| 00:00.04 | ix33 | what is your dtmfmode in sip.conf? |
| 00:00.07 | [TK]D-Fender | jack_sparo, PASTEBIN the entire inbound call attempt at CLI at verbose 10 |
| 00:00.07 | [TK]D-Fender | ~pb |
| 00:00.08 | 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 |
| 00:00.09 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 00:00.20 | iceyp | if I call my mobile phone for example and press DTMF on my mobile, I can hear it on the SIP phone, but if I press DTMF on the SIP phone the mobile hears distorted |
| 00:00.43 | iceyp | rfc2833 |
| 00:00.45 | ix33 | er, then that's not a pure SIP comm, eh? |
| 00:01.04 | iceyp | ahh no, let me try internal voip to voip phone |
| 00:01.13 | *** join/#asterisk SteveTotaro (n=Administ@pool-70-17-222-90.balt.east.verizon.net) |
| 00:01.24 | iceyp | mmmmm |
| 00:02.36 | jack_sparo | ok [TK]D-Fender |
| 00:02.56 | iceyp | mmmm, it only seems to be from my linksys devices |
| 00:03.29 | jack_sparo | http://pastebin.ca/1040862 |
| 00:03.30 | iceyp | I've tried using my cisco 7640 and dtmf is clear, but if i use my linksys ata 2102 it's distorted, and if I use my linksys router which has 2 ata ports, same thing |
| 00:03.32 | iceyp | this is weird |
| 00:03.41 | ManxPower | iceyp: what codec? |
| 00:03.46 | jack_sparo | may you have a look at the paste dude |
| 00:03.49 | *** join/#asterisk spiny (n=spiny@69-196-137-127.dsl.teksavvy.com) |
| 00:04.26 | [TK]D-Fender | jack_sparo, this is your dialplan, where did all that code come from? Executing [s@from-sip-external:5] Playback("SIP/208.44.220.234-b7d156b8", "ss-noservice") in new stack |
| 00:04.28 | iceyp | I've tried both ulaw, alaw and g729 |
| 00:04.55 | iceyp | the DTMF is very load though |
| 00:05.04 | [TK]D-Fender | jack_sparo, because its nothing automated, its your dialplan. You are receiving the call (un-authed no less), and dumping it into a very obvious context. |
| 00:05.06 | iceyp | loud* |
| 00:05.11 | jack_sparo | this is when i try to call the DID that is poinyed to the system |
| 00:05.22 | jack_sparo | so what shall i do? |
| 00:05.29 | jack_sparo | i dont have any dial plans added yet |
| 00:05.34 | [TK]D-Fender | jack_sparo, No. Where did your dialplan come from? |
| 00:05.46 | [TK]D-Fender | jack_sparo, all that code you see rolling by IS DIALPLAN. |
| 00:06.05 | ManxPower | iceyp: what DTMF mode is set on the phone? |
| 00:06.37 | iceyp | AUTO in the linksys |
| 00:07.01 | ManxPower | iceyp: set it to AVT. What is the DTMF mode set in Asterisk? |
| 00:07.07 | jack_sparo | which dialplan? |
| 00:07.31 | [TK]D-Fender | jack_sparo, all of that in your pastebin IS YOUR DIALPLAN <----- extensions.conf. |
| 00:07.41 | iceyp | ManxPower let me test |
| 00:07.58 | hardwire | [TK]D-Fender: I'm half tempted to put glow in the dark paper strips behind the LCD's on these SPA-941's :) |
| 00:08.01 | ManxPower | iceyp: NO! Not TEST. Tell. What dtmfmode= did you set? |
| 00:08.08 | ManxPower | jack_sparo: that paste is all normal |
| 00:08.22 | iceyp | rfc2833 in asterisk |
| 00:08.27 | iceyp | for the extension to the linksys |
| 00:08.36 | ManxPower | its a device, not an extension |
| 00:08.42 | iceyp | yeh |
| 00:08.44 | iceyp | sorry |
| 00:09.08 | jack_sparo | so what shall i do?>/ |
| 00:09.15 | TJNII | ~book |
| 00:09.16 | 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 |
| 00:09.21 | ManxPower | jack_sparo: you should admit what GUI you are using? |
| 00:09.24 | TJNII | jack_sparo: Read it ^ |
| 00:10.24 | javawizard2539 | So I'm kind of new to asterisk |
| 00:11.33 | javawizard2539 | and I'm wondering, if asterisk (specifically Asterisk@Home) is installed on a computer without PCI ports, is it possible to install a voice modem (to be used as an FXO) in another computer that does have PCI ports, and have asterisk somehow connect to it? |
| 00:11.57 | [TK]D-Fender | javawizard2539, No, you can't use any old POS with *, just won't work. |
| 00:12.11 | javawizard2539 | why? |
| 00:12.16 | [TK]D-Fender | javawizard2539, and Asterisk@Home is an OLD solution long since replaced by Trixbox |
| 00:12.26 | [TK]D-Fender | javawizard2539, Zaptel has no driver support for it |
| 00:12.31 | javawizard2539 | That's what I mean, I keep forgetting the new name |
| 00:12.33 | javawizard2539 | hmm |
| 00:12.43 | ix33 | technically he could do that by running two *'s, though :V |
| 00:13.02 | javawizard2539 | the one with a PCI port is a win32 |
| 00:13.17 | ix33 | oh good we've gone right for the comedy option. |
| 00:13.20 | javawizard2539 | will * connect to a voice modem? |
| 00:13.23 | javawizard2539 | :) |
| 00:13.30 | [TK]D-Fender | javawizard2539, What part of "no" don't you understand? |
| 00:13.40 | javawizard2539 | refer to ix33's comments |
| 00:13.49 | [TK]D-Fender | javawizard2539, which are BS. |
| 00:14.02 | [TK]D-Fender | javawizard2539, * does not function with any old POS modem. |
| 00:14.33 | javawizard2539 | then what is the difference between a voice modem and something like an X100P besides what they're called? |
| 00:14.43 | iceyp | ManxPower if i press multiple digits for dtmf it seems to skip some, almost like there is a echo canceller or something |
| 00:14.50 | [TK]D-Fender | javawizard2539, the chipset and functional capabilities |
| 00:14.55 | ix33 | well i was talking about using PCI ports in another computer. voicemodems won't work of course |
| 00:15.09 | [TK]D-Fender | javawizard2539, for which only the X100P has a zaptel driver written for it. |
| 00:15.17 | javawizard2539 | oh well |
| 00:15.34 | javawizard2539 | in that case, would * be able to connect to an X100P on another computer? |
| 00:15.38 | javawizard2539 | oh wait, that's what ix33 was saying |
| 00:15.47 | ix33 | right. |
| 00:15.48 | [TK]D-Fender | javawizard2539, I'd love to know how thats possible... |
| 00:15.57 | ix33 | also, don't use X100P's |
| 00:15.57 | javawizard2539 | If I run * on my main computer and *win32 on the computer with PCI, that would work, right? |
| 00:16.08 | javawizard2539 | what should I use? |
| 00:16.11 | ix33 | javawizard2539: NO |
| 00:16.15 | [TK]D-Fender | ix33, so.. what Win32 program will let it share that device remotely? |
| 00:16.17 | ix33 | zaptel only works on *nix |
| 00:16.20 | TJNII | Well, no. * won't be able to connect to a card on another computer. You can run two instances of * and have them connect to each other, though. |
| 00:16.27 | ix33 | TJNII++ |
| 00:16.38 | ix33 | but only using NIX on the computer *with* pci slots |
| 00:16.38 | TJNII | Which is what I believe ix33 is saying. |
| 00:16.43 | [TK]D-Fender | javawizard2539, To my awareness there is no ZAPTEL available for Win32 so the answer would be "no" |
| 00:16.47 | ManxPower | javawizard2539: none of this helps you. |
| 00:17.06 | ManxPower | give up now before you waste more time, or get hardware that works with Asterisk |
| 00:17.24 | javawizard2539 | If I had the money I would |
| 00:17.43 | [TK]D-Fender | javawizard2539, Then do something else, because this line of thinking is a dead-end. |
| 00:17.44 | ix33 | javawizard2539 just sign up with voicepulse and forget hardware altogether |
| 00:19.26 | TJNII | javawizard2539: pap2s are cheap on eaby |
| 00:19.30 | TJNII | good starter device |
| 00:19.40 | [TK]D-Fender | TJNII, Which doesn't do FXO |
| 00:19.53 | [TK]D-Fender | TJNII, Try to follow the conversation. |
| 00:19.54 | TJNII | Oh, did he want FXO? I missed that. |
| 00:20.07 | TJNII | makes an inappropriate gesture towards [TK]D-Fender |
| 00:20.21 | ix33 | FXO more trouble than it's worth, even with a $200 TDM card. unless this is a hobby. |
| 00:20.53 | ix33 | get a PSTN telephone number from voicepulse and forget the second computer entirely |
| 00:21.10 | deeperror | what is a good price on TDM11B |
| 00:21.25 | ix33 | it's not called that anymore, but $230-ish USD |
| 00:22.35 | ix33 | but i'm assuming new. if you're talking ebay i have no idea. |
| 00:22.58 | [TK]D-Fender | deeperror, For what kind of use? |
| 00:24.30 | deeperror | just home use configuring practice |
| 00:25.14 | [TK]D-Fender | deeperror, An X100P would be fine for configuring proactice. Its not like its any different for FXO. For FXS.. well Zaptel FXS simply isn't worth it anyways. |
| 00:26.13 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 00:26.13 | *** mode/#asterisk [+o denon] by ChanServ |
| 00:26.56 | ix33 | [TK]D-Fender: why do you say that about FXS? is it just not common? |
| 00:27.11 | [TK]D-Fender | ix33, I said ZAPTEL FXS. |
| 00:27.38 | deeperror | so you could get any phone line into fxo? |
| 00:27.42 | [TK]D-Fender | ix33, for standard use ATA's are much cheaper, flexible, smaller load on your system, and less requirements. |
| 00:27.52 | [TK]D-Fender | deeperror, ... huh? |
| 00:27.58 | ix33 | well okay i was assuming that's what you meant. thanks. |
| 00:28.10 | deeperror | ~fxo |
| 00:28.11 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 00:32.22 | *** join/#asterisk javawizard2539_ (n=chatzill@c-76-23-28-244.hsd1.ut.comcast.net) |
| 00:35.13 | javawizard2539_ | from http://forums.whirlpool.net.au/forum-replies.cfm?t=462829 |
| 00:35.18 | ix33 | so does anyone here push ABE over open source for production * installs? |
| 00:35.19 | javawizard2539_ | i have now connected an external Netcomm Roadster v.92 voice modem to my pc and started asteriskwin32 and it picked up the modem automatically. |
| 00:35.21 | javawizard2539_ | Called the landline number and the asterisk answered. great |
| 00:35.22 | javawizard2539_ | I then did the echo test , and i cannot believe my results. almost instantly. |
| 00:35.48 | javawizard2539_ | which seems to indicate that what i was thinking is possible |
| 00:36.35 | [TK]D-Fender | javawizard2539Yeah, ok, fine, sure... good luck with that though. |
| 00:37.27 | ix33 | it has always been possible. there are reasons it's not widely done, i think. |
| 00:38.07 | javawizard2539_ | i would need to find a full duplex modem though... |
| 00:38.18 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:38.19 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:38.40 | ix33 | http://www.google.com/search?hl=en&q=x100+winmodem&btnG=Google+Search |
| 00:40.50 | ix33 | okaaaaay, i have gone through a certain amount of trouble to go alllll the way back to use MG2 for EC. |
| 00:42.04 | [TK]D-Fender | ix33, Did MG2 not do the job? |
| 00:42.46 | ix33 | all quality issues considered, MG2 sounds best right now. |
| 00:43.01 | [TK]D-Fender | ix33, try out OSLEC... |
| 00:43.42 | ix33 | [TK]D-Fender: thanks for the tip. i certainly will do that tonight. |
| 00:44.00 | ix33 | headed to gym, bbl |
| 00:44.35 | *** join/#asterisk bkruse (n=bkruse@216-83-246-98.static.networktel.net) |
| 00:44.35 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 00:47.19 | lmadsen | bkruse: oh snap! |
| 00:47.33 | bkruse | lmadsen: :X what's up? |
| 00:47.38 | lmadsen | nada much |
| 00:47.39 | bkruse | When you coming to visit me? |
| 00:47.57 | lmadsen | probably not for a while |
| 00:48.01 | lmadsen | AstriCon perhaps? |
| 00:48.02 | bkruse | Not a lot going on is good usually |
| 00:48.09 | bkruse | Nice. The power phailed today :/ |
| 00:48.10 | lmadsen | can't really go to the US very easily |
| 00:48.14 | bkruse | if you didn't already know |
| 00:48.15 | lmadsen | ya, I heard! |
| 00:48.24 | Qwell | bkruse: failed? |
| 00:48.37 | lmadsen | brownouts |
| 00:48.41 | lmadsen | caused a small fire |
| 00:48.41 | bkruse | Qwell: phail |
| 00:48.44 | Qwell | at Digium? |
| 00:48.46 | lmadsen | yes |
| 00:48.48 | bkruse | Yes |
| 00:48.51 | bkruse | @3 |
| 00:48.54 | lmadsen | try and keep up Qwell |
| 00:48.55 | Qwell | nice, what happened? |
| 00:48.58 | lmadsen | hell, I'm not even in the same country! |
| 00:49.06 | bkruse | This is weird, check this out, the battery backups went out first |
| 00:49.07 | lmadsen | brownout caused a water pump to phail |
| 00:49.09 | bkruse | then the normal pc's |
| 00:49.19 | lmadsen | and smoke |
| 00:49.22 | bkruse | because the battery backups switched at brownout, then died |
| 00:49.30 | Qwell | how long? |
| 00:49.32 | bkruse | ya Qwell, smoke and lots of stuff :) |
| 00:49.35 | Qwell | nice |
| 00:49.38 | bkruse | Qwell: I believe it is still out, everyone left |
| 00:49.41 | Qwell | wtf |
| 00:49.55 | *** join/#asterisk Swabby (n=dp@74-137-61-44.dhcp.insightbb.com) |
| 00:50.05 | Swabby | Is there any way to push configuration automatically to an Grandstream? |
| 00:50.12 | Swabby | like can you save the info to a file and upload it? |
| 00:50.40 | lmadsen | I need to upgrade my polycom 501 again and see if the latest version with default configs will cause it to actually stay registered |
| 00:50.44 | bkruse | ya, Qwell, idk, I got like 49 emails from jeffery.... |
| 00:50.45 | bkruse | jk |
| 00:50.50 | lmadsen | seems to only be that one phone too (out of the 4 phones on my desk) |
| 00:51.33 | lmadsen | seems to have done it ever since I updated to the 3.x bootrom... |
| 00:51.42 | bkruse | lmadsen: What are they publically at now? or are you getting the beta stuff |
| 00:52.08 | lmadsen | oh, actually I guess the bootROM is 4.1.0.0219 |
| 00:52.27 | lmadsen | I need a beer before I really start on this :) |
| 00:53.24 | lmadsen | I think 3.0.5 |
| 00:53.28 | lmadsen | I'm not using betas |
| 00:54.18 | bkruse | lmadsen: Good. heh, let me know how that goes :) |
| 00:54.29 | lmadsen | oh the beer goes down smooth |
| 00:54.44 | lmadsen | I just updated my home PBX system to 1.6.0 branch |
| 00:54.49 | bkruse | :X |
| 00:54.50 | lmadsen | I don't know why people complain it is so hard to upgrade |
| 00:54.55 | bkruse | 1.6.0 branch is where the cool kids are at |
| 00:55.01 | *** part/#asterisk korihor (n=korihor@190.199.171.145) |
| 00:55.04 | lmadsen | k00l k1dz |
| 00:55.05 | bkruse | trunk is for really cool kids |
| 00:55.30 | lmadsen | heh... I'm not that brave :) |
| 00:56.20 | lmadsen | I got a book to update :) |
| 00:58.09 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 00:59.36 | tzanger | lmadsen's a chicken shit |
| 00:59.36 | [TK]D-Fender | lmadsen, 3.0.5? I have 3.0.2, you sure its gone that high? |
| 00:59.54 | lmadsen | [TK]D-Fender: no I'm not that sure -- I just looked -- 3.0.2 is the newest |
| 01:00.14 | lmadsen | if anyone wants to suggest changes/additions for TFoT 3rd Edition, feel free |
| 01:00.54 | lmadsen | it will be based on 1.6.x |
| 01:01.00 | tzanger | lmadsen: what version is it targeted at |
| 01:01.01 | tzanger | oh |
| 01:01.02 | tzanger | you just said |
| 01:01.05 | lmadsen | :D |
| 01:01.09 | tzanger | I guess I'd have to run 1.6 to comment :-) |
| 01:01.20 | lmadsen | you and me both :) |
| 01:01.36 | tzanger | hahaha |
| 01:01.38 | bkruse | does |
| 01:01.42 | bkruse | I run trunk at my school |
| 01:01.45 | bkruse | heh :] |
| 01:03.53 | lmadsen | couple of ideas I have: expand the hardware installation instructions to include digital hardware, expand and update the GUI chapters, redo the DUNDi section, add some information about faxing, include simple dialplan examples for every dialplan app. and func. in the appendices, and show some sort of 1.2->1.6 CLI matrix (I'm skipping 1.4 because its a bit of a mess...) |
| 01:04.04 | ix33 | you know the support guy i was talking to was saying something about a water pump and 1/2 the servers being down |
| 01:04.19 | ManxPower | lmadsen: I'm also skipping 1.4 |
| 01:04.24 | lmadsen | we should probably write up some docs on how to upgrade your 1.2 system to 1.6 |
| 01:04.47 | bkruse | lmadsen: I am going to start comitting to the gui in the last couple of seconds.... |
| 01:04.56 | bkruse | in the next couple of days is what I meant, wow. |
| 01:05.03 | [TK]D-Fender | lmadsen, PROPER IVR samples with 1.2+ versions. |
| 01:05.09 | ManxPower | hands bkruse another drink |
| 01:05.31 | [TK]D-Fender | lmadsen, samples of zaptel for analog, PRI, CAS, etc. |
| 01:05.34 | lmadsen | [TK]D-Fender: yes, that was another thing I heard about -- not enough examples of IVRs |
| 01:05.42 | lmadsen | steals the drink |
| 01:06.24 | bkruse | no1 |
| 01:06.38 | bkruse | ManxPower: Just some bug fixes for FF3, I am doing no more development because the current gui bl0ws |
| 01:06.54 | bkruse | it'll get lots better, give it some time :] |
| 01:07.01 | lmadsen | bkruse: thank god |
| 01:07.14 | lmadsen | there are some things I like about it, and some I don't |
| 01:07.16 | bkruse | lmadsen: I promise :) |
| 01:07.31 | lmadsen | bkruse: I've used the GUI a lot lately for a customer, so please feel free to bounce ideas off me |
| 01:08.05 | bkruse | lmadsen: I will, actually, I have some things for you to "test" :) |
| 01:08.25 | bkruse | lmadsen: It just rocks, but we do need comments about layout, and you, would be a great candidate |
| 01:08.25 | lmadsen | heh, sweet |
| 01:08.30 | lmadsen | totally |
| 01:08.35 | lmadsen | you need a 3rd eye to look at it |
| 01:08.46 | lmadsen | after you look at it all day, you'll have no idea whether you're doing something "right" or not |
| 01:08.53 | [TK]D-Fender | lmadsen, go for it tri-clops! |
| 01:09.27 | bkruse | lmadsen: Exactly |
| 01:09.32 | bkruse | [TK]D-Fender: How you doing? |
| 01:09.38 | ManxPower | Which GUI? |
| 01:09.40 | [TK]D-Fender | bkruse, getting by... TGIF./.. |
| 01:09.42 | lmadsen | AstGUI |
| 01:09.49 | lmadsen | TMFGIMFF |
| 01:10.01 | bkruse | [TK]D-Fender: I hear you :) |
| 01:10.04 | ManxPower | Is that the one from Digium? |
| 01:10.09 | bkruse | ManxPower: asterisk gui, it's coming to a city near you |
| 01:10.19 | bkruse | ManxPower: Ya....the current open source one has not been developed on in....awhile though |
| 01:10.22 | bkruse | give it some time |
| 01:10.28 | ManxPower | bkruse: At least it's the best of the horrid class of software. |
| 01:10.57 | bkruse | ManxPower: ouch |
| 01:11.09 | ManxPower | Could you at least put in the docs "DO NOT GO TO #ASTERISK FOR SUPPORT!" all over the docs. |
| 01:11.23 | lmadsen | thanks for the reminder ManxPower |
| 01:11.29 | lmadsen | that reminded me to fix my wiki link :) |
| 01:11.33 | bkruse | ManxPower: hehe, yes |
| 01:11.39 | ManxPower | This is a technical channel, and using a GUI is not...technical. |
| 01:12.04 | lmadsen | I totally agree |
| 01:12.08 | ManxPower | we get GUI users here all the time asking for help. Usually we just smack them upside the head and insult them, but it would be better if they were never here in the first place. |
| 01:12.30 | deeperror | haha |
| 01:12.33 | bkruse | ManxPower: redirect them to #asterisk-gui, geez. |
| 01:12.41 | *** part/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:12.49 | TrentCreek | [TK]D-Fender: My ATA still does not get voice...after forwarding all traffic to the " Box and changing IP addresses for everything..the sound quality has gotten better, but no voice going to in calls. |
| 01:12.51 | bkruse | send me a pm and i'll move them, them or 10 other ops |
| 01:12.51 | ManxPower | Their excuse is almost always "But #theasteriskguiiuse is dead!" and my response is usually "If support is so bad why are you using it?" |
| 01:13.03 | bkruse | sure, whatever. |
| 01:13.07 | lmadsen | grrrrrrrr... my wiki used to work... now I can't remember what I did to make it work |
| 01:13.22 | [TK]D-Fender | TrentCreek, you'll need to describe the complete setup as it now is. |
| 01:13.39 | ManxPower | bkruse: A minority say "Oh! I'm sorry" and leave. The others whine about no support on the channel that is dedicated to their GUI or no response on their "forums" and then try to get help anyway. |
| 01:14.07 | bkruse | ManxPower: I understand, I have kind of given up hope for them, I will do a better job in the future. |
| 01:14.09 | ManxPower | BTW, most of these people are using Trixbox/AMP/FreePBX |
| 01:14.20 | ManxPower | bkruse: don't have a GUI if you can't support it. |
| 01:14.50 | TrentCreek | [TK]D-Fender: If you remember, it has not changed since the last time..just now all the IP addresses are setup on the on the same LAN |
| 01:14.50 | lmadsen | aha! I changed the link |
| 01:15.27 | bkruse | ManxPower: I think that decision is up to me and digium, but thanks |
| 01:15.29 | ManxPower | I can't tell you how many times I've helped someone for 20 mins, then ask for a pastebin of the CLI output of a failed call and then realize that there are 200 lines of complex dialplan and AGI scripts to call one number and it's AMP. |
| 01:15.57 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 01:15.59 | ManxPower | bkruse: ALL decisions regarding Digium products is up to Digium. |
| 01:16.20 | bkruse | ManxPower: I would imagine many corporations do that, especially since people purchase BE because of the gui. |
| 01:16.21 | ManxPower | But you also rely heavily on this channel for support and if the people that know stuff start leaving because of channel problems..... |
| 01:16.28 | [TK]D-Fender | TrentCreek, pastebin a call with SIP debug enabled. |
| 01:16.36 | TrentCreek | [TK]D-Fender: everything now is based on 192.168.2.xx |
| 01:16.47 | TrentCreek | ok |
| 01:17.18 | bkruse | ManxPower: You told me this, and we had this argument over 6 months ago, and you're still here. People that buy BE have support from Digium, trust me, I fix a lot of problems. |
| 01:17.23 | ManxPower | [TK]D-Fender doesn't count. I'm convinced he's insane because he never seems to need a break from the channel. 8-) |
| 01:17.51 | ManxPower | bkruse: and people here send BE people right to Digium. BE is a non-issue. |
| 01:17.56 | bkruse | ManxPower: As far as people that just check it out because we want it to be open source, I always fix bugs on the bugtracker, answer questions in #asterisk-gui, and personal emails and the forums, I do not owe an explanation to you at all, but I figured you should know. |
| 01:18.11 | ManxPower | And Asterisk-GUI is a small fraction of the GUI users we get here. |
| 01:18.26 | bkruse | ManxPower: Well in that case, I am not sure why you are jumping on my nuts about it |
| 01:18.28 | ManxPower | bkruse: I appreciate the information |
| 01:18.39 | [TK]D-Fender | yeah... dwarfed by FreePBX toting schmucks |
| 01:18.53 | [TK]D-Fender | bkruse, You working on the GUI code? |
| 01:19.00 | bkruse | ManxPower: I am just saying, I honestly do as much as I can man :/ things just overflow, and I thank you for helping me handle that |
| 01:19.30 | bkruse | [TK]D-Fender: Not really, just trying to keep open source bug fixes and stuff in my spare time |
| 01:19.36 | bkruse | my Digium job is much different |
| 01:19.38 | ManxPower | bkruse: If Asterisk-GUI gets as popular we'll have the same problem with that. I wanted to bring the issue to your attention *before* it becomes an issue. |
| 01:19.38 | lmadsen | I have no idea how [TK]D-Fender or ManxPower find as much time as they do to help on the IRC channel |
| 01:20.01 | [TK]D-Fender | lmadsen, pirated copy of chan_fluxcapacitor.so |
| 01:20.15 | ManxPower | lmadsen: my life is odd in most ways 8-) |
| 01:20.26 | bkruse | lmadsen: Exactly, I really highly applaud them, and thank them. |
| 01:20.56 | lmadsen | kicks [TK]D-Fender and ManxPower (slackers!) |
| 01:20.59 | bkruse | ManxPower: True, and help me help ok? I am often busy a lot in the day, but I will try my best, just ping me and I will eventually get into the habbit of watching ok? |
| 01:21.47 | ManxPower | I shall /msg you if an Asterisk-GUI user does not go to the correct channel when asked. |
| 01:22.23 | bkruse | ManxPower: Absolutely, I am really encourage that, even in the #asterisk-gui channel, I say to "say bkruse" so I can help |
| 01:22.40 | bkruse | ManxPower: When I do get time to watch the channel, you and [TK]D-Fender often pass me to the answer, I love it :) |
| 01:24.36 | *** join/#asterisk jicksta (n=jicksta@75-101-5-48.dsl.static.sonic.net) |
| 01:28.25 | ManxPower | Unfortunatly my knowledge is getting increasingly out of date. |
| 01:30.07 | lmadsen | you need to do more consulting, and less maintenance |
| 01:30.52 | [TK]D-Fender | ManxPower, Or just stop fearing your users. |
| 01:32.06 | lmadsen | nah... I increasingly get sucked into maintenance tasks once a project is nearing completion because I'm the only one who can do it -- I keep making sure I actively get out of that task |
| 01:33.32 | *** join/#asterisk blind32 (n=blind32@233.16.119.70.cfl.res.rr.com) |
| 01:34.25 | ManxPower | My users are a mob of bloodthirsty real estate agents |
| 01:35.20 | *** join/#asterisk RoyK (n=roy@ip-58-10-149-91.dialup.ice.no) |
| 01:35.25 | blind32 | hey all |
| 01:37.20 | lmadsen | hey |
| 01:38.42 | blind32 | is it popular to use fedora core 9 with asterisk on a pentium 4, or should i go with a lower fedora release? |
| 01:41.09 | lmadsen | blind32: it's just Fedora 9 now (drop the Core), and while it is not popular because it is new, you can use it. However, F6 and F7 are still the recommended versions for Asterisk Business Edition. Any distribution you are *personally* familiar and comfortable with is adequate for Asterisk |
| 01:41.26 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 01:41.35 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-116-111-122.rgv.res.rr.com) |
| 01:42.48 | hardwire | Strom_C: So, the NANPA rate center codes are unique per state. |
| 01:43.44 | *** join/#asterisk blinky42 (n=steveb@c-71-230-47-244.hsd1.pa.comcast.net) |
| 01:46.21 | bkruse | Corydon76-dig: you there? |
| 01:47.24 | jblack | I think I just found my weird unreachable/reachable iax problem. |
| 01:48.39 | jblack | 10.21.0.22/24 and 10.20.1.22/16 would work fine for hours at a time, then suddenly go crazy with packet loss. |
| 01:49.06 | lmadsen | oh? |
| 01:49.11 | lmadsen | what was the solution? |
| 01:49.36 | jblack | Fix 10.21.0.22/24 so it doesn't wrongly think it's in a class C, and really is in the class B. |
| 01:49.44 | jblack | Oh, pardon, the latter one is 10.21.1.22/16 |
| 01:49.51 | jblack | That may make it more apparent. =) |
| 01:50.09 | ix33 | lol |
| 01:50.17 | jblack | So, machineB was talking directly to machineA, because it thinks its on the same network. machine A was using the gateway to talk to machine B. |
| 01:52.35 | hsv-al | lmad |
| 01:52.46 | hsv-al | some sales lady in NC tried to hook me up w/ a weekend pickup |
| 01:52.59 | ix33 | wow. good sales technique. |
| 01:53.00 | hsv-al | but the 'power iss, made things hard, monday it is ;/ |
| 01:54.13 | hsv-al | ix33, what can i do this weekend if i get the card |
| 01:54.17 | hsv-al | its at my desk at work, but with an fxs chip |
| 01:54.21 | hsv-al | i just wont have an fxo till mon |
| 01:54.49 | ix33 | you can make any old analog phone get dialtone from * |
| 01:54.51 | hsv-al | slightly annoying, was hoping to have both this sat |
| 01:55.51 | hsv-al | ix33, i put the fxs in slot 2 |
| 01:56.00 | hsv-al | so the analong phone goes to slot 2, but if fxo goes in 4, then wall line in 4 |
| 01:56.08 | hsv-al | or if its in slot 1, line goes in jack 1 |
| 01:56.13 | hsv-al | so they're interchangeable like that? |
| 01:56.38 | hsv-al | as long as the #'s are matched? |
| 01:56.55 | ix33 | yes. put it in slot 4 and the other in slot 1 |
| 01:57.05 | hsv-al | well the fxs is in slot 2 now |
| 01:57.07 | ix33 | don't confuse the two or risk frying the thing. |
| 01:57.12 | hsv-al | doesnt matter hwich the fxo chip goes in |
| 01:57.15 | hsv-al | just match the #'s up right? |
| 01:57.30 | ix33 | that's true. but make sure you don't confuse fxs/fxo |
| 01:57.36 | hsv-al | that i know |
| 01:57.39 | hsv-al | but i wanted to make sure |
| 01:57.43 | hsv-al | that you didnt have to go in static sequence |
| 01:57.45 | hsv-al | 1,2,3,4 etc |
| 01:57.47 | ix33 | fxs doesn't like getting +48V |
| 01:57.55 | ix33 | hsv-al: you don't have to go in sequence |
| 01:58.38 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:01.48 | ix33 | why isn't my ztmonitor working? |
| 02:02.27 | tzanger | here's a good one... would anyone know where I'd get a recent copy of the old tor2 driver? |
| 02:03.14 | ix33 | er... doesn't that still come with zaptel? |
| 02:03.42 | tzanger | bah, you're right |
| 02:03.48 | tzanger | no wonder I can't find it anywhere :-) |
| 02:04.38 | *** join/#asterisk erojasv (n=erojasv@190.40.91.32) |
| 02:04.42 | *** join/#asterisk rcy (n=rcy@S010600003981572c.vc.shawcable.net) |
| 02:05.01 | *** join/#asterisk a1fa (n=fiddy@unaffiliated/a1fa) |
| 02:05.05 | a1fa | [TK]D-Fender |
| 02:05.07 | a1fa | sup |
| 02:09.58 | ix33 | ok it's just ztmonitor from 1.4-branch |
| 02:10.00 | ix33 | for some reason |
| 02:13.00 | lmadsen | :) |
| 02:13.07 | lmadsen | it won't come in DAHDI though |
| 02:13.12 | lmadsen | tzanger: ^^^ |
| 02:13.43 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 02:13.54 | tzanger | ? |
| 02:14.07 | lmadsen | tor2 |
| 02:14.09 | tzanger | ztmonitor is for zaptel. I wonder if they'll have dahmon or something for dahdi |
| 02:14.13 | tzanger | lmadsen: ahh |
| 02:14.25 | tzanger | I have to say dahdi is one of the dumbest-sounding names I could think to saddle a project with |
| 02:14.34 | lmadsen | you're not the only one |
| 02:14.39 | lmadsen | I say Doddy |
| 02:14.39 | tzanger | it's right up there with fucking joomla, zarafa, zimbra and that crap |
| 02:14.48 | mwalling | drupal |
| 02:14.54 | tzanger | drupal yes |
| 02:15.03 | lmadsen | microsoft |
| 02:15.08 | mwalling | ubuntu |
| 02:15.11 | mwalling | duckks |
| 02:15.21 | lmadsen | obligatory M$ bash :D |
| 02:15.22 | tzanger | open source coders should holler up the stairs and ask their moms what they think of a new name before officially naming a project |
| 02:15.28 | lmadsen | mwalling: I agree with you |
| 02:15.29 | tzanger | yes ubuntu sucks too |
| 02:15.44 | ix33 | ubuntu has a real meaning behind it though |
| 02:15.52 | lmadsen | doesn't make it sound less dumb :) |
| 02:15.57 | mwalling | ix33: so does asshole |
| 02:16.03 | mwalling | it means i had too much tbell |
| 02:16.17 | ix33 | asshole would be an awesome name for a linux distribution. |
| 02:16.21 | lmadsen | lol |
| 02:16.33 | ix33 | 'yeah but does it run asshole?' |
| 02:16.34 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 02:17.30 | hsv-al | _mre|29a, they were making some odd filtering system for an e-commerce site that uses data about aggregate user behavior to make recommendations tailored to specific user interests |
| 02:19.04 | *** part/#asterisk spiny (n=spiny@69-196-137-127.dsl.teksavvy.com) |
| 02:19.37 | *** join/#asterisk Twister (n=jlane@66.172.207.172) |
| 02:22.40 | Twister | hey all, could you take a look at this link http://pastebin.com/d63e56b8f and tell me if you can see why my trunk wont register (from the looks of things, it's not even trying) |
| 02:24.10 | lmadsen | Twister: iax2 show registry |
| 02:24.13 | lmadsen | does it show there? |
| 02:24.18 | lmadsen | did you reload your iax2 driver? |
| 02:24.46 | Twister | its sip, |
| 02:25.06 | Twister | let me turn my account to iax and try that |
| 02:25.43 | lmadsen | no no... just check the sip |
| 02:25.53 | lmadsen | I saw teliax, so just thought iax for some reason |
| 02:25.58 | lmadsen | sip show registry |
| 02:26.06 | lmadsen | and sip set debug peer teliax |
| 02:26.20 | lmadsen | you should see the registrations go out, and a response |
| 02:27.36 | Twister | Host Username Refresh State Reg.Time |
| 02:27.41 | Twister | is sip show registry |
| 02:27.43 | Twister | nothing listed |
| 02:28.02 | Twister | just enabled debug for teliax |
| 02:28.13 | Twister | seeing nothing |
| 02:28.25 | Twister | reloaded and still seeing nothing |
| 02:28.33 | TJNII | twisted: module reload chan_sip |
| 02:28.50 | Twister | reloaded |
| 02:29.02 | TJNII | Does sip show registry show it now? |
| 02:29.10 | Twister | no |
| 02:29.19 | TJNII | How about sip show peers |
| 02:29.43 | Twister | teliax/twister 74.201.8.23 N 5060 Unmonitored |
| 02:30.23 | TJNII | No messages about registration failing with verbose >= 5? |
| 02:31.24 | Twister | nope |
| 02:31.41 | TJNII | Pastebin your sip.conf |
| 02:31.44 | TJNII | ~pb |
| 02:31.46 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:31.54 | Twister | sure just a sec |
| 02:33.30 | lmadsen | ya, with 'sip show registry' you should see output, otherwise asterisk obviously won't try to register |
| 02:36.42 | Twister | http://pastebin.com/d71cd9171 |
| 02:36.56 | Twister | most of it is the example that came with, what i added is at the bottom |
| 02:37.36 | *** join/#asterisk bkruse (n=bkruse@74.95.48.254) |
| 02:37.36 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 02:39.07 | TJNII | Try moving the register line up to where it says "OUTBOUND SIP REGISTRATIONS" |
| 02:40.18 | TJNII | Also why do you have teliax and teliax-in? |
| 02:41.30 | TJNII | And it is good form to remove superflorous comments and passwords before pastebinning. |
| 02:41.43 | Twister | that did it! |
| 02:41.47 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 02:42.08 | TJNII | Twister: Now, do you know why? |
| 02:42.15 | TJNII | I |
| 02:42.38 | Twister | not a clue but im gonna make sure i understand before i go any further |
| 02:43.00 | TJNII | the register wasn't in [general] but in [6001] |
| 02:43.49 | Twister | OHHHHHHHHHHHHHHHHHHH |
| 02:45.16 | Twister | that makes sense..kind of..i had created a new sip.conf with just the extension and the trunk in it, put the register line at the top and it still didnt register it |
| 02:46.01 | TJNII | Did you include the [general] heading? |
| 02:46.04 | Twister | no |
| 02:46.23 | TJNII | Yea, that's kind of important..... |
| 02:46.49 | Twister | :) |
| 02:47.02 | Twister | musta missed that part :( |
| 02:47.45 | *** join/#asterisk macros73_ (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 02:48.28 | Twister | thank you all greatly for your help |
| 02:48.58 | TJNII | Once again, why do you have teliax and teliax-in? |
| 02:50.49 | Twister | because thats what teliax has on their page? |
| 02:51.59 | Twister | i thought it seemed a little odd too but i dont know enough to know it dont belong there |
| 02:52.01 | Twister | so i put it there |
| 02:53.51 | Twister | i removed it |
| 03:02.17 | Twister | does anyone know if teliax has a delay in changing passwords, i just changed mine and it lists on the page but my new pass wont work and niether will my new one |
| 03:03.14 | ix33 | i have ztmonitor recordings of both the incoming and outgoing zap channels with: MG2 EC, VPM EC, and HPEC EC; pre- and post-EC on each. |
| 03:04.06 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.24 | ix33 | I will provide this to Digium I guess, but I am going to need a solution on this silly VPM thing. |
| 03:04.39 | Twister | nm there it went |
| 03:05.03 | ix33 | can anyone suggest a t1 line card with built-in EC? |
| 03:05.11 | ix33 | i'm leaving MG2 on for now. |
| 03:07.08 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 03:09.34 | ix33 | what is in slot 2? |
| 03:09.48 | ix33 | bleah wrong window |
| 03:24.03 | hsv-al | 02:01.0 ethernet controller: Digium, Inc. unknown device 8005 (Rev 11) |
| 03:26.20 | hsv-al | [ 1111.049502] zaptel: Unknown symbol crc_ccitt_table |
| 03:26.30 | hsv-al | oop |
| 03:26.33 | hsv-al | s |
| 03:27.45 | *** join/#asterisk micander (n=Michael_@ip72-199-126-179.sd.sd.cox.net) |
| 03:43.29 | *** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk) |
| 03:44.26 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:44.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:54.52 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-116-111-122.rgv.res.rr.com) |
| 03:59.31 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:10.25 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 04:16.29 | *** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk) |
| 04:18.16 | *** join/#asterisk ddeano (n=deano@c-68-61-35-231.hsd1.mi.comcast.net) |
| 04:18.33 | ddeano | hellp |
| 04:20.07 | ddeano | /help./ |
| 04:20.10 | ddeano | /help/ |
| 04:20.13 | ddeano | fuck |
| 04:21.17 | truent | i'd help but im probably newer than you |
| 04:21.18 | truent | heh |
| 04:23.11 | ddeano | how do i switch channels |
| 04:23.45 | ddeano | i doubt it, i used a gui irc client years ago but it did the basics for you |
| 04:26.47 | Corydon76-dig | pokes bkruse |
| 04:34.51 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 04:36.41 | yxa | sorry this might be alittle off-topic: anyone knows of a vendor who keeps good stock on new Linksys SPA962/942 phones? quantity needed: 50. please pm me |
| 04:41.10 | coppice | It looks like chan_sip still cannot cope with things like |
| 04:41.11 | coppice | Content-Type: application/sdp; charset=utf-8 |
| 05:05.03 | *** join/#asterisk Blacraft (n=bla@203.26.40.67) |
| 05:05.18 | Blacraft | hello i;m just after a little help getting a TDM400P FXS card to work, currently i have no dial tone |
| 05:10.43 | ix33 | what does ztcfg -vv say? |
| 05:11.52 | javawizard2539_ | So I got *win32 installed and running with a voice modem hooked up to the computer, and now I'm getting an error in chan_fx.so |
| 05:12.04 | javawizard2539_ | Channel 1: Unable to open board 1 port 0 on module fxos |
| 05:12.08 | javawizard2539_ | Any ideas on what's going on? |
| 05:12.13 | ix33 | hoo boy |
| 05:12.49 | ix33 | javawizard2539_ please believe me when i say that no one in this channel is going to be able to help with that particular setup. probably much less be willing |
| 05:13.47 | javawizard2539_ | hmm |
| 05:14.00 | javawizard2539_ | There's got to be some way to get it working, others have |
| 05:14.03 | drmessano | wow |
| 05:14.16 | drmessano | Yeah, AsteriskWin32 is really unsupported here |
| 05:14.20 | ix33 | sure. but that's a very fringe kind of thing to be doing. |
| 05:14.31 | javawizard2539_ | Is there an irc channel for *win32? |
| 05:14.35 | drmessano | Which voice modem? |
| 05:14.41 | drmessano | Just any old modem? |
| 05:14.46 | javawizard2539_ | creative di5133 I think |
| 05:14.48 | drmessano | Doesnt work that way |
| 05:14.53 | drmessano | Any old modem doesnt work |
| 05:14.59 | javawizard2539_ | It should |
| 05:15.02 | ix33 | javawizard2539_: see? |
| 05:15.06 | drmessano | No, it should not |
| 05:15.07 | javawizard2539_ | at least for menus that only require half-duplex |
| 05:15.19 | drmessano | No, it should not |
| 05:15.21 | florz | coppice: I think that particular example not being supported is perfectly ok, as I don't see that a charset parameter is allowed for application/sdp |
| 05:16.29 | drmessano | AsteriskWin32 works with a very limited set of hardware to begin with, and Asterisk run on Linux only supports one "modem" which is the X100P card that started it's life as a modem and had drivers written for asterisk to make it work |
| 05:16.35 | drmessano | "Any old modem" is not gonna work |
| 05:16.44 | javawizard2539_ | I was reading at http://forums.whirlpool.net.au/forum-replies-archive.cfm/477945.html which seems to indicate it's possible but only with half-duplex |
| 05:16.48 | javawizard2539_ | which is ok for me |
| 05:17.00 | drmessano | Well, I am telling you that is not correct |
| 05:17.30 | javawizard2539_ | Is there some place then where I can get an FXO that hooks up via ethernet and allows asterisk to connect via SIP or IAX? |
| 05:18.11 | javawizard2539_ | <PROTECTED> |
| 05:18.13 | javawizard2539_ | huh? |
| 05:18.16 | drmessano | SPA-3102 |
| 05:18.25 | drmessano | Linksys |
| 05:18.32 | drmessano | That's your best option |
| 05:18.47 | drmessano | and that forum post you pasted |
| 05:19.01 | drmessano | Which I will only comment slightly on.... |
| 05:20.00 | drmessano | Does state that cards similar to the Digium X100P will work.. but finding a card like that sold as a MODEM are rare if even possible anymore.. you can get an X100P clone all over the web for $30, but they suck |
| 05:20.24 | drmessano | In THEORY, if someone writes the drivers and wastes part of their life doing it, a lot more modem could work with Asterisk |
| 05:21.42 | Qwell | there was a chan_modem at one point |
| 05:21.46 | Qwell | it sucked, and was removed. |
| 05:21.56 | Qwell | had support for like aopen and a few others. but it sucked |
| 05:22.16 | drmessano | But clone tdm400 cards are becoming cheaper and cheaper, and will eventually replace X100Ps as the cheapo knock-offs.. which, while it's not a great thing, is better than X100Ps still existing |
| 05:22.58 | javawizard2539_ | So X100Ps aren't a good thing? |
| 05:23.02 | drmessano | No |
| 05:23.09 | drmessano | They generally suck a lot of ass |
| 05:24.22 | drmessano | Get an SPA-3102 |
| 05:24.36 | drmessano | Then you don't have to worry about the hardware layer with Windows |
| 05:25.46 | *** join/#asterisk e2e5 (n=chatzill@gw-Paramon-chel.suttk.ru) |
| 05:28.36 | e2e5 | hello, is h323 asterisk thing developer here ? |
| 05:31.27 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 05:32.07 | *** join/#asterisk erojasv (n=erojasv@190.40.91.32) |
| 05:35.13 | e2e5 | What does NoCdr do ? |
| 05:35.34 | e2e5 | I route call through extensions, Dial ... Goto (on failover) ... Dial ... Goto ... |
| 05:35.55 | e2e5 | CDR is generated for each event |
| 05:36.26 | e2e5 | if I apply NoCdr before the entire chain, will it stop CDR totally ? |
| 05:39.57 | Blacraft | hello i;m just after a little help getting a TDM400P FXS card to work, currently i have no dial tone |
| 05:41.32 | Blacraft | [trixbox1.up2date.net.au ~]# ztcfg -vv |
| 05:41.32 | Blacraft | Zaptel Version: 1.4.10 |
| 05:41.32 | Blacraft | Echo Canceller: OSLEC |
| 05:41.32 | Blacraft | Configuration |
| 05:41.32 | Blacraft | ====================== |
| 05:41.32 | Blacraft | Channel map: |
| 05:41.34 | Blacraft | Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
| 05:41.36 | Blacraft | Channel 02: FXO Kewlstart (Default) (Slaves: 02) |
| 05:41.38 | Blacraft | Channel 03: FXS Kewlstart (Default) (Slaves: 03) |
| 05:41.40 | Blacraft | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
| 05:41.42 | Blacraft | 4 channels to configure. |
| 05:42.48 | ix33 | ok. pastebin your zapata.conf file |
| 05:43.28 | ix33 | actually your /etc/zaptel.conf and your /etc/asterisk/zapata.conf files |
| 05:43.38 | Blacraft | ok |
| 05:44.10 | Blacraft | # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit |
| 05:44.10 | Blacraft | # Zaptel Configuration File |
| 05:44.10 | Blacraft | # |
| 05:44.10 | Blacraft | # This file is parsed by the Zaptel Configurator, ztcfg |
| 05:44.10 | Blacraft | # |
| 05:44.11 | Blacraft | # It must be in the module loading order |
| 05:44.13 | Blacraft | # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) |
| 05:44.15 | Blacraft | fxoks=1 |
| 05:44.17 | Blacraft | fxoks=2 |
| 05:44.19 | Blacraft | fxsks=3 |
| 05:44.21 | Blacraft | fxsks=4 |
| 05:44.23 | Blacraft | # Global data |
| 05:44.39 | nick125 | sighs |
| 05:44.56 | ix33 | on that note i believe i will be heading to bed. |
| 05:45.33 | *** join/#asterisk Blacraft (n=bla@203.26.40.67) |
| 05:45.36 | Blacraft | poop |
| 05:45.42 | ix33 | ~pb |
| 05:45.43 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 05:48.01 | Blacraft | ; |
| 05:48.01 | Blacraft | ; Zapata telephony interface |
| 05:48.01 | Blacraft | ; |
| 05:48.02 | Blacraft | ; Configuration file |
| 05:48.02 | Blacraft | [trunkgroups] |
| 05:48.17 | ix33 | ok seriously now |
| 05:48.37 | *** join/#asterisk Blacraft (n=bla@203.26.40.67) |
| 05:48.46 | Blacraft | ok i know im stupid :) |
| 05:48.51 | Blacraft | http://pastebin.com/d5fbb40b9 |
| 05:49.58 | ix33 | and when you |
| 05:50.18 | Blacraft | some people have said go back to zaptel-1.2.17.1 driver |
| 05:51.26 | ix33 | when you say you have no dialtone, what actually is connected where? |
| 05:51.49 | ix33 | you should have analog phones connected to either ports 1 or two, right? |
| 05:52.27 | Blacraft | correct and i have a pstn line connected to port 3 |
| 05:52.38 | Blacraft | so phone connected to port 1 pstn to port 3 |
| 05:53.51 | ix33 | what does 'zap show status' say? |
| 05:55.50 | Blacraft | what do you mean |
| 05:56.14 | Blacraft | Channel map: |
| 05:56.15 | Blacraft | Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
| 05:56.15 | Blacraft | Channel 02: FXO Kewlstart (Default) (Slaves: 02) |
| 05:56.15 | Blacraft | Channel 03: FXS Kewlstart (Default) (Slaves: 03) |
| 05:56.15 | Blacraft | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
| 05:56.15 | Blacraft | 4 channels to configure. |
| 05:57.17 | ix33 | in asterisk? |
| 05:57.21 | ix33 | in the CLI |
| 05:58.32 | Blacraft | trixbox1*CLI> zap show status |
| 05:58.33 | Blacraft | Description Alarms IRQ bpviol CRC4 |
| 05:58.33 | Blacraft | Wildcard TDM400P REV I Board 1 OK 0 0 0 |
| 06:03.06 | ix33 | now show me 'zap show channels' |
| 06:03.57 | Blacraft | <PROTECTED> |
| 06:03.57 | Blacraft | <PROTECTED> |
| 06:03.57 | Blacraft | <PROTECTED> |
| 06:03.57 | Blacraft | <PROTECTED> |
| 06:03.57 | Blacraft | <PROTECTED> |
| 06:03.58 | Blacraft | <PROTECTED> |
| 06:05.54 | ix33 | and you're sure that when you pick up an analog phone which is connected to ports 1 or 2 it has no dialtone? |
| 06:06.36 | Blacraft | 100% |
| 06:07.24 | ix33 | ok, can you dial out through Zap/3 or /4? |
| 06:07.33 | ix33 | using a softphone or something, naturally |
| 06:09.07 | Blacraft | i haven;t setup it to dial out through the zap yet |
| 06:09.35 | Blacraft | i can receive calls through port 3 or 4 though |
| 06:09.45 | ix33 | ah ok good enough |
| 06:09.54 | ix33 | but you can't Dial(Zap/1) or /2? |
| 06:10.00 | ix33 | what happens when you do? |
| 06:10.12 | Blacraft | ahh nothing |
| 06:10.21 | ix33 | trippy. |
| 06:10.38 | Blacraft | nooo i'm wrong |
| 06:10.50 | Blacraft | i just tried to dial it and it rang |
| 06:11.00 | Blacraft | lol didn't do that before |
| 06:12.32 | Blacraft | sort of rang it's not the phones normal ring, it's more like a beep |
| 06:13.10 | ix33 | fine. what does the CLI say when you *pick up* a non-ringing phone on ports 1 or 2 |
| 06:13.43 | ix33 | come to think of it, did you connect the supplemental power cable to your tdm4xx when you installed it? |
| 06:14.08 | Blacraft | yeah it was connected |
| 06:14.15 | ix33 | are you sure it's live? |
| 06:14.50 | Blacraft | i gather it is, because the green lights are on |
| 06:16.07 | Blacraft | now there is a dial tone |
| 06:16.12 | Blacraft | i didn't do anything |
| 06:16.53 | Blacraft | i un plugged it and plugged it in again, now the dial tone has gone |
| 06:17.07 | ix33 | alrighty then. |
| 06:17.59 | Blacraft | i redialed it from my voip phone then hung up the dial tone is back again |
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| 06:18.18 | Blacraft | now it is gone again :) after leaving it for a bit |
| 06:18.48 | Nasra | elastix |
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| 06:19.24 | ix33 | well that there is flaky. |
| 06:19.32 | ix33 | good news is that you appear to be configured correctly |
| 06:19.48 | ix33 | the bad news is that this seems like it's a physical layer problem |
| 06:20.09 | ix33 | and as a computer scientist, i am forbidden by union bylaws from diagnosing physical layer problems |
| 06:20.41 | Blacraft | i don't want it to be faulty. i bought the thing over ebay |
| 06:20.46 | ix33 | plus i am exhausted and will now be headed to bed. |
| 06:21.39 | Blacraft | ok thanks for your help. |
| 06:22.14 | ix33 | no problem. i'm sorry i couldn't be more helpful. |
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| 06:24.22 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 06:30.08 | Blacraft | anyone got any idea why my TDM400P FXS doesn't have a dial tone, though if i ring it, it rings. Then after that i have a dial tone for a minute, though i can't ring out |
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| 06:49.49 | coppice | florz: its perfectly valid, and it *is* used by some applications |
| 06:51.01 | florz | coppice: valid as per what standard? |
| 06:51.37 | coppice | its normal mime stuff. if they want to stop it being used for SIP they would have to specifically disallow it |
| 06:52.01 | coppice | more importantly, some systems actually send it |
| 06:52.02 | florz | coppice: where do the MIME RFC allow that?! |
| 06:52.23 | coppice | the charset parameter is a perfectly normal mime parameter |
| 06:52.25 | florz | coppice: Well, that would be a reason for at least ignoring parameters, yeah |
| 06:52.51 | florz | coppice: Well, I say that it's not, not there - so, where in the RFC does it say that it's allowed to appear there? |
| 06:53.08 | florz | or rather, RFCs - there are quite a few to consult |
| 06:53.38 | coppice | a normal mime content type line can carry a charset field |
| 06:53.41 | florz | coppice: Neither in the SIP, nor in the HTTP, nor in the MIME, nor in the DSP RFC do I see any provision that would allow that |
| 06:54.12 | florz | coppice: Well, that's what you are claiming, yes - now, what part of what RFC is the basis for that claim? |
| 06:54.20 | florz | erm, s/DSP/SDP/ |
| 06:55.15 | ManxPower | florz: I've never actually seen coppice be wrong. |
| 06:55.35 | florz | ManxPower: Well, then maybe this is the first time? =:-) |
| 06:56.02 | coppice | have you ever seen HTTP sent *without* a charset field? |
| 06:56.18 | coppice | RFC2047 |
| 06:56.43 | florz | coppice: Erm yeah, sure - how would a charset make any sense for, say, video/mpeg ? |
| 06:57.21 | bkruse | coppice: There is an apache "security vulnerability" now, which isn't actually a vulnerability, but other than that, no, because you wouldn't know what you are getting |
| 06:57.33 | coppice | that's not a textual mime type, but I don't think the charset is actually prohibited there, and I do klnow its often sent |
| 06:58.00 | florz | coppice: but RFC2047 is for encoding of non-US-ASCII-values in headers!? |
| 06:58.38 | coppice | whatever, right now asterisk can't talk to microsoft office |
| 06:58.59 | Blacraft | anyone got any idea why my TDM400P FXS doesn't have a dial tone, though if i ring it, it rings. Then after that i have a dial tone for a minute, though i can't ring out |
| 06:59.25 | florz | coppice: plus, BTW, HTTP does actually mandate that you can even leave out the charset, in which case the client has to assume that the entity is iso-8859-1 for text subtypes |
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| 07:00.00 | coppice | sure, but then the charset is implicit, so its still relevant |
| 07:00.37 | florz | coppice: yeah, that was just to your question whether I've ever seen HTTP being sent without a charset parameter |
| 07:02.08 | florz | coppice: but, after all, application/sdp is not a text format in the traditional sense - for example, it does allow mixing character encodings, in which case you couldn't even label it with any correct "entity character set" |
| 07:02.32 | coppice | The IETF's own software tools use lines like |
| 07:02.34 | coppice | Content-Type: application/SDP; charset=ISO-10646 |
| 07:02.36 | coppice | |
| 07:02.56 | florz | coppice: plus, for the overall document, utf-8 is the only allowed encoding, so declaring it isn't of much use, I'd think |
| 07:03.28 | coppice | my above comment seems to disagree with that? are you saying the IETF tools are wrong? |
| 07:04.09 | florz | coppice: uh? ISO-10646? wasn't the charset parameter, at least for text types, for specifying a character encoding rather than a character set (despite its name)? |
| 07:04.38 | florz | coppice: at least from my understanding of the matter I'd think so - what tool is that? |
| 07:04.41 | coppice | charset should actually specify a character set |
| 07:05.48 | coppice | try googling for "application/sdp charset" and you'll get various hits for IETF stuff. the one I was looking at was actually for ISUP handling |
| 07:07.30 | florz | coppice: http://www.w3.org/Protocols/rfc2616/rfc2616-sec3.html#sec3.4 |
| 07:07.59 | florz | coppice: there, it says that the "charset" actually means what otherwise is called a "character encoding" |
| 07:08.26 | florz | coppice: and IMO that's the only option that makes sense - after all, what does it help the character set if you don't know the encoding |
| 07:09.14 | florz | coppice: and ISO-10646 is not an encoding (unless IANA maybe defines it as an identifies for one?), so it really doesn't make much sense ... |
| 07:09.24 | florz | erm, s/identifies/identifier/ |
| 07:10.01 | florz | s/does it help the/does it help to know the/ |
| 07:10.53 | florz | coppice: (all that partially assuming that the charset paramater for application/sdp does have the same semantics as the one for text types) |
| 07:13.25 | coppice | Whatever. I believe there is something in the mime spec about tolerating parameters you don't understand, and certainly variable amounts of white space should be OK. Neither is Ok with chan_sip right now. More importantly its just plain incompatible with real world apps |
| 07:14.37 | coppice | ISO-10646 is a much or as little of a character set as ISO-8859-1 |
| 07:14.52 | coppice | how you apply the term "encoding" is a bit vague |
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| 07:15.07 | florz | well, yeah, chan_sip certainly is pretty broken, plus the sip rfc does in the BNF allow parameters to the content-type header (even though it doesn't specify and), so it probably would be wise to at least tolerate them ... |
| 07:15.21 | florz | s/and/any/ |
| 07:16.44 | florz | coppice: well, no, iso-8859-1 is both a character set and an encoding - it does specify both a mapping from code points to "elements of alphabets" and from code points to bit patterns |
| 07:16.56 | florz | coppice: ISO-10646 only specifies the former |
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| 07:18.17 | florz | coppice: thus, if you have a heap of bits labeled as "ISO-10646", you can't really know how to make unicode code points from those bits |
| 07:19.28 | florz | ahh, the iana charset registry indeed does contain ISO-10646 |
| 07:19.45 | florz | as an alias for ISO-10646-Unicode-Latin1 |
| 07:20.15 | coppice | well, I'd say ISO-10646 is UCS-2, which might then get remapped into UTF-16 or UTF-. Similar ISO-8859-1 gets remapped to other things when the path is not 8 bit. its all rather vague in the specs |
| 07:20.28 | NovceGuru | anybody happen to be using ironvoice |
| 07:23.41 | coppice | those character sets are a brain dead mess, cooked up by people looking for jollies to meetings in other countries. they even shove things like "Latin" into sets that support Chinese. |
| 07:23.43 | florz | coppice: according to rfc1815 it's the same character set as in iso-8859-1 (or at least pretty close to it), but encoded in ucs-2, yep |
| 07:24.32 | florz | coppice: well, yeah, in part that's true - but in part, it's also the implementations that are a mess, beyond what's brain dead in the standards ;-) |
| 07:25.08 | coppice | Unicode was intended from the start to be a war zone, and to promote standards committee meetings in exotic locations |
| 07:26.58 | florz | coppice: now, charset=ISO-10646 on application/sdp makes even less sense if that iana registry really is what applies here - since sdp as per its own rfc is fixed to being utf8-encoded for the overall document - so, this would be specifying a different character set than the document can be in ... |
| 07:27.07 | coppice | I remember the old mailing list for unicode definition, with messages like "we couldn't really FAX you an image of these new glyphs. We need to meet at the Bali Hilton to discuss them |
| 07:27.20 | florz | *g* |
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| 07:28.01 | florz | (pardon me, a different encoding ;-) |
| 07:29.20 | coppice | well, its all totally irrelevant. unless microsoft does something totally broken that can't be lived with, you gotta live with it. personally I can't see anything wrong with what they have done, and a proper mime parser would have no problem with it. |
| 07:35.06 | florz | well, no, not the mime parser, but the interface between the mime parser and the sdp parser - the mime parser probably will have to produce characters if it is to somehow apply that character encoding, while the sdp parser can only parse octet sequences. So there is no way to feed the output from the mime parser into the sdp parser, as they are of incompatible types. |
| 07:36.36 | florz | or rather, as the sdp parser's input type is incompatible to the mime parser's output type |
| 07:38.09 | florz | well, unless you go and encode the output from the mime parser into utf-8 before feeding it to the sdp parser - but I guess that that's not what the sender of those messages really mean ... |
| 07:42.49 | coppice | why make things so complex? if the only defined form is utf-8, just be happy if the ask for UTF-8, and whine if they ask for anything else |
| 07:45.35 | florz | well, indeed, if they declare utf-8 one may be able to interpret the RFC such that one can legally ignore that ;-) |
| 07:45.52 | florz | that was more referring to that ISO-10646 case |
| 07:54.01 | coppice | you can't cure all the world's ills. try sticking to a useful subset :-) |
| 07:58.56 | WildPikachu | oh florz , great work on the hfc-s :) |
| 07:59.25 | florz | that was probably part of the useful subset ;-) |
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| 08:06.36 | drmessano | hmmm |
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| 08:22.39 | mkl1525 | Hi, I've a regex _XXX in a context and an included context that has a 050 extension is there any way that the 050 from the included context is used not the _XXX - cause the regex should only be a catchall |
| 08:36.32 | *** join/#asterisk oej (n=olle@91.143.127.196) |
| 08:39.03 | rootlogin | wow i am surprised how easy to compile svn-1.4 is .. i remember it was a lot more time consuming about 2 years ago :) good work ! |
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| 08:57.57 | und3r | hello |
| 09:00.21 | und3r | can anyone help me with "freecall.com" configuration? |
| 09:11.43 | und3r | i can register but when i try to call by Dial(SIP/${EXTEN:1}@freecall) it don't work... |
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| 09:42.05 | und3r | ciao af_ |
| 09:48.10 | *** join/#asterisk Asterisk-nob (n=Nothing@117.20.21.66) |
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| 10:07.57 | rootlogin | an an *SVN-1.4 i get the WARNING[24747]: app_meetme.c:774 build_conf: Unable to open pseudo device is that device /dev/zap/pseudo ? |
| 10:12.26 | mvanbaak | rootlogin: yup |
| 10:12.44 | mvanbaak | rootlogin: do you have zaptel installed ? |
| 10:12.49 | mvanbaak | rootlogin: and a zaptel card ? |
| 10:12.59 | mvanbaak | rootlogin: if you dont have zaptel hardware, use ztdummy |
| 10:13.08 | rootlogin | thanks .. the device exists though ... zaptel is loaded but i dant have a card |
| 10:13.18 | mvanbaak | do you have ztdummy loaded ? |
| 10:13.34 | rootlogin | mmm i think no ... sec |
| 10:13.42 | mvanbaak | lsmod | grep ztdummy |
| 10:14.54 | rootlogin | no .. i get ztdummy: Unknown symbol rtc_register .. but i switched from 1.2.xx to 1.4-svn .. maybe thats an old module ? |
| 10:15.11 | mvanbaak | erm |
| 10:15.19 | mvanbaak | ok, what steps did you do ? |
| 10:15.43 | mvanbaak | you should do it in this sequence: complile and install zaptel, load zaptel modules, compile and install asterisk |
| 10:15.54 | mvanbaak | if you do it the other way around, it's not going to work |
| 10:16.05 | rootlogin | the error was the same with 1.2 yesterday ...ahm .. ok |
| 10:16.34 | rootlogin | i did compile zaptel, libpri, asterisk .. and then loaded the modules .. also wrong ? |
| 10:16.45 | mvanbaak | nope, should work |
| 10:16.55 | mvanbaak | you can check with: lsmod | grep ztdummy |
| 10:17.25 | rootlogin | hmm .. ak but i need the ztdummy anyway in 1.4 ? so i just need to fix the missing symbols ? |
| 10:17.56 | mvanbaak | I think the symbols are fine |
| 10:18.07 | mvanbaak | I get those as well from time to time |
| 10:18.13 | mvanbaak | just check if the module is loaded |
| 10:18.39 | rootlogin | sorry its not :) |
| 10:19.36 | mvanbaak | you have the zaptel 1.4 source ? |
| 10:19.40 | rootlogin | do the modules get compiled when running make in the asterisk source-dir ? |
| 10:19.45 | rootlogin | yes |
| 10:19.45 | mvanbaak | ./configure && make menuselect |
| 10:19.52 | mvanbaak | make sure ztdummy is selected there |
| 10:20.03 | rootlogin | in the asterisk-dir |
| 10:20.06 | mvanbaak | no, you have to compile the zaptel source first |
| 10:20.12 | mvanbaak | in the zaptel-dir |
| 10:20.16 | rootlogin | ah ok |
| 10:20.26 | mvanbaak | run that |
| 10:20.40 | mvanbaak | make menuselect will give you a list of modules etc you can select to compile |
| 10:20.55 | mvanbaak | I cant remember if ztdummy is enabled by default |
| 10:21.06 | rootlogin | thanks :) having a look now |
| 10:22.21 | *** join/#asterisk RoyK (n=roy@ip-58-10-149-91.dialup.ice.no) |
| 10:22.30 | rootlogin | yep they are all enabled .. i will try to recompile and see if i load the correct object files |
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| 10:34.14 | mvanbaak | bbl |
| 10:34.51 | rootlogin | mvanbaak: cu |
| 10:36.04 | Asterisk-nob | mvanbaak: just want to confirm |
| 10:36.07 | Asterisk-nob | opss he is gone |
| 10:36.11 | Asterisk-nob | anyone else please |
| 10:36.12 | Asterisk-nob | http://phpfi.com/322632 |
| 10:36.37 | Asterisk-nob | just want to cofirm if that is correct what i did for time schedule in the pastbin |
| 10:39.19 | Strom_L | Asterisk-nob: first off, your extension shouldnt change names on the last line |
| 10:39.45 | Strom_L | second, your priorities should be 1,n,n,n,n not 1,2,n,n,n,n |
| 10:40.04 | Strom_L | third, never answer() the call unless you have an explicit reason to |
| 10:40.54 | Strom_L | fourth, you want to be open from 8:00-20:59 so that you stop taking calls at 21:00 on the dot and not at 21:01 on the dot |
| 10:41.20 | Strom_L | fifth, are open1 and open2 separate extensions within the same context? it's not clear from your pastebin |
| 10:42.07 | Strom_L | and if so, your third gotoiftime is completely unnecessary |
| 10:42.11 | Asterisk-nob | Strom_L: thanks ... let me give me clear pastebin |
| 10:42.46 | Asterisk-nob | Strom_L: see here now: http://phpfi.com/322633 |
| 10:43.24 | Strom_L | ok...still, fix all the problems I just pointed out, ESPECIALLY your priority numbering |
| 10:45.39 | Asterisk-nob | Strom_L: i didn't understand the point >>> <Strom_L> Asterisk-nob: first off, your extension shouldnt change names on the last line |
| 10:46.06 | Strom_L | Asterisk-nob: look closely at the first extension in your group |
| 10:46.28 | Strom_L | especially how the last line differs from all the other lines |
| 10:48.22 | Asterisk-nob | ooo ic |
| 10:48.28 | Asterisk-nob | lemme fix all then i'll verify |
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| 10:49.36 | Strom_L | we would appreciate it if you turned your brain to the "on" position before continuing |
| 10:49.39 | Strom_L | thank you! |
| 10:49.46 | Strom_L | 053T |
| 10:50.35 | Asterisk-nob | Ok :) |
| 10:53.37 | *** join/#asterisk Dr-Linux|work (n=Nothing@202.125.139.198) |
| 10:54.04 | Dr-Linux|work | sorry i was disconnected |
| 10:54.17 | Dr-Linux|work | is Asterisk-nob |
| 10:55.20 | Dr-Linux|work | Strom_L: maybe i miss something from you, my last msg was pastebin > http://phpfi.com/322637 |
| 10:56.19 | Dr-Linux|work | Strom_C: i'm concerned with your fourth point |
| 11:03.34 | *** join/#asterisk Dr-Linux|home (n=Nothing@117.20.21.66) |
| 11:03.49 | Dr-Linux|home | getting disconnection again and again |
| 11:04.16 | Dr-Linux|home | Strom_L: :) |
| 11:04.39 | Dr-Linux|home | did you give any guidance for my last pastbin? :) |
| 11:05.10 | mvanbaak | backe |
| 11:05.19 | *** part/#asterisk BeerSerc (n=dries@unaffiliated/beerserc) |
| 11:06.28 | mvanbaak | he didn't |
| 11:09.24 | Dr-Linux|home | hhmm... |
| 11:09.42 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 11:09.42 | Dr-Linux|home | mvanbaak: can i show you? |
| 11:09.42 | mvanbaak | show me what ? |
| 11:09.47 | *** join/#asterisk cabbiepete (n=cabbiepe@host81-158-127-243.range81-158.btcentralplus.com) |
| 11:10.17 | Dr-Linux|home | mvanbaak: http://phpfi.com/322644 |
| 11:10.37 | Dr-Linux|home | please see if anything looks wrong |
| 11:11.30 | mvanbaak | exten => 4565566,n,GoToIfTime(21:00-07:59|*|*|*?open2,1) |
| 11:11.30 | mvanbaak | exten => 4565566,n,Goto(open2,1) |
| 11:11.37 | mvanbaak | that last gotoiftime is useless |
| 11:12.08 | mvanbaak | basically what this does is: if it's between 21:00 and 7:59, go to open2, otherwise, go to open2 |
| 11:12.13 | mvanbaak | so the result is the same |
| 11:12.23 | mvanbaak | so you can remove that last GotoIfTime |
| 11:13.39 | mvanbaak | and the second strftime in the monitor_filename lacks the epoch variable |
| 11:13.54 | Dr-Linux|home | mvanbaak: i was doing that way before, in that case it jumps n+101 |
| 11:14.15 | mvanbaak | n+101 is old |
| 11:15.07 | Dr-Linux|home | yeah but whatever is alternative |
| 11:15.17 | mvanbaak | erm |
| 11:15.21 | mvanbaak | you totally dont need it |
| 11:15.26 | Dr-Linux|home | it doesn't go to next priority that way |
| 11:15.42 | Dr-Linux|home | okey |
| 11:15.54 | Dr-Linux|home | since i made some changes so i didn't verify that, lemme remove it |
| 11:16.22 | Dr-Linux|home | also what should be proper variable for epoch there? |
| 11:16.42 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 11:18.07 | *** join/#asterisk quazzmarsh (n=quazzmar@62.8.93.2) |
| 11:19.19 | mvanbaak | EPOCH is a sysvar |
| 11:19.24 | mvanbaak | it will be set to the current timestamp |
| 11:19.28 | quazzmarsh | asterisk on Skype anyone? |
| 11:21.31 | Dr-Linux|home | mvanbaak: since EPOCH is a system variable then what is the variable can be used for asterisk time? :S |
| 11:23.12 | mvanbaak | erm, you did do the STRFTIME |
| 11:23.17 | mvanbaak | that's how you should do it |
| 11:23.31 | mvanbaak | try it |
| 11:24.07 | mvanbaak | maybe it's a good idea to setup a testbox so you can actually see for yourself how it works out |
| 11:25.10 | Dr-Linux|home | hhm... |
| 11:25.13 | Dr-Linux|home | yeah |
| 11:25.22 | Dr-Linux|home | it is a test box |
| 11:25.50 | Dr-Linux|home | mvanbaak: i'm using this: {STRFTIME(${EPOCH},,%Y%m%d)} |
| 11:26.09 | Dr-Linux|home | but i didn't understand what you said here: |
| 11:26.16 | Dr-Linux|home | <mvanbaak> erm, you did do the STRFTIME |
| 11:26.16 | Dr-Linux|home | <mvanbaak> that's how you should do it |
| 11:26.34 | Dr-Linux|home | what should i use instead |
| 11:26.54 | quazzmarsh | ASTERISK SKYPE |
| 11:27.01 | quazzmarsh | anyone! |
| 11:27.45 | mvanbaak | Dr-Linux|home: nothing, this is correct |
| 11:28.05 | mvanbaak | Dr-Linux|home: I meant, you have a second STRFTIME on that line where you forgot the EPOCH as first argument |
| 11:28.21 | mvanbaak | quazzmarsh: NOBODY |
| 11:28.32 | Dr-Linux|home | oo yeah :) |
| 11:28.54 | Dr-Linux|home | mvanbaak: since you have seen the pastebin so i'd try to ask some further question |
| 11:30.52 | Dr-Linux|home | mvanbaak: since billing is always difficult thing for me. Now i wanna bill this, as when call goes on open1/ZAP it should be billed, and when it goes to open2, both times should be save in DB |
| 11:31.23 | Dr-Linux|home | mvanbaak: i'm not suure what variables should i use get proper time for these two type of call routing |
| 11:34.00 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 11:34.00 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 11:36.16 | mvanbaak | Dr-Linux|home: you should use the cdr stuff for that |
| 11:36.19 | mvanbaak | wb murf |
| 11:38.20 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 11:38.32 | Dr-Linux|home | mvanbaak: CDR doesn't show the call time when i forwarded to open1 |
| 11:39.17 | Dr-Linux|home | mvanbaak: and again what thing should i find open1 call time in CDR? |
| 11:40.39 | mvanbaak | it's totally unclear what you want to do |
| 11:41.05 | quazzmarsh | need to integrate skype ON ASTERISK |
| 11:41.09 | quazzmarsh | BEEN THROUGH |
| 11:41.14 | quazzmarsh | ALL THE forums |
| 11:41.22 | quazzmarsh | www.voip-info.org and the likes |
| 11:41.26 | quazzmarsh | nada |
| 11:41.31 | TJNII | Skype ises PROPRIETRY CODECS |
| 11:41.37 | TJNII | which they WON'T RELEASE |
| 11:41.41 | quazzmarsh | oh! |
| 11:41.47 | TJNII | so it is NOT OFFICIALLY SUPPORTED |
| 11:41.54 | TJNII | and all the solutions are HACKS |
| 11:42.02 | quazzmarsh | need a hack |
| 11:42.08 | mvanbaak | google it |
| 11:42.13 | quazzmarsh | ouch |
| 11:42.15 | quazzmarsh | ! |
| 11:42.19 | quazzmarsh | just did that |
| 11:42.44 | mvanbaak | well, then there's your answer |
| 11:43.01 | quazzmarsh | all guiys |
| 11:43.02 | quazzmarsh | *guys |
| 11:43.06 | mvanbaak | in here we cannot support you, because there's no supported skype setup with asterisk |
| 11:43.06 | quazzmarsh | www.yeaster.com |
| 11:43.34 | quazzmarsh | ok |
| 11:43.38 | quazzmarsh | cool |
| 11:44.14 | mvanbaak | if you have questions about yeastar, send them a mail |
| 11:44.28 | quazzmarsh | its on windows |
| 11:44.37 | quazzmarsh | and i need a linux solution |
| 11:46.40 | mvanbaak | you dont need skype :) |
| 11:55.20 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 11:55.32 | whymarkwh | hi there anyone active? |
| 11:59.55 | gr0mit | depends who's asking, whymarkwh ! |
| 12:02.51 | mvanbaak | whymarkwh: depends on the question you have |
| 12:02.53 | mvanbaak | ~ask |
| 12:02.54 | jbot | i guess ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:05.20 | whymarkwh | te220 instalation on asterisk, stuck |
| 12:05.32 | *** join/#asterisk awk (n=awk@65.111.177.74) |
| 12:06.08 | whymarkwh | i cant get this thing to work |
| 12:07.44 | awk | kick it |
| 12:07.56 | whymarkwh | i used a demo trixbox install just to check the settings in zapata and zaptel conf files took that settings and placed them on my asterisk box with no luck |
| 12:08.04 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:08.10 | whymarkwh | i am about to jump on it |
| 12:08.17 | awk | do you have the correct module loaded into the kernel? |
| 12:08.31 | whymarkwh | to expensive to rum over it with the car |
| 12:08.49 | whymarkwh | brb |
| 12:17.33 | *** join/#asterisk RoyK (n=roy@ip-16-29-149-91.dialup.ice.no) |
| 12:26.41 | whymarkwh | what module are you refering to? |
| 12:43.03 | d00gster | I have a weird audio media issue. it seems to be related to my network topology but I can't think of a reason. network is a 48port dell switch where all the PCs and Cisco 7940 phones are directly connected. also connected to the switch is my * server and my routers. The router is the dhcp gateway and my asterisk server's tftp serves the config files to the phones. with this setup everything works. I then all a link to my swit |
| 12:53.34 | rootlogin | i registered a linksys SPA901 to asterisk 1.4 ... when i start a call from that phone there is a gap (silence & no logs) of 10s after dialing .. incoming calls to that phone are processed promptly .. any idea ? |
| 12:55.23 | *** join/#asterisk spiny (n=spiny@69-196-137-127.dsl.teksavvy.com) |
| 12:56.28 | rootlogin | after 10s i see the log-output of the extension Executing [10@default:1] Dial("SIP/7..... and the other side starts ringing .. that also happens when i call to a meetme extension |
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| 13:55.19 | und3r | nobody use freecall? |
| 13:55.38 | und3r | or voipstunt? |
| 13:58.07 | lmadsen | not I |
| 13:59.47 | coppice | I thought freecell was one of the world's most popular games :-\ |
| 14:00.08 | coppice | or was it "Freecell with every crime convicted"? |
| 14:02.38 | und3r | no, www dot freecall dot com |
| 14:07.10 | lmadsen | some people have no humor |
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| 14:37.12 | NovceGuru | und3r: I think there's tons and tons of voip providers that popup and fail |
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| 14:43.19 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
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| 15:15.35 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 15:16.28 | pif | hi, how can I use GotoIf within a list of statements all marked with 'n' (not numbered)? |
| 15:17.15 | russellb | you can label them |
| 15:17.27 | russellb | exten => 1234,n(mylabel),NoOp |
| 15:17.34 | russellb | Goto(mylabel) |
| 15:17.39 | lmadsen | GotoIf($[foo = bar]?mylabel) |
| 15:17.59 | pif | cool |
| 15:18.13 | pif | and what will happen in "exten => 1234,n(mylabel),NoOp" once it's executed? |
| 15:18.32 | pif | will the dialplan continue at the next non-labelled statement? |
| 15:18.36 | russellb | yes |
| 15:18.46 | lmadsen | the label is just a method of marking a specific line |
| 15:18.54 | pif | thanks |
| 15:19.14 | lmadsen | it is the alternative to using numbered priorities |
| 15:19.24 | lmadsen | (and a much better alternative at that!) |
| 15:20.40 | pif | let's say I have "exten => s,n,GotoIf($[${MACRO_EXTEN} != "fax"]?monitor)" |
| 15:21.09 | pif | then: exten => s,n(monitor),MixMonitor(... |
| 15:21.45 | pif | and then: exten => s,n,MoreRegularStuff |
| 15:21.58 | pif | does that play? |
| 15:22.08 | russellb | the MoreRegularStuff? yes |
| 15:22.15 | russellb | it will happen after the MixMonitor ... |
| 15:22.23 | pif | great |
| 15:22.57 | pif | and any MACRO_CONTEXT which is not fax will be Monitor'ed |
| 15:23.48 | *** join/#asterisk anthm (n=anthm@70-9-98-98.area4.spcsdns.net) |
| 15:24.30 | lmadsen | you also want to put the double quotes around the variable ${MACRO_EXTEN} |
| 15:24.58 | pif | oh, but if MACRO_CONTEXT = "fax", then GotoIf will fail and fall through to the n(monitor) line, no? |
| 15:25.01 | lmadsen | you want to put the double quotes on both sides when comparing strings |
| 15:25.06 | pif | ok |
| 15:25.12 | lmadsen | in the case you provided -- yes |
| 15:25.28 | pif | so I should do the reverse |
| 15:25.34 | lmadsen | if there is no 'false' path provided by the GotoIf(), then it'll fall through |
| 15:26.13 | lmadsen | GotoIf($["${MACRO_EXTEN}" = "fax"]?skip_monitor) |
| 15:26.21 | pif | exten => s,n,GotoIf($["${MACRO_EXTEN}" = "fax"]?nomonitor) |
| 15:26.23 | pif | :) |
| 15:26.26 | lmadsen | s,n,MixMonitor() |
| 15:26.31 | lmadsen | s,n(skip_monitor),NoOp() |
| 15:26.37 | pif | much better :) |
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| 15:37.37 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 15:43.42 | Twister | anyknow know of some good voip testing software/hardware, i have alot of calls breaking up and i want to trace where the problem is..im pretty certain its with my isp but i need to prove it |
| 15:53.25 | deeperror | Twister, try mtr |
| 16:01.53 | hsv-al | hello all |
| 16:02.08 | hsv-al | are we all looking forward to a long & glorious weekend of irc/asterisk addiction since its 100farenheit outside |
| 16:02.08 | hsv-al | HEH! |
| 16:03.39 | jaytee | it's only 69F here and pouring rain |
| 16:03.47 | hsv-al | well its nearly 100 here |
| 16:03.51 | hsv-al | its tempting to run 5 miles in this |
| 16:03.58 | hsv-al | even w/ a bottle of water, since i'll lose about 4-6 lbs |
| 16:04.28 | hsv-al | I lost 4.9 lbs yesterday running 5 miles in 92farenheit |
| 16:04.32 | hsv-al | 3 miles in about 25min |
| 16:05.18 | *** join/#asterisk ruied (n=ruied@89-180-123-137.net.novis.pt) |
| 16:06.05 | jaytee | wow |
| 16:07.24 | TJNII | Only 85 degrees here |
| 16:07.38 | TJNII | But the humidity is above 70%. Bleh. |
| 16:07.53 | TJNII | is running the A/C to dry the house, not cool it. |
| 16:11.39 | truent | hsv-al, where ya at? |
| 16:11.42 | ruied | Hi! I'm having a problem in the dst field of my cdr. If I place a call from A to B then place another call from A to C, and when I transfer the call from B to C (in A phone), the cdr, sets a new record with an 's' in the dst field instead of the dst of the transfered call (B) |
| 16:12.36 | ManxPower | ruied: What version of Asterisk? |
| 16:13.22 | ruied | ManxPower, 1.4.19 |
| 16:14.19 | ManxPower | ruied: are you using a macro? |
| 16:14.32 | hsv-al | . /lib/modules/2.6.24-18-generic/kernel/lib/crc-ccitt.ko |
| 16:14.36 | hsv-al | . /lib/modules/2.6.24-16-generic/kernel/lib/crc-ccitt.ko |
| 16:14.40 | hsv-al | . /lib/modules/2.6.24-17-generic/kernel/lib/crc-ccitt.ko |
| 16:14.47 | hsv-al | ;/ |
| 16:15.13 | *** join/#asterisk cabbiepete (n=cabbiepe@host81-158-127-243.range81-158.btcentralplus.com) |
| 16:15.34 | hsv-al | Linux asterix-desktop 2.6.24-18-generic |
| 16:15.42 | hsv-al | that pisses me off |
| 16:15.48 | hsv-al | that ubuntu silently upgraded to a new kernel |
| 16:16.02 | hsv-al | hence i had to get help recompiling zt last night ;/ |
| 16:16.24 | jaytee | I never let my 'buntu updates apply automatically. |
| 16:16.38 | jaytee | but I don't run * on 'buntu. just desktop |
| 16:16.47 | hsv-al | im using * on 8.04 |
| 16:16.57 | jaytee | with * it's CentOS 5 or RHEL 5 |
| 16:17.00 | hsv-al | with 411P fxo/fxs at home |
| 16:17.04 | hsv-al | ideal training setup at home |
| 16:17.33 | lmadsen | ruied: there must be some sort of macro or something delivering the call, because the 'dst' field will contain the last extension that was executed |
| 16:18.01 | lmadsen | ruied: also read the latest posts on the -dev mailing list which are talking about CDRs and transfers |
| 16:18.07 | *** join/#asterisk SteveTotaro (n=Administ@pool-70-17-222-90.balt.east.verizon.net) |
| 16:18.48 | jaytee | ideal training setup for me at home would involve 3 Asterisk servers with one pretending to be the Telco, one pretending to be a Nortel Option 11C and a primary Asterisk server for all internal customers. |
| 16:21.32 | jaytee | and that's with two TE212P cards |
| 16:22.02 | jaytee | or four actually |
| 16:26.09 | lmadsen | I bought a quad-core box and installed VMware-server and now I can run 4-8 asterisk servers on it |
| 16:26.38 | hsv-al | sed -e 's/guano/johndoe/g;s/pigeon/johndoe2/g' FILENAME |
| 16:26.52 | hsv-al | woudl that replace all occurances of guano with johndoe, and all occurances of pigeon with johndoe2, in a given txt file? |
| 16:28.24 | ruied | this is the output that I've mentioned about the transfered call: http://pastebin.com/ma01c768 |
| 16:28.37 | Strom_L | jaytee: so wait a second, why is one pretending to be the Option 11C? |
| 16:28.40 | hsv-al | lmadsen, how does asterisk run |
| 16:28.49 | hsv-al | inside of vmware, mainly; audio quality |
| 16:28.52 | hsv-al | and assumed loss of quality |
| 16:29.05 | lmadsen | works fine for me, but its a test box, I don't run 100s of calls through it |
| 16:29.23 | lmadsen | for me when I'm building on vmware, its not the audio quality I care about -- its the logic |
| 16:30.23 | jaytee | Strom_L, because I currently have an Option 11C running at work. I've integrated the Asterisk and Nortel dialplans using Asterisk as a primary call router for my inbound/outbound calls over PRI |
| 16:32.07 | Strom_L | ok...that still doesn't actually answer my question |
| 16:32.22 | Strom_L | (perhaps I'm just dense?) |
| 16:32.44 | jaytee | so that I can emulate my dialplan at work? and model changes, etc. |
| 16:33.23 | TJNII | jaytee: Basically because you have a legacy box you arn't ready to get rid of that you want to emulate, correct? |
| 16:33.35 | Strom_L | what I think perhaps I'm fishing for is why that needs to be a separate box and not merely a function of one of the other boxes |
| 16:33.42 | jaytee | TJNII, exactly |
| 16:35.07 | jaytee | Strom_L, I could probably emulate it with another function on one other box but I prefer to have real T1 spans that match my setup at work as closely as possible. Not the cheapest approach. |
| 16:35.29 | Strom_L | alright...that seems reasonable ;) |
| 16:35.41 | jaytee | because my dialplan logic is "unigue" to say the least. |
| 16:35.44 | hsv-al | jaytee |
| 16:35.46 | hsv-al | found a good url |
| 16:35.47 | hsv-al | http://www.student.northpark.edu/pemente/sed/sed1line.txt |
| 16:37.47 | jaytee | right now all calls inbound from my telco come through my Nortel. Any extensions that are VOIP exist as a "phantom TN" which is remote call forwarded to the outbound PRI to a 555XXXX number. * strips the 555 and routes the call to the SIP phone. All other 7 and 10 digit calls plus 911, 411, 011 etc go out the outbound PRI on * to the telco. Ugly, ain't she? :-) |
| 16:38.44 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 16:39.04 | *** part/#asterisk spiny (n=spiny@69-196-137-127.dsl.teksavvy.com) |
| 16:40.19 | Strom_L | jaytee: I would perhaps suggest a different unassignable prefix than 555, since 555 is occasionally used for things |
| 16:40.35 | jaytee | not in Marion County, Indiana |
| 16:40.52 | jaytee | and that's an easy change |
| 16:41.46 | jaytee | I just don't think any of the "prepackaged" flavors of * would have met my peculiar needs as well as standard * does. |
| 16:45.04 | Strom_L | yeah |
| 16:46.04 | Strom_L | i'm thinking about the numbering thing, and there are two solutions that seem more workable in the long term than 555 |
| 16:46.34 | Strom_L | pick an unassignable area code like 311, since 1-311-NXX-XXXX isn't ever going to do anything when dialed on the PSTN |
| 16:46.41 | Strom_L | or |
| 16:46.56 | Strom_L | instead of 555, use one of the vertical service codes reserved for local assignment |
| 16:49.42 | seanbright | ~vsc |
| 16:49.42 | jbot | it has been said that vsc is Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 16:49.54 | seanbright | was schooled on those by Strom_L |
| 16:50.01 | Strom_L | :) |
| 16:52.20 | jaytee | hmmmm, I think I need more coffee |
| 16:53.38 | Strom_L | more coffee eh? See ITU-T Q.850 about that one !!! |
| 16:58.49 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:59.48 | Strom_L | ~itu |
| 16:59.48 | jbot | somebody said itu was the International Telecommunication Union. Current versions of ITU-T recommendations (Q,931, T.38, V.32, et cetera) are available for free in PDF format from their website: http://www.itu.int/rec/T-REC/e |
| 17:00.01 | Strom_L | there we goes |
| 17:01.29 | seanbright | ~seanbright |
| 17:01.30 | jbot | from memory, seanbright is a girl with standards |
| 17:01.37 | seanbright | i never get tired of that |
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| 17:05.49 | lmadsen | lol |
| 17:05.53 | lmadsen | ~lmadsen |
| 17:05.53 | jbot | hmm... lmadsen is dreamy http://blogs.dfw.com/startle_grams/images/leif75_4.jpg |
| 17:06.02 | lmadsen | hahaha oh ya! |
| 17:06.49 | Strom_L | hehhe |
| 17:08.31 | outtolunc | hot like a monkeys uncle eh <G> |
| 17:08.54 | ManxPower | looks more like an 80's rock star 8-) |
| 17:11.19 | jaytee | mmmm, this Kenya AA peaberry is delightful. Effervescent and citrusy with distinct lemon notes. |
| 17:13.32 | ManxPower | it has bubbles in it?? |
| 17:13.58 | Strom_L | lemon...notes? |
| 17:14.36 | ManxPower | I like my coffee dark roast, you don't get any of those subtle flavours then. |
| 17:14.51 | Strom_L | im trying to resolve that phrase, but keep thinking it sounds like someone took a bunch of Db and E and Ab and put them into a drink |
| 17:15.11 | jaytee | hehehe |
| 17:15.48 | jaytee | this is a full city + roast so it's kind of on the dark side |
| 17:16.26 | jaytee | I've got some really nice Kona I'm going to roast a batch of later this afternoon. |
| 17:17.10 | Strom_L | "Here's a drink written by Depeche Mode" |
| 17:17.21 | jaytee | rofl |
| 17:18.05 | Strom_L | your own / personal / citrus / someone to quench your thirst (etc) |
| 17:19.30 | jaytee | Life is too short to drink bad coffee |
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| 17:21.03 | hsv-al | ugh, wtf, i was drinking iced tea in a cup, and when i was done sipping, i looked in the cup, and bugged out for a second ,thinking a slug was sliding down the cup |
| 17:21.06 | hsv-al | but it was just an ice cube hahah |
| 17:21.45 | Strom_L | you really should stop smoking all that crack |
| 17:22.01 | lmadsen | wow.. |
| 17:24.32 | hsv-al | I just have fxs now, on 411P, but.........if my analong phone is connected to the fxs port, shouldnt it be picking up a emualted ring tone from * or no? i thought i could dial sip destinations w/ my analog phone at least for now? |
| 17:25.18 | Strom_L | asterisk should generate alerting tone, yes |
| 17:25.26 | hsv-al | its blank |
| 17:25.34 | hsv-al | any ideas? my phoen has no tone |
| 17:25.36 | Strom_L | what kind of sip equipment are you dialing? |
| 17:26.01 | hsv-al | just a normal |
| 17:26.11 | hsv-al | (SIP/5002@x.x.x.x) |
| 17:26.19 | Strom_L | "a normal" isn't a brand of equipment |
| 17:26.29 | hsv-al | via extension 500, but when I pick up the phone, no tone |
| 17:26.39 | Strom_L | do you get no dial tone? |
| 17:26.42 | hsv-al | nope |
| 17:26.42 | TJNII | hsv-al: most sip devices create a dialplan when they are registered |
| 17:26.53 | TJNII | And no dialtone when they can't register |
| 17:27.03 | Strom_L | TJNII: he's not calling from a SIP device |
| 17:27.08 | Strom_L | please read the problem and try again |
| 17:27.10 | Strom_L | thank you for using AT&T |
| 17:27.12 | Strom_L | *bong* |
| 17:27.17 | Strom_L | so anyway |
| 17:27.31 | TJNII | is 0/2 this week...... |
| 17:27.40 | Strom_L | hsv-al: pastebin your zaptel.conf and zapata.conf and extensions.conf |
| 17:27.45 | hsv-al | ok |
| 17:27.52 | Strom_L | and also verify that your phone's receiver element is working |
| 17:31.14 | Strom_L | also, do you have any other telephony cards in your system besides the 411P (I assume you mean TDM411P and not TE411P) |
| 17:31.28 | hsv-al | yes one sec still |
| 17:31.35 | hsv-al | just a tDM411P , with 1 fxs for now |
| 17:31.38 | Strom_L | ok |
| 17:31.58 | Strom_L | (sorry for the barrage of questions) |
| 17:32.24 | outtolunc | *BONG* <G> |
| 17:33.01 | ManxPower | Just remember if you plug a phone LINE into a Digium FXS port it will blow the port when a call comes in. Make sure you didn't do that. Also make sure you have power to your analog card. |
| 17:33.32 | ManxPower | Also it is easy to mix up the ports on the Digium card. Make sure the module is in the port you think it is in. |
| 17:33.51 | hsv-al | it is. |
| 17:34.00 | hsv-al | fxs in slot 2 on card, and slot 2 light is green on back |
| 17:34.07 | hsv-al | and the analong phone is into the green light slot on card |
| 17:34.34 | Strom_L | analong! |
| 17:34.39 | Strom_L | as opposed to anashort |
| 17:34.47 | Strom_L | (anamedium?) |
| 17:34.58 | hsv-al | http://lists.digium.com/pipermail/asterisk-users/2004-October/062970.html |
| 17:34.59 | hsv-al | reading this |
| 17:35.05 | hsv-al | and i hope im not facing this same situation |
| 17:35.09 | hsv-al | may boot into the bios |
| 17:35.37 | Strom_L | hsv-al: can you please just pastebin already instead of trying to do other things while i'm trying to help you troubleshoot? |
| 17:37.08 | hsv-al | i found it, im still using the example zapata.conf |
| 17:37.18 | hsv-al | pasting |
| 17:37.29 | hsv-al | i have to switch up the channel #'s i believe |
| 17:37.52 | hsv-al | http://pastebin.com/m45c6669a |
| 17:37.54 | hsv-al | = zapata.conf |
| 17:38.03 | Strom_L | now the others I requested |
| 17:38.59 | hsv-al | http://pastebin.com/m43ab98a4 = zaptel.conf |
| 17:39.31 | ManxPower | shouldn't the zones be before the channels? |
| 17:39.32 | Strom_L | get rid of the fxsks stuff |
| 17:39.41 | Strom_L | i dont believe it matters, ManxPower |
| 17:39.56 | hsv-al | http://pastebin.com/m6cbabf9f = extensions |
| 17:40.02 | hsv-al | i havent inputted the sip extension connection stuff yet |
| 17:40.11 | hsv-al | but i know where to put it, so any of these i hope helps why i cant get a tone |
| 17:40.21 | Strom_L | hsv-al: yeah, delete all the FXO port related nonsense (since you dont have one) |
| 17:40.42 | ManxPower | hsv-al: I can't imagine your config not generating errors. |
| 17:40.44 | Strom_L | you should have fxoks = 2 in your zaptel.conf and your fxo_ks definition being applied to channel 2 in your zapata.conf |
| 17:41.29 | hsv-al | zaptel.conf does have |
| 17:41.30 | hsv-al | fxoks=2 |
| 17:42.40 | hsv-al | commented out the last 3 lines in zapata.conf |
| 17:42.59 | Strom_L | you need to do more than just comment out lines |
| 17:44.27 | ManxPower | I usually try to get ztcfg -vvv to not have errors before I do ANYTHING else. |
| 17:45.15 | hsv-al | ok, i just removed the fxo part from zaptel.conf |
| 17:45.19 | hsv-al | and ztcfg -vvvv |
| 17:45.26 | hsv-al | and channel 02: fxo kewlstart (default) (slaves: 02) |
| 17:45.27 | hsv-al | 1 chan to config |
| 17:45.54 | hsv-al | ko |
| 17:45.55 | hsv-al | dial tone now |
| 17:45.56 | Strom_L | ok |
| 17:45.58 | Strom_L | now |
| 17:46.01 | Strom_L | what about zapata.conf? |
| 17:46.10 | Strom_L | did you fix the channel assignment? |
| 17:46.13 | hsv-al | yah |
| 17:46.18 | hsv-al | it said in the cli: |
| 17:46.25 | hsv-al | -- starting simple switch on 'Zap/2-1' |
| 17:46.28 | hsv-al | -- hung up 'zap/2-1' |
| 17:46.29 | Strom_L | congrats |
| 17:47.14 | hsv-al | ahhh, so this works now |
| 17:47.24 | hsv-al | i just have to write down, to put the other stuff back in when the other chip arrives |
| 17:47.32 | Strom_L | it's not a chip |
| 17:47.44 | Strom_L | it's a circuit board |
| 17:47.54 | Strom_L | ~brain |
| 17:47.55 | jbot | it has been said that brain is a very useful device. It is advised that you ensure your brain is switched to the "on" position before continuing. If you need assistance, please consult your directory. This is a recording. |
| 17:48.06 | hsv-al | well they're chips to me, because I dip fxo's in salsa |
| 17:48.12 | hsv-al | thanks for this :) |
| 17:48.18 | Strom_L | ...that was the lamest joke ever |
| 17:48.20 | Strom_L | sorry |
| 17:48.26 | hsv-al | i know what doesnt float your boat |
| 17:48.47 | outtolunc | 2 fxs and 1 fxo modules walk into a bar... |
| 17:49.56 | errr | lol |
| 17:50.11 | hsv-al | is there a way to delay the echo |
| 17:50.11 | Strom_L | and the one says to the other "OOOOOOOOOOOO" and the first says "BEEP BOOP BOOP BEEP BEEP BOOP BEEP" and the second says "KACHEEP BRRRRR BRRRR BRRRRR BRRRR" |
| 17:50.14 | hsv-al | in the test echo application? |
| 17:50.19 | hsv-al | instead of 1-3ms.. |
| 17:50.22 | Strom_L | this is the last time I buy comedy at wal-mart |
| 17:50.34 | Strom_L | hsv-al: no -- that's what an echo test is supposed to do |
| 17:51.37 | Strom_L | if you want a delayed test, have asterisk make a recording and then play it back to you |
| 17:52.21 | hsv-al | well time to make dunkin donuts cofee and read another 80 pages |
| 17:52.29 | hsv-al | thx, this got it working. |
| 17:52.46 | Strom_L | hsv-al: read this |
| 17:52.48 | Strom_L | ~101 |
| 17:52.48 | jbot | from memory, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 17:53.14 | hsv-al | so far the book hasnt went over what "Zap/2-1" means, but i see it in this channel alot the talk |
| 17:53.16 | hsv-al | what does Zap mean? |
| 17:53.25 | hsv-al | , ie: my cli printouts |
| 17:53.35 | Strom_L | Zaptel |
| 17:53.41 | Strom_L | i.e. zapata telephony project |
| 17:53.49 | hsv-al | I know, but when i got a dialtone |
| 17:53.58 | hsv-al | Echo("Zap/2-1", "") showed |
| 17:54.01 | hsv-al | in CLI |
| 17:54.04 | Strom_L | ...look, turn your brain on for just a second |
| 17:54.12 | Strom_L | your phone is connected to a zaptel interface |
| 17:54.18 | Strom_L | specifically, port 2 |
| 17:54.25 | Strom_L | and therefore.... |
| 17:54.35 | Strom_L | points right at the answer, hoping you'll see it |
| 17:54.35 | outtolunc | its your DADHI <G> |
| 17:55.02 | hsv-al | well i guess thats the way the verbal standards were made |
| 17:55.08 | hsv-al | it ref's: ports on the card |
| 17:56.09 | hsv-al | well ill probably read that PDF you posted |
| 17:56.19 | hsv-al | has alot of terms im not familiar with, I havent worked in telephone industry, so its new |
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| 18:36.16 | jaytee | <PROTECTED> |
| 18:37.04 | Strom_L | OH JEAN-CHARLES YOU HAVE SAT ON THE INTERNET PUMP AGAIN |
| 18:37.08 | Strom_L | NOW WE ARE OFFLINE FOREVER AND EVER |
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| 18:40.05 | jaytee | "I have looking everywhere in ze tubes for ze packet, but she is nowhere!" |
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| 19:09.42 | hsv-al | bleh |
| 19:09.45 | hsv-al | off 4 mile run |
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| 19:22.27 | ManxPower | Isn't it DAHDI? |
| 19:22.54 | ManxPower | "Dahdi, your new car is just divine!" |
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| 19:29.27 | LuisTorres | Howdy |
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| 19:31.41 | chrisM8 | hi, does anyone know how can i get "musiconhold - start/stop" event in * manager ? |
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| 19:43.15 | *** mode/#asterisk [+ooo codefreeze twisted Corydon76-dig] by irc.freenode.net |
| 19:43.15 | drmessano | I guess today is attack day |
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| 20:00.48 | *** mode/#asterisk [+ooo codefreeze twisted Corydon76-dig] by irc.freenode.net |
| 20:02.57 | rootlogin | my * is behind a NAT firewall and registers to an "outside" SIP peer ... i have set externip but * still sends its internal IP in the REGISTER request |
| 20:03.37 | seanbright | ~nat |
| 20:03.38 | jbot | hmm... nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 20:03.40 | seanbright | ~sipnat |
| 20:03.41 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:06.24 | rootlogin | hmmm well those are still open in my browser ;) .. localmask is not set .. nevertheless i will look again |
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| 20:19.31 | hsv-al | heh |
| 20:19.48 | hsv-al | no one talking |
| 20:19.48 | hsv-al | all passed out from the heat |
| 20:20.55 | Blake` | heat? |
| 20:20.55 | hsv-al | yah |
| 20:20.55 | hsv-al | i just got bak running 5 miles, and almost passed out when i got in my apartment |
| 20:20.55 | hsv-al | it was so cold |
| 20:21.30 | hsv-al | then realized thermometer said outside was 102farenheit |
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| 20:21.34 | Blake` | heh, nice |
| 20:21.34 | Blake` | and freenode seems to be acting up today |
| 20:21.54 | tzafrir_laptop | temprature down? |
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| 20:25.32 | drmessano | WHERE IS YOUR DIALPLAN NOW? |
| 20:25.38 | hsv-al | heh |
| 20:26.55 | mvanbaak | I lost it in the netsplits |
| 20:27.07 | Blake` | mmm netsplits |
| 20:27.10 | Blake` | w/ chocolate sauce |
| 20:27.20 | mvanbaak | gheh |
| 20:27.34 | mvanbaak | The Texas Chainsaw Massacre on tv |
| 20:27.58 | drmessano | I dont know why, but "WHERE IS YOUR GOD NOW?" has become one of my fav memes |
| 20:28.31 | mvanbaak | there is no god |
| 20:28.33 | drmessano | Especially after seeing the one with the Burger King "King" standing next to the dude in his kitchen with that very caption |
| 20:28.36 | mvanbaak | only asterisk |
| 20:29.11 | drmessano | http://blogtown.portlandmercury.com/files/2007/03/theapocalypseisdelicious.jpg |
| 20:29.13 | drmessano | There |
| 20:29.15 | drmessano | pwned |
| 20:29.55 | mvanbaak | gheh |
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| 20:31.07 | drmessano | http://www.igloolounge.net/pictures/april06/whereisyourgodnow2.jpg |
| 20:31.41 | DrukenLPY | that's a good shot of that chick.. hehehe |
| 20:31.46 | hsv-al | when is that horse race on |
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| 20:48.20 | *** mode/#asterisk [+ooo codefreeze twisted Corydon76-dig] by irc.freenode.net |
| 20:48.49 | whymarkwh | hi there does anyone know what the piouts are for a crossover cable for 2 port pri diguim board? |
| 20:48.50 | Strom_M | whymarkwh: yes |
| 20:48.50 | Strom_M | T1 crossover |
| 20:49.06 | jaytee | yep, just cross pin 1 with pin 4 and pin 2 with pin 5 |
| 20:49.15 | whymarkwh | thx |
| 20:49.21 | jaytee | yw |
| 20:51.26 | Strom_M | whymarkwh: you may also want to look into a crossover adapter so you can use premanufactured patch cables. smatronix makes a T1 crossover adapter. |
| 20:51.32 | whymarkwh | one pri i want to connect to my telco and the other to my philips pbx with asterisk in the middel ,,do i have to setup both pri's on asterisk as cpr or net? |
| 20:51.47 | Strom_M | whymarkwh: the port that faces the telco you set as CPE |
| 20:51.53 | Strom_M | the port that faces your other PBX you set as NET |
| 20:52.10 | jaytee | the pri to the telco needs to be pri-cpe and the pri to your philips pbx needs to be pri-net |
| 20:52.11 | whymarkwh | i take it you have done this before? |
| 20:52.16 | Strom_M | I don't think you can set asterisk to do cardiopulmonary resuscitation on a PRI :) |
| 20:52.18 | Strom_M | whymarkwh: yes |
| 20:53.18 | jaytee | whymarkwh, my * box sits as a "man-in-the-middle" between my telco and my Nortel Option 11c |
| 20:53.34 | whymarkwh | i have never worked on pri only bri, tell what do you get from your telko if calls come in on a did on pri:? |
| 20:53.47 | Strom_M | whymarkwh: that'll depend on how the telco has DNIS set up |
| 20:53.57 | Strom_M | usually they pass you the dialed number |
| 20:54.17 | Strom_M | but you should verify with your telco and/or just try it out |
| 20:54.44 | whymarkwh | i guess NoOp is the best way to find out? |
| 20:54.53 | jaytee | yes, NoOp is your friend |
| 20:54.58 | *** join/#asterisk l0verb0y (n=l0verb0y@124.107.127.238) |
| 20:55.05 | Strom_M | exten => _X.,1,NoOp(Telco is sending ${EXTEN} as DNIS) |
| 20:55.32 | jaytee | that guy really ought to write a book |
| 20:55.37 | Strom_M | what guy |
| 20:55.44 | jaytee | pity there's no money in it. |
| 20:56.19 | jaytee | you |
| 20:56.29 | drmessano | I wouldnt read it |
| 20:56.29 | Strom_M | heheh |
| 20:56.32 | drmessano | Then again |
| 20:56.36 | drmessano | No one reads asterisk books |
| 20:56.40 | drmessano | So thats not an insult |
| 20:56.50 | Strom_M | just buy my time instead :) |
| 20:56.53 | drmessano | Most people buy/download them, and hope osmosis kicks in |
| 20:57.31 | jaytee | I wonder if lmadsen got more than 10 cents from O'Reilly from the printed edition of *, TFOT 2ndEd. I bought |
| 20:57.32 | whymarkwh | one more thing if the card is installed and i can see all the channels in the cli> is there any indication on the card(meaning led) that should be on , i have nothing connected yet but there is no light? |
| 20:57.48 | Strom_M | whymarkwh: which card is it? |
| 20:57.49 | lmadsen | jaytee: at least 12 |
| 20:58.08 | jaytee | 12 cents? |
| 20:58.18 | lmadsen | aye |
| 20:58.24 | Strom_M | whymarkwh: you should have a flashing red light on the back if the span is configured correctly with nothing plugged into it |
| 20:58.25 | lmadsen | don't worry... I'm not living in the lap of luxury because of the book :) |
| 20:58.31 | jaytee | whoa! these nets be acting up and such, doncha be thinking? |
| 20:58.33 | Strom_M | it should go green if you hard loop the port |
| 20:58.33 | whymarkwh | te220 |
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| 20:59.05 | lmadsen | heads back downstairs to get the beer |
| 20:59.07 | whymarkwh | nothing flahing |
| 20:59.19 | Strom_M | whymarkwh: perhaps you havent configured it correctly |
| 20:59.27 | Strom_M | pastebin your zaptel.conf and your zapata.conf |
| 21:02.31 | d00gster | I have a weird audio media issue. it seems to be related to my network topology but I can't think of a reason. network is a 48port dell switch where all the PCs and Cisco 7940 phones are directly connected. also connected to the switch is my * server and my routers. The router is the dhcp gateway and my asterisk server's tftp serves the config files to the phones. with this setup everything works. I then all a link to my swit |
| 21:03.53 | Strom_M | d00gster: question is truncated at "a link to my swi" |
| 21:04.16 | d00gster | ok |
| 21:04.36 | d00gster | I have a weird audio media issue. it seems to be related to my network topology but I can't think of a reason. network is a 48port dell switch where all the PCs and Cisco 7940 phones are directly connected. also connected to the switch is my * server and my routers. The router is the dhcp gateway and my asterisk server's tftp serves the config files to the phones. with this setup everything works. I then all a link to my |
| 21:04.38 | d00gster | <PROTECTED> |
| 21:05.14 | Strom_M | there's a NAT device between asterisk and the phones? |
| 21:05.23 | d00gster | no |
| 21:05.32 | d00gster | they're on the same dell switch |
| 21:06.09 | Strom_M | is the media not getting from endpoint to endpoint? |
| 21:06.20 | Strom_M | or are the calls working, just with media not touching asterisk? |
| 21:06.31 | d00gster | later |
| 21:06.38 | Strom_M | the phones are doing what's called a reinvite |
| 21:06.39 | Strom_M | that's normal |
| 21:06.44 | d00gster | I expect rtp to go through * |
| 21:06.54 | d00gster | I have noreinvite |
| 21:06.56 | Strom_M | you can disable it by saying "canreinvite=no" in the sip configuration file on asterisk |
| 21:07.01 | d00gster | have that |
| 21:07.07 | Strom_M | pastebin your sip.conf |
| 21:07.49 | d00gster | see when I disconnect the second router and make the first router the dhcp and only gateway all works |
| 21:08.22 | d00gster | I see RTP going through * |
| 21:08.55 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
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| 21:08.57 | *** mode/#asterisk [+ooo codefreeze twisted Corydon76-dig] by irc.freenode.net |
| 21:08.58 | d00gster | when I add the second router and make it the DHCP for all other devices ... rtp don't work |
| 21:08.58 | Strom_M | d00gster: well, i don't know then...I'm a telephony engineer, not an IP networking engineer |
| 21:09.24 | d00gster | ok. thanks |
| 21:09.24 | lmadsen | sounds like your routes are setup wrong then for the RTP |
| 21:09.29 | d00gster | how so? |
| 21:09.30 | lmadsen | once you add the 2nd router (I didn't see the scrollback, so I'm not entirely sure what you did :)) |
| 21:09.45 | lmadsen | well, if you're not getting RTP, then the route must be wrong... no? |
| 21:09.59 | d00gster | well sip is making it to * |
| 21:10.09 | d00gster | rtp is configured to to through * |
| 21:10.29 | whymarkwh | k thats wierd crossover cable installed now i have one green light |
| 21:12.47 | Strom_M | whymarkwh: green is good |
| 21:12.59 | Strom_M | it means that the T1/E1 is synched up |
| 21:15.07 | d00gster | I think I better put a sniffer and see that's going on ... thanks gents |
| 21:19.19 | whymarkwh | what is that site to past again? |
| 21:19.58 | jaytee | ~pastebin |
| 21:19.59 | jbot | [~pastebin] 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 |
| 21:20.05 | whymarkwh | thx |
| 21:20.21 | Strom_M | paste |
| 21:20.22 | Strom_M | not past |
| 21:20.26 | Strom_M | ENGLISH PEOPLE |
| 21:20.39 | whymarkwh | yes thanks for the the spell check |
| 21:20.51 | whymarkwh | cold here in south africa |
| 21:24.58 | whymarkwh | http://pastebin.com/m33192c3f if you don't mind having a look Strom_M after boot no lights |
| 21:25.04 | jaytee | yeah, your summer is our winter here in the Northern hemisphere |
| 21:27.00 | jaytee | you've got pri_cpe signalling on both groups, shouldn't one of them be pri_net? |
| 21:27.12 | Strom_L | whymarkwh: the channel => line in zapata.conf needs to go /below/ the context and group lines |
| 21:27.19 | Strom_L | and what jaytee said about signalling |
| 21:27.36 | Strom_L | also, what happens when you run ztcfg -vv from the bash prompt |
| 21:28.38 | whymarkwh | well just testing with a crossover do i need to change it now? |
| 21:28.53 | Strom_L | do you want help, or do you want to argue? |
| 21:29.02 | Strom_L | :) |
| 21:29.20 | whymarkwh | i want help |
| 21:29.31 | Strom_L | then follow our advice |
| 21:30.08 | jaytee | if you loop 2 T1/E1 circuits with a x-over and both are set to pri_cpe or pri_net then you can't make test calls, one needs to be net and one needs to be cpe |
| 21:31.21 | whymarkwh | http://pastebin.com/m75e9afaa here is the output doing ztcfg -vvv |
| 21:31.37 | Strom_L | ok, good |
| 21:31.47 | Strom_L | now do you have red and/or green alarms on your card? |
| 21:32.34 | mvanbaak | pink |
| 21:32.50 | Strom_M | mvanbaak: the technical term for that one is "girl alarm" |
| 21:33.08 | mvanbaak | yeah |
| 21:33.38 | whymarkwh | no lights what so ever |
| 21:34.44 | Strom_M | what happens when you hard-loop the ports |
| 21:35.01 | whymarkwh | still no lights |
| 21:35.58 | Strom_M | you're using a loopback plug? |
| 21:36.11 | jaytee | a crossover cable |
| 21:36.11 | whymarkwh | yess |
| 21:36.15 | Strom_M | jaytee: no |
| 21:36.23 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:36.23 | Strom_M | a loopback plug != a crossover cable |
| 21:36.32 | whymarkwh | i have a errorcrossover |
| 21:36.46 | Strom_M | whymarkwh: you need to build a loopback plug |
| 21:36.50 | jaytee | true, it doesn't. he asked earlier about a crossover cable. |
| 21:37.12 | whymarkwh | te4xxp version syncronization error |
| 21:37.41 | Strom_M | are you sure that's the right driver? |
| 21:38.07 | whymarkwh | dont know |
| 21:38.59 | jaytee | did you compile it or install from packages? |
| 21:39.20 | whymarkwh | yes libpri and zaptel and asterisk |
| 21:39.24 | jaytee | that error indicates the module isn't compiled with the right sources for your current kernel. |
| 21:41.53 | tzafrir_laptop | jaytee, what error? |
| 21:42.17 | Strom_M | PUFFIES |
| 21:42.19 | jaytee | version synchronization loading the te4xxp |
| 21:42.32 | mvanbaak | if I see how much trouble ppl have with t1/e1 cards I'm so darn happy with my pure-voip setups |
| 21:43.04 | tzafrir_laptop | yeah, right. That one that never has jitter issues, not to mention connectivity issues |
| 21:43.32 | mvanbaak | indeed |
| 21:43.36 | tzafrir_laptop | And that funny thing called QOS |
| 21:43.47 | jaytee | and TOS |
| 21:43.51 | mvanbaak | tzafrir_laptop: maybe if you have to deal with adsl |
| 21:44.00 | mvanbaak | we are all 10mbit fiber or better here |
| 21:44.39 | tzafrir_laptop | yeah. and SIP is a pleasent thing to debug |
| 21:44.57 | mvanbaak | you can get 100mbit managed dark fiber for 300 us $ here |
| 21:45.13 | mvanbaak | tzafrir_laptop: there's this neat thing called iax |
| 21:45.14 | jaytee | is dark fiber made from dark matter? |
| 21:45.28 | tzafrir_laptop | Which is also very nice |
| 21:45.30 | mvanbaak | jaytee: it's a product of the dark master |
| 21:46.01 | tzafrir_laptop | whymarkwh, what exactly is the issue you have here? |
| 21:47.08 | *** join/#asterisk s0lid (n=s0lid@58.69.1.79) |
| 21:47.13 | mvanbaak | tzafrir_laptop: why the hell are you even using asterisk if sip and iax and anything voip is so crappy in your opinion ? |
| 21:47.42 | whymarkwh | can see the channels in cli> |
| 21:47.45 | tzafrir_laptop | I didn't say they were. Just compared them to PRI |
| 21:47.49 | jaytee | settle down, you two! |
| 21:47.53 | whymarkwh | but no indication lights on card |
| 21:48.04 | tzafrir_laptop | whymarkwh, what is the output of: cat /proc/zaptel/* |
| 21:48.17 | mvanbaak | tzafrir_laptop: it's a fact that most trouble come from ppl who cannot get pri hardware to work corretly |
| 21:48.38 | mvanbaak | just look at the logs for the last 6 months |
| 21:48.44 | tzafrir_laptop | that's also because it's under-documented |
| 21:49.04 | mvanbaak | possible, but that does not make it any better |
| 21:49.05 | tzafrir_laptop | Maybe people at digium support enjoy getting extra support calls |
| 21:49.22 | tzafrir_laptop | judging by the quality of documentation Digium provides for its hardware |
| 21:49.42 | jaytee | I've come in here but I've never had to ask Digium for help on their T1 cards. I got it working fairly easy. |
| 21:50.04 | mvanbaak | the docs are good enough to get you setup with a working pri line |
| 21:50.24 | jaytee | there are good howtos on voip-info and asteriskguru |
| 21:50.36 | jaytee | and the pdfs from Digium |
| 21:50.46 | *** join/#asterisk voipenthusiast (n=rodrigo@67-207-133-226.slicehost.net) |
| 21:51.13 | mvanbaak | the pdfs from digium are ok |
| 21:51.34 | tzafrir_laptop | They are quite obsolete |
| 21:51.48 | drmessano | Well, here is my two cents.. since, nobody asked for it.. |
| 21:51.53 | voipenthusiast | freeswitch rules, asterisk sucks |
| 21:52.11 | drmessano | I buy a modem, I get a basic manual.. no one tells me how to connect to the internet or how to debug it |
| 21:52.46 | drmessano | I buy an external USB har drive, same thing |
| 21:52.55 | mvanbaak | yeah |
| 21:52.56 | drmessano | The documentation for the T1 cards is FAR more than that |
| 21:53.01 | mvanbaak | external usb is even better |
| 21:53.13 | tzafrir_laptop | drmessano, If noone tells you you might actually call support. And thus waste time of a support personel |
| 21:53.16 | mvanbaak | connect it, run this exe, and you are all setup |
| 21:53.17 | drmessano | So it doesnt come with a fucking book on kernel debugging.. wtf? |
| 21:53.25 | mvanbaak | I mean, 'run this exe' ??? |
| 21:53.37 | tzafrir_laptop | Or even worse, you might misunderstand something and never use the product |
| 21:53.53 | whymarkwh | here is the output of cat /proc/zaptel/* http://pastebin.com/m523aeb55 |
| 21:54.00 | drmessano | If you can't get a T1 card working, you shouldn't be using Asterisk.. |
| 21:54.05 | tzafrir_laptop | And when the product costs over 200$, it's not that easy to dismiss complaints of customers |
| 21:54.53 | mvanbaak | drmessano: you shouldn't use pri |
| 21:55.02 | drmessano | Price has nothing to do with it.. Complexity does.. if you can't handle the complexity of it, you shouldn't be installing it |
| 21:55.20 | tzafrir_laptop | smiles silently |
| 21:55.21 | mvanbaak | drmessano: I can setup iax links etc without trouble, but the t1 stuff always manages to break |
| 21:56.03 | drmessano | mvanbaak: The card isn't changing.... soo..... |
| 21:56.15 | drmessano | Sounds like a software problem.. not a hardware setup issue |
| 21:56.28 | drmessano | If it works at any one point.. the hardware is set up |
| 21:56.32 | tzafrir_laptop | And who provides the software? |
| 21:56.33 | mvanbaak | no, the problem is between chair and keyboard |
| 21:56.51 | drmessano | mvanbaak: Software... as in config, install, etc |
| 21:57.07 | mvanbaak | if I have to use pri connections I always call a friend of mine, he fixes it in 10 minutes all the time |
| 21:57.13 | mvanbaak | I just dont want to mess with it |
| 21:57.37 | mvanbaak | I'm from the ip generation. pri and stuff is before my time |
| 21:57.48 | drmessano | tzafrir_laptop: It's not digiums fault that someone learns to install Linux, configs apache, and now things they can build a complex PBX with asterisk |
| 21:57.52 | drmessano | Give me a break |
| 21:58.02 | drmessano | thnks* |
| 21:58.06 | drmessano | thinks* |
| 21:58.09 | mvanbaak | erm |
| 21:58.21 | mvanbaak | pri etc is something totally different from ip based stuff |
| 21:58.31 | tzafrir_laptop | Configuring apache is actually quite complex, isn't it? |
| 21:58.35 | drmessano | It is? |
| 21:58.36 | mvanbaak | I do complex asterisk setups etc |
| 21:58.54 | adeel | tzafrir_laptop, not really |
| 21:58.56 | mvanbaak | but not with pri because I dont know about the protocols and dont want to learn it |
| 21:59.06 | mvanbaak | tzafrir_laptop: apache is easy |
| 21:59.13 | tzafrir_laptop | Most of us don't really see it because we don't build and configure apache |
| 21:59.22 | mvanbaak | actually, it's the second most simple service to setup |
| 21:59.27 | mvanbaak | bind is even easier |
| 21:59.33 | mvanbaak | but apache is close to that |
| 21:59.38 | tzafrir_laptop | Apache is a complex peice of software with many options and pitfalls |
| 21:59.41 | Guggemand | telnet is a lot easier :P |
| 21:59.53 | mvanbaak | Guggemand: uhhuh, you simply dont set it up ;) |
| 22:00.00 | DrukenLPY | anyone here know asp.net ?? i'm having some issues with it... (i don't know it at all) |
| 22:00.06 | Strom_M | ISDN is dead simple |
| 22:00.18 | mvanbaak | s/ simple// |
| 22:00.19 | tzafrir_laptop | DrukenLPY, http://asp.net/ ? |
| 22:00.20 | drmessano | The fact that Asterisk is software gives every wannabe the indication that it's just as easy as apache or PHP to set up |
| 22:00.38 | Strom_M | mvanbaak: don't be a biased prick |
| 22:00.39 | mvanbaak | drmessano: and it is |
| 22:00.57 | DrukenLPY | tzafrir_laptop: i don't want to learn it... just want a single file fixed... :) |
| 22:01.09 | mvanbaak | drmessano: as long as you read the docs and start with a testbed (same goes for apache, php, whatever_software> |
| 22:01.28 | DrukenLPY | and i'm not willing to pay the company that developed the software 80 an hour to do it for me... |
| 22:01.54 | mvanbaak | asterisk is not that hard to get running, just read the stuff in docs/ |
| 22:02.46 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 22:03.15 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:03.35 | tzafrir_laptop | whymarkwh, ping, cat /proc/zaptel/* |
| 22:04.10 | mvanbaak | cant wait for the confbridge stuff in file's branch tho |
| 22:04.18 | mvanbaak | no more need for zaptel timing to do conferencing |
| 22:05.48 | jaytee | tzafrir_laptop, he posted it way back. you must have missed it http://pastebin.com/m523aeb55 |
| 22:06.53 | tzafrir_laptop | ok, so his channels are in alarm, and thus Asterisk would not dial through them |
| 22:07.15 | tzafrir_laptop | (and the channels are in alarm because the span is in alarm) |
| 22:07.41 | tzafrir_laptop | What is blue alarm? |
| 22:07.53 | tzafrir_laptop | always thought blue to be a calm kind of color |
| 22:08.28 | mvanbaak | it means the span has drunk to much |
| 22:08.38 | tzafrir_laptop | and how can a span have both red and yellow alarm? |
| 22:08.42 | whymarkwh | lol |
| 22:09.06 | mvanbaak | hhmm, a span having two types of alarm ? |
| 22:09.10 | mvanbaak | that's ....... weird |
| 22:09.24 | *** join/#asterisk CVirus (n=GoD@196.205.192.18) |
| 22:10.03 | tzafrir_laptop | whymarkwh, do the spans still have alarms? |
| 22:11.10 | *** join/#asterisk kash (i=nibblet@24.182.112.23) |
| 22:11.25 | kash | hi, is there a way to echo the time asterisk thinks it is to the console? |
| 22:12.26 | whymarkwh | yes |
| 22:12.35 | kash | what application? |
| 22:13.39 | kash | oh |
| 22:13.40 | kash | found it. |
| 22:13.40 | kash | heh |
| 22:14.10 | *** join/#asterisk obnauticus (n=obnautic@c-71-193-156-193.hsd1.wa.comcast.net) |
| 22:15.44 | tzafrir_laptop | whymarkwh, so is it connected to the PSTN, or as a loopback? |
| 22:16.00 | kash | <PROTECTED> |
| 22:16.05 | kash | shouldn't that work? |
| 22:16.11 | tzafrir_laptop | If loopback, what do you have on /etc/zaptel.conf ? |
| 22:16.36 | kash | <PROTECTED> |
| 22:16.41 | kash | that's all i see in the console though |
| 22:17.04 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 22:17.04 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 22:17.09 | tzafrir_laptop | kash, there's also the function STRFTIME |
| 22:18.13 | *** join/#asterisk Vorbote (n=vorbote@unaffiliated/vorbote) |
| 22:20.48 | tzafrir_laptop | seanbright, build dependencies accross distributions? we already have a beginning of such thing (the prereq script) |
| 22:22.56 | seanbright | tzafrir_laptop: awesome, where can i grab a copy to have a look? |
| 22:23.16 | tzafrir_laptop | I think it is already included under contrib/scripts |
| 22:23.26 | seanbright | ohh |
| 22:23.29 | seanbright | you mean install_prereq |
| 22:23.32 | seanbright | gotcha |
| 22:23.34 | tzafrir_laptop | right |
| 22:23.58 | *** join/#asterisk LiNeTuX (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 22:24.16 | tzafrir_laptop | I don't know how to better integrate such information with autoconf |
| 22:24.28 | kash | i have another issue where i have two accounts on the same provider, different registrations for each one in sip.conf with different peer sections |
| 22:24.42 | kash | but both accounts go into the same context |
| 22:24.48 | kash | (it's not the default context, either) |
| 22:25.06 | kash | i mean, when i dial the two separate DIDs, they both go to the same context. |
| 22:25.18 | ManxPower | I'm not seeing the problem here. |
| 22:25.44 | kash | i want them in separate contexts. in the friend (sorry, not peer) section, they are configured to go to different contexts. |
| 22:26.02 | ManxPower | kash: then set different context= lines. |
| 22:26.06 | kash | i did |
| 22:26.19 | ManxPower | then the call coming does not match as you expect |
| 22:26.46 | kash | eh? |
| 22:27.45 | kash | i have two asterlink accounts -- one is set up to use the context sinplace-in, and the other is kash-in. both friends are searching in kash-in though. |
| 22:28.01 | kash | so the one from sinplace doesn't work unless i add its rules to the kash-in context |
| 22:29.53 | mvanbaak | kash: direct them both to [from-remote] and do the going ot seperate contexts there |
| 22:29.58 | mvanbaak | [from-remote] |
| 22:30.15 | mvanbaak | exten => kash-in,1,Goto(kash,incoming,1) |
| 22:30.37 | kash | k. |
| 22:30.38 | mvanbaak | exten => not_kash-in,1,Goto(not_kash,incoming,1) |
| 22:30.42 | mvanbaak | something like that |
| 22:30.50 | mvanbaak | that's how I do it on our hosted setups |
| 22:32.03 | kash | hmm. |
| 22:32.36 | kash | i changed the context on kash-in to something that doesn't exist, and called the DID from the other, and now it says extension not found on both |
| 22:32.36 | kash | heh |
| 22:32.51 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 22:32.53 | kash | so asterisk is basically ignoring the context for the one because they connect to the same server? |
| 22:33.29 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:35.14 | ManxPower | no, I suspect the ITSP is always sending the auth info the same no matter which DID is called. |
| 22:36.06 | ManxPower | On the other hand, maybe using type=friend is not what you want. |
| 22:36.28 | mvanbaak | just use type=peer |
| 22:36.33 | ManxPower | later versions of asterisk stupidly try to auth incoming type=friend connections against the host= setting |
| 22:36.46 | ManxPower | mvanbaak: no, type=user |
| 22:36.55 | ManxPower | you send calls TO peers, you accept them FROM users. |
| 22:37.04 | mvanbaak | gheh |
| 22:37.14 | ManxPower | so a peer would have host=dynamic or host=ip, but a user would never have those. |
| 22:38.28 | *** join/#asterisk [tasty]freeze (n=dmhardis@cs.roanoke.edu) |
| 22:38.41 | mvanbaak | ManxPower: I only have 'type=peer' in my sip.conf |
| 22:38.54 | ManxPower | mvanbaak: 1.4 stupidly tried to merge them |
| 22:39.08 | [tasty]freeze | I am getting an error that registration failed because of an ACL error. Where are the access control settings located. |
| 22:39.11 | mvanbaak | 2 xs4all itsp connections, one speakup connection and one budgetphone |
| 22:39.13 | mvanbaak | all itsp |
| 22:39.19 | [tasty]freeze | I didn't realize I had any. |
| 22:39.21 | ManxPower | [tasty]freeze: and the ACTUAL error is? |
| 22:39.24 | mvanbaak | ManxPower: trunk here |
| 22:39.43 | ManxPower | mvanbaak: exactly, so the host= makes it act like friend rather than peer |
| 22:39.45 | [tasty]freeze | NOTICE[13777] ACL error (permit/deny) |
| 22:39.53 | ManxPower | [tasty]freeze: so fix it. |
| 22:40.02 | [tasty]freeze | alright. |
| 22:40.22 | ManxPower | too bad it does not say WHICH config file the problem is in. |
| 22:40.35 | ManxPower | I'd check manager.conf as well as sip.conf and iax.conf if you are using it. |
| 22:40.39 | mvanbaak | ManxPower: hhmm, I thougth oej killed the 'type=user' |
| 22:40.43 | kash | ManxPower, i'm confused. outgoing is working because of the register => line, right? |
| 22:40.55 | ManxPower | kash: no. |
| 22:41.00 | mvanbaak | kash: no, incoming will work because of register=> lines |
| 22:41.20 | kash | mvanbaak, so if I switched to type=user, then outgoing won't work? |
| 22:41.26 | ManxPower | ALL a register does is tell the remote server what user/pass/(optional extension) is associated which which IP address. It does nothng |
| 22:41.32 | mvanbaak | kash: a register line makes your asterisk register with the remote box, so the remote box knows what ip address to route the calls to when they are for you |
| 22:41.33 | ManxPower | kash: I said to split them. |
| 22:41.43 | ManxPower | put your sip.conf on pastebin.ca and I'll look at it. |
| 22:42.22 | ManxPower | at least for testing, just comment out the host= lines for that entry and reload and try, if that fixes it then it's the "host= is overloaded in the code" problem |
| 22:44.18 | rootlogin | i would like to use Playtones() in my dialzlan but there is no such application .. is that depricated ? do i need to compile additional sources ? |
| 22:44.38 | kash | can't i specify a context in the register line? |
| 22:46.04 | rootlogin | kash: afaik no .. only in the incoming [xxxx] block i think |
| 22:46.45 | ManxPower | kash: since the SIP protocol has no concept of context, no you cannot. |
| 22:47.00 | ManxPower | asterisk sets the context when it gets a matching incoming call for that user/passs |
| 22:47.30 | kash | hmm. |
| 22:47.51 | kash | what does fromuser do, exactly |
| 22:49.38 | kash | i think it may have to do with insecure=port,invite on both the friend sections |
| 22:49.46 | kash | oh well, i'll just do what mvanbaak said |
| 22:50.13 | *** join/#asterisk Blacraft (n=bla@203.26.40.67) |
| 22:50.49 | Blacraft | Hello everyone just after a little help getting my TDM400P FXS ports to work |
| 22:51.49 | ManxPower | fromuser only applies to outgoing connection |
| 22:52.12 | ManxPower | Blacraft: does ztcfg -vvv generate any errors? |
| 22:52.20 | Blacraft | nope |
| 22:52.40 | ManxPower | good |
| 22:52.57 | Blacraft | i can sort of ring the fxs ports from my voip phone, though the ring is wierd, like there isn't enough power getting to the phone. |
| 22:53.13 | Blacraft | the power is connected, i have changed PSU also |
| 22:53.14 | ManxPower | you have the power connected to the TDM card? |
| 22:53.35 | ManxPower | Have you tried a different phone? What country are you in? |
| 22:53.56 | Blacraft | yeah tried different phone. australia |
| 22:54.04 | ManxPower | do you get dialtone when you pick up the phone on the FXS port? |
| 22:54.32 | Blacraft | no dial tone, though one of the ports get a dial tone for a minute or so after i ring it with the voip phone |
| 22:55.35 | ManxPower | Blacraft: I have no idea what could cause the problem. Get dialtone working on the phone and I suspect all your other issues will go away |
| 22:55.56 | ManxPower | when the fxs phone goes off hook, do you see it in the CLI? |
| 22:56.07 | Blacraft | yeah well thats my problem :), no dial tone. |
| 22:56.22 | Blacraft | nothing happens in t he CLI when i have no dial tone |
| 22:56.42 | ManxPower | Blacraft: put /etc/zaptel.conf and /etc/asterisk/zapata.conf on pastebin.ca |
| 22:56.55 | ManxPower | Then give us the resulting URL(s). |
| 22:56.59 | Blacraft | ok |
| 22:57.07 | ManxPower | ~pb |
| 22:57.07 | 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 |
| 22:59.00 | s0lid | how do i tell asterisk to do outbound call only within this time period? |
| 23:00.26 | Blacraft | http://pastebin.com/d6abbddd4 |
| 23:00.52 | Blacraft | ManxPower http://pastebin.com/d6abbddd4 |
| 23:07.36 | rootlogin | i fetched * 1.4 from SVN and i cant use/find the application Playtones() anywhere .. its mentioned/called in the source and milliwatt() complains that its missing .. where is this app supposed to be ? |
| 23:11.34 | ManxPower | looking at it now |
| 23:11.48 | Blacraft | thx |
| 23:12.21 | ManxPower | Blacraft: I really can't help you with GUI installs of Asterisk |
| 23:13.24 | Blacraft | nps, could it be something with Trixbox |
| 23:13.32 | ManxPower | It always is. |
| 23:13.40 | ManxPower | ~trixbox |
| 23:13.41 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 23:14.11 | Blacraft | so you think the card it's self would be fine |
| 23:14.26 | jaytee | try sacrificing a cow to the "Space God Karnak" and pray alot. |
| 23:15.22 | ManxPower | Blacraft: I can't tell. |
| 23:15.39 | ManxPower | Try using the correct support forums for the product you are using. |
| 23:16.58 | jaytee | I'm gonna write a truly skank-i-fying version of * with a nasty girly-man GUI with pin-striping and all kinds of frilly edges and package it as "SkuzVox" |
| 23:17.24 | ManxPower | jaytee: Someone already did that, it's called "AMP" |
| 23:17.30 | jaytee | lol |
| 23:18.28 | ManxPower | The Window Icon Mouse Pointer System (WIMPS) |
| 23:18.44 | jaytee | I've gotta go place my lottery bet for the night |
| 23:18.58 | ManxPower | You're bad at math? |
| 23:20.18 | jaytee | all I can picture in my head is the scene from "In and Out" where the supermodel girlfriend of Matt Dillon is having a panic attack because she can't figure out how to dial a rotary phone to call her agent. |
| 23:21.54 | jaytee | ManxPower, bad at math? mmmm, no not really. just full of whimsy and a belief in serendipity. |
| 23:23.53 | *** join/#asterisk LiNeTuX (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 23:25.38 | ManxPower | The lottery is a tax on people that are bad at math. |
| 23:26.32 | Blacraft | a lottery is designed so people good at math will never have a chance of winning it |
| 23:27.12 | jaytee | "We're counting cards." - Dustin Hoffman |
| 23:28.36 | rootlogin | ManxPower: lol :) thats right |
| 23:29.10 | jaytee | I really don't mind pissing away 2 bucks on the idea of "gee, whiz. Maybe lightening will strike and I'll be able to retire on Monday" |
| 23:29.24 | jaytee | when the odds are so far against it |
| 23:32.01 | *** part/#asterisk Blacraft (n=bla@203.26.40.67) |
| 23:35.27 | *** join/#asterisk LapTop006 (n=laptop00@gemini.chriskaine.com.au) |
| 23:36.10 | LiNeTuX | jaytee: that only works in Florida (lightening capital of the world) |
| 23:36.12 | adeel | jaytee, you might as well bury that 2 bucks into the ground and see if money really does grow on trees |
| 23:47.37 | *** join/#asterisk asdx (n=diego@adsl-139-2.click.com.py) |
| 23:52.33 | *** join/#asterisk Entranced (n=entrance@191.23.119.70.cfl.res.rr.com) |
| 23:58.09 | drmessano | I'd rather spend two dollars on the lottery than hold out all hope I will never be rich enough to retire from IRC |
| 23:58.24 | drmessano | Because I dont know about you, but if I get rich, someone else is building my PBXs |
| 23:58.30 | drmessano | The heck with you guys |
| 23:59.35 | drmessano | I'll pay ManxPower to be my PBXbitch.. I'll even give him 1.5 times his hourly rate... But I get priority 24/7 on-call |
