| 00:00.52 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:04.32 | eric2 | least cost routing + real time... what's the best way to do this? in the dial plan? |
| 00:05.15 | eric2 | I have x long distance providers, based on the number dialed, I want to compare all the rates from the providers and then dial through the one with the lowest rate |
| 00:06.10 | ac1djazz | file; how? :-D |
| 00:07.05 | dkwiebe | eric2: It could be done with sql queries from the dialplan but it would probably most often be done with agi scripts or fastagi scripts |
| 00:07.36 | eric2 | damn, now I have to learn agi stuff... hmm |
| 00:08.38 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:08.40 | dkwiebe | eric2: unless you use something prebuilt |
| 00:09.44 | eric2 | nah, I'd rather write it myself |
| 00:10.51 | dkwiebe | K, if you're using perl I can give you a hand sometime if you'd like it. It all depends how fancy you want to get. Basic LCR is fairly straight forward. |
| 00:11.13 | eric2 | I never did learn perl |
| 00:11.20 | dkwiebe | :) |
| 00:11.21 | *** join/#asterisk Trapa (n=no@207.230.238.90) |
| 00:11.23 | jjshoe | you can write agi's in many languages. |
| 00:11.29 | eric2 | java? |
| 00:11.30 | jjshoe | or you can pay someone in here to do it for you too I'm sure. |
| 00:11.32 | jjshoe | eric2 yes. |
| 00:11.37 | eric2 | ok, java it is!! |
| 00:11.40 | eric2 | :) |
| 00:11.44 | eric2 | java's a piece of cake |
| 00:11.46 | dkwiebe | or you can do it in C and make an asterisk module. |
| 00:11.52 | dkwiebe | I think that's what we did with ours. |
| 00:12.28 | Trapa | Hello, i'm having some trouble setting up my asterisk server, and i'm looking for somone / some people to help me troubleshoot the problem. Would this be the right place to come or is there a better location for that? |
| 00:16.11 | Dovid | Trapa: We can try to help. what issue are you having ? |
| 00:17.05 | Trapa | David: Well, first, i'm a super asterisk newbie, so i haven't done much to the default config provided by debian. We can't make oubound calls, but i'm not too concerned about that until we can make internal calls .. which we can only sort-of do. |
| 00:17.22 | Dovid | did you read the book ? |
| 00:17.23 | Trapa | We make the call .. and the other person accepts it , but no audio is transmitted |
| 00:17.25 | Dovid | ~book |
| 00:17.25 | 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 |
| 00:17.36 | Dovid | probably NAT |
| 00:17.40 | Dovid | ~nat |
| 00:17.41 | jbot | i heard nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 00:17.42 | Trapa | Yeah i've read some of the book. BUt not all. |
| 00:17.50 | Dovid | read the book first |
| 00:17.53 | Dovid | it should help you a lot |
| 00:18.10 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 00:18.11 | Trapa | But the internernal extensions, and not natted, it's machine -- server -- machine all on the same ip subnet (within the nat) |
| 00:18.37 | Dovid | true but if you have bad nat settings it will afect the calls even internally. |
| 00:18.42 | Dovid | also did u set up the codecs ? |
| 00:19.15 | Trapa | No .. i've only edited sip.conf and extensions.conf as per many tutorials that i have found. |
| 00:19.35 | [hC] | are enum lookups to e164.org broken? |
| 00:19.38 | Dovid | sip.conf would have the settings for the clodecs |
| 00:19.44 | Dovid | can you pb your sip.conf ? |
| 00:19.45 | Dovid | ~pb |
| 00:19.45 | 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 |
| 00:20.05 | Trapa | sure |
| 00:23.29 | [hC] | Anyone here using e164.org for enum lookups? It seems that the enumlookup function is busted for e164.org's returned data. |
| 00:25.41 | Trapa | sighs.. I hate vista so very very much ... Give me a minute so i can download a editor that understand n |
| 00:27.15 | Trapa | http://pastebin.ca/996521 is sip.conf |
| 00:27.56 | Trapa | http://pastebin.ca/996522 is extensions.conf |
| 00:30.50 | *** join/#asterisk riddlebox (n=james@ppp-70-250-54-184.dsl.stlsmo.swbell.net) |
| 00:31.05 | ac1djazz | exten => s,1,Read(dialed,conf-getconfno) |
| 00:31.06 | ac1djazz | exten => s,2,SayDigits(${dialed}) |
| 00:31.09 | ac1djazz | why dont i hear the digits dialed? |
| 00:37.35 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 00:39.41 | ManxPower | ac1djazz: standard troubleshooting, put "noop(dialed is ${dialed}) as priority 2 and make the current priority 2 be priority 3. See why we use "n" priorities now? |
| 00:40.05 | ac1djazz | i changed to n |
| 00:40.05 | ac1djazz | ok |
| 00:40.15 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 00:40.33 | ac1djazz | exten => s,1,Read(dialed,conf-getconfno) |
| 00:40.34 | ac1djazz | exten => s,n,noop(dialed is ${dialed}) |
| 00:40.34 | ac1djazz | exten => s,n,SayDigits(${dialed}) |
| 00:40.34 | ac1djazz | exten => h,1,hangup() |
| 00:40.36 | ManxPower | if it says "dialed is " in the CLi then your READ failed |
| 00:40.43 | ac1djazz | ok |
| 00:40.51 | ac1djazz | i think its working cuz i see -- User entered '1888' |
| 00:41.00 | ManxPower | In this specific instance you could see if it failed by looking at the CLI output for SayDigits |
| 00:41.27 | ac1djazz | oh ok sweet |
| 00:42.34 | Dovid | Trapa |
| 00:42.49 | Dovid | under the general section take out the comments for dissalllow=all |
| 00:42.54 | Dovid | and allow=ulaw |
| 00:42.58 | Dovid | reload ansterisk and try again |
| 00:43.04 | ManxPower | Most of the time I'm troubleshooting IF statements, etc so I would only see the evaluated output (0 or 1) rather than the text, so I woul dneed a noop |
| 00:43.09 | Dovid | brb |
| 00:43.14 | ManxPower | Dovid: it's best if you spell it right |
| 00:43.31 | Dovid | Manx: You have a point. I am a tad tired |
| 00:43.38 | *** join/#asterisk gerphimum (n=trekkie@cpe-24-167-100-36.satx.res.rr.com) |
| 00:44.07 | *** join/#asterisk justdave_ (n=dave@unaffiliated/justdave) |
| 00:44.22 | Dovid | Trapa: Uncoment these 2 lines (take out the ; from in the begining of the line) and reload asterisk |
| 00:44.22 | Dovid | # |
| 00:44.22 | Dovid | ;disallow=all ; First disallow all codecs |
| 00:44.22 | Dovid | # |
| 00:44.22 | Dovid | ;allow=ulaw ; Allow codecs in order of preference |
| 00:45.07 | ac1djazz | does this look right exten => s,n,GoToIf(${dialed} > 1000 && ${dialed} < 9000, enterconf,s,1) |
| 00:46.22 | Trapa | tries that. |
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| 00:49.50 | Trapa | Dovid: Okay so tried that,.. now i notice that The thing i noice is that both softphones seem to be jumpin at the out: in: |
| 00:50.15 | Trapa | pcmu seems to be going on and off .. i thought that was maybe because it tries not to transmit silence .. but i'm no expert |
| 00:53.52 | Trapa | Dovid: I'll work some more on this and come back again if i am still haveing problems |
| 00:53.58 | ac1djazz | [Apr 24 17:32:35] WARNING[31814]: pbx.c:1832 pbx_extension_helper: No application 'Play' for extension (enterconf, s, 1) |
| 00:54.08 | Trapa | Thankyou very much for your assistance. |
| 00:54.30 | [hC] | Doesnt _X. also match just a single digit? (in a dial plan) |
| 00:55.58 | Dovid | # |
| 00:55.58 | Dovid | ;disallow=all ; First disallow all codecs |
| 00:55.58 | Dovid | # |
| 00:55.58 | Dovid | ;allow=ulaw ; Allow codecs in order of preference |
| 00:56.11 | plik | deja vu |
| 00:56.14 | Dovid | Trapa: Try Playback or Background |
| 00:56.23 | Dovid | play is not a calid command |
| 00:56.37 | Dovid | vaild* |
| 00:56.55 | Dovid | http://www.voip-info.org/wiki/view/Asterisk+cmd+Playback |
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| 00:57.14 | Dovid | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Background |
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| 01:03.36 | Dovid | hC: try it |
| 01:03.39 | Dovid | i dont think so |
| 01:03.44 | Dovid | for one digit do _X, |
| 01:04.12 | [hC] | Huh. Yeah i was asking because _X. wasnt matching one digit. I'd like to have a catch all that matches 1 or more digits, but of course _. is crazy. |
| 01:04.30 | Dovid | _. will make issues |
| 01:04.37 | [hC] | Of course. |
| 01:04.40 | Dovid | so do _X, and _X., |
| 01:04.51 | [hC] | I think i saw something a few days ago that provided a single exten for 1 or more digits, but i cant recall where. |
| 01:05.11 | Dovid | let me know what it is. |
| 01:05.17 | Dovid | i can use it too ;) |
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| 01:34.56 | [hC] | wtf, ftp upload from the digium appliance reboots the box |
| 01:34.57 | [hC] | sigh |
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| 01:58.35 | Kyoshi | Ast Real Time with MSSQL help please... http://www.pastebin.ca/996276 |
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| 02:12.47 | mgroman | Kyoshi, Do you know junshen2004? |
| 02:14.15 | *** join/#asterisk killmel8tr (n=IceChat7@ip-198-22-65-131.quickconnectusa.com) |
| 02:16.18 | killmel8tr | Hi, I just installed a TC400B and have the latest zaptel and asterisk compiled and it works just dandy but then asterisk sort of just hangs... some commands still work in CLI but it doesnt accept any new connections at all and the channels that were open appear to stick open... anyone have any ideas on this? Its in this state now and I was wondering what I could check before restarting it..... |
| 02:17.04 | mgroman | killmel8tr, Whatever you do, just do it before i kill you later |
| 02:17.10 | killmel8tr | :) |
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| 02:24.44 | [hC] | Qwell: ping? |
| 02:30.13 | Kyoshi | <mgroman> Kyoshi, Do you know junshen2004?, |
| 02:30.14 | Kyoshi | nopes |
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| 02:53.17 | lmadsen | JayTee52: can you file that as an error to errata@oreilly.com ? |
| 02:53.35 | JayTee52 | the part about the IAX2 stuff? |
| 02:53.43 | lmadsen | yah |
| 02:53.47 | JayTee52 | sure |
| 02:53.54 | lmadsen | I think someone else might have pointed that out to me too |
| 02:53.58 | lmadsen | but I had forgotten |
| 02:54.07 | JayTee52 | funny, I've seen you in here lots but never made the connection till I read the cover today |
| 02:54.11 | lmadsen | I think it's a copy/paste error |
| 02:54.14 | lmadsen | :) |
| 02:54.23 | JayTee52 | still a great book though. |
| 02:54.24 | lmadsen | ya, I try not to brag (too much) hehe |
| 02:54.30 | lmadsen | thank you! |
| 02:54.53 | JayTee52 | I have the print version cuz 1) I hate trees and 2) reading on the "throne" is intellectually stimulating :-) |
| 02:55.35 | JayTee52 | and I'm just kidding about #1 |
| 02:56.36 | JayTee52 | I just installed 1.6.0-beta8 today on a test box to mess with. |
| 02:57.59 | *** join/#asterisk BeeBuu (n=beebuu@219.132.189.63) |
| 02:59.21 | lmadsen | nice! |
| 02:59.38 | lmadsen | ya, that's the thing about books.. no one likes to actually read it online... they only use it for the search functionality |
| 03:00.06 | JayTee52 | and you can't annotate it or add bookmark tabs |
| 03:00.06 | lmadsen | anyone who only reads the online version probably wouldn't have bought the printed version anyways, or those who do, may not have if they hadn't read the online version |
| 03:00.17 | lmadsen | so I figure releasing it to a PDF actually increases sales |
| 03:01.55 | JayTee52 | I'm still pissed at O'Reilly because they haven't given me any indication when T1: A Survival Guide will be back in stock and because they advertised a book called Asterisk Cookbook that was supposed to be out in August of 07 but still isn't in print yet. |
| 03:02.49 | JayTee52 | I bought Switching to VOIP when I ordered *, TFOT 2nd Ed. That's a pretty good book |
| 03:03.38 | [hC] | goes crazy |
| 03:03.46 | [hC] | asterisk appliance has filesystem problems |
| 03:03.49 | [hC] | yet... NO FSCK!! |
| 03:04.26 | JayTee52 | did you try doing touch /forcefsck ? |
| 03:04.38 | [hC] | well... that wont do much if there's no fsck binary :) |
| 03:04.53 | JayTee52 | oh, yeah. that's odd |
| 03:05.09 | JayTee52 | it's an embedded system? |
| 03:05.15 | [hC] | yeah, digium's appliance |
| 03:05.27 | [hC] | i got e2fsck on there, but its claiming that there are features that it cant read |
| 03:05.28 | JayTee52 | time to get on the phone |
| 03:05.36 | [hC] | it was ext3, of course. |
| 03:05.41 | [hC] | digium support is closed. :) |
| 03:07.06 | JayTee52 | lmadsen want to hear about the project I'm doing with Asterisk at my work? |
| 03:07.40 | JayTee52 | I think it would be good subject matter for an article |
| 03:08.23 | killmel8tr | anyone using the tc400b? |
| 03:09.15 | JayTee52 | nope |
| 03:09.33 | JayTee52 | why are you using it? |
| 03:10.03 | killmel8tr | capacity issue really.... its really nice to just let the hardware handle the transcoding |
| 03:10.29 | JayTee52 | ah, how many phones? |
| 03:10.32 | killmel8tr | problem is, asterisk sort of randomly just stops accepting connections after a couple hundred calls. |
| 03:10.57 | JayTee52 | couple hundred simultaneous or over time? |
| 03:11.03 | killmel8tr | I use it against a genband m6 for recording and monitoring for call centers.... its got about 80 calls on it at any given time |
| 03:11.15 | killmel8tr | couple hundred over time |
| 03:11.29 | JayTee52 | on a single server? |
| 03:12.07 | killmel8tr | I guess I should say 40, because its 2 legs for each call... |
| 03:12.11 | killmel8tr | ya on 1 box |
| 03:12.19 | killmel8tr | cpu never goes over 30% |
| 03:12.25 | JayTee52 | have you monitored memory usage? |
| 03:12.45 | killmel8tr | no I havent, guess maybe I should, eh? |
| 03:13.00 | JayTee52 | yeah, sounds like something is bleeding ram and not giving it back |
| 03:13.35 | JayTee52 | you might try loading a Nagios agent and throwing Nagios on another system to monitor it remotely. |
| 03:13.56 | killmel8tr | it sucks cause I was fine at g711 but I decided I wanted to give people more bandwidth on thier Ts |
| 03:14.03 | killmel8tr | hence the 729 card. |
| 03:14.38 | killmel8tr | I'll call support tomorrow |
| 03:15.21 | killmel8tr | chanspy was even working fine with monitor |
| 03:15.48 | killmel8tr | which are 2 things i have heard are bad to use together |
| 03:15.55 | JayTee52 | haven't messed with chanspy yet |
| 03:16.10 | killmel8tr | the whisper functionality is cool |
| 03:18.14 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:18.22 | JayTee52 | yeah, handy in call centers. I used to manage a large PBX in a 200 person call center with a Nortel Option 61C using Link and CCR with two IBM RS6000 AIX servers running the CTI application. |
| 03:18.34 | killmel8tr | anyone think there is any point in setting up a spoofcard competitor (like for 5 cents a minute vs. thier 16)? Or is there even a market for that? |
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| 03:20.50 | killmel8tr | I mean it is kinda shady... |
| 03:21.46 | JayTee52 | can't imagine the market for that is that big |
| 03:22.25 | JayTee52 | but there is an abundance of unscrupulous telemarketers out there so I could be wrong |
| 03:23.06 | killmel8tr | I have a customer that does mystery shopping for like gm and banks and other stuff.... they do phone shopping. They were using spoof card and I am saving them a ton |
| 03:23.59 | killmel8tr | its a tiny market though i am sure |
| 03:30.37 | lmadsen | JayTee52: ya sorry about that... I was supposed to write that book with jim and a couple others too, but books don't pay well, so paying customers seem to have taken precident |
| 03:30.39 | vitasoy | is there a black list of phone numbers that are known to be phone spammers? |
| 03:30.55 | lmadsen | however, I've been thinking about the cookbook again, and I want to come up with a good outline so I can start working on that again |
| 03:31.01 | vitasoy | analogous to spam blocking lists? |
| 03:31.35 | killmel8tr | I dont think it would be very effective. |
| 03:31.45 | killmel8tr | kinda like blocking email addresses |
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| 03:31.52 | vitasoy | it could be if enough people contributed |
| 03:32.03 | vitasoy | well realistically yes - would be difficult |
| 03:32.14 | JayTee52 | lmadsen, I'm going to be integrating a Nortel Option 11C and and Asterisk server using two Digium TE212P cards and Asterisk is going to perform as a "Man in the middle" voip call router. |
| 03:32.17 | killmel8tr | i mean, most voice spammers just make up random phone numbers |
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| 03:35.11 | Kyoshi | Ast Real Time with MSSQL help please... http://www.pastebin.ca/996276 |
| 03:41.09 | lmadsen | Kyoshi: anytime I've run into that it was because of a column type being set wrong, or a missing column (or misnamed column) |
| 03:42.04 | lmadsen | especially if your CDR works, which means that your connection is probably fine, but has something to do with the table layout |
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| 04:01.53 | Kyoshi | lmadsen, how can i find out if there are no errors? |
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| 04:07.00 | jameswf-home | if there are no errors your verbosity is too low |
| 04:07.18 | jameswf-home | check your logger.conf as well |
| 04:07.26 | jameswf-home | all else fails DEBUG |
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| 04:26.14 | Kyoshi | verbositry is 23 |
| 04:26.32 | Kyoshi | serously i worked this over very carefully |
| 04:27.56 | killmel8tr | ok, I have tracked my tc400b issue down to a "Bad EIP value".... Could this be an interrupt thing? |
| 04:28.55 | Kyoshi | looks like the problem is that asterisk is using sp_prepare where that has been obsoleted in mssql 2k5 |
| 04:29.35 | Kyoshi | as well as sp_execute has been obsoleted by sp_executesql and sp_prepare caches and sk5 doesnt perform cached dynamic sp's |
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| 04:50.53 | vitasoy | YES.. i can dial out. |
| 04:51.28 | vitasoy | though, now that i look back, i'm really only demonstrating that i can read and follow directions |
| 04:51.53 | [TK]D-Fender | vitasoy, congratulations on your newfound mediocrity :) |
| 04:52.43 | vitasoy | it was a fun battle. I'm quite impatient and I ran into a few problems because I was skimming around |
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| 04:59.53 | drmessano | You mean, half assing doesnt always work? |
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| 05:09.32 | vitasoy | half assing is okay as long as you go back and clean up |
| 05:12.50 | C4away | so then |
| 05:13.03 | drmessano | lol |
| 05:13.07 | drmessano | No, half assing is never ok |
| 05:13.11 | [TK]D-Fender | Like they say "wipe twice, flush once" |
| 05:13.12 | C4away | I have an issue |
| 05:13.18 | drmessano | ROFL |
| 05:13.24 | [TK]D-Fender | C4away, www.drphil.com |
| 05:13.28 | C4away | yea |
| 05:13.34 | C4away | not THAT kind of problem |
| 05:13.39 | C4away | well, I probably have those too |
| 05:13.44 | C4away | but this is asterisk-related |
| 05:13.50 | drmessano | Oh, so a www.oprah.com problem? |
| 05:13.57 | C4away | heh |
| 05:13.57 | C4away | she knows asterisk/ |
| 05:13.57 | drmessano | Oh, you said Asterisk |
| 05:14.00 | drmessano | Not Xenu |
| 05:14.02 | drmessano | My bad |
| 05:14.03 | vitasoy | you don't have an issue, you have a future success! |
| 05:14.20 | drmessano | vitasoy: WIZARD! |
| 05:14.25 | drmessano | vitasoy: I banish you! |
| 05:14.29 | C4away | hah |
| 05:14.43 | drmessano | ~wizard |
| 05:14.47 | jbot | extra, extra, read all about it, wizard is enchancement to howto's |
| 05:14.51 | drmessano | ~wizard! |
| 05:14.52 | jbot | somebody said wizard was enchancement to howto's |
| 05:14.59 | drmessano | Uh, ol |
| 05:15.01 | drmessano | ok* |
| 05:15.04 | C4away | so, I found a bug patch svn thingie that has a feature I want ... and I downloaded the modified files and put them in my asterisk src tree in the proper folders and now asterisk doesn't compile |
| 05:16.11 | C4away | is there a way to see if the files in the revision are for 1.6 or something? |
| 05:16.13 | C4away | the bug is for 1.4 |
| 05:16.45 | drmessano | was it patched against trunk? |
| 05:16.46 | C4away | http://svn.digium.com/view/asterisk?view=rev&revision=114487 |
| 05:16.54 | *** join/#asterisk shital (n=shital@122.167.64.192) |
| 05:16.55 | C4away | probably |
| 05:17.06 | C4away | I think I'm doing it wrong |
| 05:17.07 | shital | hello all |
| 05:17.15 | drmessano | Trunk is post 1.6 I think |
| 05:17.34 | C4away | well, that doesn't help me much |
| 05:18.11 | drmessano | You could always download and compile trunk |
| 05:18.13 | shital | i have TDM400p card with 1 FXO and 1 FXS module, so i how many extensions i can connect to this?? |
| 05:18.38 | vitasoy | you can make a gazillion and one extensions |
| 05:18.52 | vitasoy | but me, i have 2 so far |
| 05:18.55 | drmessano | shital: 42 million |
| 05:19.33 | drmessano | You can connect the PSTN to the FXO |
| 05:19.41 | drmessano | One wired extension to the FXS |
| 05:19.42 | C4away | " for i = 1 to $[1gazillion + 1] " isn't working |
| 05:19.54 | drmessano | and 41,999,999 SIP/IAX |
| 05:20.15 | C4away | I think I may have an INT overflow |
| 05:20.26 | drmessano | Give or take a billion |
| 05:21.10 | drmessano | "how much longer is it going to take to finish this house.... and DONT TELL ME TWO WEEKS" |
| 05:21.18 | drmessano | "Ok ok ok.. One week..... give or take a week" |
| 05:21.22 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 05:21.41 | C4away | that's a good question, will asterisk break if you define more than the limit of the integer datatype extensions? |
| 05:21.48 | shital | i wanted to use only for office use, by making use of analog telephone receivers |
| 05:22.14 | drmessano | shital: You can hook as many analog office phones as you want to the FXS.. its just ONE extension though |
| 05:22.31 | drmessano | It will be like having ONE phone line |
| 05:22.38 | drmessano | A total and complete waste, if you ask me |
| 05:23.01 | drmessano | "Ed, hang up, I need to make a call" |
| 05:23.03 | C4away | you can hook as many phones as you want as long as you don't exceed the REL limit of the fxs port |
| 05:23.06 | *** join/#asterisk JenniferAkemi (n=akemi@69-196-139-106.dsl.teksavvy.com) |
| 05:23.27 | drmessano | C4away: REL is irrelevant if you never get a call |
| 05:23.30 | *** join/#asterisk f3ew (n=F3ew@203.199.114.33) |
| 05:23.44 | drmessano | lol |
| 05:24.08 | C4away | each analog phone has a Ringer Equivalance Load defined for it, if the FXS puts out 3.0 REL then you can hook 3 phones with a REL of 1 or 6 phones with an REL of 0.5 |
| 05:24.22 | C4away | heh, yea |
| 05:24.35 | drmessano | You can put 100 on there for that matter |
| 05:24.42 | drmessano | Just make sure only 6 ring |
| 05:25.08 | C4away | well, cordless phones draw the REL load regardless of whether the ringer is enabled or not |
| 05:25.31 | drmessano | Depends on if it's disabled at the base or not |
| 05:25.31 | C4away | only old REL A devices (copper coil, physical bell) physically disconnect the ringer |
| 05:25.50 | C4away | the electronics will still be enabled to detect ringing even if the switch is set to "off" |
| 05:26.14 | drmessano | I guess it depends where the switch is |
| 05:26.30 | drmessano | Switch could be the Piezo, or the whole circuit |
| 05:27.46 | Qwell | REN ^^ |
| 05:28.00 | drmessano | REN is really the common term |
| 05:28.01 | C4away | yea that |
| 05:28.11 | drmessano | Ive seen it called REL too.. not sure what the difference is |
| 05:28.16 | Qwell | Ringer Equivalency Number |
| 05:28.19 | C4away | older phones like Western Electric desk phones just said "Ringer Equivalence" |
| 05:28.27 | drmessano | Linksys devices call it REL |
| 05:28.29 | Qwell | I asked Strom what the REN of my phone was once :( |
| 05:28.37 | C4away | like the one I have on my desk says "Ringer Equivalance 1.0A" |
| 05:28.39 | Qwell | I felt like an idiot when he bluntly stated "...exactly one" |
| 05:28.45 | drmessano | LOL |
| 05:28.58 | C4away | haha |
| 05:29.01 | Qwell | the phone I have is apparently what they base REN *on* |
| 05:29.02 | C4away | one A or B? |
| 05:29.17 | drmessano | What phone? |
| 05:29.19 | C4away | does it require 20-30 HZ or will it ring off of 60HZ? |
| 05:29.26 | drmessano | 60HZ? |
| 05:29.28 | C4away | yea |
| 05:29.33 | drmessano | Who uses 60HZ? |
| 05:29.39 | C4away | I did |
| 05:29.44 | drmessano | lol |
| 05:29.49 | C4away | plays do |
| 05:30.21 | C4away | the common method of ringing a telephone on stage is to take two AC-AC adapters back to back to get about 90 VAC 60Hz |
| 05:30.21 | drmessano | 60hz must be a wicked ring |
| 05:30.31 | C4away | not on a B device |
| 05:30.35 | C4away | sounds exactly the same as 20hz |
| 05:30.39 | Qwell | http://www.oldphoneworks.com/images/phones/DF939RFEA45YLHN.JPG |
| 05:30.41 | C4away | A devices are not rated for 60hz |
| 05:30.51 | C4away | they sound very distorted if they ring at all on 60hz |
| 05:31.03 | Qwell | it is, ironically, the phone in which Strom took his namesake |
| 05:31.07 | Qwell | or, the company |
| 05:31.35 | C4away | that would most definently be an A device |
| 05:31.36 | drmessano | That phone is made by Assclown Industries? |
| 05:31.41 | C4away | haha |
| 05:31.45 | Qwell | paid a whopping $2 for it at a consignment store :D |
| 05:32.10 | C4away | I got a full Western Electric speaker phone with "remote" for $5 from goodwill |
| 05:32.15 | Qwell | http://www.oldphoneworks.com/antique-phones/by-type/phone.asp?currency=USD&PhoneType=2&Phone=266_6 |
| 05:32.36 | *** join/#asterisk Corydon76-dig (i=silver@pdpc/supporter/bronze/Corydon76-home) |
| 05:32.36 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 05:32.46 | outtolunc | we had a phone like the first one in our barn, it was yellow tho <G> |
| 05:32.49 | Qwell | mine is way cooler though. it's VoIP |
| 05:32.53 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 05:32.59 | drmessano | heh |
| 05:33.01 | C4away | the remote is about the size of a pack of ciggarettes with a giant "off/on" toggle, a volume slider and a red glowing light |
| 05:33.19 | drmessano | A charlies angels phone? |
| 05:33.21 | C4away | the wire coming out of it is the diameter of a #2 pencil |
| 05:33.23 | C4away | YES |
| 05:33.26 | C4away | exactly |
| 05:33.33 | drmessano | Lucky bastard |
| 05:33.36 | drmessano | I want one |
| 05:33.38 | C4away | hehe |
| 05:33.42 | C4away | I'll sell you mine for $50 |
| 05:33.48 | drmessano | :( |
| 05:33.59 | drmessano | What I meant was |
| 05:34.07 | drmessano | "id sure love to find one for $5" |
| 05:34.19 | drmessano | or $8.50 |
| 05:34.42 | C4away | lol |
| 05:35.00 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-89ee461ac3fb89c1) |
| 05:35.20 | C4away | mine is VoIP too |
| 05:35.24 | Qwell | or $12.75 |
| 05:35.31 | C4away | thanks to a sipura 2100 ATA |
| 05:35.39 | C4away | or 2101 or something |
| 05:35.50 | C4away | 2102? |
| 05:35.57 | C4away | whatever |
| 05:36.04 | drmessano | Thanks to my 2 crates of PAP2s, all my phones are voip.. even the chinacom $4 phone I got from the dollar store |
| 05:36.23 | C4away | chinacom? |
| 05:36.31 | drmessano | taiwanico |
| 05:36.42 | C4away | lol |
| 05:36.48 | drmessano | whatever |
| 05:36.48 | C4away | I want 2 crates of pap2s |
| 05:36.51 | C4away | =( |
| 05:36.52 | drmessano | japanica |
| 05:37.01 | drmessano | I have like 40 of those damn things. |
| 05:37.05 | C4away | SunComCorp |
| 05:37.06 | drmessano | They have come in so handy |
| 05:37.30 | C4away | yea |
| 05:37.39 | drmessano | Nothing like watching someone use an analog phone and have no clue at the awesome power of Asterisk behind their call |
| 05:37.41 | *** join/#asterisk unpaidbill (i=bill@66.90.73.20) |
| 05:37.48 | C4away | lol |
| 05:37.49 | Qwell | C4away: is the phone itself voip, or is it just on an ATA? |
| 05:37.55 | C4away | it's just on an ata |
| 05:37.56 | Qwell | because mine actually is voip :p |
| 05:38.08 | Qwell | I cracked up an iaxy and stuck that sucker in there |
| 05:38.16 | drmessano | lol |
| 05:38.18 | C4away | I have another old analog phone I want to gut and upgrade |
| 05:38.23 | Qwell | it's got a wicked blue/orange glow from the led |
| 05:38.29 | C4away | lol |
| 05:38.39 | drmessano | Not only is it VoIP, but native IAX2 too |
| 05:38.42 | Qwell | I put it right where the hookswitch is, so it shines through quite nicely |
| 05:38.42 | drmessano | Showoff |
| 05:38.51 | C4away | http://www.chinacom-ltd.com/ |
| 05:38.59 | C4away | lol |
| 05:39.36 | drmessano | The chinese do such a great job with their knockoffs |
| 05:39.39 | *** join/#asterisk _gm (n=gmustafa@117.20.28.50) |
| 05:39.44 | drmessano | I cant wait for them to make a knockoff asterisk |
| 05:39.51 | C4away | I hear the chinese PSP is better than the SONY one |
| 05:40.21 | drmessano | Sure.. it will be Asterisk trunk with Zaptel 1.2.x grafted on it |
| 05:40.26 | drmessano | But it will be a knockoff |
| 05:40.59 | C4away | where is the profit in a knockoff of free software? |
| 05:41.07 | C4away | unless you improve it |
| 05:41.13 | C4away | then it's not really a knockoff anyway |
| 05:41.14 | drmessano | As if that ever stopped them before |
| 05:41.18 | C4away | hah |
| 05:41.38 | Qwell | bed |
| 05:41.45 | C4away | night |
| 05:41.46 | drmessano | Hmmm |
| 05:42.09 | C4away | yea |
| 05:42.10 | *** join/#asterisk andrew` (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 05:42.43 | drmessano | http://www.callweaver.org <----- oh, they already have |
| 05:44.06 | C4away | haha |
| 05:45.06 | drmessano | On that note... got another 16 hour day ahead |
| 05:45.08 | drmessano | Night all |
| 05:45.09 | C4away | you all have seen this one right? |
| 05:45.10 | C4away | http://www.ubergizmo.com/15/archives/2006/08/portorotary_cellphone.html |
| 05:45.16 | C4away | not voip, but still cool |
| 05:45.27 | C4away | night drmessano |
| 05:45.32 | drmessano | Ha.. thats awesome |
| 05:45.38 | drmessano | Later c4 |
| 05:53.32 | *** join/#asterisk apollonx (i=kit@193.19.189.38.STATIC.ISP.KZ) |
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| 06:04.35 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
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| 06:23.08 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:28.29 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 07:05.32 | schreckdebase | Hi |
| 07:06.34 | *** join/#asterisk Rico29 (n=rico@free.lycee-ledantec.ac-rennes.fr) |
| 07:06.37 | Rico29 | hi |
| 07:07.14 | *** join/#asterisk foenichs (n=erlang@222.124.158.150) |
| 07:09.50 | schreckdebase | Got something strange... |
| 07:10.05 | schreckdebase | When a X-Lite client on Windows registers... |
| 07:10.06 | schreckdebase | Apr 25 08:26:03 ERROR[3545]: chan_sip.c:11102 handle_request_subscribe: Got SUBSCRIBE for extension 0388675116@internal from 192.168.119.1, but there is no hint for that extension |
| 07:10.10 | schreckdebase | For example... |
| 07:10.15 | *** part/#asterisk Mr_Bond (i=db@danny.core1-osl.nsn.no) |
| 07:10.28 | *** join/#asterisk jmacz (n=jmacz@190.26.154.31) |
| 07:11.37 | *** join/#asterisk xenonex (n=xenonex@88.204.243.88) |
| 07:11.50 | Chris-NB | hi |
| 07:11.59 | *** join/#asterisk asdx (n=diego@adsl-148-95.click.com.py) |
| 07:12.23 | Chris-NB | schreckdebase, you have 0388675116 in your contacts (X-Lite) and activated presence |
| 07:12.53 | Chris-NB | schreckdebase, deactivate presence for that entry and the msg's should disappear. |
| 07:13.40 | schreckdebase | Chris-NB: Okay, thanks for your answer |
| 07:13.44 | schreckdebase | I'll try this ! :) |
| 07:14.01 | Chris-NB | k |
| 07:15.54 | schreckdebase | I thought it was something caused by X-Lite |
| 07:16.37 | schreckdebase | As when you start X-Lite, it makes a connection to 64.34.98.139 |
| 07:18.06 | *** join/#asterisk shinao1 (n=shinao1@41.219.248.67) |
| 07:18.11 | Chris-NB | *puh no idea what 64.34.98.139 is |
| 07:20.05 | schreckdebase | Chris-NB with http://64.34.98.139 You'll know it ! ;) |
| 07:20.08 | *** part/#asterisk foenichs (n=erlang@222.124.158.150) |
| 07:20.28 | Chris-NB | okay : D |
| 07:20.33 | schreckdebase | I don't like those softwares that connect silentely... |
| 07:20.35 | schreckdebase | :/ |
| 07:25.13 | Chris-NB | schreckdebase, you got your X-Lite from that website? |
| 07:27.19 | schreckdebase | No |
| 07:27.20 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 07:27.40 | schreckdebase | Can't remember this one |
| 07:28.10 | *** join/#asterisk eddy122 (n=irc@91.73.203.98) |
| 07:29.03 | eddy122 | anyone alive |
| 07:29.57 | eddy122 | anyone alive here guys? |
| 07:31.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:32.28 | schreckdebase | Yes, but not an expert |
| 07:32.30 | schreckdebase | Apr 25 08:52:25 WARNING[3545]: chan_sip.c:9787 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"It's Me" <sip:xxxxxxxxxx@freephonie.net>;tag=as227c40bd' |
| 07:32.48 | schreckdebase | That's what I got when dialing with X-Lite |
| 07:32.59 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 07:33.00 | *** join/#asterisk gardo (n=gardo@210.213.101.130) |
| 07:33.01 | schreckdebase | I have to make an Asterisk restart to be able to dial again |
| 07:33.03 | schreckdebase | :/ |
| 07:34.44 | eddy122 | schreckdebase U KNOW HOW TO LINK 3 SERVERS TOGETHER? |
| 07:35.02 | schreckdebase | No, sorry eddy122 |
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| 07:41.54 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 07:41.55 | Telemac | Hello |
| 07:42.29 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 07:44.27 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
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| 07:45.46 | Telemac | Is there any known issue between chan_misdn and app_queue ? I've a exten that use queue application, when called from internal phone all is ok but from isdn I get a hangup after only 6sec, without error message. |
| 07:46.46 | schreckdebase | I've changed the /etc/default/asterisk to run Asterisk at boot... But there is no effect... Any other file to change ? |
| 07:48.16 | Telemac | schreckdebase: I don't know about your distro but I guess you have some command to add it as init script (rc-update add, chkconfig ...) |
| 07:48.47 | schreckdebase | Telemac: Thanks, I'm quite noob |
| 07:48.48 | schreckdebase | ^^ |
| 07:56.59 | eddy122 | Telemac, CAN U HELP ME OUT? |
| 07:57.42 | eddy122 | i have 3 pbxs, one of them is in USA where all ports are open, the other 2 cant use sip. i have setup everything to link them. i want the 2 pbx to be able to call out from the sip account that i have already setup in the system in us using iax protocol. something is wromg im getting all circuits are busy now.. any help? |
| 07:58.59 | Telemac | eddy122: I'm not (yet) an asterisk expert at all. Check in asterisk console debug if you can see some error or warning |
| 07:59.16 | *** join/#asterisk shital (n=shital@122.167.110.180) |
| 07:59.26 | shital | Hello all |
| 08:00.13 | shital | i am using TDM400P with 1 FXO and 1 FXS modules, is that require 12V power supply? |
| 08:01.13 | BeeBuu | shital: if you want your phone which connetc to the FXS port ring... |
| 08:02.27 | BeeBuu | you must plug the power connect on. |
| 08:03.36 | Telemac | (#1)http://openpaste.org/en/6268/ | (#2)http://openpaste.org/en/6269/ | First (#1) is when I call by isdn and go to a queue ('Queue' app), second (#2) is when I call by isdn go it go to a 'Dial' app. In first case I get a hangup after ~6sec (if nobody in queue answer) whereas in second case it work and keep ringing the phone until normal timeout or answer. I've no error message, the only difference I can see between debug of these two is "P[ 1] misdn_wr |
| 08:06.04 | shital | i mean m using one incoming line and 1 extension phone, so to make and receive call do i need to plug 12V supply for that? |
| 08:12.21 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 08:13.47 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 08:13.49 | Kernel_Core | hi all |
| 08:15.56 | Telemac | Oh I was needing a 'Answer' application as first so that queue was properly joined from isdn |
| 08:16.43 | Kernel_Core | I am looking for Speech Recognition feature in asterisk... imagine , I record an MP3 or Wave File , when my ZAP channel dials out , if the recorded MP3 or Wave File matches the speech, after that executes something ! |
| 08:17.31 | Kernel_Core | is it possible with asterisk (1.4.X ) ? |
| 08:20.56 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
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| 08:22.05 | Telemac | How can set set 'fr' as default language for queue so message were properly localized ? |
| 08:22.33 | Telemac | Set(LANGUAGE()='fr') in exten of is there something with queue definition ? |
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| 08:42.38 | schreckdebase | http://pastebin.org/31880 |
| 08:43.00 | schreckdebase | It's the output of the start script... |
| 08:43.18 | schreckdebase | But when I check if the process runs ==> Nothing ! :( :/ |
| 08:43.21 | schreckdebase | Any idea ? |
| 08:48.02 | *** join/#asterisk keulin (n=cray@80.15.251.6) |
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| 09:08.01 | ^shark_ | hi friends, i have just installed asterisk on freebsd 6.2 with a dubious tld (.voc) and i am getting this warning in the /var/log/asterisk/messages file >> WARNING[791] pbx_dundi.c: Unable to look up host 'jupiter.joc.voc' |
| 09:08.06 | ^shark_ | i dont know what dundi.c is about but i want to have no such warning, can someone help guide me on this? |
| 09:09.12 | ^shark_ | is this b'se of my /etc/resolv.conf file or something else, does dundi.c look up resolv.conf file? |
| 09:09.55 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:12.01 | C4away | ^shark_ : try disabling the dundi module maybe? |
| 09:13.07 | C4away | could be resolve related |
| 09:13.12 | C4away | make sure DNS is working on your box |
| 09:13.26 | C4away | nslookup www.google.com |
| 09:14.17 | ^shark_ | ok i dont have a dns server but i can resolve. |
| 09:14.36 | ^shark_ | what is the dundi really for anyways? |
| 09:15.00 | C4away | it is a number routing discovery protocol for sip |
| 09:15.02 | *** join/#asterisk Corydon76-dig (i=beige@pdpc/supporter/bronze/Corydon76-home) |
| 09:15.02 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 09:15.22 | C4away | for example, I have my main number registered in e164.org's database to point to my SIP URI for my server |
| 09:15.36 | C4away | oh, wait |
| 09:15.43 | C4away | that's e164 |
| 09:15.52 | C4away | dundi is like e164 but different |
| 09:16.21 | C4away | if you had 5 servers in different locations you could group them all together with dundi and they could figure out which server is responsible for a specific number |
| 09:16.39 | C4away | if none are, then you send it out PSTN or ITSP connection |
| 09:16.56 | ^shark_ | koay not a bad warning to have for now, thanks for the information |
| 09:17.01 | ^shark_ | okay* |
| 09:17.10 | shital | i m getting error while configuring TDM card as ZT_CHANCONFIG failed on channel 1: can any one help me? |
| 09:17.18 | C4away | correct, nothing is wrong if you get dundi errors |
| 09:17.51 | C4away | I don't know TDM hardware very well shital |
| 09:17.55 | C4away | and I"m off to bed |
| 09:18.12 | C4away | it is getting late here and i'm done with the server maintenance I was doing |
| 09:18.48 | ^shark_ | C4away: can i just show you one file and you guide me on the warnings please.. |
| 09:19.03 | ^shark_ | C4away: its here >> http://pastebin.com/m4ce5df22 |
| 09:20.09 | ^shark_ | the warning for moh is understandable b'se i haven't configured moh yet, but the others are very new to me, kindly look this up for me and advice before you go to bed ;) |
| 09:20.41 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6cc24879a3683943) |
| 09:21.12 | C4away | unless you are using ael you don't need to worry about the notices relating to that |
| 09:22.08 | C4away | same with SDMI, if you are not using it then you don't need to worry about that warning |
| 09:22.51 | C4away | if you really want to simplify your configuration you can disable many applications and modules, although they don't really do much if you are not using them so there is no real reason to do so |
| 09:23.32 | C4away | as far as the warnings nothing jumps out at me as a problem |
| 09:24.35 | ^shark_ | C4away: thanks alot man, have a wonderful time |
| 09:24.42 | C4away | this is a little odd though: [Apr 25 11:26:45] WARNING[858]: chan_mgcp.c:4236 reload_config: Unable to get our IP address, MGCP disabled |
| 09:25.02 | C4away | do you have externip= and localnet= set in your sip.conf? |
| 09:25.17 | C4away | or included in sip_nat.conf (if you are using a distro that uses sip_nat.conf) |
| 09:25.55 | C4away | if you are not using Cisco phones with the MGCP channel type then you don't need to worry that MGCP is not loading, but unable to get IP address may be an issue |
| 09:26.25 | C4away | anyway, I'm off to bed |
| 09:26.27 | C4away | night all |
| 09:28.43 | shital | please any body can help me to rectify this error? "ZT_CHANCONFIG failed on channel 1:" |
| 09:32.45 | *** join/#asterisk af_ (n=getsmart@88-149-230-19.dynamic.ngi.it) |
| 09:34.28 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 09:38.11 | mvanbaak | do the TE110p and TE221p use the same driver ? |
| 09:41.16 | *** join/#asterisk shinao1 (n=shinao1@41.219.242.104) |
| 09:43.34 | *** part/#asterisk Kernel_Core (n=I@85.133.155.134) |
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| 09:53.57 | EmleyMoor | Are there any softphones suitable for use with Asterisk that can dial from Outlook contacts? |
| 09:57.30 | Dovid | EmleyMoor: You can try snap dial. its an applicaton that works with asterisk. u goto pay for it but it is well worth it |
| 09:58.40 | *** join/#asterisk dominikholling (n=chatzill@pD957CEA1.dip.t-dialin.net) |
| 10:00.06 | *** join/#asterisk af_ (n=getsmart@88-149-230-19.dynamic.ngi.it) |
| 10:00.29 | EmleyMoor | Dovid: Can you point me to it? |
| 10:02.33 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:03.07 | dominikholling | hi guys, I have a quick question. Is it somehow possible to intercept a call going to Voicemail? By this I mean the following scenario: a persons phone rings, the person isn't fast enough to answer and the call goes to voicemail, now the persons wants to get this call. Is there a possibility to do this? I found the pickup command but that doesn't seem to do the trick. Google didn't output... |
| 10:03.08 | dominikholling | ...anything and the oreilly book on asterisk.org wasn't much help either. Can you help me? |
| 10:04.12 | dominikholling | I am using asterisk 1.4 on a debian linux system |
| 10:06.06 | *** join/#asterisk Corydon76-dig (i=black@pdpc/supporter/bronze/Corydon76-home) |
| 10:06.06 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 10:07.02 | mvanbaak | Corydon76-dig: do you happen to know if the TE110p and TE221p use the same kernel module ? |
| 10:15.34 | *** join/#asterisk idc-dutch (n=idc-dutc@195-144-082-145.dyn.adsl.xs4all.be) |
| 10:17.15 | dominic1 | one question: Mysql or Postgres for my little asterisk? |
| 10:18.30 | mvanbaak | whatever you know best |
| 10:18.44 | ac1djazz | mysql |
| 10:18.58 | ac1djazz | postgres more like failgres |
| 10:18.58 | dominic1 | what about the performance? |
| 10:19.10 | dominic1 | what is failgres? |
| 10:19.22 | *** join/#asterisk buliwyf (n=buliwyf@email.stwhas.de) |
| 10:19.25 | buliwyf | re |
| 10:20.18 | buliwyf | Where can i get some infos about trunking asterisk with an alcatel omnipcx |
| 10:26.36 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
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| 10:41.21 | JT | dominic1: postgresql |
| 10:41.26 | JT | it's far better than mysql |
| 10:42.03 | shasta | ac1djazz, bullshit :) |
| 10:47.51 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 10:54.22 | ac1djazz | shasta: 100% true |
| 10:54.39 | ac1djazz | lol jt |
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| 11:00.23 | saftsack | exten => s,1,Dial(SIP/52, 10) exten => s,2,Dial(SIP/52&SIP53) |
| 11:01.06 | saftsack | hi this doesnt work for me. 52 stops for a second ringing |
| 11:01.53 | ruied | I'm trying to compile mISDN-1_1_7_2 and it reports "CFLAGS was changed in ...Makefile... ...Fix it to use EXTRA_CFLAGS", but the make file already has the "EXTRA_CFLAGS" setted, what could be the problem? |
| 11:07.51 | *** join/#asterisk fmueller (n=user@p548F789F.dip.t-dialin.net) |
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| 11:13.39 | puzzled | hi |
| 11:19.18 | *** join/#asterisk TheH (n=tommy@86.43.150.138) |
| 11:19.54 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:19.58 | JT | mysql is a poor excuse for an rdbms compared to postgresql |
| 11:20.04 | TheH | .... anyone has a suggestion for a BRI card that can be used in north america ? i have always used the b410p but it does not support north american bri so i have to... change :( |
| 11:20.06 | JT | mysql has always been playing catchup |
| 11:20.22 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:20.31 | JT | TheH: there is nothing available for use with asterisk that is guaranteed to work with north american bri |
| 11:20.43 | JT | some people have done their own testing and got some levels of success |
| 11:20.58 | TheH | :( ah shit... thats gonna be a pain.. |
| 11:21.27 | TheH | would verizon be able to help me out or are they anti asterisk |
| 11:29.24 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584477.dsl.bell.ca) |
| 11:30.17 | JT | i dont see how they could help |
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| 11:39.53 | tzafrir_home | JT, Zap/BRI will work with American BRI |
| 11:40.02 | tzafrir_home | tested |
| 11:40.48 | tzafrir_home | Actually, as it is National and PtP, chances are it will work with plain Asterisk. At least with 1.6 |
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| 11:53.27 | spjuden | tzafrir_home, i experienced the strange call state problem again.. this time got a nice screen log too- it does seems to be caused by call waiting, although this confuses me even more because last time we set callwaiting and threeway=no for both the fxo and fxs's.. |
| 11:54.07 | spjuden | s/threeway/threewaycalling to be clear |
| 11:54.47 | tzafrir_home | "State 6" == "Up" (a call is running) |
| 11:56.03 | tzafrir_home | Is this something you can reproduce? |
| 11:56.16 | idc-dutch | tzafrir_home: will the astribank bri change firmware for 1.6? |
| 11:56.43 | tzafrir_home | idc-dutch, The Astribank driver is part of Zaptel |
| 11:56.59 | tzafrir_home | And tends to be the same in latest Zaptel 1.2 and 1.4 |
| 11:57.00 | spjuden | tzafrir_home, i can reproduce the three way with it being disabled...ill have to enlist some help from somebody here to try to reproduce the strange state error |
| 11:57.11 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 11:57.30 | idc-dutch | hmm, so basically bristuff get included in plain asterisk now? ;) |
| 11:57.50 | tzafrir_home | idc-dutch, the Astribank driver and its firmware is not part of bristuff |
| 11:58.19 | tzafrir_home | bristuff (at the Asterisk level) has so far not been updated to Asterisk 1.6 |
| 11:58.33 | idc-dutch | yeah, I know, but wondering if something will change on the astribank side firmware side to make it work with asterisk 1.6 |
| 11:58.54 | idc-dutch | aka, when bri gets supported in plain asterisk, do old astribanks still work ;) |
| 11:59.01 | tzafrir_home | If you don't need a BRI NT, you can try using the native Asterisk chan_zap/libpri, which have much improved support for BRI in 1.6 |
| 11:59.08 | *** join/#asterisk aa3dvark (n=sdfa@c-67-163-106-10.hsd1.va.comcast.net) |
| 11:59.24 | idc-dutch | we only use the TE modes indeed |
| 11:59.28 | tzafrir_home | right, it's the same |
| 12:00.53 | idc-dutch | ok, but only with asterisk 1.6? Or does the astribank TE mode work without bristuff in 1.4 also already? |
| 12:01.48 | spjuden | http://pastebin.com/m23d8a142 |
| 12:01.49 | tzafrir_home | idc-dutch, IIRC you can get it to work in a very minimal way, at least, with 1.4 and a small patch |
| 12:01.54 | tzafrir_home | But only ptp |
| 12:02.02 | tzafrir_home | And this is not really tested |
| 12:02.16 | idc-dutch | kk, still bristuff for now then.... |
| 12:03.17 | idc-dutch | implementing another astribank bri 8 next week, 6x ptp, but 2x ptmp |
| 12:03.27 | idc-dutch | all TE tho |
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| 12:08.19 | idc-dutch | tzafrir_home: On another possible project we might have to connect 120 analog devices to asterisk. Will 5 24poorts analog astribanks do the job without problems? Can we connect them over a USB hub? |
| 12:12.18 | *** join/#asterisk Tili (n=tili@58.27.144.247) |
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| 12:12.48 | dominikholling | hi guys, I have a quick question. Is it somehow possible to intercept a call going to Voicemail? By this I mean the following scenario: a persons phone rings, the person isn't fast enough to answer and the call goes to voicemail, now the persons wants to get this call. Is there a possibility to do this? I found the pickup command but that doesn't seem to do the trick. Google didn't output... |
| 12:12.49 | dominikholling | ...anything and the oreilly book on asterisk.org wasn't much help either. Can you help me? |
| 12:13.53 | tzafrir_home | idc-dutch, Well, you can also use units of 32, which will probably cheaper and more efficient. In that case you could use 4 units. You can use a USB hub. |
| 12:16.36 | dominic1 | where can I get .sql 's for my mysql database? I want to run asterisk with mysql. |
| 12:17.54 | idc-dutch | ah you, forgot about the 32 ports... will use them ofc |
| 12:19.48 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 12:22.59 | dominic1 | Can I find the template for mysql in the sources? |
| 12:23.09 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
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| 12:27.23 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:28.44 | spjuden | tzafrir, aha- i can duplicate it! but with a lot of cell phones |
| 12:30.15 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 12:30.36 | spjuden | tzafrir, from the asterisk phone i call out, flash, call out to another number...flash and i have a three way call (despite threewaycalling=no).. i hang up on the zap line, the 2 cell phones are still bridged. call in with a 3rd cell phone and * still answers as it should.. hang up on the cell phones and the calls still show active in "core show channels", try to call in from one of the cells and get the strange state 6 error and no answer |
| 12:41.05 | tzafrir_home | spjuden, flash should not work if you have threewaycalling=no |
| 12:41.19 | spjuden | ok so threewaycalling=no in zapata.conf, but =yes in users.conf, so that was overriding it.... |
| 12:42.14 | spjuden | i restart and now i can't three way call, but yet for some reason when i hit flash it still gives me a dialtone (despite as soon as i dial a number it sends nothing to asterisk and just puts me back to the call i was already on) |
| 12:42.43 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:42.43 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:43.54 | RoyK | hei, leif |
| 12:45.27 | [TK]D-Fender | na na, na na na.... |
| 12:50.25 | *** join/#asterisk pukkita (n=pukkita@137.Red-80-59-10.staticIP.rima-tde.net) |
| 12:50.34 | pukkita | hi |
| 12:52.02 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 12:53.16 | mknerd | morning |
| 12:54.53 | mknerd | hookflash a zap fxo port from a sip extension so that the sip extension can send DTMF direct to the external PBX ? |
| 12:54.55 | mknerd | anyone? |
| 12:55.15 | *** join/#asterisk shido6 (n=shido6@74-130-61-41.dhcp.insightbb.com) |
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| 12:55.51 | pukkita | I'm setting up a new asterisk box to replace an analog Bosch PBX. The bosch is connected to a switchbox, the switchbox is connected to an E1. I have a 4 primary ports card, and want to "insert" the asterisk box in-between the telco primary and the switchbox. The port to the switchbox is in MASTER clock timing mode, I have used a pri xover cable (1-4, 2-5), the port connected to the switchbox lights up as connected, but the switch |
| 12:55.54 | lmadsen | mknerd: maybe a combination of the Flash() application and application map might work? |
| 12:56.19 | mknerd | flash() only seems to work on the originating channel |
| 12:56.53 | lmadsen | you can tell the application map which channel to execute the application on |
| 12:56.58 | lmadsen | self or peer |
| 12:57.03 | mknerd | ah |
| 12:57.18 | lmadsen | the documentation in features.conf for the application map is pretty good |
| 12:57.23 | lmadsen | I just had to learn it yesterday |
| 12:57.50 | mknerd | thx, i think thats just what I needed |
| 12:58.14 | [TK]D-Fender | mknerd: http://www.voip-info.org/wiki-Asterisk+config+features.conf |
| 12:59.22 | [TK]D-Fender | pukkita: Sounds more like you're jsut putting * inbetwen the telco & PBX> What do you want * to do for you? |
| 13:02.03 | pukkita | the asterisk box will have SIP phones, the idea is to phase out the old phones |
| 13:02.33 | mknerd | lmadsen, is that something that would require user intervention? |
| 13:02.50 | lmadsen | mknerd: ya... isn't that what you want? |
| 13:02.52 | pukkita | I already do that in that same box with another telco (mobile). the asterisk box is also trunked to a remote office via IAX |
| 13:03.50 | mknerd | well, originally I just needed to handle hookflash a call originating from the pbx side to give it back to the pbx to control |
| 13:04.13 | mknerd | now they want me to be able to contact a pbx extension from a * extension |
| 13:04.16 | dominic1 | Hello guys, I am looking for the mysql templates for asterisk. Can anybody tell me where I can get these templates? In the contrib directory the templates doesn't seem to be complete |
| 13:04.26 | mknerd | was hoping to be able to put it in the dialplan |
| 13:04.28 | pukkita | the problem with the PBX is it only has a primary port (mobile calls), So the regular ones go to a E1 -> multiple ISDN port converter, dunno if switchbox is the proper name for it |
| 13:04.42 | mknerd | switchbox is a brand of appliance |
| 13:05.03 | pukkita | oh |
| 13:05.06 | mknerd | just recently purchased by digium |
| 13:05.44 | lirakis_work | ping |
| 13:05.50 | pukkita | let's call it a E1 to multiple ISDN port converter then. Don't know the proper term in english (I'm in Spain) |
| 13:05.50 | mknerd | pong |
| 13:05.53 | lirakis_work | nice |
| 13:06.13 | [TK]D-Fender | pukkita: "BRI Channel Bank" might be more appropriate |
| 13:06.34 | M1s3ry | mknerd, i think you mean switchvox |
| 13:06.40 | pukkita | thanks [TK]D-Fender! |
| 13:06.40 | [TK]D-Fender | dominic1: "templates" for what? |
| 13:06.53 | mknerd | oops |
| 13:06.55 | mknerd | :-) |
| 13:07.18 | [TK]D-Fender | pukkita: Your first message got cut off. Did you have a problem you're looking to solve in there? |
| 13:09.06 | pukkita | yes |
| 13:09.29 | pukkita | the port to the telco E1 is up and fine, so guess I know the parameters the BRi channel bank is using |
| 13:09.38 | pukkita | or expecting |
| 13:09.57 | mknerd | lmadsen, i think your right, that is what i need for this second part, I have it setup in features.conf, I am not sure how the usage works though |
| 13:10.02 | pukkita | çI'm using pri_net signalling, clock in master and a xover cable (1-4 and 2-5) |
| 13:10.17 | lmadsen | mknerd: you dial the DTMF sequence you define.... |
| 13:10.25 | pukkita | but the BRI Ch bank display the alarm leds |
| 13:10.29 | lmadsen | the call has to be active on the zap channel and the sip channel though |
| 13:10.37 | lmadsen | i.e. you called something, and the channel is active |
| 13:10.45 | mknerd | hmm |
| 13:11.00 | mknerd | so I need a random number to dial beforehand |
| 13:11.05 | [TK]D-Fender | pukkita: what model of card, is this your first time using it? Pastebin your zaptel.conf & zapata.conf |
| 13:11.07 | [TK]D-Fender | ~pb |
| 13:11.09 | 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 |
| 13:11.11 | [TK]D-Fender | ^^^^^^^^^^^ |
| 13:11.19 | pukkita | no, is not the first time, is a sangoma AFT104 |
| 13:11.27 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) |
| 13:11.35 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:11.36 | [TK]D-Fender | pukkita: And then your wanpipe1 & 2 as well |
| 13:11.46 | pukkita | i have another box with a 2-pri sangoma working fine |
| 13:12.13 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:12.20 | hackeron | hey, can someone please suggest a way to add a visual indicator to all extensions when both receptionists are on the phone? |
| 13:13.07 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:13.33 | pukkita | [TK]D-Fender: wanpipe1 : http://pastebin.com/m75b2a779 |
| 13:13.53 | [TK]D-Fender | hackeron: run presence on your phone that checks both of them. |
| 13:14.14 | pukkita | [TK]D-Fender: wanpipe2: http://pastebin.com/m46693b36 |
| 13:14.45 | hackeron | [TK]D-Fender: hmm, how? :) -- I'm just looking on the wiki, maybe something like exten => 0,hint,SIP/reception1&SIP/reception2 would work? |
| 13:15.14 | [TK]D-Fender | hackeron: No, that will light up it EITHER are in use, not specifically for BOTH. |
| 13:15.28 | [TK]D-Fender | hackeron: So 2 DIFFERENT presence subscriptions are needed. |
| 13:16.01 | [TK]D-Fender | hackeron: Or you could run a daemon off AMI that will check on events as to their state and use DevState to toggle a single indicator |
| 13:16.18 | pukkita | [TK]D-Fender: zapata.conf http://pastebin.com/m2134ead6 (E1 port is port 2, BRI channel bank port 4 |
| 13:16.31 | tzafrir_home | pukkita, how many BRI ports you need? There are plenty of quad-port cards, and several octo-port ones |
| 13:16.40 | tzafrir_home | ah, ok |
| 13:16.57 | pukkita | [TK]D-Fender: zaptel.conf: http://pastebin.com/m2da75104 |
| 13:17.08 | pukkita | tzafrir_home: 16 |
| 13:17.20 | tzafrir_home | scary :-) |
| 13:17.27 | pukkita | tzafrir I don't want to spend more on the old PBX |
| 13:18.19 | hackeron | [TK]D-Fender: what do you mean by 2 different presence subscriptions? -- 2 accounts for every phone that subscribe to 2 different hints? |
| 13:18.42 | disposable | is there any way of using global variables in context names? |
| 13:18.45 | pukkita | [TK]D-Fender the wanpipe1.conf i linked before is wanpipe2.conf and wanpipe2.conf is wanpipe4.conf, I'm using port 2 for the E1 and port4 for the Bri channel bank |
| 13:19.52 | *** join/#asterisk shinao1 (n=shinao1@41.219.217.60) |
| 13:20.19 | tzafrir_home | disposable, no. You can use them in Goto commands, though |
| 13:21.28 | [TK]D-Fender | pukkita: span=2,0,0,ccs,hdb3,crc4 <- should be 2,1,0 so that it takes timing from the telco |
| 13:22.34 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 13:25.31 | *** join/#asterisk ccvp (n=ccvp@66.0.46.210) |
| 13:26.07 | pukkita | I using two different telcos, should it be 2,1 or 2,2? |
| 13:26.39 | [TK]D-Fender | pukkita: one should be "1" and the other "2". Order si largely irrelvent |
| 13:27.02 | [TK]D-Fender | hackeron: Meaning you do 1 BLF for each |
| 13:27.29 | [TK]D-Fender | hackeron: and your users would look to see if both are lit. |
| 13:27.42 | ccvp | d-fender |
| 13:27.51 | ccvp | I almost joined a guild in memory of you |
| 13:28.02 | ccvp | "Tha killaZ" - i saw one named that on WoW last night heh |
| 13:28.20 | pukkita | [TK]D-Fender thanks |
| 13:28.29 | Katty | ccvp: Undead Poet Society |
| 13:28.36 | Katty | ccvp: Blood Bath and Beyond |
| 13:28.41 | Katty | ccvp: Gnomeland Security |
| 13:28.49 | hackeron | [TK]D-Fender: ok, thanks, I'll try |
| 13:28.52 | mknerd | I need to send a hookflash as part of Dial() |
| 13:29.00 | ccvp | I quit wow 5 months ago, until that level 80 expansion comes out, did that get cancelled? |
| 13:29.04 | pukkita | [TK]D-Fender so on the ports to telcos one will be 1,1, the other 2,2 and the one to the bri bank should be 4,0? |
| 13:29.35 | Katty | ccvp: no idea. |
| 13:30.21 | [TK]D-Fender | pukkita: Your telco ones should be numbered "1" and "2" for timing, your CB's should be "0" |
| 13:31.08 | ccvp | katty, i found out this company in town where my friend works: www.dr-inc.com, their entire phone system is asterisk, 256-895-0881 |
| 13:31.22 | ccvp | it was put together by some dude who does it "part time", but listen to their menu system, seems like its done well |
| 13:32.13 | ccvp | can this software support mp3's, and wav files, and other formats |
| 13:32.16 | ccvp | as precorded snipits? |
| 13:32.42 | [TK]D-Fender | ccvp: Yes |
| 13:32.52 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 13:33.22 | ccvp | well, thats neat thing he can do it and not get paid for it, he saved their company like $30,000 instead of going w/ some local provider ITC deltacom |
| 13:33.34 | ccvp | if someone knows it like him, he should setup a little LLC and strictly do asterisk deployments for companies. |
| 13:35.05 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.231.245) |
| 13:35.11 | Katty | ccvp: that's what i do, sorta. |
| 13:35.16 | Katty | ccvp: but i don't freelance or anything. |
| 13:35.22 | anonymouz666 | Katty! |
| 13:35.25 | Katty | ccvp: i work for an INC. |
| 13:35.27 | Katty | anonymouz666: !!! |
| 13:35.30 | Katty | huggles anonymouz666 |
| 13:35.34 | anonymouz666 | what about the FOP blinking? |
| 13:35.36 | anonymouz666 | :D |
| 13:35.50 | ccvp | katty, so if you know the software pretty well |
| 13:35.56 | ccvp | how good is your networking background? |
| 13:36.07 | ccvp | ie, cisco certs. I have a ccna, ccvp, and almost ccie voice, but I know zilch about sterisk |
| 13:36.17 | Katty | anonymouz666: fix it.. |
| 13:36.23 | Katty | anonymouz666: turned out to be a bug with adobe. |
| 13:36.36 | anonymouz666 | kaldemar: how did you fix it? |
| 13:36.40 | anonymouz666 | damn |
| 13:36.41 | Katty | ccvp: my networking background is nothing fancy... have a couple mcps, that's it |
| 13:36.44 | anonymouz666 | Katty: how did you fix it? |
| 13:36.59 | Katty | anonymouz666: rmed it, and downloaded the snapshot |
| 13:37.15 | Katty | anonymouz666: instead of the whatever release version i'd gotten before |
| 13:37.32 | anonymouz666 | I just hate FOP, but the customers do like. |
| 13:37.40 | Katty | indeed. |
| 13:37.47 | Katty | they like isymphony too, if you've not had a look at it |
| 13:37.51 | Katty | it's also Pretty(tm) |
| 13:38.47 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:39.30 | ccvp | i think all i need is coffee right, now my brain is friend, i almost gave up my masters in CS attempt..they are having us use Chi-Square statistics for IP packets |
| 13:39.41 | ccvp | for a "networking fundamentals class", how asinine is that |
| 13:39.47 | ccvp | s/friend/fried |
| 13:39.49 | grandpapadot | Spoon! |
| 13:39.50 | Katty | mad |
| 13:39.58 | Katty | hugs anthm |
| 13:40.03 | Katty | i still have problems with DNS |
| 13:40.08 | Katty | and linux is still a bit foggy |
| 13:40.23 | Katty | i don't know it as well as i know microsoft platforms |
| 13:40.40 | Katty | run levels, init, etc is still....confusing |
| 13:40.46 | hackeron | [TK]D-Fender: I've just tried adding hints and core show hints is teling me: "There are no registered dialplan hints" -- I followed this guide: http://www.voip-forum.com/asterisk/2008-01/sip-subscriptions/ - any ideas? |
| 13:41.05 | anthm | hiya |
| 13:41.09 | *** join/#asterisk IceHeat (n=ammar@77.110.81.31) |
| 13:43.39 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:45.04 | [TK]D-Fender | hackeron: Well since you're not showing me what you did, I can't tell you where you screwed up. |
| 13:45.43 | [TK]D-Fender | anthm: 'lo |
| 13:46.02 | IceHeat | hello all |
| 13:46.16 | IceHeat | I'm having some trouble configuring asterisk to forward calls |
| 13:46.35 | grandpapadot | iceheat: You just Dial() the number to forward. |
| 13:46.38 | [TK]D-Fender | IceHeat: * doesn't "forward" calls. |
| 13:46.44 | grandpapadot | forward to ... rather |
| 13:46.49 | IceHeat | I'm trying to forward incoming calls on a zaptel interface to an extenssion on a different asterisk server |
| 13:46.59 | IceHeat | using IAX |
| 13:47.12 | [TK]D-Fender | IceHeat: lookup "asterisk dual servers" on the WIKI |
| 13:47.15 | [TK]D-Fender | ~wikis |
| 13:47.16 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 13:47.23 | IceHeat | <PROTECTED> |
| 13:47.25 | hackeron | [TK]D-Fender: :) -- one sec, let me paste what I've done on rafb |
| 13:47.26 | IceHeat | thanks |
| 13:47.31 | IceHeat | I'll look it up |
| 13:47.45 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 13:49.37 | hackeron | [TK]D-Fender: here: http://rafb.net/p/rQP1Vo48.html |
| 13:50.02 | [TK]D-Fender | hackeron: exten => 0,2,hint,SIP/reception1&SIP/reception2 |
| 13:50.15 | [TK]D-Fender | hackeron: "hint" is a PRIORITY, not an APPLICATION. |
| 13:50.37 | hackeron | [TK]D-Fender: oh, lol |
| 13:50.39 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-245-162.balt.east.verizon.net) |
| 13:51.07 | Katty | hai |
| 13:52.04 | hackeron | [TK]D-Fender: thanks! - now core show hints works, sip show subscriptions isn't showing any subscriptions though :( |
| 13:52.26 | [TK]D-Fender | hackeron: You'll need to restart your phone to it to subscribe. |
| 13:53.16 | jblack | <PROTECTED> |
| 13:55.09 | [TK]D-Fender | jblack: He's fixed that... |
| 13:55.21 | hackeron | [TK]D-Fender: restarted phone, still no subscription :( -- it's a Polycom 301 if that helps |
| 13:55.36 | [TK]D-Fender | jblack: And neither of your 2 samples are correct :) |
| 13:56.15 | [TK]D-Fender | hackeron: Where are you looking for an indication of subscription? What did you do to "subscribe"? |
| 13:56.28 | hackeron | [TK]D-Fender: in sip show subscriptions |
| 13:56.46 | hackeron | [TK]D-Fender: and I just restarted the phone |
| 13:56.55 | [TK]D-Fender | hackeron: And what did you do on the phone to "subscribe to it? |
| 13:57.20 | [TK]D-Fender | hackeron: and use "show hints" |
| 13:57.24 | hackeron | [TK]D-Fender: hmm, nothing :) - let me check the manual - should I look for subscribe? |
| 13:57.40 | hackeron | [TK]D-Fender: core show hints seems to work |
| 13:57.48 | [TK]D-Fender | hackeron: then by default your phone is not even enabled to subscribe. |
| 13:58.22 | [TK]D-Fender | hackeron: So "core show hints" shows a watcher? |
| 13:58.31 | hackeron | [TK]D-Fender: no, core show hints shows no watchers |
| 13:58.41 | hackeron | [TK]D-Fender: 0@incoming : SIP/reception1&SIP/r State:Idle Watchers 0 |
| 13:59.10 | [TK]D-Fender | hackeron: Then if you've done nothing in your polycom provisioning then it will not even support your hints. |
| 13:59.19 | [TK]D-Fender | hackeron: It won't work "out of the box" |
| 13:59.38 | hackeron | [TK]D-Fender: hmm, where can I find more info on what to do on the polycom? -- should I contact polycom? |
| 13:59.56 | [TK]D-Fender | hackeron: What actions on your phone are you taking that you believe would lead to subscribing to this hint in the first place? |
| 14:00.18 | hackeron | [TK]D-Fender: I thought it was part of the SIP specification and should "just work" |
| 14:00.33 | [TK]D-Fender | hackeron: Not an answer to my question, try again... |
| 14:00.42 | hackeron | [TK]D-Fender: nothing :) |
| 14:00.58 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
| 14:01.07 | [TK]D-Fender | hackeron: You mean you expect your phone to magically know it should be looking at that hint? |
| 14:01.57 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 14:02.10 | hackeron | [TK]D-Fender: hmmm, kinda, lol - the wiki says it "works" on polycom phones and I can't find any details on what to do on the phone so far - I'll keep looking |
| 14:02.27 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:02.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:02.34 | grandpapadot | hackeron: "Watch Buddies" |
| 14:03.00 | [TK]D-Fender | hackeron: You didn't even tell your phone to ask * for that info. |
| 14:03.54 | [TK]D-Fender | hackeron: Yes, go read the WIKI on how to provision your phone and set up presence. And BTW, on a 301 you are not going to get any "live" indication on-screen for this once you do configure it correctly anyways. |
| 14:04.07 | pukkita | [TK]D-Fender: the alarm on the BRI bank still lights... and HDLC Bad FCS (8) on Primary D-channel of span 4 |
| 14:04.14 | [TK]D-Fender | hackeron: Your users will have to actually go through a screen or 2 to see it if they care. |
| 14:04.26 | [TK]D-Fender | pukkita: Thats typically a timing error.... |
| 14:05.09 | pukkita | [TK]D-Fender is to be extected to get timing from it then? should i put span=4,2? |
| 14:05.21 | hackeron | [TK]D-Fender: oh? - what can I use to get a live indication? |
| 14:05.44 | [TK]D-Fender | pukkita: your CB's should be "X,0,0", and your telco's "X,[1,2,3,4],0" |
| 14:06.02 | [TK]D-Fender | hackeron: Another phone. |
| 14:06.41 | pukkita | [TK]D-Fender: then where could that timing error come from? |
| 14:06.52 | hackeron | [TK]D-Fender: hmm, the manual says I can use SLA to show live indications for this kind of thing - is that the only way? :( |
| 14:07.02 | *** join/#asterisk qdk (n=qdk@195.184.98.6) |
| 14:07.17 | [TK]D-Fender | pukkita: uncertain. I think I've reached the limit of my ability to advise you. |
| 14:08.16 | [TK]D-Fender | ANYONE : pukkita here has a 4-port E1 PRI crad with 2 telco, and a BRI CB. PB's are : http://pastebin.com/m75b2a779 , http://pastebin.com/m46693b36 , http://pastebin.com/m2134ead6 , http://pastebin.com/m2da75104 |
| 14:09.11 | [TK]D-Fender | hackeron: * doesn't support SLA, and the 301 will not display your presence info live, PERIOD |
| 14:09.19 | pukkita | [TK]D-Fender: thanks a lot anyway :) |
| 14:09.37 | santoshr | I am using asterisk 1.4.18.1 on a pentium 4 3.00GHz machine. running only asterisk and gnugk. with 2gb men and 1 gb swap. No X or any such strong application. the problem is that asterisk dies very often. |
| 14:10.01 | santoshr | Is there a way for it to dump the cpu utilization or mem info before it dies. |
| 14:10.02 | [TK]D-Fender | pukkita: Yeah, at least I kept your PB's close together. Your setup is very clean so it shouldn't be too hard for someone more knowledgble to pick things up from here. |
| 14:10.09 | hackeron | [TK]D-Fender: I see, but is there anyway to use a different indicator on the phone for this like I donno, the voicemail indicator? |
| 14:10.30 | [TK]D-Fender | hackeron: Nothing vaguely realistic. |
| 14:10.49 | grandpapadot | hackeron: What phone? |
| 14:10.58 | hackeron | grandpapadot: Soundpoint IP 301 |
| 14:11.02 | hackeron | grandpapadot: (polycom) |
| 14:11.32 | grandpapadot | hackeron: Well, you're not going to have a lot of fun with what you're trying to do and that phone. Your best bet is to get a 601 and the associated Operator Console. |
| 14:12.11 | pukkita | [TK]D-Fender: I think I found it!!! gonna test in seconds, wanpipe4.conf needed the TE_REF_CLOCK to 1 in order to pass the timing from the telco to the channel bank |
| 14:12.25 | [TK]D-Fender | pukkita: Must have missed that... |
| 14:12.53 | hackeron | grandpapadot: the 601 is so huge though :( - and for my need I just want it to light an LED or show some kind of indicator on the screen - I don't really need the functionality of the 601 phone |
| 14:13.08 | grandpapadot | hackeron: You basically setup "watch buddies" with the phones contacts and then setup associated hints to their extensions. When asterisk sees an extension in use, it essentially notifies the phone. If the phone is 'subscribed' via a contact's watch buddy option being enabled, and that contact is assocaited with a SLA, then the indicator (if any) next to the SLA will illuminate. |
| 14:13.29 | M1s3ry | ccvp, I looked at the link you sent... that right down the road from digium |
| 14:13.32 | grandpapadot | hackeron: Yes, you DO need the functionality of the 601, that's *exactly* what' your tring to do ... |
| 14:13.42 | M1s3ry | s/that/they are |
| 14:13.47 | [TK]D-Fender | grandpapadot: Not bad... |
| 14:13.53 | [TK]D-Fender | grandpapadot: You're learning :) |
| 14:14.03 | grandpapadot | lol, it can be tought! |
| 14:14.07 | grandpapadot | taught |
| 14:14.09 | hackeron | grandpapadot: well, isn't there a smaller/cheaper phone than the 601 that can light an LED on demand, lol? |
| 14:14.14 | [TK]D-Fender | "it" indeed |
| 14:14.33 | [TK]D-Fender | hackeron: IP 430 /501 (on-screen, but not LED) |
| 14:15.11 | [TK]D-Fender | hackeron: I would advise the IP 501 in this case. |
| 14:15.15 | hackeron | [TK]D-Fender: wow, the 430 looks identical to the 310 - so they just released the same phone without crippled firmware? |
| 14:15.59 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 14:16.09 | [TK]D-Fender | hackeron: Lit indicators, PoE native, Pixel display, MicroBrowser capable. |
| 14:17.36 | hackeron | [TK]D-Fender: but can't lit indicators be done in firmware? |
| 14:18.05 | [TK]D-Fender | hackeron: Sure.... YOU go rwite your own firmware for your phone.... |
| 14:18.13 | hackeron | [TK]D-Fender: lol, or whine to polycom about it |
| 14:18.18 | EmleyMoor | I know you can set the caller pres (with SetCallerPres(prohib) for example) but is there any way you can read that setting? |
| 14:18.37 | [TK]D-Fender | hackeron: and "onscreen" yes, but LIT, no. there are no LED's for your line keys on a 301 |
| 14:19.20 | hackeron | [TK]D-Fender: hmm, there's a green LED that lights up for line1 and line2 whenever it's in use, and it blinks red when there's a call on hold |
| 14:19.34 | hackeron | [TK]D-Fender: (by in use I mean I'm using it, not SLA) |
| 14:19.57 | [TK]D-Fender | hackeron: Was there? |
| 14:20.17 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 14:20.24 | hackeron | [TK]D-Fender: oh wait a second, I'm such an idiot, these are IP430 phones!! |
| 14:21.06 | hackeron | woops, lol |
| 14:21.25 | [TK]D-Fender | hackeron: Yes, and Yes. |
| 14:21.26 | [TK]D-Fender | ~cluebat hackeron |
| 14:21.27 | jbot | ACTION pulls out a ClueBat (tm) and thwaps hackeron. |
| 14:21.37 | hackeron | deserves it |
| 14:21.57 | [TK]D-Fender | hackeron: Now go read the provisioning guide on the WIKI for enabling presence and setting up your buddies |
| 14:22.34 | [TK]D-Fender | There goes another 25 minutes of my life I'll never get back... |
| 14:22.42 | grandpapadot | rofl |
| 14:23.04 | hackeron | [TK]D-Fender: lol, sorry :( -- do you have a paypal account? -- let me buy you a beer or something for my utter stupidity |
| 14:23.46 | [TK]D-Fender | hackeron: Save it up, and when you need something bigger (like a course on all of this) you can add that in. |
| 14:24.30 | hackeron | [TK]D-Fender: :) - yes, and an asterisk book for dummies |
| 14:26.48 | keith4 | wow, what little confidence I had in telephony depot is fading fast |
| 14:28.04 | keith4 | I ordered a TDM410P with 2 FXO ports... they sent me the card and a quad FXO module for the TDM2400 |
| 14:28.44 | keith4 | maybe I should just sell it on ebay and buy 2 FXO modules |
| 14:29.14 | EmleyMoor | SetCallerPres - is it effective on conections other than Q931? |
| 14:29.48 | EmleyMoor | It certainly appears to! |
| 14:30.51 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 14:31.34 | lmadsen | keith4: if you can make money off of it, that might make sense :) |
| 14:32.05 | keith4 | well, they sell the quad FXO for $175... and the single FXOs are about $44 |
| 14:32.51 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 14:33.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:33.47 | keith4 | does anyone want to buy a quad fxo module for about $120? ;-) |
| 14:33.59 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:34.30 | keith4 | it looks like there isn't much of a market for these on ebay |
| 14:35.44 | *** join/#asterisk koukai (n=cel@77.193.132.57) |
| 14:36.15 | M1s3ry | keith4, tell em they messed up... they should replace it for the correct items |
| 14:36.23 | M1s3ry | hopefully that is |
| 14:36.43 | keith4 | yah, i did... i'm just wondering if I can profit from this somehow |
| 14:36.56 | M1s3ry | ahh... |
| 14:37.10 | keith4 | like, return it, and buy the 2 FXOs instead |
| 14:37.24 | EmleyMoor | If you could get a quick sale on it, you could |
| 14:37.36 | EmleyMoor | keith4: Have they documented that they've sent you that module? |
| 14:37.45 | *** join/#asterisk asdx (n=diego@adsl-148-95.click.com.py) |
| 14:37.51 | keith4 | yes. the packing slip shows what they sent |
| 14:38.01 | keith4 | although the confirmation email from my order shows otherwise |
| 14:38.06 | EmleyMoor | In that case they ought to refund on it |
| 14:38.11 | *** join/#asterisk blq (n=Bl@dslb-088-066-254-210.pools.arcor-ip.net) |
| 14:38.12 | keith4 | i.e. it shows what I actually ordered |
| 14:38.26 | EmleyMoor | Could be tricky |
| 14:38.53 | keith4 | it's almost a moot point. the hardware is for a client, and i don't have time to dick around anyway |
| 14:38.56 | M1s3ry | wonders how they thing a quad fxo would work with a TDM410 |
| 14:38.57 | grandpapadot | Take 1000 and add 40 to it. Now add another 1000 . |
| 14:38.57 | grandpapadot | Now add 30 . |
| 14:38.57 | grandpapadot | Add another 1000 . Now add 20 . Now add another 1000 |
| 14:38.57 | grandpapadot | Now add 10 . What is the total? |
| 14:38.59 | EmleyMoor | Certainly, they have to resolve it without cost to you - as for money back, I don't know |
| 14:39.08 | grandpapadot | oh sh*t, sorry, wrong window. |
| 14:39.21 | keith4 | 5000? |
| 14:39.23 | EmleyMoor | 4100 |
| 14:39.28 | grandpapadot | lol, 4100 |
| 14:39.30 | grandpapadot | sorry ... |
| 14:39.46 | grandpapadot | I got 5000 the first time too ... |
| 14:39.56 | EmleyMoor | It's easily done |
| 14:40.24 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 14:41.02 | *** join/#asterisk bOngFrOG (n=winston@c-67-166-37-20.hsd1.co.comcast.net) |
| 14:41.14 | keith4 | oh |
| 14:41.16 | keith4 | that's tricky |
| 14:43.26 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 14:43.41 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 14:43.54 | EmleyMoor | Is there a way to see what the current setting of CallerPres is? |
| 14:44.19 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 14:44.22 | *** join/#asterisk juanjoc (n=juanjoc@host30.190-31-245.telecom.net.ar) |
| 14:45.07 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 14:45.07 | *** mode/#asterisk [+o twisted] by ChanServ |
| 14:45.42 | *** join/#asterisk Marquel (n=Marquel@port-247.pppoe.wtnet.de) |
| 14:45.53 | Zeeek | tired but happy |
| 14:45.58 | Zeeek | thanks for asking |
| 14:46.11 | Marquel | morning |
| 14:46.23 | IceHeat | I'm trying to forward all calls from a zap channel to default extension of another asterisk server. I keep getting this message on the sending server and I can't figure it out. -- Executing [s@incoming:3] Dial("Zap/1-1", "IAX2/IAX393/310| 60") in new stack |
| 14:46.23 | IceHeat | <PROTECTED> |
| 14:46.23 | IceHeat | <PROTECTED> |
| 14:46.23 | IceHeat | <PROTECTED> |
| 14:46.45 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 14:46.58 | IceHeat | I tried the guide on dual asterisk server but it is not working for me |
| 14:47.59 | IceHeat | my extensions.conf have [incoming] |
| 14:48.00 | IceHeat | ; incoming calls from the FXO port are directed to this context |
| 14:48.00 | IceHeat | ;from zapata.conf |
| 14:48.00 | IceHeat | exten => s,1, NoOp(Hello) |
| 14:48.00 | IceHeat | exten => s,n, Answer() |
| 14:48.00 | IceHeat | exten => s,n, Dial(IAX2/IAX393/310, 60) |
| 14:48.35 | EmleyMoor | What is IAX393? |
| 14:48.40 | IceHeat | and I have [IAX393] |
| 14:48.41 | IceHeat | type=friend |
| 14:48.41 | IceHeat | host=dynamic |
| 14:48.41 | IceHeat | trunk=yes |
| 14:48.41 | IceHeat | secret=****** |
| 14:48.41 | IceHeat | mailbox=1234 |
| 14:48.43 | IceHeat | context=incoming on my iax.conf |
| 14:49.08 | [TK]D-Fender | IceHeat: PASTEBIN <---- |
| 14:49.10 | [TK]D-Fender | ~pb |
| 14:49.10 | 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 |
| 14:49.15 | [TK]D-Fender | IceHeat: do not spam like that again. |
| 14:49.26 | IceHeat | sorry |
| 14:49.42 | [TK]D-Fender | IceHeat: show us that IAX393 is registered. Then if it is, do the call again with IAX debug enabled |
| 14:51.09 | [TK]D-Fender | \o/ -- Friday counter-productivity day |
| 14:51.18 | Zeeek | spam is not the correct terminology for flooding |
| 14:51.48 | Zeeek | that would be like calling all VoIP SIP |
| 14:51.57 | EmleyMoor | "spew" is closer |
| 14:52.12 | Zeeek | flood is perfectly adequate |
| 14:53.24 | [TK]D-Fender | Zeeek: Yes, by any other name.... |
| 14:53.33 | keith4 | spams |
| 14:53.47 | [TK]D-Fender | Zeeek: its a Brand-hijacking anyways, what a minor liberal stretch? |
| 14:54.01 | Zeeek | sends penis enlargement information unencrypted to [TK]D-Fender |
| 14:54.25 | Zeeek | Today, I am tasting Tyrells home cooked beet root, parsnip and carrot chips for you. And I find them darn good, albeit a little greasy |
| 14:54.30 | [TK]D-Fender | Zeeek: As long as it isn't compressed ;) |
| 14:54.48 | Zeeek | np, tyhe balls per second rate is very low |
| 14:55.00 | *** join/#asterisk TheSov (n=a@204-16-231-218-risingmedical.pkh.ord.sparkplugbb.net) |
| 14:56.02 | Zeeek | Interesting: http://demo.voiceroute.org |
| 14:56.08 | TheSov | how does one create a pool of random outbound voip registrations to use when dialing out? |
| 14:57.01 | TheSov | and if not random just ones not being used |
| 14:57.09 | [TK]D-Fender | TheSov: its your dialplan, go pick a random # and dial out your provider of choice based on it. |
| 14:57.56 | TheSov | how do i select the outbound line? |
| 14:58.20 | Zeeek | why random? |
| 14:58.36 | [TK]D-Fender | TheSov: "select"? You just "Dial" |
| 14:58.57 | Marquel | i'm trying to figure out a "best mode" for switching on and of an unconditional forward to an (only seldom re)programmed. I'd favor a way with only one number to dial for switching the forward on and off, and i'd like to have a blf-lamp on a phone displaying the state of the forward. any hints? |
| 14:59.00 | TheSov | doesnt have to be random, just ones not being used right now my voip provider just charged for having 2 outbound connections on the same sip line |
| 14:59.36 | [TK]D-Fender | TheSov: "sip line"? |
| 14:59.57 | TheSov | yeah u know voip/voip no pots lines |
| 15:00.26 | [TK]D-Fender | TheSov: Well go check whatever you have the check to decide which you want to use, and use use it. |
| 15:00.41 | Zeeek | Marquel I'd love to see more lamps on phones that we could do use as desired |
| 15:01.02 | [TK]D-Fender | Marquel: in 1.4 use DevState in conjunction with something llike AstDB to store your state and forwarding info. |
| 15:01.03 | [TK]D-Fender | ~devstate |
| 15:01.04 | jbot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/ |
| 15:01.52 | Marquel | Zeeek: so far i've found some bits which might build a solution, but since i'm rather new to asterisk, i'm not sure how to implement it fitting my needs... :/ |
| 15:03.37 | Marquel | [TK]D-Fender: this is 1.2 but w/ bristuff so should have DevState too. and i've seen something using AstDB, but it was unfortunately only a snippet of code :/ |
| 15:04.17 | [TK]D-Fender | Marquel: set a value in AstDB to say "yup, I'm forwarded", and then change the BLF accordingly. |
| 15:04.53 | [TK]D-Fender | Marquel: Set(DB(forwarding/myphone)=yup) |
| 15:05.02 | Zeeek | To second [TK]D-Fender, that's what I do with a zillion little flags on our pbx, like how many rings, which phones to ring, when to allow DISA, etc |
| 15:05.19 | Zeeek | astdb is one of the greatest features of asterisk |
| 15:05.30 | Marquel | [TK]D-Fender: understood, and i think i will figure out how to do that (and reset the db), but i'm not sure how to call that codesnippet... |
| 15:05.43 | [TK]D-Fender | Marquel: GotoiIf($["${DB(forwarding/myphone)}"="yup"]?100) |
| 15:06.20 | [TK]D-Fender | Marquel: Now how to actually SET the BLF via BRIStuff DevState, well that you're going to have to hit oher docs for. |
| 15:06.28 | [TK]D-Fender | Marquel: There are a few docs on the WIKI for this |
| 15:06.29 | Zeeek | then for output use a status extension that reads you the flags and talkis |
| 15:07.34 | Marquel | [TK]D-Fender: as said: i think i can figure out setting the db, but how to actually call that code (and how to tell asterisk to use it on incoming calls)? |
| 15:07.44 | [TK]D-Fender | Zeeek: I made a 300-line or so super macro (and setup stuff too) that used a LOT of AstDB to let users completely configure how to forward, set rings, deal with no-answer / busy, etc. All dynamic |
| 15:08.13 | [TK]D-Fender | Marquel: I jsut showed you how to check it on incoming calls in the GotoIf line I gave you |
| 15:09.12 | Zeeek | I use flags sometimes in order to reduce the number of db reads |
| 15:09.51 | Marquel | [TK]D-Fender: ah, okay. sorry, misunderstood you there. i understand to set the db is done in AEL, is it sufficient to simply put a regexp into extensions.ael to call it from the user? |
| 15:09.58 | Zeeek | for example we have two lines 1 and 2 if both are enabled, the value returned is 3 |
| 15:10.22 | [TK]D-Fender | Marquel: AEL isn't anything special. Its all just dialplan. |
| 15:11.40 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 15:12.03 | lmadsen | ~non-root |
| 15:12.03 | jbot | methinks non-root is what you should irc as |
| 15:12.13 | lmadsen | ~asterisk-non-root |
| 15:12.47 | lmadsen | jbot: asterisk-non-root is an article written by Leif Madsen at http://www.taug.ca/node/115 |
| 15:12.47 | jbot | lmadsen: okay |
| 15:13.20 | lmadsen | jbot: no, asterisk-non-root is how you should be running your Asterisk boxes in production. A useful article is available at http://www.taug.ca/node/115 |
| 15:13.21 | jbot | lmadsen: okay |
| 15:13.24 | *** join/#asterisk IceHeat (n=ammar@77.110.81.31) |
| 15:13.56 | Marquel | [TK]D-Fender: so if i put something like "_*21X. => {[...]}" into extensions.ael it'll be executed whenever a user dials '*21something' on his phone? |
| 15:14.04 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 15:14.07 | IceHeat | I do apologize again for the earlier flooding, |
| 15:14.36 | lmadsen | [TK]D-Fender: think that is a useful tag? 'asterisk-non-root' ? |
| 15:15.26 | IceHeat | I did find that IAX393 is not registered, the server is returning to me with "unable to load config iax.conf" |
| 15:16.00 | IceHeat | I'll try to figure that out, if I can't, I might need your help again |
| 15:16.07 | IceHeat | thanks for the tip |
| 15:16.15 | Zeeek | don't be root, be beetroot |
| 15:16.34 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:16.34 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:16.39 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:16.56 | [TK]D-Fender | ~asterisk-non-root |
| 15:16.57 | jbot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 |
| 15:17.30 | [TK]D-Fender | Marquel: I neither use, nor advise AEL |
| 15:17.40 | [TK]D-Fender | lmadsen: Indeed, and improved :) |
| 15:18.05 | lmadsen | [TK]D-Fender: nice! |
| 15:18.11 | Zeeek | [TK]D-Fender do you use alcohol? |
| 15:18.20 | lmadsen | only in my dragsters |
| 15:18.58 | [TK]D-Fender | Zeeek: Sure, gets those tough spots off my scanner glass :) |
| 15:19.16 | Zeeek | so how can anyone take advice seriously? |
| 15:19.35 | Marquel | [TK]D-Fender: hmmm... wouldn't that make it more complicated to read and act according to database? |
| 15:19.50 | [TK]D-Fender | Zeeek: I'm perfectly serious. Perhaps you should refine your questions more :) |
| 15:20.03 | [TK]D-Fender | Marquel: Not really. |
| 15:20.50 | *** join/#asterisk bschrock6962 (n=bschrock@66-195-4-130.static.twtelecom.net) |
| 15:20.58 | Zeeek | I have no time for the socratic method |
| 15:21.28 | Zeeek | That's Leif! |
| 15:21.40 | [TK]D-Fender | na na, na na na |
| 15:22.26 | EmleyMoor | I am currently rewriting my dial plan due to a price change and the arrival of some new numbers. |
| 15:23.34 | *** join/#asterisk gormux (n=julien@asuka.genux.info) |
| 15:23.36 | gormux | hello |
| 15:23.42 | Zeeek | I can never remember all the extens I created to do stuff on my pbx |
| 15:23.57 | Zeeek | including sending it SMS to call me back with vmail or DISA prompts |
| 15:24.37 | Zeeek | all those speed dials with automon |
| 15:25.00 | gormux | i have a strage thing here : i have 2 asterisk boxes running, one has port 5060 in tcp and udp, and another one just opened on udp |
| 15:25.19 | gormux | does someone know what could cause that ? |
| 15:25.32 | [TK]D-Fender | gormux: * 1.4- does not support SIP over TCP |
| 15:25.35 | *** join/#asterisk xacatecas (n=jkroon@dsl-241-130-116.telkomadsl.co.za) |
| 15:25.56 | xacatecas | hi folks, hope all are well. |
| 15:26.18 | Zeeek | considering we're all gonna die |
| 15:26.50 | [TK]D-Fender | Zeeek: Life is a sexually transmitted disease which is in all cases fatal. |
| 15:27.06 | gormux | 1.2.12.1-1mdv2007.0 for the one that has both, and SVN-branch-1.2-r96931 for the one that has just udp |
| 15:27.24 | [TK]D-Fender | gormux: Hence no TCP. |
| 15:27.50 | gormux | so i should use another version of *... |
| 15:27.53 | gormux | ok |
| 15:27.55 | Zeeek | RFCs seem so pathetically irrelevent in that light, [TK]D-Fender |
| 15:28.24 | *** join/#asterisk pgarcia (n=pedgarci@yoda.kanatek.com) |
| 15:28.35 | [TK]D-Fender | gormux: Only 1.6 supports TCP and its nowhere near stable or in full release yet |
| 15:28.35 | Marquel | [TK]D-Fender: thanks for your time and advice - i'll come back to you when i'm dead for my customer crushing my head (or any other life-important organ of mine)... |
| 15:30.18 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:30.21 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
| 15:31.09 | xacatecas | i've got a strange situation, so i'm hoping someone can help. three asterisk servers involved, cl1 (public IP) <--> switch (static IP) <--> cl2 (private IP: NAT). cl1 only supports speex, switch does both speex+g729, cl2 does both speex+g729. When I make a call from cl2 -> cl1, the switch does g729 to cl2, and then transcodes to speex on the other end ... is there any way to avoid that? speex is supported right through, so why n |
| 15:31.09 | xacatecas | ot use it right through? |
| 15:31.29 | *** join/#asterisk mmelbert (i=d1c3877a@gateway/web/ajax/mibbit.com/x-9cdc9e17ac2d1d4f) |
| 15:33.25 | gormux | [TK]D-Fender: this one is a 1.2 and listens on tcp 1.2.12.1-1mdv2007.0 |
| 15:33.27 | Zeeek | asterspam: http://x2z.eu in 30 minutes, conference on DRUID 1.0.2 very cool demo. |
| 15:34.08 | mmelbert | I am having trouble with some of my phone not UNregistering when I change the extension info on them. I have a polycom phone that I am changing the extension from 211 to 252 using the polycom gui interface. After I change the extension on the phone I can call both the old extension and the new extension and it rings to that phone. I do not have this problem with my dlink-dph-14s phone. We are using openasterisk 1.4. |
| 15:34.31 | Zeeek | Theremme you need to edit the config file |
| 15:35.34 | [TK]D-Fender | gormux: Makes no sense unless you're running some non-mainstream patch. |
| 15:35.44 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 15:36.05 | Zeeek | mmelbert the polycom needs its XML config file changed |
| 15:36.16 | gormux | [TK]D-Fender: ah i just found that /etc/asterisk/version was certainly from an old install |
| 15:36.37 | gormux | version is 1.4.11-2mdv2008.0 in fact |
| 15:37.01 | mmelbert | Zeeek: we are not using a tftp server to have the polycoms get their config...it should just be getting it from the internal settings |
| 15:37.32 | Zeeek | but the polycom gui is the world's worst |
| 15:37.42 | Zeeek | at least use ftp |
| 15:37.44 | dominic1 | I have a problem: My new asterisk isn't listening to any ports |
| 15:37.47 | [TK]D-Fender | xacatecas: You should restrict your codecs and set the preference order |
| 15:38.09 | [TK]D-Fender | dominic1: Go check if your channel driver modules are even laoded. |
| 15:38.29 | pgarcia | Hi, I remember to test/use *call parking* (700 ext) long time ago and when I put someone on parking, I could hear the park slot number (701,etc). Today I'm testing it again and it does't announce the slot anymore (The call was parked correctly). Anyone is aware about this? (Asterisk 1.4.18.1 zaptel 1.4.9.2). |
| 15:38.39 | dominic1 | how can I check that? |
| 15:38.47 | xacatecas | I've got "disallow=all", "allow=g729" and "allow=speex" on both ends of the trunk supporting both, and exactly the same except for the g729 line on the trunk that doesn't. |
| 15:39.09 | mmelbert | Zeeek: thanks for your help...but is there any reason that the phone shouldn't be unregistering? |
| 15:39.28 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 15:39.30 | Zeeek | you may need to treload SIP |
| 15:39.50 | Zeeek | IOW, once the poly is rebooted |
| 15:40.14 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 15:40.26 | xacatecas | does the poly unregister on reboot? |
| 15:40.35 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:40.37 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:40.47 | dandre | hello, |
| 15:41.04 | [TK]D-Fender | dominic1: Go load the module and see if ti starts listening. |
| 15:41.05 | Zeeek | I'm no expert but I'd have though that a SIP reoload on the asterisk side woul dsolve that |
| 15:41.07 | dandre | can an misdn port bellong to more than one group? |
| 15:41.24 | b11d` | if I had 10 analog phones to connect to an asterisk box, and the client refused to go voip (wanting to keep his desk phones) -- what kind of ATA should I be looking at? Rhino T1 based CB's seem like overkill |
| 15:41.25 | [TK]D-Fender | Zeeek: If chan_sip isn't loaded he's not going to even have that options. |
| 15:41.25 | rupa | sip registrations are persisted |
| 15:41.49 | Zeeek | wait I'm confused. What is the phone registering *to* ? |
| 15:41.53 | [TK]D-Fender | b11d`: Linksys SPA-8000 + SPA-2102 |
| 15:42.43 | *** join/#asterisk axisys (i=iqbala@otaku.freeshell.ORG) |
| 15:42.46 | xacatecas | ok, swapping the two allow lines around fixes that case, now I've got the opposite problem. my upstream break-out for every number that doesn't belong to one of my clients only supports g729, if the caller has a g729 codec I'd prefer to use that. |
| 15:42.47 | b11d` | how many lines does that support? or just a single line per device? or should I STFU and go look it up? |
| 15:42.48 | [TK]D-Fender | b11d`: SPA-8000 = 8 FXS @ $235. SPA-2102 = 2 FXS @ $70 |
| 15:42.56 | b11d` | ahhh |
| 15:43.26 | b11d` | thanks for the advice TK |
| 15:43.42 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 15:43.47 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:44.11 | xacatecas | ok, so then, is it possible to say something like "if two clients calls each other, prefer speex, if a client breaks out, or an external call comes in for a client, prefer g729"? |
| 15:44.52 | Zeeek | this will the be almost last call for participants for the VoIP Users Conference in 15 min. http://x2z.eu. It began life as the Asterisk USers Conference. Asterisk is a registered trademark of Digium. Thanks for all the fish. 43 |
| 15:44.59 | mmelbert | ok...here is the situation. I originally have the phone registered as 211. I then go in and change the phone, using the gui, to 252. After the phone reboots, and I never see an unregister during the reboot, I do a si show peers and I see both 211 and 252 registered to the same IP address. And no..I am not configuring line 2. |
| 15:45.45 | xacatecas | mmelbert, 211 will time out eventually. |
| 15:46.34 | Zeeek | will a SIP reload NOT remove it? |
| 15:47.12 | xacatecas | no. |
| 15:47.34 | mmelbert | do you know how long the timeout is? |
| 15:47.37 | xacatecas | imagine it did ... your asterisk server suddenly forgets where all the extensions are. |
| 15:47.45 | xacatecas | mmelbert, depends on the registration period. |
| 15:47.54 | xacatecas | show sip peer 211 should tell you. |
| 15:48.18 | Zeeek | xacatecas good point! |
| 15:48.26 | xacatecas | sip show peer 211 sorry |
| 15:48.42 | Zeeek | not "core ..." ? |
| 15:49.08 | lmadsen | why core? |
| 15:49.10 | lmadsen | sip is a module |
| 15:49.21 | lmadsen | <module> <verb> <arguments> |
| 15:49.24 | mmelbert | so the "expire" setting is the timeout? |
| 15:49.26 | Zeeek | I dunno I still run 1.2 :) |
| 15:50.52 | mmelbert | it still is disconcerting that the polycom phones are not sending a "unregister" notification |
| 15:51.05 | russellb | Zeeek: who is talking |
| 15:51.47 | Zeeek | mz right now |
| 15:52.09 | russellb | mz o.O |
| 15:52.17 | Zeeek | me |
| 15:52.20 | russellb | ah |
| 15:54.00 | Zeeek | and DRUID OSE 1.0.2 |
| 15:57.59 | mmelbert | so no ideas on why asterisk is not getting polycom's unregister? The answer is just wait for the other extension to timeout? |
| 15:58.00 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:59.00 | [TK]D-Fender | mmelbert: Just remove the register info from AstDB |
| 15:59.41 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 15:59.44 | Telemac | Hello |
| 16:00.47 | lmadsen | anyone happen to know if a variable is set when automon is utilized by a phone? |
| 16:01.04 | lmadsen | i.e. some method to know that someone started recording a call with automon? |
| 16:01.25 | [TK]D-Fender | lmadsen: think an AMI message might be sent out... |
| 16:01.29 | Telemac | I've read that timeout option for queue is the time to ring on each member/agent. Is there any way to force the queue "global timeout" after which * go to next priority ? |
| 16:01.43 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:01.53 | lmadsen | [TK]D-Fender: hrmm... I wanna do this in the 'h' extension since I need to execute dialplan after the call hangs up to clean up the recording |
| 16:02.10 | lmadsen | Telemac: core show application queue |
| 16:02.29 | lmadsen | there is a [timeout] option you can pass to the Queue() application |
| 16:02.48 | [TK]D-Fender | lmadsen: do a dump on the channel.... don't forget a lot of stuff can't be accessed under "h" |
| 16:03.08 | [TK]D-Fender | Telemac: "core show application queue" |
| 16:03.24 | xacatecas | ok, so no solutions for the codec preference thing based on routing? |
| 16:03.32 | lmadsen | [TK]D-Fender: I know, I just need to know that a recording happened though... but you're right, there might be a variable set with the filename or something |
| 16:03.38 | Telemac | lmadsen: I know but it keep ringing after that tmout, even with 'n' (no timeoutrestart) |
| 16:04.29 | [TK]D-Fender | Telemac: You mean if its rining agent #3 and the timeout gets hit it jsut keeps ringing that person indefinitely, but leaves the "queue"? |
| 16:05.43 | Telemac | I mean it doesn't seem to leave the queue, next priority isn't fired |
| 16:05.53 | lmadsen | [TK]D-Fender: TOUCH_MONITOR_OUTPUT... perfect :) |
| 16:07.04 | [TK]D-Fender | Telemac: show us your queue line and "show application queue in a PASTEBIN |
| 16:07.06 | [TK]D-Fender | ~pb |
| 16:07.07 | 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 |
| 16:07.10 | [TK]D-Fender | ^^^^^^^^^ |
| 16:07.15 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-155-11.dsl.stlsmo.sbcglobal.net) |
| 16:07.46 | LemensTS | hey i have an asterisk that went down and wont come back up. I did ztcfg....then safe_asterisk....it keeps trying to restart over and over |
| 16:08.34 | [TK]D-Fender | LemensTS: Sopt in, and run it manually to see what the error is |
| 16:10.00 | LemensTS | tkd-fender: i dont follow you? |
| 16:10.06 | LemensTS | sopt in? |
| 16:10.24 | [TK]D-Fender | LemensTS: Stop safe_asterisk. Then run * MANUALLY to see what the error is |
| 16:10.32 | [TK]D-Fender | *stop |
| 16:10.46 | LemensTS | btw this is asterisk@home heh |
| 16:10.59 | LemensTS | rpath i think |
| 16:11.03 | *** join/#asterisk ManxPower (n=manxpowe@196.sub-70-222-150.myvzw.com) |
| 16:11.28 | *** join/#asterisk crakrjak (n=merc@cpe-75-187-108-229.insight.res.rr.com) |
| 16:11.36 | *** part/#asterisk ManxPower (n=manxpowe@196.sub-70-222-150.myvzw.com) |
| 16:11.59 | *** join/#asterisk ManxPower (n=manxpowe@196.sub-70-222-150.myvzw.com) |
| 16:12.06 | LemensTS | when i do: asterisk_safe stop, Asterest ended with exit status 1, Asterisk died with code1. Automatically restarting asterisk. |
| 16:12.11 | LemensTS | over and over and over |
| 16:12.19 | [TK]D-Fender | LemensTS: Kill the process |
| 16:12.49 | ManxPower | I didn't think safe_asterisk suppoered a "stop" option |
| 16:13.12 | [TK]D-Fender | ManxPower: Welcome to "Invented Syntax World"! |
| 16:13.45 | ManxPower | [TK]D-Fender: Nifty! /me invents 5 million dollars in his bank account! |
| 16:13.52 | ManxPower | Hrm.. Why does that never work for me? |
| 16:14.35 | [TK]D-Fender | ManxPower: Planetary alignment failure.... next event in 281 years... |
| 16:15.04 | b11d` | lol |
| 16:15.19 | gormux | [TK]D-Fender: hum, i installed 1.4.19.1, and still no tcp, and there is one on 1.4.11-2mdv2008.0 ... |
| 16:15.33 | gormux | is mandriva version patched ? |
| 16:15.48 | [TK]D-Fender | gormux: as I've said numerous times, only * 1.6+ truely supports TCP |
| 16:15.51 | ManxPower | gormux: patches? |
| 16:16.37 | ManxPower | [TK]D-Fender: Oh! It's ignore the smart people day today! |
| 16:16.56 | [TK]D-Fender | ManxPower: Its not just a "day" anymore... |
| 16:16.59 | *** join/#asterisk fadumpt (n=chatzill@adsl-070-154-035-081.sip.gsp.bellsouth.net) |
| 16:17.23 | fadumpt | Is it possible to disable incoming calls on a zaptel trunk? |
| 16:17.32 | smash- | durka |
| 16:17.35 | ManxPower | fadumpt: yes and no. |
| 16:17.39 | smash- | sup tk |
| 16:17.41 | fadumpt | :) |
| 16:17.47 | fadumpt | where would I start looking for that? |
| 16:17.56 | LemensTS | pheww i killed safe_asterisk now what |
| 16:17.57 | fadumpt | I'd really like for all calls to go out over the voip trunk |
| 16:17.59 | ManxPower | fadumpt: you can disable the channel, but you can't enable it without restarting Asterisk |
| 16:18.15 | ManxPower | fadumpt: Huh? You said INCOMING then you said OUTGOING. Which is it? |
| 16:18.36 | fadumpt | incoming on zaptel is bad, outgoing on voip and zaptel is good |
| 16:18.52 | ManxPower | fadumpt: you cannot just disable one direction. |
| 16:19.06 | [TK]D-Fender | LemensTS: .. |
| 16:19.08 | [TK]D-Fender | ~nowwhat |
| 16:19.09 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 16:19.14 | *** join/#asterisk bmg505 (n=leon@196-209-77-112-tbnb-esr-2.dynamic.isadsl.co.za) |
| 16:19.15 | ManxPower | you could make the incoming context for the zaptel channels bea context that just hangs up the calls |
| 16:20.22 | fadumpt | well, my situation is that the PSTN line is used by a house as well as a business, they have distinctive ring on the line so it rings differently depending on whether it's for house or business but if I plug it into the asterisk box, it will send all calls to IVR |
| 16:20.46 | ManxPower | fadumpt: not if you don't tell it to send calls to IVR |
| 16:21.01 | ManxPower | fadumpt: you have a dialplan design problem, not a disable only incoming channels |
| 16:21.02 | *** join/#asterisk quaqo (n=quaqo@89-97-102-198.ip17.fastwebnet.it) |
| 16:21.08 | [TK]D-Fender | fadumpt: And what calls SHOULD * be answering? |
| 16:21.20 | ManxPower | sent incoming zap calls to a context that just hangsup |
| 16:21.25 | LemensTS | tkd-fender: i tried safe_asterisk again after i killed it same thing. asterisk -r asks if that .ctl exists |
| 16:21.47 | ManxPower | LemensTS: Asterisk is not running. |
| 16:21.55 | [TK]D-Fender | LemensTS: and I told you to run * DIRECTLY. |
| 16:22.01 | ManxPower | asterisk -r reconnects to an EXISTING INSTANCE OF ASTERISK |
| 16:22.03 | fadumpt | so if i say all calls from the zaptel channel just go to a ring group, then the calling party shouldn't be the wiser about the fact that their call was just sent to regular phones and a phone system? |
| 16:22.29 | [TK]D-Fender | fadumpt: Indeed your problem is a dialplan conception failure. |
| 16:22.36 | ManxPower | fadumpt: I really can't help you if you keep changing what you want to do. I told you what will solve the issue you are having. |
| 16:22.42 | [TK]D-Fender | fadumpt: Stop sending them to an IVR and do what it is you want it to do. |
| 16:22.52 | fadumpt | alright, trying that now |
| 16:22.54 | fadumpt | thanks |
| 16:22.57 | LemensTS | tkd-fender: Ok i guess i have never started it manually, rpath is different. I would usually /etc/init.d/asterisk start |
| 16:23.14 | b11d` | SPA-8000 and 2102 on their way.. thanks TK |
| 16:23.24 | ManxPower | LemensTS: did you do a "make config" Asterisk won't install it's init scripts if you don't do that |
| 16:23.30 | [TK]D-Fender | LemensTS: No, that is an init script which has nothing to do with starting * manually. |
| 16:23.32 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 16:23.58 | ManxPower | LemensTS: "asterisk -cvvv" starts Asterisk manually. run that command |
| 16:24.00 | [TK]D-Fender | b11d`: You're welcome. jblack picked up an SPA-8000 on my proposing it and loves it. |
| 16:24.14 | [TK]D-Fender | b11d`: Its jsut like any other SPA, only bigger & with more connectors :) |
| 16:24.19 | b11d` | you steered me right with the Polycoms before, so I trust you.. :) |
| 16:24.20 | b11d` | cool |
| 16:24.45 | ManxPower | [TK]D-Fender: maybe it's time for me to retire from the channel if people are not listening to advice anymore. |
| 16:24.57 | [TK]D-Fender | b11d`: I have not had my hands on one personally, but I remote adminned it to help him. |
| 16:25.17 | [TK]D-Fender | ManxPower: Or I could run amok KB-ing people.... |
| 16:25.28 | [TK]D-Fender | ManxPower: but NAH.... tha'd ruin my rep... |
| 16:25.31 | ManxPower | [TK]D-Fender: naw, too much drama |
| 16:25.38 | mort_gib | Are the Polycoms better for * |
| 16:25.53 | ManxPower | Are oranges better for health? |
| 16:26.01 | b11d` | I love the Polycoms from an admin Point of view, and my 300 users love them from the users point of view.. |
| 16:26.02 | ManxPower | you'll notice the lack of "better than WHAT" in that question |
| 16:26.02 | [TK]D-Fender | ManxPower: No drama required ; I could do it without warning or commentary. |
| 16:26.04 | b11d` | i love the polycoms |
| 16:26.05 | mort_gib | In fact they are :-) |
| 16:26.24 | mort_gib | Ok, better than Snom ?? |
| 16:26.26 | LemensTS | manxpower: that started it but asterisk -r wont connect to it |
| 16:26.31 | ManxPower | ~phines |
| 16:26.32 | b11d` | I just wish I could get a Polycom with a built-in ATA |
| 16:26.33 | ManxPower | ~phones |
| 16:26.34 | jbot | it has been said that phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 16:26.34 | b11d` | :) |
| 16:27.37 | [TK]D-Fender | ManxPower: I seriously have to get around to rebuilding his dated list... |
| 16:27.51 | *** join/#asterisk nirz (n=nnscript@bzq-79-182-149-111.red.bezeqint.net) |
| 16:28.19 | mort_gib | I have done a few installs with Snom, it works fine |
| 16:28.50 | mort_gib | Few things are quirky |
| 16:28.58 | ManxPower | Too bad Cisco bought Linksys and not Polycom |
| 16:29.08 | *** join/#asterisk qdk (n=qdk@195.184.98.6) |
| 16:29.16 | Nugget | no doubt. |
| 16:29.19 | b11d` | what? |
| 16:29.23 | mort_gib | -don't much like Cisco |
| 16:29.24 | b11d` | dont you think they might do bad to Polycom? |
| 16:29.31 | [TK]D-Fender | ManxPower: No, GOOD THINK they didn't |
| 16:29.38 | b11d` | yeah exactly |
| 16:29.40 | b11d` | that would have been shitty |
| 16:29.43 | [TK]D-Fender | ManxPower: Stops them from $#ing up a GOOD product |
| 16:30.26 | [TK]D-Fender | ManxPower: By buying Linksys they have a chance to elevate them a bit. Polycom they could only drag down. |
| 16:30.46 | b11d` | any word when Polycom will EOL the 501 |
| 16:30.47 | b11d` | ? |
| 16:31.03 | mort_gib | I had a Cisco certified retard telling me that the command line in Linux is depreciated, while telnetting into the border router |
| 16:31.03 | Nugget | telnet is eeeeeeevil! |
| 16:31.14 | mort_gib | telnet telnet |
| 16:31.25 | [TK]D-Fender | b11d`: Not yet... |
| 16:31.28 | b11d` | all my cisco gear supports SSH thank god |
| 16:31.33 | b11d` | cool |
| 16:31.36 | [TK]D-Fender | b11d`: They are getting a bit dumb with their lineup... |
| 16:31.45 | b11d` | yeah it seems like its getting kind of bloated |
| 16:31.53 | mort_gib | It all does, but even the pro's don't enable it... |
| 16:32.36 | ManxPower | we turn off telnet and enable ssh on every router we have (all but one supports ssh) |
| 16:33.02 | mort_gib | I do the same, but I'm still not a fan |
| 16:33.10 | b11d` | I do wish that Windows would include an ssh client by default.. that'd be nice.. |
| 16:33.21 | b11d` | i'd still switch to putty, but it'd at least have the presence |
| 16:33.55 | b11d` | i suspect that the lack of a windows ssh client is the reason why telnet is still so commonly used by cisco admins.. |
| 16:33.56 | mort_gib | Yeah, but the Windows guys ARE slow in the uptake |
| 16:33.59 | b11d` | i could be wrong though |
| 16:34.18 | b11d` | lack of an included windows ssh clinet, i mean to say |
| 16:34.20 | mort_gib | Come on, Windows includes HyperTerminal |
| 16:34.22 | b11d` | lol |
| 16:35.24 | mort_gib | That REALLY gets to me! Try to run sho int on the tiny screen !! |
| 16:35.25 | b11d` | i've gotta go fix a credit card machine.. bbl lads |
| 16:35.42 | b11d` | mort_gib.. i end up capturing all output to text and looking there |
| 16:36.03 | mort_gib | Yes, but it WOULD be nice to just maximize.... |
| 16:36.27 | ManxPower | hugs PuTTY |
| 16:36.34 | LemensTS | when i do top, thier is not asterisk command running, even after i did the asterisk -cvvv |
| 16:37.08 | ManxPower | LemensTS: um, Asterisk -cvvv will start asterisk in the foreground. When you exit the app it exits |
| 16:37.19 | ManxPower | LemensTS: you don't know a damn thing about Linux, do you? |
| 16:37.34 | LemensTS | manxpower: im no expert no |
| 16:37.36 | ManxPower | ps -ax | grep asterisk |
| 16:37.38 | ManxPower | try that |
| 16:37.59 | ManxPower | while "asterisk -cvvv" us running and you are at the Asterisk console prompt |
| 16:37.59 | *** join/#asterisk draygon (n=Dustin@gateway5-pnap.exigo.com) |
| 16:38.37 | draygon | happy friday! |
| 16:39.21 | ManxPower | LemensTS: unfortunatly it takes you so long to do anything, I doubt anyone will be able to help you to a solution |
| 16:42.01 | LemensTS | manxpower: thanks for boosting my confidence |
| 16:42.06 | LemensTS | heh |
| 16:42.37 | mort_gib | LemensTS: what are you trying to do?? |
| 16:42.40 | ManxPower | *shrug* Thanks for expecting us to wait around for free. |
| 16:43.27 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 16:43.33 | ManxPower | LemensTS: Asterisk is a highly complex piece of software. To use Asterisk effectively you need a good understanding of not only Asterisk, but Linux, NAT, TCP/IP networking and telecom. |
| 16:45.02 | [TK]D-Fender | ManxPower: Oh please... he's running a GUI, so he doesn't have to know anything, right? :) |
| 16:45.04 | mort_gib | And how specific phones works, and |
| 16:45.29 | ManxPower | Yes, and understanding the protocol you decide to use (usually SIP) |
| 16:45.51 | ManxPower | [TK]D-Fender: if he's running a GUI why are we trying to help him at all? |
| 16:45.59 | LemensTS | manxpower: ive grown up on windows the past 15 years, ive been moving to linux/bsd only for a year or so....no i dont use gui |
| 16:46.30 | mort_gib | LemensTS: What are you trying to do?? |
| 16:46.34 | draygon | anyone looking for dedicated servers? I over ordered and have some extra ones.. they are on strcitly internap bandwidth |
| 16:46.36 | ManxPower | LemensTS: we are still waiting for you to report the ps output while running what I said to run |
| 16:46.43 | [TK]D-Fender | [12:10]<LemensTS>btw this is asterisk@home heh |
| 16:46.45 | [TK]D-Fender | ^^ |
| 16:47.00 | ManxPower | [TK]D-Fender: that ends my help right there. |
| 16:47.05 | ManxPower | Have a great weekend, everyone! |
| 16:47.08 | *** part/#asterisk ManxPower (n=manxpowe@196.sub-70-222-150.myvzw.com) |
| 16:47.09 | [TK]D-Fender | eek |
| 16:47.38 | outtolunc | stepped in a GUI .. eww |
| 16:47.46 | LemensTS | tkd-fender: a@h sucks i know i run all my other asterisk on debian and gentoo |
| 16:48.09 | draygon | does anyone here need a dedicated server? |
| 16:48.24 | outtolunc | i'll take one, thanks <G> |
| 16:48.32 | LemensTS | http://pastebin.com/m2644d9c6 |
| 16:48.36 | LemensTS | here is the ouptut |
| 16:48.40 | draygon | im serious |
| 16:48.43 | draygon | ... |
| 16:49.10 | LemensTS | im not at the console only ssh |
| 16:50.51 | [TK]D-Fender | LemensTS: and I told you 3 times to start * manually and see what the error is,. |
| 16:51.33 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:52.20 | LemensTS | tkd-fender: asterisk -cvvv http://pastebin.com/m8ce9670 it runs through this like it starts |
| 16:52.38 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 16:54.43 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.231.245) |
| 16:54.46 | *** part/#asterisk anonymouz666 (n=anonymou@201.19.231.245) |
| 16:55.11 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:58.03 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:06.25 | *** join/#asterisk Trapa (n=no@207.230.238.90) |
| 17:07.29 | *** join/#asterisk dahunter3 (n=dahunter@static-71-189-78-103.lsanca.dsl-w.verizon.net) |
| 17:10.26 | [TK]D-Fender | LemensTS: and thats the last line you see> |
| 17:12.04 | Trapa | Hello, I'm having a trouble with my asterisk server and sip. When two sip phones (softphones ekiga) call each other, the connection is made but no voice goes through. When the sip phone calls teh asterisk server (for say the echo test) the sip phone can hear the instructions and such, but no audio back. I suspect it might be a channels issue, but i'm a newb so i don't know |
| 17:13.23 | [TK]D-Fender | Trapa: Where are your Ekiga's located relative to your server? |
| 17:13.28 | Trapa | Same subnet |
| 17:14.14 | [TK]D-Fender | Trapa: Check your firewall on your * server, then make sure your speaker & mic are fully functional for your softphones. |
| 17:14.37 | Trapa | Is there any way to make the softphones call each other directly? And rule out the phones themselves? |
| 17:14.45 | ac1djazz | how do i list all active calls in teh CLI/AGI ? |
| 17:14.47 | Trapa | No firewall on the server |
| 17:15.03 | JayTee52 | no iptables on the server? odd |
| 17:15.50 | [TK]D-Fender | ac1djazz: "core show channels concise" |
| 17:16.17 | Trapa | Let me make sure |
| 17:16.36 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:16.55 | Trapa | iptables is there but there are no rules |
| 17:17.08 | Trapa | Input, output, forward are all ACCEPT |
| 17:18.22 | [TK]D-Fender | Trapa: If the phones are all local, then its either a local firewall on the PC's, or a speaker/mic problem. |
| 17:18.52 | Trapa | Hrm interesting. |
| 17:19.03 | Trapa | we are using vista .. *shudder* ... sigh .. |
| 17:19.12 | Trapa | well one machine is ubuntu at least |
| 17:20.06 | Trapa | The firewall on the ubuntu machine is also ACCEPT for input, output, and forward |
| 17:20.28 | Trapa | The Other machine that we're trying to call .. /shouldn't/ have firewall issues, because it can connect to other sip services and make calls no problem |
| 17:20.30 | [TK]D-Fender | Trapa: Then test that one alone with Echo |
| 17:21.10 | Trapa | *tests* |
| 17:21.47 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 17:21.50 | Skarmeth | hi all |
| 17:22.05 | Skarmeth | I am in troubles with zaptel (1.4.10) and Digium TE220B |
| 17:22.31 | Trapa | Wow .. okay that woirks .. but it's insanely .. jittery? |
| 17:22.54 | [TK]D-Fender | Trapa: that can be Ekiga in general. Try something like Linkphone,e tc for contrast |
| 17:22.55 | tzafrir_home | Skarmeth, could you please be more specific? |
| 17:23.13 | Skarmeth | I can't get Alarms green... I pluged out a Loop in each port and it don't get green... I pluged it on a E1 link and it don't get green |
| 17:23.43 | Skarmeth | it just stay blinking |
| 17:23.46 | Skarmeth | in red |
| 17:23.47 | *** join/#asterisk SamuraiDio (n=diovani@201.41.41.235) |
| 17:24.17 | SamuraiDio | do someone knows what causes this errors? http://pastebin.com/d13161e24 |
| 17:24.21 | Trapa | Fender: Whats a good softphone for windows? |
| 17:24.23 | *** join/#asterisk ka0zoz (n=ka0zoz@66.172.193.233) |
| 17:24.30 | lmadsen | Trapa: x-lite |
| 17:24.36 | lmadsen | Trapa: or zoiper |
| 17:25.09 | tzafrir_home | Skarmeth, what do you see on /proc/zaptel/* ? |
| 17:26.12 | Skarmeth | my two channels info |
| 17:26.38 | tzafrir_home | Could you pastebin that? |
| 17:27.12 | *** join/#asterisk timgws (n=TimGroe@202.172.97.51) |
| 17:27.13 | Skarmeth | look at http://pastebin.com/d139a448 |
| 17:27.20 | timgws | Hey |
| 17:27.33 | timgws | I have the following issue when calling a meetme conference: |
| 17:27.33 | timgws | [Apr 26 03:26:19] WARNING[20174]: chan_sip.c:3625 sip_write: Asked to transmit frame type 64, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256) |
| 17:27.42 | [TK]D-Fender | ~zoiper |
| 17:27.43 | jbot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 17:27.49 | timgws | would anybody know what the issue could possibly be? |
| 17:28.07 | [TK]D-Fender | timgws: Can't transcode G.729. Guess you don't have enough licenses. |
| 17:28.14 | timgws | ah! thanks :) |
| 17:28.16 | ac1djazz | graci [TK]D-Fender |
| 17:28.27 | ac1djazz | now how in the AGI/CLI would i knock off/remove/disconnect a caller? |
| 17:28.46 | *** join/#asterisk Tili (n=tili@58.27.144.247.wateen.net) |
| 17:29.08 | [TK]D-Fender | ac1djazz: "soft hangup [channel]" |
| 17:29.09 | ac1djazz | [TK]D-Fender: btw i re-wrote all that stuff we were complaining about yesterday into the dialplan and it works fine |
| 17:29.14 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 17:29.18 | drfreeze | Hi |
| 17:29.23 | [TK]D-Fender | ac1djazz: Glad to hear |
| 17:29.44 | drfreeze | Does anyone here know if a polycom 550 phone have versions that are not backlit? |
| 17:30.04 | drfreeze | I'm looking at: http://www.ipphone-warehouse.com/ProductDetails.asp?ProductCode=2200%2D12550%2D001&Show=TechSpecs |
| 17:30.14 | drfreeze | and it doesn't mention backlit |
| 17:30.19 | drfreeze | But other sites do |
| 17:30.34 | grandpapadot | They are all backlit |
| 17:30.48 | drfreeze | grandpapadot: thx |
| 17:30.53 | draygon | anyone here using akick start machine to do network installs? |
| 17:31.46 | drfreeze | grandpapadot: does polycom have other phones that are backlit other than the 550 and 650? |
| 17:32.11 | grandpapadot | I don't think so ... But check polycom.com for the word ... |
| 17:34.22 | Trapa | fender: Okay, yay, two machines here both can call the echo test and get it good .. A note tho .. when the call first starts, teh audio from the asterisk server is jittery ... then it smoothes out and is good. The second thing to note, is when the two working machines try to phone each other .. the codecs just blink on and off rapidly .. which is probably why we have no audio |
| 17:34.44 | SamuraiDio | can someone help me? |
| 17:34.46 | [TK]D-Fender | drfreeze: 550, 560, 650 |
| 17:34.53 | Nugget | SamuraiDio: down, not across. |
| 17:35.04 | Trapa | At the beginning of the echo test, the same rapid-blinking of the name of the codec (pcmu) happens but then PCMU stays on. |
| 17:36.19 | Skarmeth | tzafrir_home, any clues? |
| 17:36.24 | [TK]D-Fender | Trapa: Make sure that Vista machine doesn't screw up your comms. Also add "canreinvite=no" to all of your phone's entries |
| 17:37.06 | SamuraiDio | Nugget, what? |
| 17:37.36 | drfreeze | [TK]D-Fender: do you know why the 550 and the 560 use a 24V power adapter when they are IEEE 802.3af compliant-- which I thought was 15v |
| 17:38.24 | [TK]D-Fender | drfreeze: Easy guess : 24v PS's are cheaper for already being in production. |
| 17:38.29 | tzafrir_home | Skarmeth, and what do you have in zaptel.conf ? |
| 17:39.09 | drfreeze | [TK]D-Fender: so I guess the phones are quite tolerant of the voltage they get |
| 17:39.25 | [TK]D-Fender | drfreeze: Guess so. |
| 17:40.20 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 17:41.42 | Trapa | fender: woot! We got a call .. and things seemed to be good except at the end there suddenly we got all kinnds of noise ... |
| 17:42.09 | *** join/#asterisk lirelent (n=lirelent@70.16.147.2) |
| 17:42.37 | [TK]D-Fender | Trapa: Softphones, speakers, mics, etc can all suck.... |
| 17:43.00 | [TK]D-Fender | Trapa: Dedicated devices tend to perfom much more consistently |
| 17:43.05 | Trapa | yeah .. I want to get ip phones in here .. But not until we can get the soft phones working so that we know the system will work |
| 17:43.10 | Trapa | even if there's issues. |
| 17:43.28 | [TK]D-Fender | Trapa: well if you've got a call in at all then the answer is "yes". |
| 17:43.32 | Trapa | Well we have a machine .. it's a dual-quad core (8 cores) at 3.22ghz , with 12gb ram and 4tb of disk space. |
| 17:43.40 | [TK]D-Fender | Trapa: Time to research your options and shop around. |
| 17:43.53 | [TK]D-Fender | Trapa: And what kind of load are you looking to put on it? |
| 17:43.57 | Trapa | The other services on it are php, apache, mysql .. . and jabber .. so .. i think this machine aught to be able to handle 10 simultanious calls |
| 17:44.11 | [TK]D-Fender | Trapa: that system is MASSIVE overkill. |
| 17:44.30 | [TK]D-Fender | Trapa: But enjoy your breathing room :) |
| 17:44.46 | twisted | run a COD server on it |
| 17:45.27 | grandpapadot | BF2 |
| 17:45.45 | twisted | meh |
| 17:45.55 | GrumpyOldMan | wish there was a mac version of bf2... bf2142 is just not as fun. |
| 17:46.03 | [TK]D-Fender | DOOM 1 over IPX babY! YEAH!!!! |
| 17:46.17 | twisted | WOOHOO! |
| 17:46.29 | Qwell | hmm |
| 17:46.31 | Qwell | does linux support IPX? |
| 17:46.34 | GrumpyOldMan | marathon ? |
| 17:46.34 | twisted | yes |
| 17:46.49 | grandpapadot | GrumpyOldMan: bf2142 for mac is just the PC binaries launched with cedega, you could easily run bf2 |
| 17:46.56 | twisted | i just compiled a new kernel yesterday, there is still IPX/SPX support |
| 17:47.08 | Qwell | twisted: fun |
| 17:47.42 | twisted | i want to implement sip over ipx |
| 17:47.43 | GrumpyOldMan | grandpapadot: no cause of the cd check crap. |
| 17:47.44 | twisted | that'd be fun |
| 17:47.54 | grandpapadot | GrumpyOldMan: Ahh... |
| 17:48.02 | GrumpyOldMan | grandpapadot: why play bf2 if I can't play online... |
| 17:48.06 | Skarmeth | tzafrir_home, http://pastebin.com/d3e82a0ea |
| 17:48.09 | grandpapadot | true, true |
| 17:48.40 | Qwell | GrumpyOldMan: so...use the CD? |
| 17:48.58 | Qwell | (this is, of course, quite off-topic) |
| 17:49.04 | tzafrir_home | twisted, there is support there for appletalk, decnet and whatever |
| 17:49.35 | Trapa | <PROTECTED> |
| 17:49.36 | grandpapadot | @Qwell, not really, we use asterisk to host our own in-game meetme, lol |
| 17:49.50 | *** join/#asterisk GoldenFish (n=is@AC9E4665.ipt.aol.com) |
| 17:50.36 | *** join/#asterisk troy- (n=troy@worldnet.tauri.ca) |
| 17:50.37 | GoldenFish | ~thebook |
| 17:50.38 | 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 |
| 17:51.01 | tzafrir_home | Skarmeth, hmm... no idea. |
| 17:52.26 | Skarmeth | tzafrir_home, http://pastebin.com/d1a0a0b30 |
| 17:52.42 | Skarmeth | my zaptel don't load tone zones.... |
| 17:54.04 | tzafrir_home | Skarmeth, what version of Zaptel drivers? cat /sys/module/zaptel/version |
| 17:54.21 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 17:54.59 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 17:55.18 | Skarmeth | 1.4.9.2 |
| 17:55.41 | Skarmeth | just zaptel and wct4xxp loaded |
| 17:55.51 | tzafrir_home | http://zaptel.tzafrir.org.il/#_past_incompatibilities |
| 17:56.00 | tzafrir_home | you installed zaptel 1.4.10? |
| 17:56.14 | tzafrir_home | hates such incompatibilities |
| 17:57.19 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 17:57.26 | lirelent | what is the difference between the discontinued TDM400P and its replacement, the TDM410P? |
| 17:57.51 | *** join/#asterisk bmg505 (n=leon@196-209-77-112-tbnb-esr-2.dynamic.isadsl.co.za) |
| 17:58.00 | tinkerghost | 410 is supposed to have better impedance matching |
| 17:58.40 | lirelent | with the phone lines? |
| 17:59.19 | tinkerghost | yes - which means less echo --- there is another 410 with hardware echo cancelling as well |
| 17:59.41 | Skarmeth | tzafrir_home, let me see |
| 18:00.43 | Mercestes | Hey guys. Have a PRI checklist from Logix, do I need a DMS 100, ATT53, or NI-2 for zaptel? |
| 18:01.01 | Mercestes | ATT5E right? |
| 18:01.31 | Skarmeth | shit... yum don't upgraded kmdl's.... |
| 18:02.06 | tinkerghost | lirelent - 410 might be mostly an upgrade adding a port to plug in the hardware echo canceling module - the specs look awful similar |
| 18:02.12 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-86-17.vif.net) |
| 18:02.23 | Skarmeth | because of http://bugs.digium.com/view.php?id=12387 I will need to reboot with the new modules |
| 18:02.37 | Skarmeth | tzafrir_home, thank's for the light's |
| 18:03.29 | lirelent | tinkerghost, yea that's what I was getting, there is also some mention of fixing compatibility issues with the few chips sets that the 400 had issues with |
| 18:03.51 | lirelent | chipsets* |
| 18:03.59 | lirelent | but I don't have any of the mentioned chipsets |
| 18:04.09 | Mercestes | no answers for me eh? No mercestes love. :P |
| 18:05.19 | Mercestes | NI-2 it is. |
| 18:05.36 | grandpapadot | Like anything by via and their sh*tty PCI bus |
| 18:06.12 | lirelent | yea I have all intel chipsets actually, at least on the machines I actually use |
| 18:06.33 | lirelent | via tends to leave something to be desired |
| 18:07.14 | lirelent | anyone know how much of a difference that hardware echo module makes? |
| 18:07.34 | lirelent | is there really noticable echo with the 400 or the 410 without that module? |
| 18:08.41 | *** join/#asterisk unpaidbill (i=bill@66.90.73.20) |
| 18:08.55 | tinkerghost | Hmm, I just put a bid in on a 400 card ... probably should check what MB chipset I use on the linux box at home |
| 18:10.04 | tinkerghost | I know the PCI slots are attrocious - have only 2 slots & 1 is IRQ'd to match the parallel port |
| 18:10.27 | lirelent | if you search for 400 on http://www.digium.com/en/products/eol.php it has some notes on why the discontinued it |
| 18:10.54 | grandpapadot | With PCI, sharing IRQ's is ok in some cases .. but you can simply disable the parallel port in your bios ... |
| 18:11.37 | lirelent | tinkerghost, http://www.digium.com/en/docs/misc/compatibility_notes.php |
| 18:11.38 | *** join/#asterisk SomethingISODD (n=dancole@S010600a0d1757bfb.cg.shawcable.net) |
| 18:12.06 | [TK]D-Fender | Mercestes: You need to confirm what signalling the telco will use |
| 18:12.18 | [TK]D-Fender | Mercestes: If you have a choice make it NI2 |
| 18:12.20 | SomethingISODD | hello all quick question i am creating a php agi script. i was wondering how i could get it so if the call is busy to try the next ip address.. all with in the agi script |
| 18:12.40 | [TK]D-Fender | SomethingISODD: ${DIALSTATUS} <---- |
| 18:12.58 | SomethingISODD | thanks [TK]D-Fender |
| 18:14.12 | [TK]D-Fender | tinkerghost: Zaptel 1.4 is much better at interrupt management. Aim for the latest stable |
| 18:14.28 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 18:14.55 | [TK]D-Fender | tinkerghost: for EC, try Zaptel's native ones first, then OSLEC, then HPEC if need be (that order) |
| 18:16.01 | Mercestes | [TK]D-Fender: Thanks. :) |
| 18:16.11 | SomethingISODD | [TK]D-Fender one more question if you dont mind, i have cheaper routes,, would I use an array to line up cheapeast to the highest? |
| 18:16.16 | Mercestes | Yea, they are giving me a choice so I picked nI-2 |
| 18:16.36 | [TK]D-Fender | SomethingISODD: This is programming. Make up your mind as to what tool severs the best |
| 18:16.51 | [TK]D-Fender | Mercestes: Its the most common in NA. |
| 18:18.27 | tinkerghost | I'm just ordering a card now - found a used TDM card for a little more than a X100P clone - figured I'd bid on it since people keep telling me it has better sound quality off the bat |
| 18:19.27 | Yourname`` | Hi. http://pastebin.ca/997578 -> What would I need to do so the please-try-again playback happens only twice instead of being on a forever loop? |
| 18:19.31 | *** join/#asterisk cayorde (n=flexable@87.18.98.72) |
| 18:19.49 | tinkerghost | Oops, just got a notice it's now more expensive to get the TDM, time to buy the clone |
| 18:20.31 | Marquel | [TK]D-Fender: small question for AstDB - what tests are available? is it allowed to do something like $["${DB(myphone/dest)}" != ""] and it will return true if that string is either empty or the key does not exist? |
| 18:21.15 | [TK]D-Fender | Yourname``: make a proper IVR, use the "t" AND "i" standard extensions, and keep a loop counter. |
| 18:21.29 | [TK]D-Fender | Yourname``: And only call waitexten once in there. |
| 18:21.29 | Yourname`` | [TK]D-Fender: How do I do the loop counter? |
| 18:21.32 | Yourname`` | Gotcha |
| 18:21.40 | [TK]D-Fender | Marquel: Sure. |
| 18:21.54 | [TK]D-Fender | Yourname``: set a variable and INCREMENT IT. |
| 18:22.32 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 18:22.37 | Katty | ohai! |
| 18:22.40 | Katty | i upgraded your pbx |
| 18:23.24 | Marquel | [TK]D-Fender: last question: how about the cost of such a query? on a phone line with maximum of four concurrent incoming calls, is it better to use a variable or ask the db twice - performancewise? ;) |
| 18:24.54 | *** part/#asterisk LemensTS (i=CustomGT@adsl-70-238-155-11.dsl.stlsmo.sbcglobal.net) |
| 18:25.15 | *** join/#asterisk point (n=litvinen@77.66.155.184) |
| 18:25.29 | [TK]D-Fender | Marquel: completely irrelevant. Code with confidence |
| 18:25.55 | Mercestes | Hai katty |
| 18:26.02 | [TK]D-Fender | Katty: U CAN HAZ CALLZ NOW |
| 18:26.06 | Marquel | [TK]D-Fender: thx, i think i now can make my customer very happy ;) |
| 18:27.40 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 18:28.58 | Skarmeth | tzafrir_home, I just upgraded zaptel to 1.4.10 and the alarms still blink red |
| 18:28.58 | Skarmeth | my /sys/module/zaptel/version says 1.4.10 |
| 18:28.58 | Katty | [TK]D-Fender: :< |
| 18:28.59 | Skarmeth | I can load tonezones right now... |
| 18:30.09 | *** join/#asterisk AmabassadortoPol (n=root@adsl-70-241-119-233.dsl.hstntx.swbell.net) |
| 18:30.40 | [TK]D-Fender | Katty: Why so grim? You've got a new PBX! |
| 18:30.53 | Katty | [TK]D-Fender: pbx :< |
| 18:31.00 | Katty | [TK]D-Fender: beach + museum :> |
| 18:31.03 | Katty | [TK]D-Fender: pbx :< |
| 18:31.08 | Katty | [TK]D-Fender: ice cream and weekend :> |
| 18:31.10 | Katty | [TK]D-Fender: pbx :< |
| 18:31.37 | Katty | [TK]D-Fender: i am grumpy right now. |
| 18:32.22 | Katty | wants to go home! |
| 18:32.36 | unpaidbill | what the heck, pbx is better than all those things combined |
| 18:32.59 | Katty | unpaidbill: you're mental. |
| 18:33.04 | unpaidbill | yes :) |
| 18:33.06 | AmabassadortoPol | not when the limited vision of office phones is all thats on display |
| 18:33.28 | Mercestes | pets Katty. |
| 18:33.30 | Mercestes | It'll be ok./ |
| 18:34.56 | AmabassadortoPol | is there only one asterisk forum? |
| 18:35.43 | M1s3ry | many there are |
| 18:36.36 | M1s3ry | I see 12 right now |
| 18:39.37 | tinkerghost | OK, any point in ordering an X100P clone instead of just hitting up the local surplus shop for an Intel 537 based card? |
| 18:39.57 | Qwell | tinkerghost: x100p is pretty much junk anymore.. |
| 18:40.24 | M1s3ry | points to the TDM410P |
| 18:40.48 | alrs | tinkerghost: surplus or Sangoma |
| 18:40.52 | tinkerghost | Quell: It's a home box to play with - can't justify the 100+ for a TDM w/ FXO module |
| 18:41.02 | alrs | either almost free/low expectations or real money/real support |
| 18:41.40 | tinkerghost | alrs: for real support I go to work & bitch about the 2 dual T1 cards |
| 18:42.22 | alrs | tinkerghost: we have 31 quad-port T1 cards |
| 18:43.28 | tinkerghost | alrs: you probably have a few more incomming lines than we do - In fact, I think you have a card for each of our incomming lines |
| 18:44.39 | alrs | tinkerghost: retail, ~40k. Ebay Value, $3k. |
| 18:45.17 | AmabassadortoPol | and office phones make all the money....... |
| 18:46.06 | ruied | Hi! Im having a problem, when I compile mISDN with my old kernel, it compiles it, but whe I compile with the 2.6.25 kernel it reports the error: "Fix it to use EXTRA_CFLAGS" |
| 18:46.50 | x86 | alrs: 31 quad-port cards of the brand I buy would cost somewhere around $80,600 |
| 18:46.58 | ruied | I I've checked the Make file and it is using the EXTRA_CFLAGS... |
| 18:47.02 | *** join/#asterisk gitguy (n=diego@adsl-148-95.click.com.py) |
| 18:47.13 | x86 | alrs: I get the Sangoma A104dx, quad-port T1/E1 with HWEC |
| 18:47.17 | alrs | SIP/2.0 200 OK |
| 18:47.17 | alrs | outtolunc: sip:999@202.172.97.51;tag=as01aa4882 |
| 18:47.17 | alrs | User-Agent: Asterisk PBX |
| 18:47.17 | alrs | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY |
| 18:47.17 | alrs | Supported: replaces |
| 18:47.18 | alrs | Contact: <sip:202.172.97.51> |
| 18:47.24 | alrs | wrong window, apologies |
| 18:48.30 | gitguy | hi |
| 18:48.43 | gitguy | I have an asterisk server on a amazon ec2 server |
| 18:48.55 | gitguy | but their network is full of NATs |
| 18:48.57 | gitguy | and I'm using SIP |
| 18:49.03 | gitguy | any ideas how can I make that work? |
| 18:49.09 | gitguy | nat=yes and I specify the ip address right? |
| 18:50.10 | disposable | is it wise to have all users of a company in one context or should i keep them in different contexts and do includes of other contexts in each context? i wand departments of the company to be able to call each other. |
| 18:50.26 | x86 | gitguy: how did you get an ec2 server? |
| 18:50.31 | x86 | gitguy: i've been wanting one :) |
| 18:50.37 | gitguy | x86: from my boss |
| 18:50.59 | gitguy | x86: any ideas how can i fix that issue? |
| 18:51.14 | gitguy | it has a public ip address |
| 18:51.20 | gitguy | and a dns name address |
| 18:51.42 | gitguy | but when i do 'ifconfig -a' I only see the private ip address |
| 18:52.34 | gitguy | it's behind a NAT |
| 18:52.49 | disposable | by users i meant personal extensions |
| 18:53.38 | tinkerghost | gitguy: usually when you are behind a NAT firewall, you only have the local IP address - things coming into the public one hit the firewall & get forwarded to the internal ip |
| 18:53.40 | x86 | gitguy: not sure... try using IAX |
| 18:54.15 | gitguy | ok |
| 18:54.17 | gitguy | thanks! |
| 18:54.25 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:56.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:56.58 | AmabassadortoPol | so what kind of things can you send out to the phone company? |
| 18:57.04 | *** join/#asterisk killmel8tr (n=IceChat7@ip-198-22-65-131.quickconnectusa.com) |
| 18:58.03 | gitguy | nope, it doesn't work with IAX too |
| 18:58.38 | gitguy | it doesn't register |
| 18:59.14 | x86 | IAX is NAT-friendly |
| 18:59.19 | x86 | only uses one port |
| 18:59.23 | gitguy | i know that |
| 18:59.24 | gitguy | amazon sucks |
| 18:59.37 | x86 | they need to forward 4569 UDP and TCP to you |
| 18:59.44 | AmabassadortoPol | so i heard what you can do unlimited cellphone family plan free minutes user to user and dial out on teliax right? |
| 19:00.27 | gitguy | x86: i have to do that? |
| 19:01.29 | AmabassadortoPol | ok i didnt get banned for that but its still kinda stuffy in here |
| 19:01.31 | x86 | if you want inbound registrations from the outside world, yes |
| 19:01.38 | AmabassadortoPol | ill have to do some more research |
| 19:01.58 | tinkerghost | gitguy: whoever is in charge of the firewal has to route incomming on that port to the IP address of your * server |
| 19:01.58 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 19:02.02 | gitguy | x86: how, iptables? |
| 19:02.32 | gitguy | is that described in the book? |
| 19:02.43 | tinkerghost | gitguy: depends entirely on what type of firewall you are running |
| 19:02.52 | gitguy | iptables i think |
| 19:02.58 | gitguy | it's a debian box |
| 19:03.07 | nny_1 | anyone using automon with web access? I am working on setting it up so customers can access the monitor directory, but hitting a wall with the permissions. I have asterisk runnin as asterisk, and apache can't control that dir. I thought about just adding apache to the group asterisk, but fear that would be a can o worms security wise |
| 19:03.13 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 19:03.22 | x86 | gitguy: you need to call amazon and have them do it for you |
| 19:03.53 | gitguy | x86: shouldn't i do that? i have root there |
| 19:04.51 | tinkerghost | gitguy: do you have root on the firewall box or on the * box |
| 19:05.08 | gitguy | * box |
| 19:05.14 | x86 | right |
| 19:05.19 | x86 | they need to do it on the firewall |
| 19:05.24 | gitguy | ok |
| 19:05.36 | tinkerghost | gitguy: whoever has access to the firewall box is who has to set up forwarding |
| 19:05.43 | *** join/#asterisk geek_cl (n=geek@200.75.18.211) |
| 19:05.47 | gitguy | right |
| 19:05.48 | gitguy | thanks guys |
| 19:05.49 | gitguy | :D |
| 19:06.13 | *** join/#asterisk _slack (n=nombre@201.243.174.130) |
| 19:06.39 | _slack | hi, where is the default codec in asterisks files? |
| 19:07.06 | tinkerghost | nny_1: try setting it up with just read access to the monitor directory for the asterisk group |
| 19:08.49 | *** join/#asterisk qdk (n=qdk@213.173.230.2) |
| 19:10.18 | tinkerghost | nny_1: 740 should be rwxr_____ & that should be close to all you need - might want to play with .htaccess for more security |
| 19:11.35 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:11.35 | nny_1 | tinkerghost: k |
| 19:12.25 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:12.28 | ruied | <PROTECTED> |
| 19:13.37 | tinkerghost | nny_1: does each of your customers get their own directory for monitored calls? if not it would be ugly to give them all access to the directory |
| 19:14.13 | nny_1 | tinkerghost: for this current setup they are using the system to capture quotes verbally, so no individual user permissions needed |
| 19:14.37 | nny_1 | tinkerghost: so your saying add apache to group asterisk? |
| 19:14.56 | nny_1 | eh well |
| 19:15.03 | nny_1 | -rw-r--r-- 1 asterisk asterisk 174444 Apr 25 15:12 auto-1209150764-8433425901-3845171.wav |
| 19:15.39 | tinkerghost | nny_1: unless someone can come up with a specific security risk, I would say add apache to the asterisk group and set group perms to read only, and other to no access |
| 19:15.46 | nny_1 | so anyone can read the file, I believe it is the drwxr-x--- 9 asterisk asterisk 1024 Dec 21 22:45 asterisk in /var/spool |
| 19:17.35 | tinkerghost | nny_1: as long as you have apache set up with an alias to the directory, you should be good |
| 19:17.50 | nny_1 | k |
| 19:17.52 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-155-11.dsl.stlsmo.sbcglobal.net) |
| 19:18.24 | nny_1 | well per the howto on voipinfo I have a symlink from /var/www/html/automon/monitor to /var/spool/asterisk/monitor |
| 19:18.57 | LemensTS | http://pastebin.com/m604cd9f zapcard is not coming up some reason, cant find a fix to this error on google |
| 19:21.06 | tinkerghost | nny_1: make sure that apache will follow symlinks - it can be configured to not follow them |
| 19:22.26 | LemensTS | zttool shows it is thier and no alarms |
| 19:22.56 | [TK]D-Fender | LemensTS: Big print : Module 0: FAILED FXS (FCC) |
| 19:23.58 | LemensTS | TKD-Fender: yea i seen that, do you think that module is bad? |
| 19:24.33 | [TK]D-Fender | LemensTS: I don't have to "think". Its telling you in flashing neon. |
| 19:25.52 | LemensTS | TKD-Fender: Ok i did _not_ know that a failed module would prevent the rest of the modules/card from working. thanks |
| 19:25.53 | [TK]D-Fender | LemensTS: Like looking at a crate where a city used to be, a nigh-unimaginably huge mushroom cloud rising from the center and wondering "Gee, was that a nuke?" |
| 19:26.53 | nny_1 | tinkerghost: hmm i have <Directory /> Options FollowSymLinks in my stock httpd.conf, yet it still gives a permission denied in logs. I have added my non root user to asterisk and can now successfully list the directory. Any way to figure out *why* apache is still giving a permission denied? |
| 19:27.01 | LemensTS | TKD-Fender: What if that person was home schooled and never made contact with tv or outside world, how would they know what a nuke was. They have to experience and learn. |
| 19:28.46 | *** join/#asterisk korihor (n=humberto@190.78.209.202) |
| 19:29.46 | tinkerghost | nny_1: 1) see if the followsymlinks is overwriten by a later nofollow |
| 19:30.03 | tinkerghost | 2) if you just updated it - restart apache |
| 19:30.39 | nny_1 | tinkerghost: i can post httpd.conf to pastebin, I see that the default is set, as well as late ron it allows it again, I don't see a no follow |
| 19:30.44 | *** join/#asterisk zackz (n=zdz@rrcs-24-123-106-250.central.biz.rr.com) |
| 19:31.12 | tinkerghost | 3) flip to 770 permissions on the directory & try again |
| 19:31.21 | zackz | anyone know why my Cisco 7940 would ring back a busy tone after a Hangup() command? |
| 19:31.28 | zackz | this is what I'm seeing in asterisk: http://pastebin.com/m1e8d68e4 |
| 19:31.50 | grandpapadot | zackz: What else should it be doing? |
| 19:32.32 | nny_1 | tinkerghost: it does appear (comments obscure this) that <Directory "/var/www/html"> Options Indexes FollowSymLinks is below the default |
| 19:33.05 | nny_1 | tinkerghost: I later have a defined area for an authfile, but it shoudl include the original options, or does it overwrite them? |
| 19:33.33 | nny_1 | tinkerghost: ha that was it |
| 19:33.45 | zackz | grandpapadot: disconnecting the SIP channel? |
| 19:33.52 | nny_1 | tinkerghost: i had password protected the root html dir, but not redefined my options for indexing etcv |
| 19:34.11 | tinkerghost | nny_1: ahh, that would do it |
| 19:34.23 | grandpapadot | zackz: What indication do you have that the sip channel is still active? Is it showing up in show channels? |
| 19:35.04 | zackz | no it doesnt |
| 19:35.16 | zackz | my Polcoms never ring busy after a hangup though |
| 19:36.22 | *** join/#asterisk oej (n=olle@75.112.130.67) |
| 19:36.46 | grandpapadot | zackz: My linux workstation has different default background colors than my Windows workstation ?? |
| 19:37.36 | grandpapadot | zackz: That's just each phone's default behavior, neither has anything to do with asterisk ... |
| 19:37.44 | zackz | ah ok |
| 19:37.52 | *** join/#asterisk qdk_ (n=qdk@213.173.230.3) |
| 19:37.57 | _slack | zackz, hi |
| 19:38.02 | zackz | hi |
| 19:38.15 | _slack | zackz, where is the default codec in asterisks files? |
| 19:38.43 | zackz | gsm? |
| 19:38.49 | _slack | yep |
| 19:39.04 | _slack | default is ulaw |
| 19:39.16 | _slack | but i need gsm as default codec |
| 19:39.28 | grandpapadot | slack: sip.conf allow=gsm,ulaw |
| 19:39.46 | M1s3ry | _slack, don't forget to disallow=all |
| 19:40.11 | _slack | add disallow=all line? |
| 19:40.39 | M1s3ry | if it's not already there... which it should be... then allow=gsm,ulaw |
| 19:42.57 | _slack | and how disable the ulaw coedc? |
| 19:43.01 | _slack | codec* |
| 19:43.19 | M1s3ry | then just use allow=gsm and thats it |
| 19:43.22 | tinkerghost | _slack: just don't include it in the allow statement |
| 19:43.28 | grandpapadot | slack: in sip.conf, disallow=all<cr>allow=gsm<cr> under the peer definition you're trying to change |
| 19:43.47 | bakermd | Anyone have a good BHCA calculator? |
| 19:43.56 | _slack | grandpapadot, right, but this not work for me |
| 19:44.22 | grandpapadot | Did you reload your sip configuration? And what end-point are you using? softphone? hardphone? another asterisk peer? |
| 19:44.27 | _slack | grandpapadot, i dont understand why |
| 19:44.29 | M1s3ry | XD |
| 19:44.46 | grandpapadot | _slack: pastbin your sip.conf |
| 19:44.46 | _slack | im use freepbx |
| 19:44.57 | grandpapadot | _slack: ahh, nevermind, see #freepbx |
| 19:45.00 | M1s3ry | .... get rid of it |
| 19:45.37 | _slack | :s |
| 19:45.53 | _slack | un #freepbx anybody answerme |
| 19:46.16 | M1s3ry | then what you're looking for would be done within the GUI of FreePBX. We cannot help you there |
| 19:47.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:47.01 | grandpapadot | freepbx is unpure, bad, bad admin |
| 19:47.02 | _slack | M1s3ry, what do you can suggest to me? |
| 19:47.21 | grandpapadot | _slack: You won't find any freepbx help here, best bet is the docs, google or #freepbx |
| 19:47.32 | _slack | i know i know |
| 19:47.47 | Qwell | bakermd: BHCA? |
| 19:47.54 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 19:47.56 | M1s3ry | _slack, if you want my honest suggestion... throw out FreePBX install opensource asterisk, and use it |
| 19:48.05 | Qwell | oh |
| 19:48.24 | grandpapadot | _slack: what M1s3ry said++ |
| 19:48.36 | Qwell | bakermd: isn't that generally pretty close to erlang? |
| 19:48.38 | LemensTS | asterisk1*CLI> load chan_zap.so |
| 19:48.38 | LemensTS | Unable to load module chan_zap.so |
| 19:48.38 | LemensTS | <PROTECTED> |
| 19:48.38 | LemensTS | <PROTECTED> |
| 19:48.41 | *** join/#asterisk goodjoke (i=1827a8fa@gateway/web/ajax/mibbit.com/x-d4a7306fc093b503) |
| 19:48.47 | LemensTS | whoops http://pastebin.com/m4fb5b5a7 |
| 19:49.42 | *** join/#asterisk VaNNi (n=VaNNi___@lgb-static-216.70.165.200.mpowercom.net) |
| 19:49.50 | ka0zoz | Need some advice on good sip phones for an office of 20, 8 fxo lines. |
| 19:49.58 | ka0zoz | anyone have a good link? |
| 19:49.59 | Qwell | ~phones |
| 19:49.59 | jbot | from memory, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 19:50.02 | grandpapadot | ka0zoz: Polycom 330 + PoE switch |
| 19:52.04 | ka0zoz | Great! I'll check them out. |
| 19:52.50 | x86 | I prefer the 501 over the 320/330 |
| 19:53.05 | x86 | only a little cost difference between the two |
| 19:53.15 | grandpapadot | Both are good phones... |
| 19:53.43 | grandpapadot | The 330 has a smaller footprint, better speakerphone, and updated look. It also has a smaller screen. The 501 has a bigger screen, a hollow sounding speakerphone. |
| 19:53.43 | gitguy | is asterisk + SIP possible on a server that is behind a NAT? |
| 19:54.01 | ka0zoz | with STUN I believe |
| 19:54.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:54.30 | *** join/#asterisk Tourinho (n=tourinho@201.37.118.16) |
| 19:55.17 | gitguy | ka0zoz: does asterisk supports that? |
| 19:56.08 | grandpapadot | gitguy: Yea, you need to forward SIP port 5060, and RTP ports 20000-30000 to your internal private ip from your firewall (these ports are defined in sip.conf and rtp.conf) |
| 19:56.39 | disposable | ka0zoz: with the newest firmware sip3.0.x they support ldap and conference management |
| 19:56.45 | disposable | ka0zoz: the polycoms |
| 19:57.03 | Winkie | disposable: they support LDAP? |
| 19:57.08 | Winkie | natively? |
| 19:57.16 | Winkie | that's pretty awesome |
| 19:57.27 | disposable | Winkie: yes, but you have to wait until they release the final version of sip3.0 |
| 19:57.31 | Winkie | we're unfortunately spending $70k on intertel equipment instead of asterisk :( |
| 19:57.38 | Winkie | so it is of little use to me at the moment |
| 19:57.41 | x86 | grandpapadot: 501's speakerphone sounds great to me |
| 19:57.50 | ka0zoz | wow, no stun support in asterisk. I'm still learning. |
| 19:57.52 | gitguy | grandpapadot: ok, thx |
| 19:57.55 | x86 | grandpapadot: actually I can't tell the difference between my 501's and 330's as far as sound quality |
| 19:58.06 | mcab | You need to buy an extra licence to get the LDAP and n-Way conferencing features with SIP 3.0 |
| 19:58.22 | grandpapadot | x86: imho, yours obviously may vary ... The 330's I've used seem to have a better microphone ... |
| 19:58.25 | disposable | mcab: that's unfortunately true |
| 19:58.37 | x86 | grandpapadot: i've not had a single problem |
| 19:58.49 | grandpapadot | x86: I didn't say it was a problem, just an opinion ... |
| 19:59.00 | Winkie | the 330s/320s we have are quality enough |
| 19:59.08 | mcab | disposable: at least it's still an option though |
| 19:59.12 | Winkie | we bought a selection of low to high range voip phones and tested them |
| 19:59.17 | x86 | Winkie: yeah don't get me wrong, 320/330 is a decent phone |
| 19:59.18 | Winkie | the 320s/330s were right up there |
| 19:59.32 | x86 | does asterisk support g722 yet? |
| 19:59.51 | [TK]D-Fender | x86: 1.6 yes, 1.4 partial |
| 19:59.53 | disposable | x86: the ip650 has "HD sound" printed on the receiver |
| 19:59.58 | disposable | x86: :) |
| 20:00.05 | disposable | x86: it's got to be better |
| 20:00.06 | Winkie | x86: the only one we had superior was the linksys sipura phone but i forget the model number |
| 20:00.21 | x86 | disposable: err, that's g722 ;) |
| 20:00.25 | grandpapadot | lol, yea, but it's only going to be as good as what you're pumping into them, i.e., a pri is still only 64k ulaw |
| 20:00.44 | x86 | Winkie: by "superior" you mean "inferior" right? :) |
| 20:01.03 | grandpapadot | HD voice is still a marketing poly imho, there's no real-world use for it that I'm aware of .. |
| 20:01.10 | grandpapadot | poly = ploy |
| 20:01.13 | disposable | x86: wideband stuff |
| 20:01.22 | x86 | grandpapadot: sure there is, it sounds better on internal calls |
| 20:01.29 | grandpapadot | x86: touche |
| 20:01.36 | x86 | grandpapadot: great for site-to-site conferencing over a WAN |
| 20:01.36 | Winkie | x86: i don't, one second |
| 20:01.57 | grandpapadot | x86: At 256kbps? I think I may have better use for my bandwidth ... |
| 20:02.23 | Winkie | x86: apparently they don't sell it anymore |
| 20:03.03 | Winkie | x86: the spa942 is the one i was talking about |
| 20:04.05 | x86 | 942's are decent, but not the quality of a polycom for sure |
| 20:04.14 | ka0zoz | are there many of these suggested phone floating on ebay that I can buy them cheap to try before I buy? |
| 20:04.48 | x86 | ka0zoz: what did your ebay search tell you? |
| 20:04.57 | ka0zoz | nothing yet |
| 20:04.59 | *** part/#asterisk pgarcia (n=pedgarci@yoda.kanatek.com) |
| 20:05.17 | ka0zoz | someone must have tried this |
| 20:05.38 | grandpapadot | ka0zoz: 330's or 501's will serve you well for many years. |
| 20:06.05 | ka0zoz | Ok. I'll go searching for those first grandapa |
| 20:06.09 | disposable | spa942s have a very high fault rate. almost 30% get returned |
| 20:06.48 | disposable | but still better than grandstream :) |
| 20:06.58 | grandpapadot | ~grandstream |
| 20:06.59 | jbot | [grandstream] the Yugo of VoIP hardware. Run. Run away now. |
| 20:07.07 | disposable | the hell i went through with them, i will never forget |
| 20:07.30 | grandpapadot | crapola |
| 20:07.31 | M1s3ry | whenever you get a chance... go into the source directory for asterisk, do "make menuselect" and hit i once in there |
| 20:07.52 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 20:08.11 | ka0zoz | polycom advertises M$ right on their front page.. ewwww |
| 20:09.20 | tinkerghost | grandpapadot: we've lost a bunch of 330's in the last 18 months - 6 of 45 - where either the display died or the whole phone just curled up & died |
| 20:09.48 | *** join/#asterisk unbkbl (n=work@static-adsl201-232-88-87.epm.net.co) |
| 20:09.50 | grandpapadot | tinkerghost: Wow. That's odd. Poor PoE implementation? We have thousands of them in use and I rarely hear of issues ... |
| 20:10.32 | grandpapadot | generally speaking, I think most in here would agree, the Polycom phones are at the top of the list from a quality and useability standpoint ... |
| 20:10.43 | disposable | agreed |
| 20:10.44 | b11d` | i would back that up |
| 20:10.50 | tinkerghost | oops sorry 301 not 330, done using power adapters not POE |
| 20:11.34 | disposable | ip300 is my darling. it's so ugly and simple it's beautiful |
| 20:11.36 | grandpapadot | Big difference between 301's and 330's ... The 330 is a *much* better phone |
| 20:12.01 | [TK]D-Fender | IP 501 is advised for those not looking for PoE and in need of a more mid-range phone. IP30X is completely unadvised. IP 320 is advised wired on its own subnet VS IP330 ($ pays for wiring). |
| 20:12.50 | [TK]D-Fender | IP560 = bad call, only for the desperate. IP 601 not so great, spend a bit more for the IP 650. IP 550 =mah |
| 20:13.04 | b11d` | I wish I could buy 501s with the handset from the 650.. the audio quality on those 650 handsets is so much better.. and they totally work on the 501.. |
| 20:13.19 | tinkerghost | we've never had a problem with the 601s we have |
| 20:13.27 | [TK]D-Fender | b11d`: I like my home IP 501 personally. |
| 20:13.37 | tinkerghost | ohh, Ubuntu is finished upgrading .... time to reboot |
| 20:13.41 | grandpapadot | [TK]D-Fender: What are you guys using at work? Cisco? |
| 20:13.45 | disposable | btw, has anybody ever had any issues with Snom phones? i admit they need getting used to but they can probably make you a cup of coffee if you know how to ask |
| 20:13.51 | b11d` | im just saying, there is a noticable improvement in quality with the handset from the 650.. |
| 20:14.01 | [TK]D-Fender | grandpapadot: After that speil you're still asking? |
| 20:14.10 | b11d` | lol |
| 20:14.19 | grandpapadot | lol, no, I thought you worked at digium and I know they use a lot of Cisco handsets ... |
| 20:14.24 | [TK]D-Fender | grandpapadot: Polycom 100%. Most people here thought I worked for Polycom at one point or another |
| 20:14.34 | M1s3ry | huh? |
| 20:14.43 | M1s3ry | Digium uses polycom |
| 20:14.49 | *** join/#asterisk juso (n=adamb@216.52.163.145) |
| 20:15.33 | juso | Hey guys, any out the box asterisk spinoff with web interface? Need to just get one up and running fast, big wigs want a web interface. |
| 20:15.34 | grandpapadot | I just seem to see a lot of chatter regarding digium's Cisco's and chan_skinny, dunno, just a passing observation |
| 20:16.16 | b11d` | trixbox i assume is all web interface.. |
| 20:16.18 | b11d` | never used it.. |
| 20:16.22 | M1s3ry | we definitely do testing on them, but I would guess around 90% (give or take a couple) use polycom |
| 20:16.31 | b11d` | isnt there the AsterNOW gui and whatnot? |
| 20:16.36 | b11d` | i dunno |
| 20:16.48 | M1s3ry | yes |
| 20:16.49 | grandpapadot | juso: Swichvox? |
| 20:17.01 | juso | hah so many! heh |
| 20:17.01 | M1s3ry | AsteriskNOW and Switchvox are supported by digium |
| 20:17.10 | grandpapadot | M1s3ry: I stand corrected ... But you just affirm my case, buy Polycom, lol |
| 20:17.22 | b11d` | there is no reason to not purchase Polycoms.. |
| 20:17.29 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 20:17.35 | M1s3ry | grandpapadot, np :D |
| 20:18.11 | M1s3ry | but I really don't have much experience with other phones so, i'm a bit of a nub in that feild |
| 20:18.18 | b11d` | Btw TK.. I got my SPA-800 for $215 and my 2102 for $61 |
| 20:18.26 | b11d` | err 8000 |
| 20:19.28 | disposable | b11d`: is that the ata for 8 phones? |
| 20:19.36 | b11d` | yes.. |
| 20:19.41 | b11d` | the 2102 for an additional 2 phones |
| 20:19.56 | M1s3ry | an ATA for multiple phones... interesting |
| 20:20.21 | disposable | the 8 utp cables into one thingy connector cost almost as much as teh device itself. |
| 20:20.37 | b11d` | what? |
| 20:20.47 | [TK]D-Fender | b11d`: Good price... congrats |
| 20:20.49 | b11d` | thanks |
| 20:21.04 | disposable | b11d`: nevermind.. we've some weird cabling with it |
| 20:21.08 | b11d` | ahh ok |
| 20:21.11 | *** join/#asterisk metfan2007 (n=metfan20@201.103.3.182) |
| 20:21.22 | [TK]D-Fender | b11d`: You've matched the per-port pricing of a PAP2 with much less hassle and better processor |
| 20:21.31 | [TK]D-Fender | b11d`: You should be very happy with it. |
| 20:21.51 | b11d` | Im certain I will be.. Thanks again for the advice. |
| 20:22.06 | [TK]D-Fender | disposable: the SPA-8000 has RJ11 jacks AND RJ-21 |
| 20:22.10 | ka0zoz | ok next question.. I need to add wireless headset support.. just a generic one work? |
| 20:22.18 | b11d` | to what kinda phone? |
| 20:22.23 | ka0zoz | with a polycom |
| 20:22.28 | ka0zoz | 600 seriew |
| 20:22.28 | metfan2007 | Hi all, I have a question, I'm receiving a lot of "Remote UNIX connection" and "Remote UNIX connection disconnected" in my Asterisk CLI. I don't have any external app that consult Asterisk, how can I know what's the problem? |
| 20:22.47 | b11d` | i like the plantronics ones, but I HATE THE LIFTERS!@!@! |
| 20:23.03 | [TK]D-Fender | b11d`: jsut get a straight amp like the M22 then |
| 20:23.16 | b11d` | I understand Polycom has some new kind of electronic hook now that you can use wireless bluetooth headsets and not need a lifter.. |
| 20:23.17 | [TK]D-Fender | metfan2007: Its eiter AMI, or CLI./ |
| 20:23.20 | b11d` | M22 eh.. |
| 20:23.22 | Qwell | b11d`: people were talking the other day about electronic hookswitch support in some sets |
| 20:23.26 | [TK]D-Fender | b11d`: yup |
| 20:23.32 | ka0zoz | we currently have some old partner phones and they showed me the wireless option with the lifter and I laughed. |
| 20:23.36 | metfan2007 | [TK]D-Fenfer: CLI |
| 20:23.38 | EmleyMoor | My "call a mobile" macro is now incredibly complicated - 224 lines! |
| 20:23.47 | metfan2007 | [TK]D-Fender: CLI |
| 20:24.06 | [TK]D-Fender | Qwell: Yes, Polycom now supports Jabra's answer signalling standard. On the 550/560 I believe, not sure if others can as well... |
| 20:24.14 | b11d` | interesting.. |
| 20:24.15 | disposable | [TK]D-Fender: we use the rj-21 out of which 8 utp cables come out. they're custom made and expensive |
| 20:24.24 | [TK]D-Fender | metfan2007: No, I'm saying what can CAUSE it. You DO have a process hitting up * for info. |
| 20:24.37 | [TK]D-Fender | disposable: Your choice. |
| 20:24.51 | [TK]D-Fender | disposable: I can get the stuff cheap. |
| 20:24.52 | EmleyMoor | Still, if I am moving over to "as much VoIP as possible", I have to get this right |
| 20:25.18 | disposable | [TK]D-Fender: probably not in Britain |
| 20:25.19 | metfan2007 | [TK]D-Fender: I just did a ps -fea and I don't see any strange process..., is the a way to debug that connections to guess what could be? |
| 20:26.23 | [TK]D-Fender | metfan2007: tcpdump , netstat, etc |
| 20:26.59 | disposable | [TK]D-Fensillynick: where do you get the cables from? |
| 20:27.35 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 20:27.47 | disposable | [TK]D-Fender: the rj-21 ones |
| 20:29.11 | [TK]D-Fender | ok, heading home, later all |
| 20:29.36 | killmel8tr | dont get in a fender bender fender |
| 20:30.03 | x86 | dont go getting your bender fended |
| 20:30.15 | juso | Anyone in the USA and familiar with numbers that use *67 to block and then call a asterisk number.. any way to display the number anyways its called "call capture" |
| 20:30.44 | *** join/#asterisk koukai (n=cel@ABordeaux-256-1-41-239.w90-11.abo.wanadoo.fr) |
| 20:31.50 | *** join/#asterisk AsteriskRo (n=rosadesa@190.36.190.79) |
| 20:32.19 | AsteriskRo | helluu |
| 20:32.49 | eric2 | using realtime in the dial plan, is there a way to get the smallest numeric value of some column? |
| 20:32.51 | eric2 | exten => _011X.,1,Set(BESTRATE=${REALTIME(ldrates|city_code|416)}) |
| 20:33.01 | AsteriskRo | please, anybody can help me to activate features???? |
| 20:33.05 | *** join/#asterisk tinkerghost (n=eric@host-64-179-18-177.spr.choiceone.net) |
| 20:33.29 | eric2 | I have 5 LD providers, each with different rates for 1-416 and need to find which has the cheapest rate |
| 20:33.54 | eric2 | it automatically returns the highest rate |
| 20:34.07 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:34.15 | b11d` | anyone know of a reliable voip termination service in Minnesota? |
| 20:35.01 | AsteriskRo | on CLI i see the features, but they don't work when i dial :( |
| 20:35.44 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 20:36.38 | *** join/#asterisk ruied (n=ruied@89-180-124-138.net.novis.pt) |
| 20:37.04 | *** join/#asterisk oej (n=olle@12.179.175.2) |
| 20:38.30 | AsteriskRo | <PROTECTED> |
| 20:38.47 | AsteriskRo | on CLI i see the features, but they don't work when i dial |
| 20:39.02 | b11d` | is res_features.so loaded? |
| 20:39.03 | b11d` | I guess/ |
| 20:39.04 | b11d` | ? |
| 20:39.18 | jblack | You have to enable them in the dialplan. |
| 20:39.35 | jblack | something in general like DYNAMIC_FEATURES = thing thing thing |
| 20:39.41 | koukai | hi all, is that there is somebody who know the role of the priorityjumping in the file extensions.conf |
| 20:40.01 | AsteriskRo | thanks, i'll check |
| 20:40.38 | b11d` | pretty sure it means it'll go from s,1, to s,3 (example) if there is no s,2, |
| 20:40.42 | b11d` | where otherwise it will stop.. |
| 20:40.51 | b11d` | im guessing though |
| 20:41.04 | b11d` | ~b11d |
| 20:41.07 | jbot | b11d is a constant source of misinformation... |
| 20:41.07 | b11d` | remember |
| 20:41.12 | b11d` | see |
| 20:41.27 | tinkerghost | AsteriskRo: remember to restart Asterisk after editing features.conf - reload doesn't reset features |
| 20:41.29 | juso | b11d`: you terminating in the USA? |
| 20:41.36 | b11d` | i want to give it a try.. |
| 20:41.40 | b11d` | looking for someone in MN.. |
| 20:41.53 | juso | why in MN? |
| 20:42.01 | b11d` | because im in MN.. figured the closer you were the better I guess.. |
| 20:42.10 | juso | lol no man.. |
| 20:42.19 | b11d` | less latency means better connections doesnt it? |
| 20:42.20 | juso | just get anyone in the US you will be fine. |
| 20:42.20 | jackson__ | b11d`, where about in MN are you? I'm in Rochester |
| 20:42.26 | b11d` | no way.., my wife is in Roch right now.. |
| 20:42.30 | b11d` | im in Hibbing up north |
| 20:42.34 | *** part/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 20:42.47 | jackson__ | heh, nice weather we're having today :) |
| 20:42.49 | b11d` | juso.. thanks.. |
| 20:42.51 | juso | b11d`: no latency isn't the only factor |
| 20:42.51 | b11d` | yeah rainy as fuck |
| 20:43.14 | b11d` | starting to turn to snow up here man.. |
| 20:43.14 | jackson__ | yeah, I heard that on the radio :( |
| 20:43.14 | b11d` | juso.. cool.. any providers you recommend? |
| 20:43.20 | juso | naah i been out the loop for a while.. |
| 20:43.25 | b11d` | I guess I'd just like to give it a shot for a little bit.. just see how it works |
| 20:43.38 | b11d` | jackson__.. rochester rocks though.. best city ever |
| 20:44.04 | b11d` | my wife went to Mayo HS.. |
| 20:44.15 | b11d` | she always yaps it up |
| 20:44.18 | koukai | thanks b11d |
| 20:44.25 | b11d` | koukai? |
| 20:44.34 | b11d` | oh.. yeah..was I right? |
| 20:44.39 | jackson__ | Yeah, Rochester's nice. I'm not one to bitch about the lack of nightlife. For my hobbies (aviation) it's nice. |
| 20:45.05 | b11d` | it seems fine to me.. i go to the same bar.. its always dead, which is how I like it.. |
| 20:45.15 | jackson__ | yeah, me too. |
| 20:45.27 | b11d` | seems like a nice, relatively clean city.. rich assholes and whatnot :) |
| 20:45.33 | b11d` | little traffic problems, which is nice |
| 20:45.37 | jackson__ | anyway, speaking of latency, what's acceptable in terms of traceroute times? |
| 20:45.45 | b11d` | i have no idea.. |
| 20:45.51 | b11d` | never done voip across the internet yet |
| 20:46.03 | b11d` | im all voip internally, but i push my shit out across a local PRI i have on site.. |
| 20:46.20 | juso | Anyone in the USA and familiar with numbers that use *67 to block and then call a asterisk number.. any way to display the number anyways its called "call capture", if im not mistaken its ani |
| 20:46.22 | grandpapadot | jackson__: < 150ms will yeild toll-quality audio assuming adequate bandwidth both directions |
| 20:46.28 | grandpapadot | yield, even |
| 20:46.29 | jackson__ | I have one hop that's ~ 264ms, most others are around ~80ms |
| 20:46.48 | b11d` | ~80ms would be perfect IMHO |
| 20:47.04 | jackson__ | I have someone complaining about audio breaking up. |
| 20:47.14 | b11d` | what codec are you using? |
| 20:47.17 | b11d` | straight ulaw? |
| 20:47.18 | grandpapadot | Of course, the codec plays a part. If you're using ulaw, your latency will be higher and you will consume more bandwidth. If you're using g729, there will be less latency and less bandwidth consumed |
| 20:47.19 | jackson__ | ulaw |
| 20:47.29 | b11d` | might want to try something else |
| 20:47.34 | grandpapadot | jackson__: g729 |
| 20:47.44 | b11d` | is g729 licence/patent encumbered at all? |
| 20:47.45 | jackson__ | we have 4 t1 bundles (6Mbit), and we're only at ~3mbit today. |
| 20:48.05 | Qwell | b11d`: VERY much so |
| 20:48.08 | b11d` | :( |
| 20:48.26 | jackson__ | grandpapadot, I'll see if our voip provider will do g729 and give that a shot. |
| 20:48.35 | grandpapadot | jackson__ who's your provider? |
| 20:48.38 | b11d` | any reasonable alternatives? |
| 20:48.46 | C4away | anyone know if there is a wait option in zapata.conf for FXOs |
| 20:48.47 | jackson__ | Vocalocity, a hosted pbx provider. |
| 20:48.50 | C4away | wait for dialtone |
| 20:48.59 | jackson__ | we're not using my new shiney Asterisk box yet. |
| 20:49.02 | C4away | I thought there was but all I find is callprogress=yes |
| 20:49.08 | C4away | which is a different thing |
| 20:49.11 | grandpapadot | No, they don't do g729. IronVoice will (http://ironvoice.com). |
| 20:49.44 | C4away | can you tag on a few pauses to the beginning of the dial string? |
| 20:49.50 | jackson__ | grandpapadot, you're familliar with Vocalocity then? I had never heard of them before a week or so ago. |
| 20:50.20 | grandpapadot | Yes, very familiar. |
| 20:50.36 | grandpapadot | They are basically a hosted asterisk solution. |
| 20:50.40 | jackson__ | you connecting up to them with *? |
| 20:50.42 | b11d` | can anyone recommend a reliable voip terminator ? |
| 20:50.48 | grandpapadot | Not bad service, you could do worse (i.e., Packet 8) |
| 20:50.53 | b11d` | that maybe will give me a demo account for a couple days or somehting? |
| 20:51.07 | b11d` | i just want to "experience" it firsthand without subscribing long-term |
| 20:51.19 | AsteriskRo | hi again |
| 20:51.23 | grandpapadot | ulaw is 64k, that's a *lot* of upstream data to push on typical US broadband. |
| 20:51.25 | jackson__ | b11d`, I've been playing with voxprime.net - pay as you go termination and origination services. |
| 20:51.36 | b11d` | cool.. i'll check them out |
| 20:51.52 | grandpapadot | Which is why iLBC and g729a work so much better, they have facilities to deal with packet loss and because they compress, less bandwidth ... |
| 20:51.56 | b11d` | yeah I definatly do not want to use ulaw across the internet.. |
| 20:51.58 | AsteriskRo | i see the features on my CLI console, i want to use the "Attended transfer" and it has assigned the *2, but doen't do anything |
| 20:51.59 | jackson__ | jackson__, then I'm also playing with les.net termination - also pay as you go. |
| 20:52.09 | jackson__ | er b11d` ^^^ |
| 20:52.12 | b11d` | cool |
| 20:52.21 | b11d` | voxprime.net doesnt exist :( |
| 20:52.26 | jackson__ | I'm an asterisk/voip newbie, but I've been reading the book :) |
| 20:52.39 | b11d` | .com works ;) |
| 20:53.38 | jackson__ | b11d`, opps, sorry on the .net :) |
| 20:54.40 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:55.43 | b11d` | its ok.. i forgive you my friend :) |
| 20:55.51 | AsteriskRo | i have to set something in extensions.conf to use basic features like atxfer ? |
| 20:58.03 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 20:58.50 | AsteriskRo | please, do I have to set something in extensions.conf to use basic features like atxfer ??? |
| 20:59.47 | *** join/#asterisk s0lid (n=s0lid@210.213.198.112) |
| 21:02.17 | b11d` | well those provders look promising.. their websites are a little shady L) |
| 21:02.21 | *** join/#asterisk XnOSX (i=53e75a2c@gateway/web/ajax/mibbit.com/x-969b3a012560d9d8) |
| 21:02.34 | juso | if I get an incoming call thats is blocked, (*67)I should still get the number in CDR right? depending on my sip provider I guess? |
| 21:02.59 | XnOSX | hello friends, what can i do for enter to CLI asterisk console in color mode? |
| 21:03.00 | jackson__ | b11d`, yeah - I agree. I plopped ~$10 via paypal into each on to play around with. lasts a while at 1.6cents/min. |
| 21:03.43 | grandpapadot | les.net has pretty good service ... |
| 21:04.07 | b11d` | cool |
| 21:04.11 | b11d` | im going to do the same I think :) |
| 21:04.16 | jackson__ | so grandpapadot - back to Vocalocity, have you used them as a sip trunk with Asterisk? |
| 21:04.19 | C4away | juso: it depends on if they are using the ANI for CID or just CID as presented |
| 21:04.20 | b11d` | did they provide you with a DID too? |
| 21:04.25 | *** part/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 21:04.44 | jackson__ | b11d`, yeah les.net has some MN numbers, none for my area code. Not that it really matters I spose. |
| 21:04.57 | C4away | if they are connected to the SS7 network directly they will get real ANI info and can place that as CID |
| 21:05.28 | juso | C4away: "they" as in my sip provider that terminates, or they as in the originating caller? |
| 21:05.35 | C4away | many ITSPs are not CLECs so they pay another company for the SS7 integration and therefore have no control over ANI |
| 21:05.40 | grandpapadot | Vocalocity, much like IronVoice, are hosted PBX solutions bundled with your phone service to give you a seamless "Just Add Phones" way to take advantage of low cost Internet based VoIP. Neither one of those services is good for termination. You would be better off with vitelity, les.net, voicepulse, or any of the other dozen or so ITSP's |
| 21:05.46 | C4away | the ITSP |
| 21:06.32 | C4away | your sip provider |
| 21:06.34 | juso | Ahh ok, so what do you suggest I do if I need to go about setting a system up that will capture phone number that call in, blocked or not.. email some ITSP and ask? |
| 21:06.47 | C4away | I have a provider that I could set you up with |
| 21:06.58 | C4away | 303/720 area codes only I think |
| 21:07.07 | juso | yeah? niice msg me them and I will signup this week. |
| 21:07.15 | C4away | probably toll free as well though |
| 21:07.32 | juso | sweet. |
| 21:11.13 | jackson__ | So, a question that's been at me for a while. Say my company goes wtih my Asterisk box using a ITSP for SIP trunking. We would like a way to integrate a pstn line in with the mix, so that when the situation occurs that our ITSP is unreachable (our internet is down), our incoming calls are routed to us via the pstn line. |
| 21:11.55 | jackson__ | are there ITSP's who'll call forward our main SIP provided number to the pstn number? |
| 21:12.51 | *** join/#asterisk RobH (n=RobH@dsl017-048-227.sfo4.dsl.speakeasy.net) |
| 21:14.38 | [TK]D-Fender | jackson__, call them and ask |
| 21:15.17 | [TK]D-Fender | AsteriskRo, Not if you're dealing with a decent phone. |
| 21:17.01 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 21:19.47 | bakermd | I'm trying to use AGI to originate a call - getting the following error in the console - any ideas? :: Starting SIP/outboundprovider-08dfc698 at outboundprovider,12125551212,1 failed so falling back to exten 's' |
| 21:21.03 | [TK]D-Fender | bakermd, Clearly where you are sending the call is not getting a match |
| 21:21.12 | *** join/#asterisk VaNNi (n=VaNNi___@lgb-static-216.70.165.200.mpowercom.net) |
| 21:21.32 | bakermd | TK - Right, what config would you like to seE? |
| 21:23.31 | [TK]D-Fender | bakermd, I don't need anything. Look where you are sending your call. |
| 21:23.42 | [TK]D-Fender | bakermd, Its a dialplan error, go LOOK for yourself |
| 21:24.06 | bakermd | so, issue is in my extensions.conf most likely? |
| 21:25.30 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 21:25.40 | [TK]D-Fender | bakermd, .... |
| 21:25.50 | bakermd | nm - I'm working on it |
| 21:26.03 | bakermd | Should not be asking to be spoon-fed ;) |
| 21:26.34 | bakermd | As always, I appreciate your time, TK. You are the best at this |
| 21:28.38 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 21:29.04 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 21:42.22 | bakermd | [TK]D-Fender: Got it working ;) Sometimes we need ppl to just tell us "..." to let us know we are asking a dumb question :) |
| 21:43.12 | carrar | I can do that |
| 21:43.18 | bakermd | lol |
| 21:43.44 | carrar | lets see... on join say ... |
| 21:43.52 | bakermd | hahaha! |
| 21:44.45 | C4away | on *:JOIN:#asterisk: /say $nick ... |
| 21:47.29 | *** join/#asterisk matthew_i (n=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 21:49.10 | carrar | It's been a LONG time since I've written any irc scripts |
| 21:49.39 | *** join/#asterisk frieze (n=frieze@cpe-69-203-15-230.nyc.res.rr.com) |
| 21:49.54 | SwK | \wii C4away |
| 21:50.00 | SwK | heh |
| 21:50.06 | SwK | hmmm |
| 21:50.27 | SwK | not who I thought it migh tbe |
| 21:50.37 | frieze | If all I want to add is a doorbell (no doorphone) is there some solution out there that doesn't cost more than an IP phone? |
| 21:51.00 | SwK | friedrich|, a smimple door box |
| 21:51.08 | SwK | those are pretty cheap |
| 21:51.12 | carrar | just hang two wires out next to the door and tell people to cross them to ring your doorbell |
| 21:51.38 | SwK | frieze, actually do you have an extra port on a PAP2 or SPA2000 |
| 21:52.04 | frieze | Don't actually know what that is |
| 21:52.40 | SwK | frieze, one of those cheap ATAs |
| 21:52.49 | b11d` | pilaster |
| 21:53.01 | frieze | ah |
| 21:53.04 | frieze | lets say I get one |
| 21:53.18 | carrar | Put a FLIR outside and ring your doorbell based on someone stading in front of your door with a human heat signature |
| 21:53.23 | frieze | fxs adapters not being crazily expensive |
| 21:53.43 | SwK | then you just need a really cheap door box... they have one button that just picks up the line... the PAP2 and SPA2000 has a 'hotline' feature that will dial a number as soon as the port goes off hook |
| 21:53.45 | frieze | I was thinking landmines. Chinese ones are cheap now, but I really like my doormat |
| 21:53.52 | SwK | and poof it calls everyone in the house |
| 21:54.20 | SwK | just get a cheap "analog call box" |
| 21:54.34 | frieze | SwK: thanks |
| 21:54.50 | *** join/#asterisk quigon (n=matias@2-17-17-190.fibertel.com.ar) |
| 21:55.04 | frieze | now to find an indoor one that doesn't look like crap, or an outdoor one that costs less that $500 |
| 21:55.35 | SwK | frieze, thats easy... go to greybar ... theres a few locations around the city |
| 21:56.37 | C4away | or a crappy analog phone you get at the thrift store for $1 |
| 21:56.48 | SwK | they should have them in stock its a common item... and if you are deploying it in any area where you IP says you probably are, you can even get stainless enclosure ones for outside so people dont screw it up |
| 21:57.27 | C4away | set dtmfmode=none or point them to an IVR that only has so many options and nothing that could dial out |
| 21:58.00 | C4away | an apartment building would need an IVR for unit number, but a single family dwelling would not need any DTMF |
| 21:59.02 | *** join/#asterisk mgroman (n=stansmit@d60-65-93-136.col.wideopenwest.com) |
| 21:59.10 | Juggie | mog! |
| 21:59.11 | jjshoe | frieze i've seen the door stuff to fxo, or was it fxs... like intercom thingy |
| 22:00.02 | *** part/#asterisk mgroman (n=stansmit@d60-65-93-136.col.wideopenwest.com) |
| 22:00.35 | jjshoe | http://www.ablecomm.com/doorinvoicvi.html |
| 22:01.00 | *** part/#asterisk AsteriskRo (n=rosadesa@190.36.190.79) |
| 22:01.42 | frieze | yeah I don't really need an intercom. All I need is a doorbell. |
| 22:02.23 | jjshoe | heh |
| 22:02.23 | jjshoe | http://www.voip-info.org/wiki/view/Asterisk+phone+door |
| 22:03.12 | frieze | yeah those all are doorphones and most seem to weigh in around $500 |
| 22:03.20 | jjshoe | http://www.voip-info.org/wiki/view/HowTo%3A%20%20Doorbell/Doorphone |
| 22:03.48 | jjshoe | i think you haven't done a whole lot of research... |
| 22:05.51 | jjshoe | W-2000A can be had for $150 from what I can see |
| 22:06.32 | *** join/#asterisk dworkin (n=asdf@cpc2-barn4-0-0-cust177.brnt.cable.ntl.com) |
| 22:09.49 | dworkin | is it possible to use a macro instead of manually adding 200 entries in sip.conf or are macros only for extensions.conf? |
| 22:12.17 | *** join/#asterisk Trapa (n=no@207.230.238.90) |
| 22:13.42 | matthew_i | dworkin, you could use a #exec directive |
| 22:13.44 | C4away | dworkin: you can use a template |
| 22:13.56 | C4away | [peer](!) |
| 22:13.59 | C4away | details for all of them |
| 22:14.05 | C4away | [peer1] |
| 22:14.10 | C4away | details specific to this peer |
| 22:14.11 | Trapa | good evening people. I thank this channel for all the support they have given me lately, and want you all to know i appriicate it. The latest problem i'm experienceing is when i'm trying to make a call to a remote extenion exten => _7XX,1,Dial(SIP/pbx.somdomain.com/${EXTEN}) ... It's passing my domain name instead of a proper username that i've registered with, with the register command. |
| 22:14.30 | *** join/#asterisk DrkShadow (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 22:14.41 | matthew_i | #exec /usr/bin/leet-sip.conf-generator.sh |
| 22:15.05 | matthew_i | dworkin, that would execute the script and include it's output in the config file |
| 22:15.30 | dworkin | so either way, i'll still end up with hundreds of lines in my sip.conf, right? |
| 22:15.52 | DrkShadow | I'm trying to work with a provider that is sending calls to username@mysipbox.com ... and I have a context in sip.conf for [username] telling it what context to start the extension in, and host=dynamic, but the calls never go to that context -- always the context set up in the generic section of sip.conf. What's up? |
| 22:15.54 | C4away | realtime |
| 22:16.03 | C4away | generate the dialplan from sql server |
| 22:16.16 | matthew_i | dworkin, no, my way you only have that 1 #exec line |
| 22:16.40 | dworkin | matthew_i, thanks, where can i find more info about this? |
| 22:16.45 | matthew_i | dworkin, hmmm |
| 22:16.50 | matthew_i | dworkin, sample config files? |
| 22:17.29 | matthew_i | dworkin, I basically gave you all the information you need, #exec runs the given script, and it's output is treated as if it was written in the config file |
| 22:19.36 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 22:19.50 | *** part/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 22:19.53 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 22:19.54 | juso | was thinking of putting my 800 number on a telco and forwarding to my voip setup, anyone recommend a major telco to get toll free number without hassels? |
| 22:19.55 | dworkin | matthew_i, so i could use #exec to execute a shell script that generates my sip.conf lines? (voip-info isn't very helpful) |
| 22:20.08 | matthew_i | dworkin, you are correct |
| 22:20.20 | matthew_i | dworkin, really any script, shell, perl, c, lua... |
| 22:20.58 | *** join/#asterisk gitguy (n=diego@adsl-149-162.click.com.py) |
| 22:21.00 | gitguy | hi |
| 22:21.09 | gitguy | can someone help me with setting up asterisk/SIP behind NAT? |
| 22:21.12 | matthew_i | hmm, using #exec, you could replace static realtime configs... |
| 22:21.12 | *** join/#asterisk RoyK (n=roy@ip-150-20-149-91.dialup.ice.no) |
| 22:21.13 | gitguy | please |
| 22:21.14 | dworkin | matthew_i, thanks now i've got to learn to do those loops properly.. |
| 22:21.30 | *** part/#asterisk RoyK (n=roy@ip-150-20-149-91.dialup.ice.no) |
| 22:21.37 | DrkShadow | Does anyone know how to set the context calls go to when they're coming from a service provider that doesn't have a fixed hostname? |
| 22:21.40 | gitguy | I have nat=yes and externip=<ip address> in my sql.conf |
| 22:21.42 | gitguy | errr |
| 22:21.44 | gitguy | sip.conf |
| 22:22.09 | [TK]D-Fender | gitguy, read up : |
| 22:22.11 | matthew_i | gitguy, you may need to specify localnet as well |
| 22:22.11 | [TK]D-Fender | ~sipnat |
| 22:22.12 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:22.17 | [TK]D-Fender | ^^^^^^ |
| 22:22.35 | gitguy | my asterisk is in Amazon EC2 |
| 22:22.36 | C4away | sql.conf? |
| 22:22.39 | gitguy | [TK]D-Fender: ok i will read that |
| 22:22.53 | gitguy | C4away: my mind doesn't work too well after hours of no sleep + coffee |
| 22:22.54 | matthew_i | DrkShadow, if they are coming from a specific IP address you can create a sip user to do it |
| 22:22.58 | [TK]D-Fender | DrkShadow, It should match against a peer with the appropriate user & pass. You can leave "host=dynamic" for that. |
| 22:23.01 | C4away | lol |
| 22:23.21 | [TK]D-Fender | DrkShadow, but it should match an entry |
| 22:23.28 | matthew_i | DrkShadow, [192.128.0.1] type=user; context = special-provider; |
| 22:23.33 | C4away | I shouldn't even be here anyway, I have service calls to take care of |
| 22:23.37 | DrkShadow | tk: I can give a user, but I can't give a pass.. and even with [user]type=user username=<user>, it doesn't send me to the context.. |
| 22:23.52 | DrkShadow | matthew: I'm not sure the IP is the same every time. I'm not going to trust that. |
| 22:24.25 | matthew_i | DrkShadow, in that case, [TK]D-Fender's solution might work (with only a user name), but im not sure |
| 22:24.53 | [TK]D-Fender | DrkShadow, pastebin the sip debug and your peer config |
| 22:24.55 | [TK]D-Fender | ~pb |
| 22:24.56 | 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 |
| 22:24.57 | [TK]D-Fender | ^^^^^^^^^^^ |
| 22:25.14 | DrkShadow | tk: host.. checking that real quick.. I think I left out host=dynamic |
| 22:25.25 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 22:26.47 | DrkShadow | how... do I turn off sip debug again? |
| 22:27.09 | matthew_i | sip set debug off |
| 22:27.10 | DrkShadow | got it. |
| 22:27.13 | DrkShadow | ty ;-) |
| 22:27.45 | DrkShadow | omfg. I think i hit well over a hundred messages :-/ |
| 22:29.00 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 22:29.00 | *** join/#asterisk Acxty (n=x3rus@201.220.132.138) |
| 22:29.50 | gitguy | [TK]D-Fender: i already did what that guide said, but still, my softphones wont connect to my asterisk |
| 22:30.09 | matthew_i | gitguy, what do you mean won't connect? |
| 22:30.19 | gitguy | doesn't register |
| 22:30.22 | matthew_i | gitguy, does tcpdump on port 5060 show any traffic at all? |
| 22:30.22 | gitguy | i have |
| 22:30.34 | matthew_i | gitguy, also does sip debug show any traffic? |
| 22:30.36 | gitguy | in general, externip, localnet, nat=yes |
| 22:30.47 | gitguy | matthew_i: it doesnt |
| 22:31.01 | matthew_i | gitguy, in that case you have a networking problem |
| 22:31.12 | matthew_i | gitguy, make sure your firewall has port 5060 open |
| 22:31.34 | matthew_i | gitguy, and also try pinging the phone from that box to see if they can at least see eachother |
| 22:31.38 | gitguy | matthew_i: i used to use asterisk on another server that didn't had NAT |
| 22:31.41 | gitguy | it had a public ip |
| 22:31.50 | gitguy | the problem is on that server |
| 22:31.52 | gitguy | amazon |
| 22:31.55 | *** join/#asterisk edibrac (n=edibrac3@netblock-66-245-252-161.dslextreme.com) |
| 22:31.59 | matthew_i | gitguy, if asterisk is completely behind the nat make sure your router is forwarding the proper ports |
| 22:32.57 | gitguy | matthew_i: i'm behind the nat too, i was using Asterisk+SIP just fine before, with another asterisk server |
| 22:33.07 | gitguy | matthew_i: so it can't be my router right? |
| 22:33.36 | matthew_i | gitguy, hmm, well if sip debug does not show any registration attempts, it's probably not your asterisk configuration (unless you told asterisk to listen on 127.0.0.1 or something) |
| 22:34.10 | matthew_i | i don't have enough information to tell you if it is your router or not |
| 22:34.23 | gitguy | it's not my router |
| 22:34.29 | gitguy | it's probably a firewall issue on amazon |
| 22:34.44 | matthew_i | oh, this box is on amazon's cloud thingy |
| 22:34.49 | gitguy | yeah |
| 22:35.18 | gitguy | i'm better off putting asterisk somewhere else |
| 22:35.29 | gitguy | but my boss loves amazon |
| 22:38.03 | gitguy | http://developer.amazonwebservices.com/connect/thread.jspa?messageID=56003 |
| 22:38.11 | gitguy | they have some weird nat stuff going on there |
| 22:39.40 | *** join/#asterisk pukepail (n=pukepail@pool-72-69-75-242.chi01.dsl-w.verizon.net) |
| 22:39.48 | gitguy | "You do not see your public IP because your public IP address is mapped to your private address using 1:1 NAT. " |
| 22:40.01 | matthew_i | i looked through that, it was confusing |
| 22:40.11 | matthew_i | i stopped reading about halfway through |
| 22:40.14 | gitguy | yeah |
| 22:41.14 | gitguy | I shouldn't have to deal with firewall stuff, I guess I will tell my boss to move somewhere else |
| 22:42.01 | matthew_i | gitguy, or you could just run without the NAT |
| 22:42.37 | gitguy | mtnguy: without the nat, what do you mean? |
| 22:43.25 | matthew_i | gitguy, you should be able to run an EC2 instance on a public IP address |
| 22:47.45 | *** join/#asterisk bminish (n=bminish@brenbox.home.minish.org) |
| 22:49.32 | gitguy | matthew_i: really? |
| 22:49.37 | [TK]D-Fender | gitguy, pastebin your sip.conf masking only passwords |
| 22:49.37 | gitguy | matthew_i: where did you read that? |
| 22:49.47 | gitguy | [TK]D-Fender: ok |
| 22:49.50 | DrkShadow | Fender: are you still around? the relevant section, I believe, is here: http://rafb.net/p/Dxe8Uf78.html -- Asterisk says "Found no matching peer or user for '<ipaddr>:port'. It seems to not make any difference what the username I specify in <user>@<myhost>.com is in the sip routing for the number.. |
| 22:50.50 | [TK]D-Fender | DrkShadow, who are these people? |
| 22:51.44 | matthew_i | gitguy, well if NAT is a new feature im assuming you can run it with out nat, I don't really know, I don't have much experience with ec2 |
| 22:52.12 | gitguy | [TK]D-Fender: http://pastebin.com/m13798a03 |
| 22:52.29 | gitguy | matthew_i: ok |
| 22:52.32 | gitguy | matthew_i: i will ask them |
| 22:53.27 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 22:54.06 | JayTee52 | can someone help me decipher this line, I've read through the Dial function in the book but I'm still confused. |
| 22:54.09 | JayTee52 | exten => _XXXX,1,Dial(Zap/g1/www,15,mD(${EXTEN})) |
| 22:54.19 | [TK]D-Fender | gitguy, So * is behind NAT, and your phone is behind its own as well? |
| 22:54.36 | EmleyMoor | JayTee52: What confuses you about it? |
| 22:54.40 | DrkShadow | fender: as in the company? .... I'm not sure. The number was bought through TTNC -- The Telephone Number Company -- and they use the "Novero Platform". TTNC is a UK company. |
| 22:54.44 | [TK]D-Fender | JayTee52, What are you dialing out on? |
| 22:54.47 | DrkShadow | fender: so, any ideas? |
| 22:54.50 | JayTee52 | I'm pattern matching on any 4 digit extension and dialing on a zap channel on group 1 but what does the www and the mD do? |
| 22:55.07 | gitguy | [TK]D-Fender: yeah, * is behind NAT and my phone is behind NAT too |
| 22:55.09 | JayTee52 | the mD seems to be music on hold default |
| 22:55.27 | [TK]D-Fender | DrkShadow, Could be they send calls to you completely un-authed. add "allowguest=yes" under [general] and do a catch-all in the context your spcific under [general] |
| 22:55.50 | DrkShadow | fender: that's essentially what I've done.. I just hoped for a better way :-/ |
| 22:55.52 | [TK]D-Fender | gitguy, you also need "qualify=yes" for your phone's entry. You left that out. |
| 22:55.55 | EmleyMoor | The w implies "wait" to me... |
| 22:55.58 | gitguy | [TK]D-Fender: i don't think is my router because i was using my phone behind NAT before, with my asterisk being on slicehost (without NAT) |
| 22:56.03 | matthew_i | JayTee52, that line opens the line then pulses out some digits |
| 22:56.04 | gitguy | [TK]D-Fender: ok |
| 22:56.12 | [TK]D-Fender | DrkShadow, Doesn't seem to be a way yet if you have no way to ID the incoming attemp |
| 22:56.39 | DrkShadow | fender: I'd love to ID it by phone number... so I gave them the phone number as the username ;-) |
| 22:56.39 | [TK]D-Fender | JayTee52, What kind of line is that? |
| 22:56.42 | JayTee52 | so the 3 w's mean wait or the 15 means wait? |
| 22:56.45 | matthew_i | JayTee52, the www (chan_zap specific) causes chan_zap to wait a few seconds, then the D() makes it pulse digits once it is answered |
| 22:57.01 | JayTee52 | [TK]D-Fender, it's from a 1.2 install and part of it got carried over to my 1.4 install. |
| 22:57.02 | matthew_i | JayTee52, the 15 is the timeout for no answer |
| 22:57.10 | DrkShadow | though if I give a textual username, Asterisk says "Extension <textual username> not found" |
| 22:57.11 | JayTee52 | ah, ok |
| 22:57.16 | [TK]D-Fender | JayTee52, what kind of LINE is that? ZAPTEL <- |
| 22:57.26 | matthew_i | JayTee52, you could also write it like: |
| 22:57.37 | matthew_i | exten => _XXXX,1,Dial(Zap/g1/www${EXTEN},15,m) |
| 22:57.39 | matthew_i | i think |
| 22:57.52 | JayTee52 | [TK, it was for an FXO line but now I'm trying to setup outgoing on a PRI |
| 22:58.08 | [TK]D-Fender | JayTee52, you don't DO DTMF to dial out a PRI |
| 22:58.09 | JayTee52 | oops, meant [TK]D-Fender . |
| 22:58.21 | *** part/#asterisk crakrjak (n=merc@cpe-75-187-108-229.insight.res.rr.com) |
| 22:58.35 | [TK]D-Fender | JayTee52, exten => _XXXX,1,Dial(Zap/g1/${EXTEN},15,m) |
| 22:58.42 | JayTee52 | ok |
| 22:58.53 | [TK]D-Fender | JayTee52, And you typically don't dial 4-digit numbers on them, so adjust your pattern as needed |
| 22:59.33 | JayTee52 | we are passed 4 digits from our telco. I've got my two PRI spans looped so I can make test calls outbound then back in on the other span just for testing. |
| 22:59.38 | [TK]D-Fender | DrkShadow, Well the "To:" means they should be dialing the # you provided when you register so you can do without the catch-all and instead do an exact match on that. |
| 23:00.03 | [TK]D-Fender | JayTee52, Just because your telco sends you 4 digits IN, doesn't mean you diakl 4 to get OUT. |
| 23:00.13 | [TK]D-Fender | JayTee52, You dial a REAL NUMBER like the rest of the world. |
| 23:00.29 | DrkShadow | fender: so I would specify host=m.e164.org.uk in my sip.conf? |
| 23:00.47 | DrkShadow | hmm.. same IP address.. |
| 23:00.55 | JayTee52 | I know, I'm setting it up to send the correct amount of digits. |
| 23:01.34 | JayTee52 | when I actually send a real call out. for now I just want to mimic incoming calls by sending one out the other span and have it route to another extension. |
| 23:01.46 | [TK]D-Fender | DrkShadow, if you're sure it'll resolve the same host for these calls. |
| 23:01.58 | DrkShadow | fender: ... I really dunno :-/ |
| 23:02.29 | DrkShadow | fender: thanks, though. I now know a lot more than I did. |
| 23:03.49 | gitguy | [TK]D-Fender: so what can I do? |
| 23:04.07 | [TK]D-Fender | gitguy, add the lines I suggested, restart your phone. |
| 23:04.19 | [TK]D-Fender | gitguy, and what model of router are your 2 sides behind? |
| 23:04.27 | [TK]D-Fender | gitguy, and what have you forwarded to *? |
| 23:04.59 | pukepail | I have a bug fix for vmail.cgi, I see it is in the contrib folder, is it acceptable to put a bug report on mantis for this? |
| 23:05.15 | [TK]D-Fender | pukepail, Of course. |
| 23:06.45 | gitguy | [TK]D-Fender: my router is a d-link WBR-2310 |
| 23:06.45 | pukepail | just put it under core/general? |
| 23:06.53 | gitguy | [TK]D-Fender: i don't know what kind of router amazon uses |
| 23:06.54 | gitguy | hmm |
| 23:07.17 | [TK]D-Fender | gitguy, D-Links are often trouble with their NAT implementation. |
| 23:07.36 | gitguy | uhm |
| 23:07.44 | gitguy | [TK]D-Fender: ok |
| 23:08.01 | [TK]D-Fender | gitguy, Yes that may entail replacement. |
| 23:08.14 | gitguy | ok |
| 23:08.17 | [TK]D-Fender | gitguy, follow the rest of what I suggested and test spaw-out the router if you can. |
| 23:08.20 | [TK]D-Fender | swap* |
| 23:09.35 | gitguy | [TK]D-Fender: but if my softphone registers well with another asterisk server that is outside my network, do you still think the problem is my router? |
| 23:09.59 | gitguy | my father will get pissed if i try to remove his router |
| 23:10.01 | gitguy | :/ |
| 23:10.26 | EmleyMoor | Are there any softphones known that can dial out from Outlook contact details? |
| 23:10.37 | matthew_i | gitguy, i think the EC2 NAT is your problem |
| 23:10.44 | gitguy | yeah |
| 23:10.56 | [TK]D-Fender | gitguy, if it works fine on another * box, then its your * situation. |
| 23:11.04 | gitguy | yeah |
| 23:11.07 | [TK]D-Fender | gitguy, and I asked what you had forwarded to your server |
| 23:11.09 | gitguy | probably the nat mess in amazon |
| 23:11.20 | gitguy | [TK]D-Fender: i don't know how to answer that |
| 23:11.35 | [TK]D-Fender | gitguy, then the answer is "screwed" |
| 23:11.44 | gitguy | oops |
| 23:12.06 | gitguy | what do you mean, what i have forwarded to my server? |
| 23:12.16 | [TK]D-Fender | gitguy, port forwarding <---- |
| 23:12.18 | gitguy | should i forward something? |
| 23:12.26 | gitguy | i haven't did any port forwarding |
| 23:12.56 | [TK]D-Fender | gitguy, I told you to read the guide, clearly you didn't. |
| 23:13.03 | gitguy | I did |
| 23:13.07 | gitguy | the first one |
| 23:13.12 | [TK]D-Fender | gitguy, Poorly. |
| 23:13.22 | gitguy | let me read it again |
| 23:13.27 | EmleyMoor | I wish I could get to grips with NAT and stuff |
| 23:14.00 | [TK]D-Fender | "Our Asterisk server will also have to have ports 5060 (UDP), and the port range specified in ârtp.confâ (typically 10000-20000 UDP) forwarded to it." |
| 23:14.16 | gitguy | oops |
| 23:14.18 | EmleyMoor | My mobile can log onto my Asterisk through a neighbour's BT kit, but gets nowhere if I try from my mother's, which has a Belkin router |
| 23:14.20 | gitguy | i haven't read that |
| 23:14.25 | [TK]D-Fender | EmleyMoor, Its all spelled out between those 2 links. If you still can't cope, quit while you're ahead. |
| 23:14.38 | EmleyMoor | looks back for the two links |
| 23:14.54 | matthew_i | it is all very complex |
| 23:15.02 | matthew_i | sip was not designed with nat in mind |
| 23:15.24 | gitguy | [TK]D-Fender: should I tell Amazon guys to change their firewall or I have to do that in my vps? |
| 23:15.34 | EmleyMoor | The N95 was not designed with IAX in mind :-) |
| 23:15.46 | [TK]D-Fender | gitguy, Never dealt with them. You know what you need to do. Deal with them. |
| 23:16.12 | [TK]D-Fender | EmleyMoor, Yeah, I think they had their minds set on GSM for some reason.... |
| 23:16.47 | gitguy | [TK]D-Fender: uhm, that confused me a bit |
| 23:17.10 | matthew_i | gitguy, you can either make your firewall do the proper port forwarding, or just disable it all together and run on a public ip |
| 23:17.11 | gitguy | "when in doubt, use brute force" |
| 23:17.24 | [TK]D-Fender | gitguy, No idea how EC2 operates or if you can fix this at all. Go ask their techs, I have no answers for you on this matter |
| 23:17.37 | gitguy | ok |
| 23:17.38 | EmleyMoor | [TK]D-Fender: ... and SIP |
| 23:17.50 | [TK]D-Fender | matthew_i, its a VPS he doesn't fully control. |
| 23:18.03 | matthew_i | [TK]D-Fender, i know |
| 23:18.10 | [TK]D-Fender | Ok, GTG, later all. |
| 23:18.19 | gitguy | matthew_i: that's what I would like to do, run my vps on a public ip |
| 23:18.27 | gitguy | matthew_i: I already asked them if that was possible |
| 23:18.28 | gitguy | heh |
| 23:18.59 | EmleyMoor | Is there a guide to solving SIP NAT problems classified by symptom? |
| 23:19.56 | [TK]D-Fender | EmleyMoor, usually there is 1 symptom : no audio. |
| 23:20.10 | [TK]D-Fender | EmleyMoor, do as the first guide says and you're usually set |
| 23:20.22 | EmleyMoor | hasn't found the links |
| 23:20.33 | matthew_i | basic nat issues are pretty straightforward, but it get's complex when the server is behind a nat |
| 23:20.51 | EmleyMoor | Besides, I am getting two different symptoms... no register, and register, audio but does not ring... |
| 23:21.15 | matthew_i | does not ring? that is odd |
| 23:21.35 | [TK]D-Fender | ~sipnat |
| 23:21.36 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:21.37 | [TK]D-Fender | ^^^^ |
| 23:21.42 | matthew_i | no register, probably means bad credentials or bad packet routing |
| 23:21.48 | [TK]D-Fender | EmleyMoor, its been repeated hundreds of times... |
| 23:21.57 | EmleyMoor | Definitely not bad credentials |
| 23:22.21 | [TK]D-Fender | EmleyMoor, Yes, complete 1-way traffic is one, 1-way audio is the other. |
| 23:22.31 | matthew_i | well im out of here, later |
| 23:22.36 | *** part/#asterisk matthew_i (n=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 23:22.38 | [TK]D-Fender | ditto. |
| 23:23.11 | gitguy | I'm out of luck |
| 23:23.21 | gitguy | there is no way to have a EC2 server without NAT |
| 23:23.24 | gitguy | amazon sucks |
| 23:23.45 | EmleyMoor | I will have to investigate this further |
| 23:24.01 | EmleyMoor | All my scenarios involve Asterisk itself out on a public IP |
| 23:24.17 | gitguy | what is a good server for asterisk |
| 23:24.31 | EmleyMoor | gitguy: You mean in hardware terms? |
| 23:24.38 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 23:25.10 | gitguy | one that I could buy on internet and have all ready to compile asterisk |
| 23:25.15 | gitguy | without nat |
| 23:26.26 | gitguy | meh |
| 23:26.37 | gitguy | I will try to beat that ugly nat |
| 23:27.22 | EmleyMoor | The thing is, I have scenarios where my N95 is quite happy to log in to my home Asterisk, and other similar ones where it fails totally... |
| 23:28.06 | EmleyMoor | I suppose the main thing to do is work out the difference |
| 23:29.21 | EmleyMoor | If I do crack it, I might want to give my mother an Asterisk box too |
| 23:29.46 | DrkShadow | How would you guys go about implementing a follow-me-sort of system? I want it to first ring the office phone, and if that doesn't pick up, I want it to ring, at the same time, office, home, cell, other, whatever else... but that causes a problem if they have an extension in the company, which doesn't need to have details recorded, and a line outside the company, which does need to have call... |
| 23:29.48 | DrkShadow | ...info recorded... thougths? |
| 23:31.52 | EmleyMoor | I think another day and a half or so should sort my dialplan |
| 23:32.12 | EmleyMoor | is planning a move as much in favour of his VoIP provider as possible soon |
| 23:33.37 | EmleyMoor | might be moving to Essex - at least all the VoIP phone numbers can go with him |
| 23:34.42 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 23:38.03 | DrkShadow | can I play music on hold until a dial() app succeeds? |
| 23:39.06 | EmleyMoor | Are any of you lot actually using rotary phones? |
| 23:39.13 | DrkShadow | ... omg man |
| 23:39.22 | DrkShadow | [wo]man |
| 23:39.28 | alrs | I've got some rotary phones of the western electric persuasion |
| 23:39.43 | EmleyMoor | I have a British Post Office one, in heavy use! |
| 23:42.13 | gitguy | IAX2 fails too |
| 23:42.18 | gitguy | when I try to register my softphone |
| 23:42.20 | gitguy | wth |
| 23:42.38 | gitguy | amazon is seriously screwed |
| 23:42.50 | alrs | EmleyMoor: I haven't used my rotary stuff much in the last year as most IVRs no longer drop-out to an operator after a minute to make sure that all of the confused old people have someone to talk to |
| 23:44.21 | EmleyMoor | 20 years ago they were still relatively common here |
| 23:44.49 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 23:44.52 | EmleyMoor | I think only Germany had a greater proportion of them at that time |
| 23:47.25 | gitguy | what do you suggest? |
| 23:47.37 | gitguy | should i get a server with public ip |
| 23:47.50 | gitguy | or i continue to try to make asterisk work behind a nat? |
| 23:48.18 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:48.45 | alrs | EmleyMoor: I remember being a 6-year-old in 1980 in the US, and kids that would come over to our house would be totally confused by our rotary phones |
| 23:49.52 | EmleyMoor | alrs: Anything other than rotary was rare over here then |
| 23:50.07 | EmleyMoor | I think 95% of phones were rotary |
| 23:50.42 | EmleyMoor | (in 1980, the Post Office still ran our phones) |
| 23:51.43 | alrs | EmleyMoor: I was born at a weird time, I'm just old enough to have gone to drive-in movie theaters in high school, to have never seen a child wearing a bicycle helmet until I was 15 |
| 23:51.55 | alrs | I remember gasoline in the late 80's selling for under a dollar a gallon |
| 23:52.14 | alrs | and yet my youth and vitality are intact |
| 23:52.22 | alrs | save for the hearing and hair loss |
| 23:53.32 | EmleyMoor | Our POTS operator was privatised in 1981 but it took quite a while longer before competitors appeared |
| 23:54.26 | EmleyMoor | It was still BT, or in Hull, the council, only, by the time I was 10 (1983) |
| 23:55.30 | EmleyMoor | Still, at least we've moved on now |
