv-al | ~book |
| 14:49.19 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:49.25 | anebi | thanks, i will read it, do i need to configure also extensions.conf ? i'm confusing with extentions.conf file |
| 14:49.27 | anebi | thanks |
| 14:49.39 | iamhrh | that book has everything you need to get started |
| 14:49.42 | hsv-al | everything is in there that covers the basic |
| 14:49.42 | iamhrh | its even a fun read |
| 14:49.53 | hsv-al | it picks up quickly around chapter 5/6 |
| 14:50.01 | *** join/#asterisk knobo (n=bohmersp@148.122.202.168) |
| 14:50.03 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 14:50.06 | anebi | perfect, thanks :) |
| 14:50.12 | spokra | if your sip.conf you will have a block for the sip account.. the context line refers to the section of extension.conf that it will be sent when a call comes in |
| 14:50.20 | [TK]D-Fender | anebi: forget the book at this point.. |
| 14:50.24 | [TK]D-Fender | ~freepbx |
| 14:50.25 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:50.28 | [TK]D-Fender | ~trixbox |
| 14:50.29 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 14:50.33 | hsv-al | anebi, the downfall to those distros is it makes people reliant on gui only |
| 14:50.43 | hsv-al | and not understanding how to manually configure *, which is great benefit to hte user |
| 14:50.53 | knobo | Can one send information to a sip phone that someone else answered the phone, so that the phone not will report unaswered call? |
| 14:51.05 | [TK]D-Fender | anebi: the FreePBX front-end that configures * for you owns your ass. |
| 14:51.20 | hsv-al | fender, i believe EVB |
| 14:51.21 | [TK]D-Fender | anebi: Go use their help resourses to learn how to use it. |
| 14:51.22 | hsv-al | uses freepbx |
| 14:51.36 | [TK]D-Fender | hsv-al: EVB? |
| 14:51.40 | hsv-al | easyvoxbox |
| 14:51.51 | [TK]D-Fender | hsv-al: Same shit, different smell |
| 14:51.54 | DavidR2008 | I have a legacy PBX that connects to my * box via a Wink start T1. I need to play dial tone when it connects while waiting for the first touch tone. Is this possible? |
| 14:52.14 | hsv-al | d-fender i used AN for a while, but now that I see I can get things working |
| 14:52.17 | x86 | dtmf wink |
| 14:52.20 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 14:52.23 | x86 | aka e&m wink |
| 14:52.24 | hsv-al | with just console, no front end, i see no point anymore into front ends. |
| 14:52.29 | x86 | asterisk supports this |
| 14:52.37 | sehh | hey people |
| 14:52.39 | yang | [TK]D-Fender: here it is the last output for the modem - http://www.pastebin.sk/en/6878/ |
| 14:52.46 | [TK]D-Fender | hsv-al: There's a point for companies where non-techs need to configure things. |
| 14:52.56 | anebi | i would like to learn how to use asterisk not only with trixbox or asterisk know, just more things for asterisk and related stuff :) |
| 14:53.09 | hsv-al | anebi, learning cli, and AEL and other baseline items |
| 14:53.11 | sehh | q: i'm having problems loading chan_gsm, asterisk reports "chan_gsm.c: Unable to initialize GSM", anyone can help me please? |
| 14:53.14 | hsv-al | will enable fuller understanding |
| 14:53.21 | hsv-al | then make the choice to use frontends like AN, etc... |
| 14:53.52 | [TK]D-Fender | yang: I see no reference to any zaptel compatibility |
| 14:53.54 | DavidR2008 | I know it's working correctly, it just not playing dial tone |
| 14:53.56 | DavidR2008 | I need asterisk to generate dialtone after wink, but before first touchtone |
| 14:53.57 | DavidR2008 | right now, it's just silence until user enters all touchtones |
| 14:53.59 | [TK]D-Fender | yang: Looks like its 100$ useless |
| 14:54.06 | hsv-al | t |
| 14:54.29 | [TK]D-Fender | DavidR2008: use "immediate=yes", and then dump the call into DISA |
| 14:55.01 | DavidR2008 | "immediate=yes" in my zapta.conf? |
| 14:55.19 | [TK]D-Fender | DavidR2008: Yes |
| 14:55.34 | [TK]D-Fender | 100% |
| 14:55.34 | iamhrh | *doh* jabber is in the resources |
| 14:55.48 | sehh | anyone using chan_gsm please? |
| 14:56.22 | *** part/#asterisk iamhrh (n=iamhrh@74.7.128.170) |
| 14:56.41 | *** join/#asterisk s0lid (n=s0lid@122.53.69.11) |
| 14:57.07 | ManxPower | DavidR2008: I have NEVER been able to have Asterisk generate a dialtone when using W&M/Wink signaling. |
| 14:57.19 | ManxPower | And really, it should not. no other equipment does, AFIK |
| 14:57.59 | ManxPower | DavidR2008: immediate=yes makes Asterisk NOT wait for digits, but to immediately jump to extension "s" when the line goes off hook. |
| 14:58.18 | DavidR2008 | perfect |
| 14:58.25 | DavidR2008 | I can use playtones in "s" |
| 14:59.23 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 14:59.24 | *** join/#asterisk talntid (n=erict@66.208.251.170) |
| 14:59.51 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:00.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:00.24 | *** join/#asterisk sp00k3y (n=chatzill@74.202.4.2) |
| 15:00.26 | yang | [TK]D-Fender: oh crap |
| 15:01.12 | DavidR2008 | [TK]D-Fender: sorry about the DISA question, I'm very new to IRC and still learning all the commands |
| 15:02.07 | mkl1525 | Hi, * 1.4.20.1, Is it possible to add a context to AddQueueMember()? exten => 995,n,AddQueueMember(it|SIP/442@agents) seems not to work " WARNING[4777]: chan_sip.c:2923 create_addr: No such host: agents" |
| 15:03.18 | [TK]D-Fender | mkl1525: what is "agents"? |
| 15:03.28 | [TK]D-Fender | mkl1525: * is trying to resolve that as a HOST for another server |
| 15:04.03 | [TK]D-Fender | mkl1525: Somthing tells me you've got all your concepts mashed up pretty badly there. |
| 15:08.13 | kannan | any one has used meet-me web control, from http://sourceforge.net/projects/web-meetme/ |
| 15:09.09 | DavidR2008 | does zap restart kill active zap channels? Google doesn't seem to know the answer |
| 15:09.32 | mkl1525 | [TK]D-Fender @agents should be the context that's used when the agent is called - have seen some examples that used this notation - any other way to do this |
| 15:10.15 | [TK]D-Fender | mkl1525: the dialplan is not a SIP channel. You should clearly not have SIP/ in front of the extension & context pair like that. |
| 15:10.39 | [TK]D-Fender | mkl1525: So what kind of channel do you think would utilize the dilapl? |
| 15:10.44 | [TK]D-Fender | dialplan* |
| 15:10.51 | [TK]D-Fender | mkl1525: Think through it. |
| 15:10.58 | tzafrir_laptop | DavidR2008, it tries to. This hsould generally only work with analog channels |
| 15:11.19 | talntid | I have 2 asterisk servers connected on the same network - they keep dropping connection from eachother when I had qualify=yes on the configs, so I removed the qualify=yes, and they still do not work properly.. any idea what the next step is? |
| 15:11.54 | mkl1525 | [TK]D-Fender http://www.repage3.de/memberdata/yakumo88/Festplatten.pdf |
| 15:12.09 | mkl1525 | [TK]D-Fender forget the link wrong copy&paste |
| 15:12.41 | [TK]D-Fender | talntid: clarify "not work properly" in the latter scenario. |
| 15:12.42 | DavidR2008 | tzafrir_laptop: sorry! |
| 15:13.03 | mkl1525 | [TK]D-Fender AddQueueMember(techsupport|SIP/3000) -> is example from voip-info.org AddQueueMember entry |
| 15:13.24 | [TK]D-Fender | mkl1525: yes, because that is is reffering to a SIP DEVICE. that is not "dialplan". |
| 15:13.31 | talntid | Well, the call does not go through to the call box with the PRI card... agents complained before it "died" that it got staticy, and then it would not dial.... |
| 15:14.24 | yang | [TK]D-Fender: Do you know about this message - [Jun 6 17:12:27] WARNING[26809]: translate.c:175 framein: no samples for gsmtolin ; also sometimes alawtolin and ulawtolin ? |
| 15:16.12 | DavidR2008 | tzafrir_laptop: I'm not clear on what "it tries to" do? does it try to kill the existing calls, or not kill them. I need to restart zap to get the configuration change I just implemented on one T1 without killing all the calls on my other T1s, if possible |
| 15:17.56 | tzafrir_laptop | yes, it kill all existing calls |
| 15:18.02 | sehh | anyone using chan_gsm please? |
| 15:18.20 | tzafrir_laptop | you can also try 'module unload chan_zap.so' and then 'module load chan_zap.so' |
| 15:19.04 | talntid | [TK]D-Fender: Is there some more information I can provide you to help solve the issue? |
| 15:19.19 | [TK]D-Fender | talntid: no clue. |
| 15:19.40 | *** join/#asterisk dwclarkNU (n=dwclarkN@h-74-0-49-250.cmbrmaor.covad.net) |
| 15:19.42 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 15:19.54 | talntid | With qualify=off, it doesn't tell me when it loses connection... ping works the whole time... |
| 15:20.16 | dwclarkNU | i have a te212p card with 2 T1s plugged in and configured...users are experiencing dropped calls...any ideas? |
| 15:20.43 | talntid | Is there an alternative to IAX2 for linking the servers? |
| 15:21.15 | DavidR2008 | dwclarkNU: do you see anything on the console with verbosity set to 3 or higher when the call drops? |
| 15:22.05 | dwclarkNU | i'll need to try to duplicate it and watch |
| 15:23.49 | dwclarkNU | DavidR2008: should i be looking for anything in particular? |
| 15:26.23 | DavidR2008 | I don't know enough yet, to be able to tell you specifics, but I would expect if it an asterisk problem there should be some sort of error message that should give you a clue |
| 15:27.50 | *** join/#asterisk Sky-Knight (n=Sky-Knig@ip68-2-137-221.ph.ph.cox.net) |
| 15:28.00 | *** part/#asterisk Sky-Knight (n=Sky-Knig@ip68-2-137-221.ph.ph.cox.net) |
| 15:28.04 | dwclarkNU | i am using digium cards, i found this article, does it make sense: http://readlist.com/lists/lists.digium.com/asterisk-users/4/22404.html |
| 15:28.23 | Zeeek | VoIP Users Conference is live to day in ~20 minutes. Info is found at http://x2z.eu - n00bs welcome, ask your questions. Your first job is getting in on your asterisk box. |
| 15:33.01 | dwclarkNU | DavidR2008: it also seems to happen to one specific user internally more than others, as well as a specific 800 line we use for a conference bridge |
| 15:35.12 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:37.32 | *** join/#asterisk ikevin (n=kevin@ANancy-257-1-9-121.w86-204.abo.wanadoo.fr) |
| 15:37.36 | ikevin | hello |
| 15:37.40 | dwclarkNU | DavidR2008: also, we when dial into another DID on another T we have going into a different system at our company, the audio is extremly choppy |
| 15:38.21 | ikevin | anyone know a soft phone that i can use under linux & asterisk who support sms? |
| 15:42.56 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 15:42.59 | [TK]D-Fender | dwclarkNU: Waht ver of * and Zaptel are you using? |
| 15:43.16 | [TK]D-Fender | ikevin: What does a softphone have to do with SMS? |
| 15:43.41 | nny_1 | i am using a ITSP for outbound sip, and noticed they need "dtmfmode=inband" in the outbound sip.conf entry in order to pass dtmf . Is this normal? |
| 15:43.51 | [TK]D-Fender | talntid: SIP, TDMoE, or hardware |
| 15:43.58 | ikevin | [TK]D-Fender, mmm sending them? |
| 15:44.02 | Zeeek | VoIP Users Conference is live NOW. Info is found at http://x2z.eu - n00bs welcome to ask your questions there |
| 15:44.05 | [TK]D-Fender | nny_1: less common, but still out there. |
| 15:44.16 | ikevin | like x-lite, so, under linux sms are not availlable |
| 15:44.19 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:44.25 | nny_1 | [TK]D-Fender: i am trying to read up as to why this is needed |
| 15:44.26 | [TK]D-Fender | ikevin: SIP has nothing to do with SMS. Thats like asking what butter has to do with golf. |
| 15:44.45 | [TK]D-Fender | nny_1: because their gear doesn't encode to RFC 2833 |
| 15:44.48 | ikevin | SIP can be used to send sms |
| 15:45.06 | [TK]D-Fender | ikevin: Feel free to enlighten me... |
| 15:45.15 | *** join/#asterisk Guza (n=guza@77.239.64.134) |
| 15:45.17 | ikevin | under win i can do that with a kiwak sip account |
| 15:45.43 | nny_1 | [TK]D-Fender: gotcha |
| 15:45.48 | nny_1 | [TK]D-Fender: makes sense ty |
| 15:45.53 | ikevin | gizmo can do that too, so i can't use asterisk with gizmo |
| 15:45.57 | Guza | hi, will someone give me php script that show zap channels status |
| 15:46.56 | [TK]D-Fender | Guza: What you're asking for is really specific so I doubt there's anything out there that will be close enough "out of the box". Go write one yourself. |
| 15:47.36 | Guza | well i dont know php language |
| 15:47.42 | tzanger | sweet that freenx thing is going to work perfectly |
| 15:47.49 | tzanger | I forget who pointed me to it here, but thank you |
| 15:48.00 | [TK]D-Fender | Guza: Go learn, or use another language. |
| 15:48.09 | Guza | well i need php |
| 15:49.18 | ManxPower | Guza: what SPECIFIC status are you looking for? |
| 15:50.38 | ManxPower | ikevin: no, SIP is NOT sending SMS. SIP is sending a message to an SMS gateway and the gateway handles the SMS |
| 15:50.52 | ManxPower | And are you looking for GSM/Cell SMS or landline SMS? |
| 15:51.16 | ikevin | no, i would to use a soft phone |
| 15:51.17 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 15:51.34 | ManxPower | ikevin: no, there are two kinds of SMS. Which kind do you need? |
| 15:51.38 | ikevin | <ManxPower> ikevin: no, SIP is NOT sending SMS. SIP is sending a message to an SMS gateway and the gateway handles the SMS <--- sms is always a message send to a gateway ... |
| 15:51.49 | ikevin | 2 kinds of sms? |
| 15:52.13 | ManxPower | ikevin: perhaps you need to read the SMS documentation in the asterisk "doc" directory in the asterisk source tree? |
| 15:52.14 | [TK]D-Fender | ikevin: Who said there was a STANDARD for this? |
| 15:52.39 | [TK]D-Fender | ikevin: And * doesnt' really support SIP IM. |
| 15:52.50 | ikevin | i use asterisk as a sip relay, can't i use it to relay message too? |
| 15:52.58 | [TK]D-Fender | Guza: Hire a consultant or go learn PHP. |
| 15:53.06 | ikevin | [TK]D-Fender, i've patched asterisk to get it working |
| 15:53.28 | ManxPower | ikevin: Asterisk is not a SIP relay. Asterisk will regenerate any messages between the two channels. What you are looking for is a SIP Proxy, Asterisk is not a SIP Proxy, Asterisk is a SIP B2BUA |
| 15:54.13 | ikevin | i obligated to use the asterisk sms functions? |
| 15:54.31 | ManxPower | ikevin: perhaps you need to read the SMS documentation in the asterisk "doc" directory in the asterisk source tree? |
| 15:54.55 | ManxPower | Asterisk only supports lanline SMS. |
| 15:55.04 | ikevin | i think i can't get real sms because my voip provider use sip for sms |
| 15:55.13 | ManxPower | And you must have access to the SMSC for your carrier. |
| 15:55.47 | ManxPower | but of course, since you refuse to read the documentation I just pointed out earlier, I cannot help you further. I wish you the best of luck. |
| 15:56.07 | coppice | * only supports the less popular of the land line SMS protocols, too, unless that has recently changed |
| 15:56.39 | ManxPower | coppice: I suspect he's using "SMS" as a generic term for "message" |
| 15:56.56 | coppice | of course, if its the one your country uses it might be considered the more popular :-) |
| 15:56.59 | Guza | ManxPower: i need output of command: zap show channels |
| 15:57.02 | ManxPower | Which is wrong on so many levels, I'm sure a baby seal dies everytime he says "SMS" |
| 15:57.21 | ikevin | <ManxPower> coppice: I suspect he's using "SMS" as a generic term for "message" <--- nop, like sms service of a real gsm phone |
| 15:57.26 | ManxPower | Guza: then execute: asterisk -rx 'zap show channels' |
| 15:57.37 | Guza | but i need on web |
| 15:57.38 | ManxPower | Guza: and capture the output into your script. |
| 15:58.06 | ikevin | so, i'll not use asterisk for that |
| 15:58.07 | Guza | i was found some scripts but i dont know to edit them |
| 15:58.10 | ManxPower | ikevin: asterisk's SMS support does not include support for SMS service for GSM phones. |
| 15:58.18 | Guza | http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+PHP |
| 15:58.27 | ManxPower | Guza: we are not here as free consultants, we are not here as PHP teachers either. |
| 15:58.45 | Guza | ok |
| 15:59.25 | coppice | I think I would die a little inside if I taught PHP |
| 15:59.26 | ManxPower | ikevin: I will put you on /ignore now. Please tell someone when you have finished reading the document I referred you to so I can take you off /ignonore |
| 16:00.01 | ikevin | sms doc don't have infos for me |
| 16:00.07 | jblack | rough day today |
| 16:00.08 | ikevin | it's not based on sip ... |
| 16:00.34 | ManxPower | jblack: worse every day Asterisk becomes more popular. |
| 16:00.51 | ikevin | so, have a nice day |
| 16:00.53 | ikevin | bye |
| 16:00.57 | *** part/#asterisk ikevin (n=kevin@ANancy-257-1-9-121.w86-204.abo.wanadoo.fr) |
| 16:02.07 | *** part/#asterisk anebi (n=anebi@87-126-80-161.btc-net.bg) |
| 16:02.26 | nny_1 | is there a official term for being able to "intercept" a call from a sip client.. like say 100 is ringing, but 101 wants to be able to dial something and get the call |
| 16:02.52 | ManxPower | nny_1: call pickup |
| 16:03.02 | ManxPower | see pickupgroup and callgroup options |
| 16:04.18 | ManxPower | it does not work for IAX channels, BTW. |
| 16:04.29 | ManxPower | It does work for at least SIP and ZAP channels |
| 16:04.51 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
| 16:05.21 | WildPikachu | his is weird. ... i have an extension 109 and have added it to a queue as member => SIP/109 , but it shows up as SIP/109 (Invalid) has taken no calls yet |
| 16:07.04 | *** join/#asterisk Bentley (n=rcourtna@70.75.7.74) |
| 16:09.06 | *** join/#asterisk AstPT (n=wewe@213.63.128.1) |
| 16:09.21 | Bentley | hi - is anyone using a GS GPX2000? The ones we have don't send dialstrings that start with 1888 to the server .. and these phones don't appear to have an internal dialplan that you can modify |
| 16:09.26 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 16:10.56 | AstPT | I have ZAP frame retransmissions with above 40 simultaneos calls! Can any one help me? |
| 16:12.20 | nny_1 | ManxPower: so i can put callgroup=1 pickupgroup=1 under the sip peers i want to be able to do this with. however, the request is that they can intercept 100 when it rings, but dont want them intercetping each other on accident. How is this controlled? |
| 16:14.19 | nny_1 | [TK]D-Fender: should i be worried if my ITSP doesn't support RFC2833? |
| 16:14.26 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 16:15.10 | nny_1 | i know thats a strange question, but i am getting ready to setup a company that uses 20000 minutes a month long distance, and is going to be using a ITSP on top of a PRI to get cheaper long distance |
| 16:15.27 | nny_1 | well ITSP and a PRI, one for local, inbound, one for long distance |
| 16:15.40 | dwclarkNU | [TK]D-Fender: sorry, 1.4.17-1 and 1.4.10 for zaptel |
| 16:15.49 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 16:16.28 | nny_1 | ManxPower: i think i figured out how to control which phones can be grabbed etc |
| 16:17.20 | kannan | for overhead paging with Asterisk, is there any way other than to connect a auto answer sip perr to the speakers system? |
| 16:17.30 | kannan | sip peer i mean |
| 16:18.04 | [TK]D-Fender | dwclarkNU: And what kind of server? |
| 16:18.20 | [TK]D-Fender | nny_1: if you're missing too many DTMF, yes. |
| 16:19.00 | hsv-al | stupid morons screwed up in the order, and didnt ship the fxo module |
| 16:19.02 | hsv-al | overnighting it here |
| 16:19.22 | dwclarkNU | TK: Dell PowerEdge 2650 |
| 16:19.27 | nny_1 | [TK]D-Fender: k i'll see what they say about it thanks |
| 16:19.41 | [TK]D-Fender | dwclarkNU: what card? |
| 16:19.57 | *** part/#asterisk Bentley (n=rcourtna@70.75.7.74) |
| 16:20.08 | dwclarkNU | TK: TE210P |
| 16:20.33 | dwclarkNU | TK: i just reproduced the problem myself, the entire asterisk application reboots |
| 16:20.54 | dwclarkNU | TK: it drops every call in teh system and reboots the asterisk core |
| 16:21.47 | dwclarkNU | TK: but it seems as though being on the conference bridge is the cause of the reboot, i was on for maybe 5 minutes...however, prior to that, i had called my cell phone and left the call open for 45 minutes with no problems |
| 16:22.18 | dwclarkNU | TK: also, the conference bridge goes out through our long distance T while my cell phone goes out through my local T |
| 16:23.23 | dwclarkNU | TK: the confernece bridge is hosted by global crossing, not within the asterisk server |
| 16:23.35 | ManxPower | dwclarkNU: how long does the Power On Self Test take when the server reboots? |
| 16:24.15 | nny_1 | [TK]D-Fender: "We primarily use inband on our network." |
| 16:24.18 | dwclarkNU | ManxPower: not hte server, just the asterisk application |
| 16:24.22 | nny_1 | [TK]D-Fender: heh hmm should i be worried? |
| 16:24.46 | ManxPower | dwclarkNU: the correct term is RESTART, and there should be a core file generated. see backtrace.txt in the Asterisk source dir. |
| 16:24.58 | ManxPower | well the doc dir in the asterisk source dir |
| 16:26.08 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584336.dsl.bell.ca) |
| 16:26.57 | LuisTorres | Howdy |
| 16:27.20 | LuisTorres | do you guys use highpriority = yes on the Asterisk.conf? |
| 16:27.43 | ManxPower | nny_1: you should be worried if they are using inband DTMF and any codec except ulaw or alaw (G711) |
| 16:28.24 | ManxPower | since only ulaw/alaw can transport DTMF without distortion |
| 16:29.08 | dwclarkNU | ManxPower: I found the backtrace.txt file, but i'm not sure where to look from tehre |
| 16:29.29 | WildPikachu | this is weird ... opening flash operator panel remotely, seems all the buttons are just flashing |
| 16:30.00 | ManxPower | dwclarkNU: follow the instructions, create a report on bugs.digium.com |
| 16:30.23 | ManxPower | WildPikachu: IIRC that is a common thing with a common fix, check the FOP FAQ |
| 16:30.39 | WildPikachu | it worked a few minutes ago :) |
| 16:30.43 | WildPikachu | just thinking out loud |
| 16:30.43 | nny_1 | ManxPower: they are using ulaw |
| 16:31.10 | nny_1 | may be time to find an alternative though |
| 16:31.27 | ManxPower | nny_1: still not great if there's any jitter not handled by the jitterbuffers |
| 16:31.37 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.108.161) |
| 16:31.48 | nny_1 | ManxPower: you mean in regards to dtmf inbound? |
| 16:33.03 | nny_1 | trying to figure out how to set sip debug for a peer, it says no such peer, but in sip show channels, the peer is listed |
| 16:33.20 | ManxPower | nny_1: ANY dtmf |
| 16:33.28 | nny_1 | ManxPower: gotcha |
| 16:33.29 | ManxPower | nny_1: what peer? |
| 16:33.42 | nny_1 | ManxPower: "vitel-outbound/atla_test |
| 16:33.49 | ManxPower | that is not a peer |
| 16:33.56 | nny_1 | k |
| 16:34.00 | ManxPower | peers are inside the [] in sip.conf |
| 16:34.24 | dwclarkNU | ManxPower: is there a place i can see why it is rebooting? |
| 16:34.25 | ManxPower | I suspect the peer is "vitel-outbound" |
| 16:34.29 | nny_1 | got it |
| 16:34.30 | nny_1 | thanks |
| 16:35.02 | ManxPower | dwclarkNU: ASTERISK CRASHED!!! You must find out why, file a bug report, install any fixes that are found by the developers! |
| 16:35.52 | ManxPower | dwclarkNU: and if you continue to insist on using the term "reboot" to describe your problem then anyone that tries to help you will help with the wrong thing. Your Asterisk is crashing. |
| 16:36.46 | ManxPower | dwclarkNU: you should expect at least a week before you are able to solve the CRASH problem |
| 16:36.50 | dwclarkNU | ManxPower: sorry i used the wrong term, but if there is an output of why it is crashing, I might be able to Google other users who are having the same problem |
| 16:37.16 | dwclarkNU | ManxPower: there might already be a fix and I dont want to re-submit a problem that already has a solution |
| 16:37.44 | ManxPower | dwclarkNU: the output is in the core file, backtrace.txt tells you how to extract the information from the .core file. I cannot help you further. Best of luck with your problem. |
| 16:38.57 | nny_1 | aweosme: http://www.afrotechmods.com/cheap/hdspeakers/hdspeakers.htm |
| 16:39.10 | nny_1 | more awesome: http://www.youtube.com/watch?v=pmfHHLfbjNQ&eurl=http://gizmodo.com/5013705/old-printers-scanners-and-hard-drives-used-to-perform-radioheads-nude |
| 16:39.12 | nny_1 | <PROTECTED> |
| 16:40.46 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 16:46.16 | DavidR2008 | dwclarkNU: sorry I disappeared, it seems like you gotten some info though |
| 16:47.01 | *** part/#asterisk LeBowlingAlley (n=derek@71.16.158.170) |
| 16:50.54 | *** join/#asterisk imcdona (n=t@c-24-19-99-60.hsd1.wa.comcast.net) |
| 16:51.59 | *** join/#asterisk spokra (n=spokra@25.sub-70-192-20.myvzw.com) |
| 16:53.24 | nny_1 | ManxPower: their response to RFC 2833 was that "most of our vendors use inband and this is what we recommend" |
| 16:53.37 | ManxPower | nny_1: then don't sweat it. |
| 16:53.43 | nny_1 | ManxPower: roger |
| 16:53.53 | ManxPower | What they REALLY mean is that Level3 won't use anything except inband because they are evil. |
| 16:54.14 | nny_1 | ManxPower: got a big client getting ready to use them, making sure I am not setting myself up for issues |
| 16:54.17 | nny_1 | ManxPower: heh yeah oh well |
| 16:57.22 | hsv-al | on the 410P which slot does the fxs chip go into, 1,2,3,4 |
| 16:59.02 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:00.36 | imcdona | hello all. Is it possible to have a t-1 provisoned to run ATM? So the t-1 connects to an adtran 604 then converts it to a dsx-1 port that I can connect to TE205? Am I making sense? |
| 17:01.53 | imcdona | or let me put it this way. Can a te205p talk to a dsx-1 port? |
| 17:03.10 | Strom_L | imcdona: i don't see why not...T1 is T1. I don't know if the ATM stuff will work though. |
| 17:05.26 | imcdona | T1 is layer one and you can run a layer 2 protocol on top of it? Say frame relay or ATM? |
| 17:07.50 | coppice | I've never seen ATM over T1. ATM usually starts at 25M |
| 17:09.31 | florz | coppice: M as Mb/s? |
| 17:09.48 | florz | +in |
| 17:09.49 | coppice | yes |
| 17:11.31 | florz | coppice: Then I must have imagined all those consumer DSLs ;-) |
| 17:12.56 | coppice | yeah, true, but that's kinda fudgy ATM. |
| 17:13.25 | florz | well, yeah, not exactly what you were thinking of, I guess ;-) |
| 17:14.14 | imcdona | so would this product run ATM over a t-1? http://www.interlinkweb.com/systemics/product.asp?sku=ADT+4203640L1%23ATM |
| 17:15.03 | nny_1 | anyone got any clever ways of retaining caller id when transfering a to another extension? |
| 17:15.10 | nny_1 | a call* |
| 17:15.24 | *** join/#asterisk spokra (n=spokra@238.sub-70-192-59.myvzw.com) |
| 17:17.27 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:18.54 | *** join/#asterisk fedya (n=fedya@rrcs-71-43-222-2.se.biz.rr.com) |
| 17:19.47 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:20.15 | Strom_L | nny_1: yeah -- don't do attended transfers :) |
| 17:20.26 | nny_1 | Strom.. heh yeah |
| 17:21.34 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:22.13 | DavidR2008 | Quick question: does * support taking "busy out" on PRI/T lines? The latest information I can find via Google is a -dev conversation back in '06 about whether or not to implement it. |
| 17:22.30 | DavidR2008 | -taking |
| 17:22.49 | LuisTorres | weird stuff.., donno if this happens with you guys, but after a restart gracefully , the queues only see the members with a reload |
| 17:24.09 | hardwire | so you have to restart gracefully then reload the queues? |
| 17:24.14 | russellb | LuisTorres: edit your modules.conf to ensure that chan_sip.so gets loaded before app_queue.so |
| 17:24.38 | LuisTorres | russellb: good thinking |
| 17:24.41 | LuisTorres | ;) |
| 17:27.21 | LuisTorres | russellb: thanks it worked ;) |
| 17:29.23 | WildPikachu | ok, most my problems sorted, except.... when calling my pbx (tdm400), i put my call into a queue, it times out and Dial()'s the operator ... but during this dial, it doesn't detect the hangup |
| 17:30.21 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:30.54 | *** join/#asterisk s0lid (n=s0lid@125.212.79.191) |
| 17:32.16 | ManxPower | DavidR2008: it does not as of 1.4 |
| 17:33.36 | nny_1 | Strom_L: hmm need to figure out how to do an "unattended transfer" in a linksys 962, seems to have a method, but mostly requires you to just hit transfer again before they pick up |
| 17:34.21 | Strom_L | no, thats not blind transfer |
| 17:34.28 | nny_1 | yeah >< |
| 17:35.47 | DavidR2008 | ManxPower: Thx! not what I wanted to hear, but thx for the answer |
| 17:36.24 | nny_1 | Strom_L: heh ok there is another way |
| 17:36.33 | nny_1 | it works, now to figure out how to move the softkeys, if i can |
| 17:36.42 | ManxPower | DavidR2008: you might be able to fake it by doing a Hangup(34) or something like that. You can, in the CLI, do a "zap destroy channel X" but you have to restart asterisk and zaptel to get them back |
| 17:36.48 | mort_gib | nny_1: I use hold on my Snom handsets and call join. dunno if you can do the same on Linksys |
| 17:37.03 | *** join/#asterisk Twister (n=mfultz@66.172.207.172) |
| 17:37.03 | russellb | LuisTorres: you're welcome |
| 17:37.19 | ManxPower | neither option really "busies out" the channels, but it might be close enough for you rneeds. |
| 17:39.10 | Twister | hey all, i am getting a problem with 1.4.20.1 where its complaining 488 not accepted here when trying to dial in, i am using teliax as a trunk provider. i found something on google about a bug in 1.2 but i tried to apply that patch (i figured it wouldnt work but wanted to try just to say i did) |
| 17:39.13 | DavidR2008 | ManzPower: the hangup option might be just what the Dr. ordered, thx! |
| 17:39.36 | Twister | any ideas, it is infact multipart sdp |
| 17:39.55 | Strom_L | DavidR2008: download a copy of Q.850 from ITU and try cause codes until you find the one that best suits your needs |
| 17:40.19 | ManxPower | Twister: you have a codec problem. |
| 17:40.38 | ManxPower | Asterisk and teliax are not agreeing on a mutual codec. |
| 17:40.46 | ManxPower | that is TRIVIAL to fix, no patch required |
| 17:41.31 | ManxPower | what is your allow= and disallow= lines for the [teliax] part of sip.conf or iax.conf |
| 17:41.53 | anonymouz666 | anyone in here already try to use chan_mobile with nokia phones? |
| 17:42.01 | jblack | Twister: Teliax does work with asterisk. I use it here. |
| 17:42.36 | Twister | Manx, its 2 lines, you want me to paste here or to a pastebin |
| 17:42.46 | ManxPower | if it's ONLY 2 lines, put it here. |
| 17:42.50 | jblack | On their configuration page, you can chose which codes teliax will try to negotiate with you. I bet you have only g.729 accepted. |
| 17:43.14 | Twister | disallow=all |
| 17:43.15 | Twister | allow=ulaw&alaw&gsm |
| 17:43.26 | ManxPower | Twister: that is NOT a valid allow line. |
| 17:43.29 | Twister | ah |
| 17:43.37 | Twister | that could be a problem then |
| 17:43.49 | ManxPower | you would normally only allow= one codec, if you REALLY need to allow more than 1 codec use multiple allow= lines. |
| 17:44.06 | ManxPower | there is a way in 1.4 to specify multiple codecs on one line, but I never use it, the old way works just fine. |
| 17:44.24 | Twister | i just want gsm |
| 17:44.45 | ManxPower | then only allow=gsm along with disallow=all |
| 17:44.55 | Twister | disallow = all first? |
| 17:44.55 | *** part/#asterisk artz22 (n=artz@190.189.0.33) |
| 17:44.57 | Strom_L | but put the disallow first :) |
| 17:45.01 | Twister | ok |
| 17:45.04 | Twister | trying |
| 17:45.18 | jblack | Also, check your teliax configuration page, and make sure you allow gsm on the configuration page. |
| 17:49.51 | nny_1 | so, i have just been informed that a client wants a way to talk to another sip client, and interrupt the call, without having to hold the first call.. any ideas? |
| 17:50.39 | [TK]D-Fender | nny_1: .... |
| 17:50.43 | nny_1 | i know |
| 17:50.54 | [TK]D-Fender | nny_1: And what will be happening on this "not held" call? |
| 17:51.10 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 17:51.14 | nny_1 | their current 10 yeard old system "beeps" and allows the front desk girl to talk to the intended transfer OVER the current conversation |
| 17:51.30 | nny_1 | which they have exclaimed is the only way |
| 17:51.45 | [TK]D-Fender | nny_1: WTF. Sounds jsut like a 3-way acll to me. |
| 17:51.48 | [TK]D-Fender | call* |
| 17:51.52 | nny_1 | as they dont want to put the caller on hold while being informed of the incoming transfer |
| 17:51.56 | nny_1 | yeah pretty much |
| 17:52.11 | [TK]D-Fender | nny_1: So have them do that then. There is no automated way of course. |
| 17:52.51 | nny_1 | not sure how it would flow |
| 17:52.52 | [TK]D-Fender | nny_1: This shows a very pooor understanding of traditional telephony on their part. |
| 17:52.57 | nny_1 | indeed |
| 17:53.03 | nny_1 | first time i heard of it |
| 17:53.29 | [TK]D-Fender | nny_1: redescribe the calling order please. |
| 17:53.36 | Twister | ok fixed the 488 error thanks, now onto my next error |
| 17:54.04 | Twister | thank you for your assistance :) |
| 17:55.49 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 17:56.21 | grandpapadot | Hi all. Anyone ever seen an issue where two channels 'combine' or 'barge' into each other? |
| 17:56.44 | [TK]D-Fender | grandpapadot: Inexplicably? |
| 17:57.04 | grandpapadot | Yea, very weird. |
| 17:58.47 | hardwire | anybody seen rj-11 biscuit blocks that allow an in-line override if it is a junction box between the NID and another RJ11 port? |
| 17:59.42 | Strom_L | hardwire: there shouldn't be any RJ-11 anywhere beyond the very end of the inside wiring |
| 18:00.05 | Strom_L | RJ-11 is *only* for connecting CPE to the inside wiring |
| 18:00.20 | Strom_L | and they put it in some NIDs as a convenient testing feature |
| 18:00.48 | grandpapadot | [TK]D-Fender: Ok, it's not asterisk (at least not our asterisk servers). It appears to be carrier related. |
| 18:00.59 | grandpapadot | Odd.. and yet very cool in some way |
| 18:01.06 | [TK]D-Fender | grandpapadot: .... |
| 18:01.07 | hardwire | Strom_L: you're refering to NID to a single RJ-11 port right? |
| 18:01.08 | [TK]D-Fender | ~wglwat |
| 18:01.09 | jbot | wglwat is, like, well, good luck with all that |
| 18:01.10 | [TK]D-Fender | :p |
| 18:01.16 | grandpapadot | lol |
| 18:01.21 | hardwire | Most houses aren't wired that way. |
| 18:01.27 | Strom_L | hardwire: um, no? |
| 18:01.40 | hardwire | It sounded like it. |
| 18:01.54 | Strom_L | NID includes an RJ-11 to make it simple for the customer to disconnect the inside wiring and test at the demarc point |
| 18:02.20 | Strom_L | but beyond that, the only place there should be RJ-11 is where the inside wiring terminates and connects to the CPE |
| 18:02.28 | sehh | anyone using chan_gsm please? |
| 18:02.37 | hardwire | ok, I'd even be happy with an RJ-45 that handled the quick disconnect |
| 18:02.39 | hardwire | or RJ-6 |
| 18:02.41 | hardwire | or RJ anything |
| 18:02.49 | Strom_L | what for, exactly? |
| 18:02.52 | hardwire | I just need an override port that's internal :) |
| 18:03.04 | Strom_L | to override what? |
| 18:03.09 | hardwire | we forward a lot of calls from remote sales offices as forward no answer/no busy |
| 18:03.16 | hardwire | instead of forward alway |
| 18:03.17 | hardwire | s |
| 18:03.30 | LuisTorres | What is the <zombie> channel? |
| 18:03.49 | LuisTorres | happens when I make a blindtransfer |
| 18:04.12 | hardwire | so if our forward destination is screwed up, or the remote sip phones at those locations are frelled, they can always plug in a handset and it starts ringing without anybody having to deactivate the forwarding |
| 18:04.13 | Strom_L | hardwire: ok, so? how does having RJ-11 factor into this? |
| 18:04.32 | Strom_L | hardwire: hm |
| 18:04.51 | Strom_L | it sounds like you're trying to fix a badly-planned network with more bad planning |
| 18:04.57 | hardwire | yup |
| 18:05.03 | hardwire | I can't get around that right yet. |
| 18:05.27 | hardwire | I'm not proclaiming genius on this one, I'm just slowly getting the ball rolling on a better network. |
| 18:05.36 | nny_1 | [TK]D-Fender: Call comes into front desk. Attendant answers, places call on hold. Attendant then (if recipient is on the phone) pages the intended recipient, and if they are on a call, they recieve the notification WITHOUT interrupting the current call |
| 18:06.07 | nny_1 | basically they have a 20 year old system that can receive pages without interrupting the current call |
| 18:06.20 | nny_1 | i dunno how do duplicate this with astersik quite yet |
| 18:06.39 | nny_1 | considered IM/ jabber etc, but they don't want notifications on the computer screen |
| 18:06.42 | [TK]D-Fender | nny_1: here's how : |
| 18:07.26 | *** join/#asterisk truent (n=truent@c-67-181-8-115.hsd1.ca.comcast.net) |
| 18:07.48 | [TK]D-Fender | nny_1: in your "user extensions" for the receptionist, have them check if they are on a call. If so, initiate chanspy + Whisper page, otherwise just dial as normal. |
| 18:08.42 | nny_1 | [TK]D-Fender: sounds cool enough.. now to figure out how :) |
| 18:10.16 | jblack | The dow is down 320 points today. nice. |
| 18:10.45 | grandpapadot | jblack: woohoo! come on $6.50 petro prices... I might just go crank up my trunk and let it run for a couple hours... |
| 18:11.29 | jblack | Yeah. Oil is hitting new highs today too. |
| 18:12.15 | jbeez | bleh |
| 18:12.22 | nny_1 | [TK]D-Fender: cool though, thanks for the start |
| 18:12.50 | nny_1 | [TK]D-Fender: meh... need to find a way to avoid future fun stuff like that popping up in inning 9 |
| 18:13.39 | grandpapadot | jblack: It totally amazes me that this country doesn't understand the most basic economic concept like supply and demand. |
| 18:14.25 | jblack | which country? |
| 18:14.35 | grandpapadot | USA, sorry |
| 18:14.36 | ManxPower | grandpapadot: because until recently the price of gas in the USA had very little to do with supply and demand. It had to do with the very high dllar. |
| 18:14.58 | jblack | That's still mostly the case, if you ask me. |
| 18:15.23 | ManxPower | jblack: I guess you've not looked at the value of the dollar over the past 6 years |
| 18:15.27 | [TK]D-Fender | Yeah, well Gas in the US costs about $1.04 / L. Here its sitting at $1.43 / L |
| 18:15.32 | [TK]D-Fender | Go f'n figure |
| 18:15.45 | [TK]D-Fender | 30-45 minute drive to the border... |
| 18:15.45 | jblack | Heh. That's just a teaser manxpower. |
| 18:16.17 | ManxPower | [TK]D-Fender: that's all taxes, I think |
| 18:16.18 | grandpapadot | ManxPower: Agreed. The value of the dollar has really tanked. Our middle class is being adjusted. |
| 18:16.21 | jblack | It's dropped only what? maybe 25-30%? I'm expecting it to drop 70% off of it's high. |
| 18:16.44 | *** join/#asterisk mkl1525 (n=some@82.193.233.108) |
| 18:16.45 | grandpapadot | Which is why I'm learning chinese ... |
| 18:16.49 | grandpapadot | early |
| 18:17.17 | [TK]D-Fender | "W" & the Fed are bankrupting the US. |
| 18:17.27 | jblack | yeah |
| 18:17.55 | Strom_L | GUYS GUYS CAN I JOIN IN TOO? I ALSO ONCE TOOK INTRO TO MICROECON AND INTRO TO POLITICAL SCIENCE |
| 18:18.35 | grandpapadot | I can't wait for Obama to get elected, he'll fix everything .. . |
| 18:18.41 | talntid | lol. |
| 18:18.59 | mkl1525 | Hi, * 1.4 - I'd like to have different moh files for different queues - do I have to create a dir for each file or is there a way to use single files direct in musiconhold.conf? |
| 18:19.02 | nny_1 | [TK]D-Fender: wondering id there is a way to send messaged to a linksys 962 |
| 18:19.06 | nny_1 | if* |
| 18:19.12 | ManxPower | [TK]D-Fender: part of it is also that the Euro is becoming more and more of an international currency. The dollar used to be the international currency and that propped up the value of the dollar |
| 18:19.18 | Strom_L | mkl1525: separate dirs, separate moh classes |
| 18:20.23 | coppice | grandpapadot: will obama help with the chinese classes? |
| 18:20.43 | mkl1525 | Strom_L thanks for the info |
| 18:22.20 | x86 | rofl! amazon.com is down |
| 18:22.39 | Strom_L | x86: old news |
| 18:22.50 | x86 | how long has it been down? |
| 18:23.02 | Strom_L | since at least this morning when I looked at it? |
| 18:23.10 | x86 | anyone know whats going on with it? |
| 18:23.27 | Strom_L | x86: I don't know for sure, but I think it might be down, or something. |
| 18:23.39 | x86 | heh |
| 18:24.12 | hsv-al | :) |
| 18:24.21 | Strom_L | lord knows I can't go more than twenty-nine seconds without buying books on amazon.com |
| 18:24.27 | Strom_L | this is such a tragedy |
| 18:24.28 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:24.32 | hsv-al | strom_L |
| 18:24.35 | hsv-al | usenet is your friend :) |
| 18:24.46 | Strom_L | hsv-al: I'm being sarcastic, stupid. |
| 18:25.41 | x86 | haha |
| 18:25.51 | jblack | Yeah. The euro is a part of why I think the Dollar has only dropped a little bit comapred to where it's going. once the euro becomes the common reserve currency for oil, the US is fucked. |
| 18:26.02 | x86 | Strom_L: you're being aweful mean today... got sand in your clam? |
| 18:26.08 | hsv-al | V? |
| 18:26.15 | Strom_L | this counts as mean? |
| 18:26.17 | Strom_L | wow |
| 18:26.26 | x86 | :p |
| 18:26.32 | [TK]D-Fender | Strom_L: Got Clam? (tm) |
| 18:26.47 | x86 | jblack: oh no kidding... the US is fucked anyway... I'm buying guns as quick as I can :P |
| 18:26.47 | Strom_L | are we talking about chowder or scientologists? I can't tell. |
| 18:27.01 | jblack | Ohh, down almost 350. I wonder if we'll hit 4 today. |
| 18:27.38 | x86 | jblack: for what? |
| 18:27.42 | jblack | The dow. |
| 18:27.47 | x86 | ah |
| 18:27.59 | x86 | -350 points in one day? wow |
| 18:28.12 | hsv-al | wolf blitzer saying earlier gas prices to |
| 18:28.16 | hsv-al | 7.00/gallon for regular by september |
| 18:28.17 | hsv-al | HEH |
| 18:28.18 | talntid | so, seeing as there is a probably depression coming, what are you guys doing to prepare? or are you? |
| 18:28.34 | jblack | 7 would surprise me. 5 wouldn't. |
| 18:28.36 | x86 | buy guns and a lot of ammo |
| 18:28.44 | Strom_L | Nothing is more amusing than clueless morons discussing economics on IRC |
| 18:28.47 | x86 | that's all the preperation you need |
| 18:28.51 | jblack | Me? I'm buying all the food I can afford. =) |
| 18:28.51 | talntid | ;P |
| 18:28.57 | Strom_L | this is better than going to the zoo |
| 18:29.01 | talntid | is silver a good investment? |
| 18:29.02 | x86 | Strom_L: then consider yourself amused... on my command |
| 18:29.05 | jblack | Yeah. |
| 18:29.19 | x86 | gold is a better investment |
| 18:29.27 | coppice | Strom_L: they learn it from CNN |
| 18:29.34 | x86 | I don't watch CNN |
| 18:29.46 | jblack | well, long term, it's good. Commodities are overinflated right now, because everyone ran to them at once |
| 18:29.59 | *** join/#asterisk tomcontr3 (n=gcontrer@190.196.17.230) |
| 18:30.04 | x86 | in fact, the only news I purposely injest comes from Slashdot or The Register... all other news is crap |
| 18:30.11 | tomcontr3 | hi, Im having a problem with IAX |
| 18:30.12 | hsv-al | "subprime" |
| 18:30.14 | jblack | That's 1/2 of your inflation right there. Futures. |
| 18:30.15 | hsv-al | used to be phrase of the year |
| 18:30.17 | Strom_L | x86: ingest? or in jest? |
| 18:30.17 | talntid | i disagree, gold is still too expensive to buy right now. I think silver will likely gain in value nearly to gold. i may be off though |
| 18:30.17 | tomcontr3 | I have a Box that has 2 pstn lines connected using a TDM400 Card... |
| 18:30.23 | tomcontr3 | Also in that box I configured an IAX Extention.. |
| 18:30.23 | tomcontr3 | <tomcontr3> from another box I configured a trunk using that IAX |
| 18:30.30 | tomcontr3 | from another box I configured a trunk using that IAX extention |
| 18:30.36 | x86 | Strom_L: ingest* |
| 18:30.38 | tomcontr3 | now the problem is that every time I make calls using that IAX extention (or any other from the BOX 1) I always hear an Incoming call sound while Im talking... which is really anoying |
| 18:30.40 | tomcontr3 | any ideas? |
| 18:30.44 | jblack | wonders where today's mail is |
| 18:30.52 | jblack | No mail yesterday, no mail yet today. |
| 18:30.53 | talntid | does too. lol |
| 18:30.55 | Strom_L | tomcontr3: uhm.....pastebin your Dial() line |
| 18:30.57 | x86 | tomcontr3: your copy and paste skills are fail |
| 18:31.10 | Strom_L | tomcontr3: but seriously, your question makes no sense at all |
| 18:31.18 | talntid | shoulda used next-hour mail service. |
| 18:31.23 | tomcontr3 | the problem is really wird |
| 18:31.27 | jblack | bah |
| 18:31.34 | hsv-al | thats insanity |
| 18:31.40 | jblack | funny thing is, fed ex was next door just an hour ago. |
| 18:31.41 | coppice | I saw a buy one get one free deal on houses somewhere in the US. If you can buy a pair of semi-detached houses like that you could break them through into somewhere pretty nice :-) |
| 18:31.42 | Strom_L | CAN'T WE JUST KILL ALL THE POLITICIANS AND REPLACE THEM WITH ENDLESS TECHNO |
| 18:31.43 | hsv-al | if my tank is $51.22 to fill up w/ a 2004 honda civic |
| 18:31.48 | tomcontr3 | and also It seems to happend only with some IP phones |
| 18:31.50 | hsv-al | at $3.92/gallon(reg) |
| 18:31.54 | tomcontr3 | others does not have the problem |
| 18:31.59 | hsv-al | almost $100+ to fill a honda civic up |
| 18:32.01 | hsv-al | at $7.00/gallon |
| 18:32.14 | hsv-al | I feel bad for those w/ Ford F350 Turbo Diesels :) |
| 18:32.26 | talntid | .... or.... 2004 dodge 2500's... :( |
| 18:32.34 | jblack | There's one upside to high prices.... I've had less trouble getting about town |
| 18:32.35 | talntid | cummins turbo diesel |
| 18:32.38 | coppice | at $7 a gallon you could be where most of europe was a generation ago :-) |
| 18:32.51 | talntid | imperial gallon? :P |
| 18:33.15 | coppice | oh, yeah. that changes it by 25%, doesn't it? |
| 18:33.24 | jblack | There's a 2003 hummer that the ford dealership is still tring to sell. I almost went in and said "I'll give you a hundred bucks for it", until I realized I didn't even want it at that price. |
| 18:33.46 | talntid | that would be worth it for the fuel resale. |
| 18:34.03 | talntid | that, and when the depression hits, you'll have a armored vehicle. :P |
| 18:34.16 | jblack | that thing is too heavy to push. |
| 18:34.21 | coppice | I saw GM might shut down the hummer businesss |
| 18:34.31 | jblack | I should get a mini, and make my daughter push me everywhere I want to go. |
| 18:34.41 | jblack | coppice: Yeah. I heard that on bloomberg yesterday. |
| 18:34.45 | talntid | i am going to purchase a motorcycle soon. 80MPG |
| 18:34.51 | hsv-al | Geo Metro |
| 18:34.57 | talntid | i have a metro |
| 18:35.00 | talntid | actually, swift. |
| 18:35.06 | talntid | same thing |
| 18:35.14 | jblack | That tempts me too. But better mileage still, is only making weekly trips. A tank of gas lasts me two months. |
| 18:35.39 | talntid | yeah. i have to travel to work and places :( |
| 18:35.45 | jblack | talntid: Did you upgrade the kernel on that domU yet? |
| 18:35.48 | hsv-al | metro's get 75mpg |
| 18:35.52 | x86 | jblack: jesus... what kind of car are you driving |
| 18:35.56 | coppice | the last tank of gas I bought was nearly 2 years ago |
| 18:35.57 | hsv-al | http://1989geometro.com/ |
| 18:36.04 | jblack | x86: It's not what I drive, it's how much I drive. |
| 18:36.08 | talntid | jblack: no. i'm not trying to break it currently ;) |
| 18:36.14 | jblack | I could drive a hummer, and still only fill the tank once every two months. |
| 18:36.25 | x86 | jblack: I drive about 30 minutes a day, 15 minutes to work, 15 minutes home.... |
| 18:36.38 | jblack | I drive well less than 3,000 miles a year. |
| 18:36.43 | jblack | Closer to 2k |
| 18:36.51 | x86 | wow |
| 18:37.09 | coppice | I've driven 0 miles in the last year |
| 18:37.09 | [TK]D-Fender | indeed |
| 18:37.13 | seanbright | are you in your golden years? |
| 18:37.23 | talntid | he's superman. he flies everywhere |
| 18:37.29 | seanbright | the autumn of your life? |
| 18:37.29 | jblack | my 2003 subaru something or other has 23000 miles on it |
| 18:37.57 | jblack | Something like that. |
| 18:38.13 | hsv-al | g37 coupe gets only 28mpg |
| 18:38.15 | talntid | you're due for a timing belt in.... 13 years. |
| 18:38.17 | jblack | My car insurance is something like 500 a year for full coverage. :) |
| 18:38.33 | talntid | gah |
| 18:38.37 | talntid | that's my 6 month rate. |
| 18:38.37 | jblack | And I always change my oil every inspection. |
| 18:38.51 | [TK]D-Fender | jblack: your cost/mile sucks.... |
| 18:38.59 | jblack | It does. |
| 18:39.01 | talntid | just think... |
| 18:39.02 | coppice | is that your major form of oil consumption? :-) |
| 18:39.13 | hsv-al | jblack, your in hong kong |
| 18:39.17 | talntid | jblack currently has fuel in his car, that was bought at 0.99c/gallon! |
| 18:39.18 | hsv-al | whats gas there? / gallon |
| 18:39.22 | jblack | hmmmm. Probably, yes. |
| 18:39.41 | jblack | I have no idea. It seems like every time I buy a new tank of gas, it's gone up 30 cents. |
| 18:39.53 | hsv-al | you american in hk on consulting work? |
| 18:40.04 | talntid | converted the racecar to e85.... 1.79/gal |
| 18:40.10 | jblack | Nah. I'm just a worthless bum. Ask tal |
| 18:40.29 | jblack | My only use to society is pushing 110 buttons, in various ordering. |
| 18:40.39 | talntid | heh. |
| 18:40.49 | talntid | in a very raped manner. |
| 18:40.54 | talntid | do'h! rapid. |
| 18:41.02 | jblack | Both are correct. |
| 18:41.03 | drako | jblack, not in here, you can fill ur tank to full with €1 |
| 18:41.14 | jblack | My primary oil use is probably replacement keycaps |
| 18:41.43 | talntid | oh hey. that's a funny thought |
| 18:41.47 | talntid | eric buys motorcycle |
| 18:41.52 | talntid | eric's lung fucks up. |
| 18:42.09 | jblack | can I have your stuff? |
| 18:42.12 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 18:42.14 | talntid | i suppose |
| 18:42.28 | talntid | i have lots of computer stuff/ |
| 18:42.30 | talntid | :) |
| 18:42.38 | ManxPower | coppice lives in a civilized country -- one with public transportation |
| 18:43.02 | coppice | yeah, but the oil prices have pushed up the bus fares recently |
| 18:43.27 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 18:43.36 | talntid | hey james, are you sure nobody has picked up your house, and put it somewhere else as a prank? maybe that's why you're not getting mail.... |
| 18:43.39 | coppice | gas is really expensive here. |
| 18:43.53 | ManxPower | idly notes he has a geo metro with new tires, new water pump, new exhaust system, and a blown engine. |
| 18:44.02 | ManxPower | looking to sell it |
| 18:44.08 | talntid | where u live? |
| 18:44.19 | talntid | please say WA |
| 18:44.22 | talntid | :) |
| 18:44.27 | ManxPower | talntid: about 1/2 way between Birmingham, AL and Huntsville, AL |
| 18:44.32 | talntid | do'h! |
| 18:44.37 | talntid | what year is it? |
| 18:44.40 | seanbright | 2008 |
| 18:44.47 | seanbright | for another 6 months or so |
| 18:44.47 | talntid | smartass. |
| 18:44.50 | hsv-al | manx, where you at |
| 18:44.51 | ManxPower | 94 I think. I'd have to go look |
| 18:44.56 | talntid | 3cyl? |
| 18:45.01 | ManxPower | yes |
| 18:45.08 | talntid | boxy body or round? manual? |
| 18:45.22 | x86 | so when is oil going to switch to the Euro? |
| 18:45.23 | drako | coppice, http://money.cnn.com/pf/features/lists/global_gasprices/ <- check Venezuela |
| 18:45.27 | ManxPower | automatic. I would not call the chassis "boxy" |
| 18:45.42 | talntid | how much u want fer it? |
| 18:45.43 | hsv-al | manxpower, get 24" rims on the metro, black tint, reverse spinners on the rims, and a chandalier hanging in the ceiling in it, and Injen exhaust |
| 18:46.02 | ManxPower | It's the Geo Metro LSI |
| 18:46.08 | Strom_L | jesus fuck on a stick -- you people need to install spell check or something. |
| 18:46.12 | talntid | reason i ask, is my sister rolled my dad's metro 3cyl the night before mothers day.. |
| 18:46.16 | ManxPower | hsv-al: I need a new engine first |
| 18:46.19 | talntid | so got great motor, rolled body :P |
| 18:46.39 | ManxPower | blown piston rod, by the sounds of it. |
| 18:47.01 | jblack | where. is. the. mail. man. |
| 18:47.26 | talntid | off spending ur $$ |
| 18:47.31 | jblack | The bastard took my stimulus payment and other associated money, and went to disneyland, I bet |
| 18:47.33 | coppice | drako: well venezuela owns the stuff. china is quite a bit cheaper than the US right now, and they import most of theirs |
| 18:48.07 | jblack | yeah. venezuela is 14 cents a gallon, no? |
| 18:48.31 | jblack | Oh, did you guys see the duke nukem forever video? |
| 18:49.15 | jblack | x86: oil is switching to the euro as we speak. Iran has already moved, which is why Bush wants to attack it |
| 18:49.29 | ManxPower | BTW, in the USA federal gas tax is 18.4/cents/gal and (for example) Alabama state gas tax is 18/cents/gal |
| 18:49.41 | jblack | That was why the US attacked Iraq. They had announced plan to move to the euro |
| 18:50.17 | coppice | are they going to attack most of mainlain europe for that reason? |
| 18:50.44 | jblack | Most of mainland europe is a net importer. Western europe, at any rate. |
| 18:50.45 | ManxPower | oil being sold in euros will TOTALLY screw the USA |
| 18:51.05 | jblack | Russia is much to large to invade, and south america would band together. |
| 18:51.17 | coppice | but they were the very first to change to the euro :-) |
| 18:51.47 | jbeez | How will oil sold in euros screw the USA? |
| 18:51.57 | ManxPower | jblack: the US attacked Iraq because Bush'd poll numbers were too low. |
| 18:52.24 | jblack | bush's numbers were pretty high before the attacks |
| 18:52.37 | ManxPower | jbeez: according to some people, oil being sold in dollars artificially props up the value of the dollar. |
| 18:52.51 | jblack | Some? That's pretty well known. |
| 18:53.03 | coppice | war is a wonderful thing for a government. Its very hard to loose an election while at war, and the constitutions of most countries gives the leaders very broad powers while at war |
| 18:53.06 | Strom_L | I'M CONFUSED NOW -- IS THIS #ASTERISK OR IS THIS #OILCHAT |
| 18:53.15 | Strom_L | LITTLE HELP HERE PEOPLE |
| 18:53.27 | talntid | its... #pissoffstrom chat :) |
| 18:53.27 | jblack | Strom_L: What do you need help with? |
| 18:53.29 | ManxPower | Strom_L: HELP ME WI TRICKSBOX! |
| 18:53.40 | Strom_L | I CAN'T INSTALL MY TRIXTHING |
| 18:53.46 | Strom_L | I MEAN BREAKFAST CANCER OR WHATEVER |
| 18:53.48 | talntid | bbiadf |
| 18:53.49 | jblack | bah |
| 18:54.22 | jblack | coppice: Then how do you explain the 2006 flip of congress? |
| 18:55.50 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:55.51 | coppice | bush didn't loose in 2004, did he? however badly most things were messing up? |
| 18:56.43 | jblack | again, how do you explain the firing of nearly every incumbant republican that ran in 2006? |
| 18:56.59 | coppice | many, like thatcher in the UK, have achieved landslide successes just a short time after heading for a historic crushing, just by starting a totally avoidable war |
| 18:57.42 | coppice | I think the war factor only tends to work well for the leader. those following can get badly screwed |
| 18:58.16 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:59.06 | *** join/#asterisk nicox (n=nicox@213-33-15-152.adsl.highway.telekom.at) |
| 19:00.11 | DavidR2008 | * Q: it seems like if I set caller ID with set(callid(all)=blah blah) then I issue a goto to another context caller id is no longer set. Is the correct, if so what should I do? |
| 19:00.16 | jackson__ | "That's why we need someone in the white house with experience" - however noone talks about someone with a solid rasp on history. |
| 19:00.27 | jackson__ | er grasp |
| 19:01.22 | *** join/#asterisk korihor (n=korihor@190.199.171.145) |
| 19:02.58 | Strom_L | DavidR2008: what happens if you set caller ID name and number separately? |
| 19:03.51 | *** join/#asterisk nicox_ (n=nicox@213-33-15-152.adsl.highway.telekom.at) |
| 19:03.59 | *** join/#asterisk spokra (n=spokra@145.sub-70-192-139.myvzw.com) |
| 19:04.19 | DavidR2008 | I spoke too soon. I wasn't setting callerid correctly! |
| 19:04.48 | x86 | http://images.icanhascheezburger.com/completestore/2008/3/28/iizgonnageti128511981945156250.jpg <-- ManxPower? |
| 19:11.30 | nny_1 | [TK]D-Fender: can i pm you real quick? |
| 19:16.28 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 19:18.18 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:18.25 | kannan | hello , i have a question about commercial call charge rates for a T-1 in California. What would be an approx cost per minute to call (1) California lanlines (2) rest of USA states? |
| 19:20.36 | ManxPower | kannan: in the USA it costs the same call a landline as a mobile phone |
| 19:20.49 | ManxPower | kannan: you need to talk to a carrier to get cost information |
| 19:21.29 | kannan | ManxPower, yes, i plan to mail Verizon, just wanted to know the approx chage |
| 19:21.32 | nny_1 | anyone here use chan spy? |
| 19:22.05 | nny_1 | i am working on implementing this "feature" an old system had (note: feature = work around for having one line by paging the person) and not sure how to implement it |
| 19:22.14 | ManxPower | kannan: anywhere from 25/cents/min to 2/cents/min, it all depends on your usage commitments, contract length, etc |
| 19:22.18 | nny_1 | i have exten => 123,1,ChanSpy(26|W) |
| 19:22.34 | nny_1 | to basically test dialing 123 and be able to whisper to 26 when they are on a call |
| 19:22.37 | nny_1 | is that right? |
| 19:23.05 | ManxPower | nny_1: I thought chanspy took a channel, not an extension. I'd have to look it up |
| 19:23.44 | nny_1 | er well |
| 19:23.48 | nny_1 | hmm |
| 19:23.50 | nny_1 | actually |
| 19:23.52 | hardwire | how could it take an extension? |
| 19:24.08 | hardwire | it would have to traverse the dialplan and get routed the same way, always. |
| 19:24.09 | nny_1 | i think it can now, vs zapbarge nto surte |
| 19:24.10 | nny_1 | sure |
| 19:24.17 | RoyK | ManxPower: chanspy takes a channel, but it can take them by group, which is nice |
| 19:24.51 | *** join/#asterisk nicox (n=nicox@213-33-15-152.adsl.highway.telekom.at) |
| 19:25.35 | ManxPower | nny_1: "core show application chanspy" would tell you |
| 19:26.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:26.12 | nny_1 | yes it grabs a channel |
| 19:26.16 | nny_1 | in this case SIP/26 |
| 19:26.26 | nny_1 | works, now to make it automagic :\ |
| 19:26.40 | ManxPower | so ChanSpy(SIP/26|W) |
| 19:26.47 | nny_1 | yeah |
| 19:27.12 | ManxPower | exten => _*99XX,1,ChanSpy(SIP/${EXTEN:3}|W) |
| 19:27.36 | nny_1 | need to make it so it only works when they are on a call, so if she trys to transfer and they are then the phone she can whisper the info to them :\ |
| 19:29.48 | nny_1 | gonna have to do it so only her extension can use the ability, which means custom default for just that sip peer right? |
| 19:36.27 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 19:36.37 | teknoprep | is there a TAPI application for asterisk.. besides asttapi ? |
| 19:36.48 | hardwire | haha |
| 19:36.55 | hardwire | who named it asttapi? |
| 19:37.00 | hardwire | cause they totally need a medal of humor. |
| 19:37.06 | _ShrikE | teknoprep: have you looked at snap dialer? |
| 19:37.42 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 19:38.03 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 19:38.11 | a1fa | what has been done about zaptel timing in xen machines? |
| 19:38.15 | teknoprep | will do |
| 19:38.22 | _ShrikE | http://www.snapanumber.com/ |
| 19:38.45 | teknoprep | i thought if you had VT extensions in your CPU... timing was perfect inside of XEN |
| 19:38.51 | ManxPower | a1fa: shouldn't you be talking to the Xen people about htat? |
| 19:39.35 | a1fa | sorry :9 |
| 19:39.40 | a1fa | whats the deal? |
| 19:40.05 | nny_1 | whats the best way to see if a person is in a call? I have hints setup for the front desk blf, any way to check that variable like if hints=foo than do this else do this?> |
| 19:41.06 | a1fa | ManxPower : whats the deal with xen and timing? |
| 19:41.09 | ManxPower | nny_1: Dial the destination, then check the value of DIALRESULT or HANGUPCAUSE |
| 19:41.54 | ManxPower | a1fa: I have no idea, but if timing does not work in Xen, then it's Xen's problem not Asterisk's problem |
| 19:42.03 | ManxPower | nny_1: you can also use chanisavail |
| 19:42.38 | nny_1 | ManxPower: the channel would be available wouldn't it? |
| 19:42.45 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 19:42.56 | ManxPower | nny_1: wny would it be available? |
| 19:42.57 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:43.05 | ManxPower | nny_1: go read the docs for that app |
| 19:44.13 | a1fa | ManxPower : someone told me timing wont work at all unless you have zaptel source |
| 19:44.25 | _ShrikE | Anyone put their hands on an Adtran IP 712? |
| 19:44.36 | x86 | what is it? |
| 19:44.41 | _ShrikE | IP phone |
| 19:44.47 | *** join/#asterisk icel (n=dan@65.200.26.108) |
| 19:44.53 | x86 | had no idear adtran even made phones |
| 19:44.55 | x86 | that's cool |
| 19:45.07 | _ShrikE | They have been rebranding polycom, but this one is theirs I believe. |
| 19:45.41 | nny_1 | ManxPower: maybe mis communicating the need. SIP peer is using one of it's registered "lines" to talk to someone over Zap. There is still both another zap channel available as well as another SIP line available on the peer. BLF knows this because when a call comes in it flips the little hint flag. I want to know when they are using the phone, even if the channel is available. I apologize if i screwed up any of these concepts |
| 19:46.01 | ManxPower | a1fa: IAX2 Trunk and MeetMe require timing, nothing else does. |
| 19:46.23 | [TK]D-Fender | _ShrikE: I don't see any retailers carrying it, and street pricing doesn't say much for its value. |
| 19:46.33 | *** join/#asterisk izaak (n=izaak@modemcable101.145-203-24.mc.videotron.ca) |
| 19:46.34 | nny_1 | thinking $HINT |
| 19:46.47 | ManxPower | nny_1: the only other option that I can think of is to use GROUP and GROUP_COUNT stuff |
| 19:47.04 | ManxPower | nny_1: well you know where all the Asterisk variables are documented. Check there. |
| 19:47.10 | [TK]D-Fender | ManxPower: and app_page, which uses Meetme (people don't tend to think what powers it.) |
| 19:47.13 | _ShrikE | 888voipstore looks like they have them |
| 19:47.18 | nny_1 | kk just brainstorming.. thanks for the input |
| 19:47.21 | ManxPower | [TK]D-Fender: I sit corrected. |
| 19:47.58 | nny_1 | any way to see what a variable value is form cli? sure there is, never tried :) |
| 19:48.16 | nny_1 | from* |
| 19:48.51 | ManxPower | nny_1: no. |
| 19:49.01 | izaak | setting up a new embedded pbx. like linux 'cause it's the well-worn path. love [open/free]bsd because of pf. qos is important. what to do? asterisk on bsd, stable enough? deal with freakin' iptables/tc? thoughts? |
| 19:49.07 | nny_1 | booo |
| 19:49.20 | ManxPower | well, creat an extension with Noop(HAPPY_VAR is ${HAPPY_VAR}) then call that extension |
| 19:49.27 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 19:49.40 | nny_1 | yeah heh was hoping for a quicker method i could use later.. will do |
| 19:49.41 | ManxPower | izaak: Asterisk is by far best run on Linux |
| 19:49.58 | ManxPower | There are the couple of oddballs that like pain and run it on BSD, but not all THAT many |
| 19:50.18 | izaak | ManxPower: that's my thought as well. but iptables/tc makes me want to puke compared to pf for qos. i think i'm going to double the costs and go for two boxes. |
| 19:50.33 | nny_1 | I run asterisk on my commodore64 |
| 19:51.01 | ManxPower | izaak: for the most part you don't need a complicated firewall. |
| 19:51.15 | ManxPower | izaak: hell, you don't need any firewall really. |
| 19:51.27 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 19:51.31 | ManxPower | Just be sure to secure your Asterisk |
| 19:51.40 | Corydon76-dig | Nothing says you couldn't make a c64 into a softphone |
| 19:51.49 | izaak | ManxPower: for an office with a regular broadband internet connection, you need qos. |
| 19:51.56 | ManxPower | izaak: and QoS is handled by the ROUTER, not ASTERISK |
| 19:51.59 | [TK]D-Fender | _ShrikE: Pricey, and they imply char matrix screen use (bad). |
| 19:52.26 | izaak | ManxPower: yeah, but if only linux had something decent like pf, you could handle both tasks on the same embedded device, easy. |
| 19:52.39 | ManxPower | izaak: True QoS can't be done on a regular boradband connection, as to do true QoS you need to control BOTH sides of the link |
| 19:52.41 | a1fa | [TK]D-Fender : so meetme is the only app requiring timing |
| 19:52.55 | a1fa | and iax trunk |
| 19:53.22 | ManxPower | a1fa: do you understand what an IAX2 trunk is and how it is different from a regular IAX2 non-trunked connection? |
| 19:53.39 | izaak | thanks for confirming that asterisk on bsd is for freaks, though. |
| 19:53.58 | ManxPower | izaak: we like to call them lunatics, but freaks works too 8-) |
| 19:54.09 | [TK]D-Fender | _ShrikE: No G.723 (less important) |
| 19:54.14 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:54.20 | [TK]D-Fender | a1fa: READ... I just added to that |
| 19:54.28 | _ShrikE | I just cant help but want one though. |
| 19:54.39 | a1fa | scrolls up |
| 19:54.43 | izaak | actually, i've been running asterisk really well on openbsd for a year and a half in my own office. however disa applications have weird drop-outs :) |
| 19:55.50 | ManxPower | izaak: hardware is where you really have to worry about. |
| 19:57.21 | ManxPower | And just remember the OXford English Dictionary defines crazy as "someone that uses only VoIP for their business". |
| 19:57.24 | izaak | i don't implement systems with fxs/fxo pci cards. i find pap2ts and the like much easier and cheaper. |
| 19:57.46 | ManxPower | izaak: I never do any analog in Asterisk. |
| 19:57.48 | ManxPower | ALWAYS PRI |
| 19:58.51 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:59.42 | a1fa | [TK]D-Fender : so anything bad about running * on xen |
| 19:59.51 | [TK]D-Fender | a1fa: You tell me. |
| 19:59.56 | ManxPower | Systems I install: Generic Server Class PC, Cisco Catalyst switches, Sangoma 2-port T-1 card, PRI from telco, Polycom phones |
| 20:00.06 | a1fa | besides this timing everyone is freaking out, and hardware issues |
| 20:00.08 | a1fa | i dont see a problem |
| 20:00.20 | a1fa | i've had it running like that for 3 years |
| 20:00.55 | a1fa | it does SIP only |
| 20:01.34 | ManxPower | And that is the design I recommend to everyone, EXCEPT perhaps using a (modern) Digium card instead of a Sangoma |
| 20:01.48 | [TK]D-Fender | a1fa: Sorry... this isn't #validation .... that's 2 doors to the left :p |
| 20:02.21 | a1fa | ;) |
| 20:02.26 | a1fa | d-fender |
| 20:02.33 | nny_1 | hmmph |
| 20:02.38 | a1fa | DR. D-Fender |
| 20:02.45 | drako | im gonna downgrade |
| 20:02.55 | drako | from debian 64bits to 32bits |
| 20:03.08 | drako | can't figure why is asterisk just locking |
| 20:03.11 | a1fa | thanks for letting one know |
| 20:04.18 | a1fa | so anyone else dumb enough to run * in virtual enviroment? |
| 20:04.43 | nny_1 | crap |
| 20:04.50 | nny_1 | crap on a crap cracker |
| 20:04.59 | nny_1 | so HINT returns SIP/10 lol |
| 20:05.07 | nny_1 | so thats not my magic variable |
| 20:08.04 | *** join/#asterisk grEvenX (n=even@pc107-102.ktv.no) |
| 20:08.22 | a1fa | [TK]D-Fender |
| 20:08.43 | a1fa | i value your opinion, whats your opinion on virutal server |
| 20:09.24 | nny_1 | so if i want to initiate say chanspy on dialing an extension, is it ricky to do say * or # first |
| 20:09.27 | [TK]D-Fender | a1fa: Buy low, sell high! |
| 20:09.29 | nny_1 | risky* |
| 20:09.52 | nny_1 | so maybe 10 is extension 10 and #10 is chanpsy on ext 10 |
| 20:09.59 | [TK]D-Fender | a1fa: If you have it, it works, you're happy, it isn't incurring any expenses or unvalidated pain, then stick with it, who cares? |
| 20:10.34 | drako | nny_1, I use * for spy and ** for whisper |
| 20:10.40 | nny_1 | drako: nice |
| 20:10.51 | a1fa | [TK]D-Fender : =) |
| 20:10.52 | nny_1 | drako: that works, and easier than the magic i am trying to perform right now |
| 20:10.53 | a1fa | thanks |
| 20:11.30 | *** join/#asterisk MACscr (n=Mark@c-98-214-107-162.hsd1.il.comcast.net) |
| 20:12.16 | nny_1 | ManxPower: your original suggestion for a chanspy extension is aweome |
| 20:12.21 | nny_1 | awesome* |
| 20:12.24 | nny_1 | radical |
| 20:12.26 | nny_1 | gnarly |
| 20:12.28 | MACscr | is there an sip address I can use for testing sip to sip calls? Need to see if I can hear them ok, and if they can here me ok. Guess it would have to record and play back what I said to it |
| 20:12.28 | nny_1 | ty |
| 20:12.46 | [TK]D-Fender | MACscr: FWD |
| 20:14.52 | a1fa | any reason why my sound in IVR soudns cracky? |
| 20:15.30 | drako | nny_1, what was the ManxPower suggestion? |
| 20:16.19 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 20:18.10 | [TK]D-Fender | a1fa: there are a hundred. With the level of information you gave so far it could be any of them. Good luck. |
| 20:18.17 | ManxPower | (2:27:12 PM) ManxPower: exten => _*99XX,1,ChanSpy(SIP/${EXTEN:3}|W) |
| 20:18.24 | nny_1 | drako: exten => _*99XX,1,ChanSpy(SIP/${EXTEN:3}|W) |
| 20:18.49 | ManxPower | nny_1: check the docs for chanspy, maybe you can try to spy by default, then if that fails, send the call as normal |
| 20:18.56 | hardwire | damn I hate that SPA94x's have no backlight |
| 20:21.37 | x86 | amazon iz teh back! |
| 20:21.40 | nny_1 | ManxPower: damn fine idea |
| 20:22.15 | [TK]D-Fender | hardwire: SPA-942 <- |
| 20:22.16 | nny_1 | ill test it |
| 20:22.16 | ManxPower | nny_1: I don't know what happens if you try to spy on a channel that does not exist, check the docs for chanspy and try it. |
| 20:22.20 | *** join/#asterisk Wangster (n=Wangster@host-253.epicnet.ca) |
| 20:22.23 | nny_1 | ManxPower: will do |
| 20:22.49 | nny_1 | ManxPower: I think though, it just goe sinto chanspy mode, and lets you pick the xtensions with * etc |
| 20:23.19 | nny_1 | ManxPower: i modified the suggestion you gave me to exten => _*XX,1,ChanSpy(SIP/${EXTEN:1}|W) |
| 20:23.24 | nny_1 | they use 2 digit extensions |
| 20:23.32 | ManxPower | If you want to see some really gnarly dialplan stuff look at http://www.fnords.org/~eric/macro-std-exten-v2.inc and http://www.fnords.org/~eric/macro-dial-result-v2.inc |
| 20:24.01 | TrentCreek | [TK]D-Fender: My ATA still does not get voice...after forwarding all traffic to the " Box and changing IP addresses for everything..the sound quality has gotten better, but no voice going to in calls. |
| 20:24.03 | nny_1 | heh BARF |
| 20:24.09 | ManxPower | nny_1: check chanisavail as well, I think there's an option to say "act as unavail if ANY of the channels are in use" |
| 20:24.23 | Wangster | Does murf hang out here? |
| 20:24.41 | nny_1 | ManxPower: k |
| 20:25.11 | hardwire | [TK]D-Fender: seriously? it has a backlight? |
| 20:25.16 | *** join/#asterisk banzaika (n=banzaika@rrcs-24-136-116-90.nyc.biz.rr.com) |
| 20:25.28 | nny_1 | yeha 952 is pretty cool |
| 20:25.31 | hardwire | [TK]D-Fender: thats reason #9812 to order them instead of the 941's which I can't stand. |
| 20:25.41 | hardwire | we have like 30 of em now. |
| 20:25.42 | Wangster | I'm testing a branch for murf and I've fond a segfault. I thought here was a web page that showed the proper way to do a trace and gather the relevant output but I can't find it now. Can anyone give me pointers on that? |
| 20:25.59 | ManxPower | Wangster: backtrace.txt in the asterisk doc dir |
| 20:26.00 | *** join/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 20:26.07 | ManxPower | (or similar file name) |
| 20:26.13 | Wangster | ManxPower: thanks. I'll find it. |
| 20:26.16 | nny_1 | lol crap |
| 20:26.19 | nny_1 | no dtmf again |
| 20:26.24 | nny_1 | gah f this ITSP |
| 20:27.07 | nny_1 | just tried to send DTMF and it ignores it unless i mash a bunch of buttons |
| 20:27.11 | nny_1 | sigh |
| 20:27.15 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 20:27.45 | nny_1 | screw em... they had their chances, and their customer support sucks.. time to start hunting again |
| 20:28.12 | hardwire | nny_1: thats nothing |
| 20:28.29 | hardwire | one of our ITSP's turned off busydetect when trunking to PRI because it was messing with something non-trivial |
| 20:29.46 | nny_1 | chaos? |
| 20:29.51 | nny_1 | boom?! |
| 20:29.57 | WildPikachu | thanks ManxPower all is working for me :) |
| 20:30.19 | hardwire | meh.. no.. it just wouldn't return back if the ITSP was returning busy or congested |
| 20:30.39 | banzaika | ~cut |
| 20:31.16 | [TK]D-Fender | ~emo |
| 20:31.17 | jbot | I just cut myself :`( |
| 20:31.53 | [TK]D-Fender | BBIAB |
| 20:32.13 | nny_1 | ManxPower: your original exten entry worked great ty again |
| 20:33.16 | banzaika | anybody know why input keys would stop working in the macro ? |
| 20:34.36 | banzaika | silence ... |
| 20:35.02 | banzaika | answered call hits a macro with extensions like ( 0, 1 ,2 ) |
| 20:36.37 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 20:39.54 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 20:40.38 | MACscr | Im getting silence on SIP to SIP calls. Is it an allowed codec issue most likely? |
| 20:41.16 | deeperror | MACscr, are both sides of the call silent? |
| 20:41.44 | MACscr | deeperror: yes, I think so |
| 20:42.02 | deeperror | any errors |
| 20:42.09 | MACscr | could be because im behind nat though. But the issue only happens with SIP to SIP, doesn't happen with SIP to PTSN |
| 20:44.18 | *** join/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 20:44.19 | deeperror | MACscr, sip set debug |
| 20:45.13 | MACscr | deeperror: I have been doing that, but it seems like information overload it at escapes my ssh window before I can even read it |
| 20:46.20 | deeperror | modify logger.conf and you will be able to save that to /var/log/*/console |
| 20:46.37 | nny_1 | does retaining callerid on transfer work on normal system without using blind transfer? or is this the standard deal for most systems? |
| 20:46.46 | pigpen | Here is an odd question: anyone know of a "foot pedal" control for a dictation solution based on asterisk? |
| 20:47.12 | deeperror | pigpen, what would that do start and stop calls? |
| 20:47.26 | pigpen | start/stop/pause/record |
| 20:47.40 | nny_1 | um |
| 20:47.41 | pigpen | depending on which end. |
| 20:47.44 | deeperror | what type of input? |
| 20:47.46 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 20:48.22 | pigpen | dunno. I have thought about a dialplan that would starting recording on dtmf. |
| 20:48.24 | deeperror | nny_1, not sure |
| 20:48.40 | pigpen | thus, a simple dtmf solution on the handset might work. |
| 20:48.48 | pigpen | I'll dig around.... |
| 20:49.01 | pigpen | just thought by a stroke of luck someone has done it. |
| 20:49.23 | deeperror | it could probably be setup if you built one ha |
| 20:49.35 | Wangster | For some reason I can't get make menuselect to run (can't find curses). Where do I manually set "DONT_OPTIMIZE". |
| 20:49.42 | deeperror | i'm sure someone else would appreciate that |
| 20:49.58 | nny_1 | any SPA962 users know if you can move the transfer buttons etc? |
| 20:50.01 | pigpen | yeah, dictation systems are very expensive. |
| 20:50.05 | deeperror | Wangster, have you tried to install curses? |
| 20:50.12 | Wangster | Makefile.rules OPTIMIZE?=-O6 ? What should that be changed to so it is like "DONT_OPTIMIZE" ? |
| 20:50.17 | Wangster | deeperror: I have curses |
| 20:50.34 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 20:50.41 | deeperror | did you just install it? |
| 20:51.03 | Wangster | > rpm -q ncurses ncurses-5.6-41 |
| 20:51.07 | *** join/#asterisk s0lid (n=s0lid@58.69.1.79) |
| 20:51.11 | deeperror | Wangster, may need to recompile asterisk with ./configure |
| 20:51.20 | Wangster | deeperror: not really. But I have done make clean and redone ./configure. |
| 20:52.17 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-8-rrdg-esr-2.dynamic.isadsl.co.za) |
| 20:52.54 | deeperror | make and make install ? |
| 20:53.50 | Wangster | deeperror: make && make install run fine. I just can run make menuselect |
| 20:53.59 | Wangster | *** Install ncurses to use the menu interface! *** |
| 20:54.23 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:54.26 | Wangster | I also have ncurses-devel-5.6-41 |
| 20:54.50 | deeperror | i'm not sure |
| 20:55.17 | Wangster | deeperror: in any case I do definately have curses because other things that require it run fine like yast (this is an opensuse box) |
| 20:55.54 | Wangster | I don't really care about the menuselect but I do need to set DONT_OPTIMIZE |
| 20:55.58 | *** join/#asterisk mascool (n=george@71.65.5.171) |
| 20:56.33 | mascool | does anyone know how I can re-send someone to a context, priority after a call has been parked 3 times? I get a "tires of being parked" and then a busy signal |
| 20:56.51 | mascool | I checked features.conf but couldn't find anything |
| 20:57.40 | mascool | I've also tried goto s-${DIALSTATUS} and defined s-busy, s-noanaswer, s-congestion |
| 20:57.56 | mascool | all with the same result |
| 20:58.20 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:58.31 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:59.09 | *** join/#asterisk jpeeler (n=jpeeler@adsl-065-005-230-151.sip.lft.bellsouth.net) |
| 21:02.28 | mascool | anyone? |
| 21:05.18 | *** join/#asterisk talntid (n=erict@66.208.251.170) |
| 21:06.17 | lmadsen | it would be s-BUSY, fyi |
| 21:06.36 | mascool | yeah sorry for the lowercase |
| 21:06.59 | lmadsen | you could probably setup a channel variable as a counter, then increase that number each time you park the call, then when you go to park it again, check the variable, and do something different when parking |
| 21:08.15 | mascool | sounds good but I would need to access the variable that lets asterisk know when a call has been parked too many times |
| 21:08.24 | ix33 | ManxPower here? |
| 21:12.04 | ix33 | any * consultants here push ABE, or is o/s * 1.4 preferred? |
| 21:12.09 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 21:12.31 | exothermc | Once I have ran make menuselect once, how do I get the same options in the next build of asterisk I do? |
| 21:12.42 | exothermc | without having to run through the whole menu again? |
| 21:13.56 | ix33 | exothermc: er oldconfig? |
| 21:14.35 | exothermc | ix33: and that would be? |
| 21:14.59 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 21:15.47 | ix33 | nevermind i was wrong |
| 21:21.46 | ManxPower | mascool: try defining s-BUSY |
| 21:21.57 | mascool | ManxPower, I've tried that |
| 21:22.05 | mascool | same result |
| 21:22.11 | ManxPower | and you would have to Goto(s-${DIALSTATUS},1) Always need the priority |
| 21:22.22 | mascool | yep have that too |
| 21:22.46 | ManxPower | mascool: look at the ACTUAL extension it's trying to go to, then define it, see the CLI |
| 21:23.19 | mascool | i've looked but it's not trying to go anywhere |
| 21:23.22 | mascool | http://pastebin.ca/1040760 - dialplan |
| 21:24.06 | MACscr | in the past, I have had my phone system at a DC, so it wasn't behind NAT. Im putting together a new one at home office until I take it to the DC in a couple weeks. I only have one extension hooked up to it which is on the same LAN as the server. I can make calls and receive calls fine with my SIP2PTSN provider, but I can't make sip to sip calls (well I can, just can't hear anything and they can't hear me). Is it most likely a firewall |
| 21:24.27 | mascool | http://pastebin.ca/1040763 - debug |
| 21:25.29 | mascool | in the debug it show it's trying to announce the parking spot and it fails because I sent it to SIP/110 which does not exists, did that in purpose because I don't want it to announce the spot, I have hints to parking spots defined on all phones |
| 21:26.25 | mascool | and it works, it parks the call and waits 20 seconds, but it only does that 3 times then it hangs up because it "got tired of being parked" |
| 21:27.00 | ix33 | ManxPower: what class server would you spec for a 30-sip phone + 1 PRI install? |
| 21:28.39 | Wangster | ix33: you don't need anything special for that setup but I always recommend getting server-class equipment for phone systems since most people expect robust uptimes. |
| 21:28.42 | *** join/#asterisk savaticus (n=savaticu@sta-206-168-231-35.rockynet.com) |
| 21:29.05 | Wangster | ix33: so for example an HP DL360 or ML350 works nicely. |
| 21:29.35 | ix33 | ah. just wondering |
| 21:30.12 | ix33 | i agreed on a dell t300 with raid5 and UPS. he wouldn't spring for 2 PSU's |
| 21:30.28 | MACscr | why in the world use raid5? use raid1 |
| 21:30.35 | andrewy | faster |
| 21:30.39 | ix33 | seems like he coul dhave gotten a DL360 for what he paid though. |
| 21:30.41 | MACscr | lol, raid5 is slower |
| 21:30.48 | andrewy | ...than raid 1? |
| 21:30.48 | ix33 | raid5 in h/w, excuse me |
| 21:30.53 | ix33 | he paid for the controller, so... |
| 21:32.05 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:32.21 | Wangster | Dell is Dell.... It's bottom end equipment but it will probably work just fine. |
| 21:32.39 | Wangster | like I said you don't need any special horsepower for a basic setup. |
| 21:33.17 | Wangster | Though the odd motherboard has timing issues with the telephony cards I don't believe there is a "known bad" list of them anywhere. |
| 21:35.12 | [TK]D-Fender | Sure there is... its on digium.com under "Compatibility" |
| 21:35.27 | ix33 | hmm.. readlly? |
| 21:36.14 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:36.15 | [TK]D-Fender | http://www.digium.com/en/supportcenter/documentation/viewdocs/TE405P |
| 21:36.26 | [TK]D-Fender | Read the BIG PRINT people.... |
| 21:37.04 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:38.49 | ix33 | that list hasn't been updated in > 2 years |
| 21:39.23 | ix33 | and it's not linked off of the single span card doc page either |
| 21:39.28 | ix33 | guess i lucked out |
| 21:39.34 | *** join/#asterisk grEvenX (n=even@pc107-102.ktv.no) |
| 21:39.53 | [TK]D-Fender | Lesson is to read in detail. |
| 21:40.01 | ix33 | here's another hindsight question: anyone using NetGear POE equipment? |
| 21:40.10 | ix33 | ducks behind desk |
| 21:40.15 | [TK]D-Fender | Zaptel is also better under 1.4 that it was under 1.2 and that has helped a lot of cases |
| 21:40.41 | [TK]D-Fender | ix33, I have 4 of their 4-port PoE switches. They work OK, but for larger I'd sooner go D-Link / Linksys |
| 21:41.15 | ix33 | i've done this install using one 48- and one 24-port netgear L3 POE |
| 21:41.25 | ix33 | may god have mercy on my soul |
| 21:42.37 | ManxPower | GOD HAS NO MERCY FOR PEOPLE THAT USE NETHEAR |
| 21:42.43 | ManxPower | and NETGEAR too. |
| 21:42.52 | ix33 | this is my fear |
| 21:43.19 | ix33 | i tried to get him to buy adtran, but it was kind of a hassle. |
| 21:43.24 | ix33 | kinda silly since we live in HSV |
| 21:45.42 | [TK]D-Fender | ix33, is it that much cheaper just for being there? |
| 21:45.52 | lmadsen | I doubt it |
| 21:46.23 | ix33 | no. i'm just saying they're right across the street from us but going through a local distributor is like pulling teeth |
| 21:46.27 | ix33 | they all universally suck. |
| 21:50.11 | ManxPower | When I've ever needed PoE I went with the switch I wanted (which ends up being non-poe) then adding a PoE injector. |
| 21:53.00 | izaak | what's wrong with netgear? |
| 21:53.30 | hsv-al | heh |
| 21:53.59 | izaak | ? |
| 21:54.21 | *** join/#asterisk fedya (n=fedya@183.12.119.70.cfl.res.rr.com) |
| 21:57.17 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 21:57.24 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 21:58.07 | *** join/#asterisk MaartenB (n=Maarten@84-105-197-212.cable.quicknet.nl) |
| 21:59.24 | MaartenB | anyone knows what "rc_avpair_new: unknown attribute 1490026597" means? |
| 22:00.16 | *** join/#asterisk javawizard2539 (n=javawiza@c-76-23-28-244.hsd1.ut.comcast.net) |
| 22:03.55 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 22:06.44 | mascool | what are ADSI parking announcements ? |
| 22:08.09 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 22:09.36 | ix33 | oooh. now that was a f*ckup. |
| 22:09.57 | *** join/#asterisk phillipk (i=d8f88f57@gateway/web/ajax/mibbit.com/x-664579beaaeee62d) |
| 22:09.59 | ix33 | i installed debian-686 on a xeon. bollocks. |
| 22:10.23 | ix33 | i guess it doesn't matter if i'm < 2GB RAM, right? |
| 22:10.35 | *** join/#asterisk CrashSys (n=kumba@216-199-37-76.tpa.fdn.com) |
| 22:11.22 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
| 22:11.43 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:12.46 | Shotygun | Hello. Could somebody please explain to me in briefly what is the meaning of "no user response" ? |
| 22:13.33 | lmadsen | ix33: no, RAM doesn't matter that much in Asterisk -- CPU power is more important |
| 22:13.59 | lmadsen | once the modules are loaded into memory, not much more RAM is used to process calls -- its all CPU at that point |
| 22:14.15 | hardwire | Shotygun: how did that message occur? |
| 22:15.05 | Shotygun | hardwire: It received in the softphones of agents during a dial-out, only for a certain period. I couldn't find anything unusual at the cdr and trying to figure out the meaning of this |
| 22:15.28 | Shotygun | They claim the occured every couple of minutes during a hour period |
| 22:15.36 | [TK]D-Fender | mascool, * advertising to ADSI phones that calls were left parked |
| 22:16.02 | ix33 | lmadsen: i noticed. uptime is 8 days live and only 650M used reported by free(1) |
| 22:16.06 | mascool | [TK]D-Fender, thanks |
| 22:17.38 | Shotygun | Does it mean no peer is reachable? |
| 22:23.58 | [TK]D-Fender | Shotygun, it means we need some context as to WHHERE you got this message. APSTEBIN is your friend. |
| 22:23.59 | [TK]D-Fender | ~pb |
| 22:24.00 | 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 |
| 22:27.32 | Shotygun | Unfortunately nothing to paste. The agents received it at the softphone ("Zoiper"/Idefisk) screen itself, which occuring to my readings, the softphone received it from the SIP channel. The context is a simple outbound call context between two peers (in order, one is fallback to another). |
| 22:29.06 | [TK]D-Fender | Shotygun, I don't mean "context" in terms of dialplan or SIP peer definition. We need to see the CLI OUTPUT of your call to see exactly what * tried to do to cause the error |
| 22:31.36 | Shotygun | I understand, which I do not have. Next time I guess.. |
| 22:31.43 | Shotygun | Thanks tho |
| 22:32.52 | Shotygun | I found it in messages: |
| 22:33.50 | Shotygun | Ok I got the answer. It makes sense that it's a result of load average block I guess |
| 22:36.41 | *** join/#asterisk km2 (n=x@c-24-23-255-173.hsd1.mn.comcast.net) |
| 22:38.16 | Shotygun | Could this be a result of call limit? |
| 22:40.37 | [TK]D-Fender | Shotygun, thats a generic message that can mean a lot of things. If you don't have CLI output, come back later when you do. |
| 22:41.16 | Shotygun | I have 'messages' output if that will do.. |
| 22:43.15 | [TK]D-Fender | Shotygun, No. |
| 23:04.00 | *** join/#asterisk jhein (n=Jeremy@adsl-70-143-99-179.dsl.pltn13.sbcglobal.net) |
| 23:05.34 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:07.40 | *** part/#asterisk jhein (n=Jeremy@adsl-70-143-99-179.dsl.pltn13.sbcglobal.net) |
| 23:17.28 | ix33 | ManxPower: what would it take to get one of your echo cancellers? |
| 23:19.11 | hardwire | ix33: your jaws not wide enough. |
| 23:20.03 | ix33 | ? |
| 23:20.30 | [TK]D-Fender | .... |
| 23:20.49 | [TK]D-Fender | hardwire, very inappropriate... |
| 23:22.01 | hardwire | [TK]D-Fender: it's not happy hour yet, eh? |
| 23:22.53 | ix33 | i have compiled zaptel, explicitly commenting out the #define VPM_SUPPORT and am running with HPEC and i still have this clipping issue. |
| 23:23.36 | hardwire | ix33: is it zap to local network or other zap? |
| 23:24.18 | ix33 | SIP->*->ZAP->PSTN |
| 23:24.22 | ix33 | is that whan you mean? |
| 23:24.28 | ix33 | *what |
| 23:24.34 | hardwire | the sip is on the local network? |
| 23:24.38 | ix33 | yes. |
| 23:24.46 | ix33 | SIP to SIP works flawlessly. |
| 23:24.48 | [TK]D-Fender | ix33, Do you get clipping even without EC? |
| 23:24.58 | ix33 | when echocancel=no, no clipping. |
| 23:25.12 | hardwire | ix33: where are you? |
| 23:25.16 | ix33 | hsv, al |
| 23:25.31 | [TK]D-Fender | ix33, Well you coul have a high-gain link at which point you could lower it a bit first. |
| 23:25.44 | hardwire | ix33: I had to do that on a local PRI |
| 23:25.48 | hardwire | that's why I asked |
| 23:25.52 | [TK]D-Fender | ix33, And then of course you could try OSLEC & Zaptel's other EC's |
| 23:25.54 | hardwire | my telco was too hot. |
| 23:26.01 | ix33 | hmm. |
| 23:26.08 | ix33 | gain settings in zapata.conf? |
| 23:26.19 | [TK]D-Fender | ix33, yup |
| 23:26.34 | hardwire | yar |
| 23:26.34 | ix33 | hardwire: you HSV also? |
| 23:26.44 | hardwire | ix33: no anc, ak |
| 23:27.12 | ix33 | so on my end i would be setting rxgain? |
| 23:27.21 | hardwire | setting it down probably |
| 23:27.25 | hardwire | you can test using ztmonitor |
| 23:27.25 | ix33 | ~book |
| 23:27.26 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:27.45 | hardwire | I did milliwat loopback tests and found a nice happy medium |
| 23:27.55 | hardwire | but I hate that you have to adjust anything with PRI |
| 23:27.57 | hardwire | it freaks me out |
| 23:28.07 | hardwire | a channel shouldn't be *hot* you know? |
| 23:28.08 | ix33 | hardwire: me also |
| 23:28.19 | hardwire | it should be happy, digital, and maybe luke warm.. never hot. |
| 23:28.58 | ix33 | well while i have wcte12 butchered, lemme try the other EC for fun |
| 23:29.38 | hardwire | I have yet to play with a board that has echocan on it |
| 23:29.44 | hardwire | I'm gonna be all excited when I get the chance. |
| 23:37.48 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 23:37.59 | iceyp | hey guys... anyone know what causes outbound DTMF to be distorted? |
| 23:38.18 | ix33 | doing it inband on a bad channel? |
| 23:38.22 | iceyp | has only started since the change from 1.2 to 1.4, also different hardware though I dont know if this would cause it |
| 23:38.33 | *** join/#asterisk _MrSeb_ (n=SebaX@87.253.113.240) |
| 23:38.34 | _MrSeb_ | Hi to all |
| 23:38.43 | iceyp | ix33 it's over a sip channel, tried both ulaw & g729 codec |
| 23:38.57 | iceyp | also tried multiple sip trunks |
| 23:39.11 | iceyp | if i go back to my old 1.2 server it seems ok |
| 23:39.14 | ix33 | purely SIP channel? |
| 23:47.16 | ix33 | shitfire why don't i just load up a bigass box with x100p's... |
| 23:52.53 | [TK]D-Fender | lol |
| 23:56.05 | *** join/#asterisk jack_sparo (n=eddy@pptp04.witopia.net) |
| 23:57.31 | jack_sparo | i am sending one DID to my box and i have created the incoming route for it, but when i call it, i see the call coming and the DID is show, but it says that the number you have dialed is not in service, any idea guys? |
| 23:58.22 | ix33 | what does * say on the CLI? |
| 23:59.28 | iceyp | ix33 yeh just sip, sorry someone was at the door |
| 23:59.35 | ix33 | np |
