| 00:07.22 | ManxPower | c|oneman: If there IS a timeout, it is in the phone, not Asterisk |
| 00:13.53 | TrentCreek | I wonder what is afflicting all the RF spectrum today. WiFi is crap as well as digital TV signals |
| 00:15.30 | jblack | There are short heavy storms on the east coast. |
| 00:15.40 | jblack | I wouldn't be surprised if satellite uplinks are affected |
| 00:18.19 | *** join/#asterisk nobesnickr (n=pmccaffr@ip72-201-157-30.ph.ph.cox.net) |
| 00:19.46 | nobesnickr | i seem to be having some very stressful issues with my Cisco 7940 and could really use some help. Everything seemed to be working perfectly yesterday but for some reason today my phone WILL NOT register to my asterisk server. I still get notify packets telling my I have voicemails but the phone will not register to make or receive calls |
| 00:20.01 | nobesnickr | when I do a SIP debug i get a 401 unauthorized error |
| 00:20.02 | TrentCreek | I am in deep south Texas..all sunny |
| 00:20.29 | nobesnickr | can anyone point me in what direction I should be looking please |
| 00:20.34 | TrentCreek | And I don;t see how satellite would afflict WiFi and local digital TV signals |
| 00:21.35 | *** part/#asterisk ejbvanc (n=eric@c-24-21-78-0.hsd1.mn.comcast.net) |
| 00:22.30 | nobesnickr | anyone? |
| 00:24.04 | [TK]D-Fender | nobesnickr, 401 = bad user/pass. You've mees one or the other up. |
| 00:25.20 | nobesnickr | I thought that also but the phone is set for two different "lines" both use different user/pass |
| 00:25.29 | nobesnickr | and I checked them against the sip.conf and they are the same |
| 00:26.37 | nobesnickr | i am behind a NAT but have nat=yes set in my sip.conf |
| 00:27.01 | nobesnickr | the weird thing is, when I run sip show peers it says NAT N by all of the peers I have set up |
| 00:27.12 | seanbright | yeah |
| 00:27.16 | seanbright | NAT N means nat |
| 00:27.19 | seanbright | not "no" |
| 00:27.23 | nobesnickr | o lol, ok |
| 00:27.32 | nobesnickr | well that is good at least |
| 00:28.22 | seanbright | (it was a silly design choice, imho) |
| 00:28.43 | nobesnickr | i agree but at least now I know its work lol |
| 00:29.13 | seanbright | yeah |
| 00:29.34 | hsv-al | bobf, you can do it by: redent bin(i): ret = []; while i: ret.append(str(i & 1)); i >>= 1;; return ''.join(reversed(ret)) |
| 00:29.43 | hsv-al | oops |
| 00:31.24 | seanbright | is that rubt? |
| 00:31.30 | seanbright | s/rubt/ruby/ |
| 00:31.36 | seanbright | or python? |
| 00:31.42 | hsv-al | .py |
| 00:32.01 | seanbright | interesting. |
| 00:32.18 | hsv-al | python is useful |
| 00:33.00 | hsv-al | seanbright, if you are looking to start: |
| 00:33.01 | hsv-al | http://mail.python.org/pipermail/tutor/2001-January/003337.html |
| 00:33.07 | hsv-al | http://www.python.org/doc/Intros.html |
| 00:33.42 | hsv-al | they updated the beginner section, invalid url, replace with: http://www.python.org/doc/ |
| 00:33.47 | seanbright | hsv-al: i have an o'reilly book on the topic |
| 00:33.53 | seanbright | hsv-al: saving it for a rainy day |
| 00:33.57 | hsv-al | thats all you need then |
| 00:34.08 | nobesnickr | does anyone have any ideas? |
| 00:34.46 | hsv-al | [07:19pm] [nobesnickr] i seem to be having some very stressful issues with my Cisco 7940 and could really use some help. Everything seemed to be working perfectly yesterday but for some reason today my phone WILL NOT register to my asterisk server. I still get notify packets telling my I have voicemails but the phone will not register to make or receive calls |
| 00:35.22 | seanbright | doesn't know |
| 00:35.39 | hsv-al | nobesnickr, start herE: |
| 00:35.40 | hsv-al | http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx |
| 00:35.59 | nobesnickr | i have that page dang near memorized |
| 00:36.02 | nobesnickr | its just this phone |
| 00:36.18 | nobesnickr | that is the weirdest part, i have other 7940s working great |
| 00:36.44 | TrentCreek | Maybe this is why I am getting bad RF signals here |
| 00:36.48 | TrentCreek | "Recent Conditions: Geophysical Activity Summary 30/2100Z to 31/2100Z : The geomagnetic field was quiet to unsettled. Solar wind velocities have remained elevated with speeds between 550 - 600 km/s. |
| 00:36.48 | TrentCreek | " |
| 00:36.50 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-7-156.lns10.syd7.internode.on.net) |
| 00:41.11 | TrentCreek | X-ray Solar Flares |
| 00:41.11 | TrentCreek | 6-hr max: A0 2355 UT May31 |
| 00:41.11 | TrentCreek | 24-hr: A0 2355 UT May31 |
| 00:42.33 | TrentCreek | A solar wind stream flowing from the indicated coronal hole will reach Earth on or about June 1st. |
| 00:44.50 | nobesnickr | i am checking the debug and i dont see any packets coming from the phone anymore |
| 00:44.57 | nobesnickr | only going to the phone from asterisk |
| 00:58.22 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:58.22 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 01:44.31 | Twister | is it possible to sign up for iaxtel anymore? |
| 01:45.05 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 01:46.47 | hsv-al | just found a good book series |
| 01:46.50 | hsv-al | http://www-cs-staff.stanford.edu/~uno/taocp.html |
| 01:47.03 | hsv-al | http://www.amazon.com/Art-Computer-Programming-Volumes-Boxed/dp/0201485419 |
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| 02:02.19 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 02:03.05 | hsv-al | .Kpom $[rt; |
| 02:04.47 | *** join/#asterisk paci`` (n=pacifico@cpe-075-182-072-065.nc.res.rr.com) |
| 02:04.49 | paci`` | anyone around? |
| 02:04.55 | paci`` | having some trouble compiling app_conference |
| 02:04.59 | paci`` | ../../../usr/asterisk/include/asterisk/utils.h:278: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?void? |
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| 02:25.24 | Strom_C | paci``: why not just use meetme? |
| 02:25.44 | paci`` | Strom_C, has some wierd problem getting the kernel source working with the zaptel dummy |
| 02:27.19 | Strom_C | is that even being developed still? |
| 02:27.21 | Strom_C | looks |
| 02:27.33 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 02:31.59 | Strom_C | paci``: the documentation provides no indication of which asterisk version app_conference is designed to work with |
| 02:32.30 | Strom_C | you're probably better off trying to solve your zaptel problem |
| 02:34.00 | paci`` | I got it working, it was because I was using beta |
| 02:34.20 | Strom_C | ok |
| 02:35.31 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
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| 03:09.13 | *** join/#asterisk Amun (n=Jonsi@c-71-205-221-174.hsd1.mi.comcast.net) |
| 03:11.01 | Amun | just a quick question, what would i need to run asterisk as a incoming/outgoing phone with all the standard features? I have a old 2.4ghz pc laying around. Do i need any PCI devices, a pay-for account with any companies, etc etc etc, or can i do this for free (absolutely no cost, just the pc and some manual labor) ? |
| 03:11.28 | Amun | also note, i have VOIP through comcast. i dont know if its possible to connect it to that or not |
| 03:13.36 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 03:13.55 | mackes | Disco |
| 03:14.03 | Strom_M | fever |
| 03:14.03 | LiNeTuX | sux |
| 03:14.10 | Strom_M | inferno? |
| 03:14.38 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
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| 03:15.42 | Strom_M | amun: you can either buy an fxo/fxs card or get a paid itsp accojnt |
| 03:16.20 | Strom_M | s/jn/un/ |
| 03:16.26 | [TK]D-Fender | Amun, You need hardware only if you want to interface with physical PSTN lines you have. |
| 03:16.49 | [TK]D-Fender | Amun, PCI cards are 1 solutions, and there are other gateway devices as well |
| 03:17.00 | [TK]D-Fender | Amun, Or you can get PSTN access via an ITSP |
| 03:17.01 | LiNeTuX | or that flux capacitor in the garage |
| 03:17.05 | [TK]D-Fender | ~itsp |
| 03:17.07 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 03:17.37 | [TK]D-Fender | Amun, that requires no special hardware and is a service you access over the internet. |
| 03:23.33 | Amun | hrmm |
| 03:23.51 | Amun | ~itsplist-us |
| 03:23.51 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 03:25.24 | coppice | flux capacitors are a fake. you need an interociter |
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| 03:27.53 | *** join/#asterisk paulproteus (n=paulprot@wide-rose.makesad.us) |
| 03:29.01 | Amun | ok, thanks for the information you guys, i'm definitely new to all this. |
| 03:29.10 | Amun | if you could answer a few more easy questions ;) |
| 03:31.35 | Amun | in a estimate, say i talk on the phone for, say, an hour a day. only 1 phone line with voicemail, no internal calling since its going to be a house phone, how much do you think it would cost us, and what company should I go through? most of our calls are long distance, but usually in the united states, and rarely out of our own state (michigan). |
| 03:33.04 | Amun | and question #2 is, am I able to set up asterisk to have its own voicemail, without using another company? and telephone prompts (theres a few people who live with me, it would be cool to have something like 'if your wishing to leave a message for blah, press 1, etc etc), and caller-id spoofing? is that possible with asterisk (a feature i'd use like... once) |
| 03:33.42 | *** join/#asterisk simprix (n=simprix@c-24-11-30-113.hsd1.mi.comcast.net) |
| 03:34.10 | [TK]D-Fender | Amun, 1hr /day * 30 = 1800 minutes @ $.02c/min = 36$ for your typical per-minute service |
| 03:34.26 | [TK]D-Fender | Amun, you can get an unlimited account for nearly half that much. |
| 03:34.43 | [TK]D-Fender | Amun, and to your last question, yes, entirely |
| 03:34.54 | Amun | what company would you suggest? |
| 03:35.17 | [TK]D-Fender | Amun, shop around for rates and see whos close, then ask between them. |
| 03:37.26 | Amun | well, the terminology is new to me, and i dont understand the difference between internal outbound and local outbound ;x |
| 03:37.32 | Amun | https://www.teliax.com/plans/5? |
| 03:37.37 | Amun | i've never heard of MANY of these terms |
| 03:38.22 | [TK]D-Fender | inbound is just that. incoming calls to a # they provide for you are free |
| 03:38.56 | Amun | but domestic and local? |
| 03:38.58 | [TK]D-Fender | local outbound is whatever is considered "local" based on the # you acquire from them. |
| 03:39.16 | [TK]D-Fender | just do the math from there... its pretty obvious. |
| 03:39.27 | [TK]D-Fender | if you understanda the concetp of LD phone calls :) |
| 03:40.07 | simprix | Does anyone know of any companies that provide sip trunks for mexico that will terminate in the US |
| 03:40.53 | [TK]D-Fender | simprix, that doesn't add up... try again... |
| 03:41.16 | jameswf-home | WOW judt got the kids a wii now we need wii points to buy an internet browser |
| 03:41.42 | *** join/#asterisk techie (n=techie@adsl-76-214-14-114.dsl.lsan03.sbcglobal.net) |
| 03:42.27 | simprix | I want to terminate some mexico numbers via sip trunks into a phone system that is in the us. |
| 03:43.06 | [TK]D-Fender | simprix, most itsp's will let you call other countries... |
| 03:43.41 | simprix | Right. Im just looking for a company that can provision mexican numbers on sip trunks. |
| 03:44.41 | Amun | [TK]D-Fender: thanks for the help. 1 last (hopefully) question. is there a service that you know of that just gives me basic phone use, so i can let asterisk handle the rest? seems like all the 'features' that these voip providers give you is useless since asterisk can do it all |
| 03:45.01 | [TK]D-Fender | simprix, ok, you are reversing your terminology. Lets see if I got this right : you want to get a MEXICAN DID delivered via SIP? |
| 03:45.47 | [TK]D-Fender | Amun, you don't need to use them for the mostpart if you don't need to, and simply ignore them. |
| 03:45.53 | [TK]D-Fender | Amun, http://www.broadvoice.com/rateplans_unlimited_us.html might be right for you. |
| 03:46.04 | Amun | we think alike. im on tht page right now. |
| 03:46.14 | simprix | [TK]D-Fender: yes |
| 03:46.54 | [TK]D-Fender | simprix, go look on the WIKI to see who's listed for Mexico. |
| 03:46.56 | [TK]D-Fender | ~wikis |
| 03:46.57 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 03:47.19 | Amun | now, for the more advanced stuff... whats SIP and whats PSTN ? |
| 03:47.31 | [TK]D-Fender | ~sip |
| 03:47.32 | jbot | it has been said that sip is http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 03:47.34 | [TK]D-Fender | ~pstn |
| 03:47.34 | jbot | rumour has it, pstn is Public Switched Telephone Network, or "please stop the nonsense" |
| 03:47.57 | [TK]D-Fender | Amun, time for you to stop and read THE BOOK. |
| 03:47.58 | [TK]D-Fender | ~book |
| 03:47.59 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:48.00 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 03:48.20 | [TK]D-Fender | Amun, it has a good intro chapter explaining in brief most forms of telephony |
| 03:48.28 | Amun | awesome. im gonna spend tonight reading it |
| 03:48.39 | Amun | im sick of paying comcast 70 bucks a month for phone service. |
| 03:48.45 | Amun | and im sure thats why all you guys are here as well ;p |
| 03:49.40 | [TK]D-Fender | Amun, something like that. |
| 03:49.55 | [TK]D-Fender | Amun, For some its about control, not cost. * is different things to different people. |
| 03:50.28 | Amun | well, thats why im interested in trying it out. flexibility + no money-hungry companies telling me what i can and cant have = win |
| 03:53.25 | [TK]D-Fender | Amun, Ok, so get reading, and enjoy. |
| 03:53.40 | *** join/#asterisk BBHoss_Laptop (n=hoss@12.180.45.140) |
| 03:53.47 | Amun | im reading, and thanks. your advice and answers have helped tremendously. |
| 03:53.58 | BBHoss_Laptop | hey whats the cheapest rate for toll free origination that you guys have seen |
| 03:59.43 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-239-103.balt.east.verizon.net) |
| 04:02.51 | hsv-al | I said, what what,itb |
| 04:06.05 | adeel | is echo cancelling done on an incoming zap -> sio call? |
| 04:06.08 | adeel | er sip |
| 04:06.43 | BBHoss_Laptop | yeah |
| 04:07.42 | adeel | hmmm...i just setup oslec, and i don't see it running on an incoming zap call |
| 04:07.58 | adeel | if i dial out through an fxs line, i see it working |
| 04:08.42 | [TK]D-Fender | adeel, you enable it per channel, and * decides if its needed per channel |
| 04:09.24 | adeel | [TK]D-Fender, i thought setting echocancel=yes globally would enable it per channel |
| 04:09.29 | adeel | and how does * decide if it's needed? |
| 04:09.34 | [TK]D-Fender | adeel, yup, should |
| 04:09.57 | [TK]D-Fender | adeel, * tests to see if there is echo to be cancelled |
| 04:10.14 | [TK]D-Fender | adeel, or more precisely, you EC routine |
| 04:11.13 | adeel | my EC is set to OSLEC...i see the confirmation when i load the zaptel module, but if i do a 'watch cat /proc/oslec/info' and make an incoming call to the zap trunk, the only output i see is 'no echo canceller being monitored - make a new call' |
| 04:12.47 | [TK]D-Fender | adeel, Sorry, can't tell you anything about making thta work specifically. |
| 04:13.20 | adeel | [TK]D-Fender, i appreciate it anyway....i think i'll check the mailing lists and post there |
| 04:18.02 | Yosam | <PROTECTED> |
| 04:18.58 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:31.20 | jblack | Now I want to kill. |
| 04:31.25 | jblack | slay yosam |
| 04:32.15 | Yosam | :( |
| 04:32.38 | Yosam | slaps jblack around a bit with a large trout |
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| 04:43.43 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 04:43.46 | Kernel_Core | hi all |
| 04:44.57 | Kernel_Core | hi all |
| 04:45.33 | BBHoss_Laptop | sup dog |
| 04:46.00 | adeel | interesting, * does not EC when the call is ZAP-->VM |
| 04:53.24 | *** join/#asterisk tekryan (n=anon@CPE-65-25-169-157.wi.res.rr.com) |
| 04:57.41 | Kernel_Core | imagine , I Have 4 port Analog FXO Card ( TDM400p ), I added channel 1-4 in group 0 (Dial 1, Zap/g0/${EXTEN}) , when I want to call out ( my first and second lines , 1,2 phone lines arenot working) , Asterisk tries the first line , instead of trying the working line ? |
| 04:58.29 | Kernel_Core | Channel 1 and 2 are Onhook ! |
| 04:58.37 | Kernel_Core | (zap show channel 1 , 2 |
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| 05:07.36 | jameswf-home | HOLY CRAP i just did like 12 rounds of wii boxing... I am way to fat for all of that... |
| 05:15.27 | Amun | jameswf-home: nice. when i did that, my arms burned for days. im skinny as fuck too |
| 05:16.33 | jameswf-home | thinks people should skip the gym memberships and spend 250 on one ofthese |
| 05:22.37 | JT | my legs are pretty sore |
| 05:22.45 | JT | walked maybe 50km in the last 2 days |
| 05:28.15 | Yosam | ok |
| 05:28.24 | Yosam | i have an audio file and a guy is punching in his number |
| 05:28.41 | Yosam | saying that i should call him back :S. how can i get his number. |
| 05:28.59 | Yosam | i know each dtmf sound has different frequencies but |
| 05:31.02 | florz | Yosam: have a look at multimon |
| 05:31.33 | Yosam | multimon? |
| 05:31.57 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 05:34.20 | [TK]D-Fender | Yosam, Feed the recording into an ivr to catch the digits |
| 05:34.23 | JT | wintone or similar |
| 05:35.13 | Kernel_Core | hi [TK]D-Fender |
| 05:36.33 | [TK]D-Fender | JT : * can do it for him |
| 05:38.06 | Kernel_Core | [TK]D-Fender: I have TDM400P (4FXO ) ,I defined channel 1-4 in group 0,the phone lines of first and second channel , zap1,zap2 isnot connected , but asterisk sends the call first to the zap1 which doesn't have phone line , and in remote side it rings ! |
| 05:38.35 | Kernel_Core | status of zap1 and zap2 in asterisk -rx zap show channel 1 2 is "ON-Hook" |
| 05:39.04 | [TK]D-Fender | Kernel_Core, * doesnt know any better. YOU put it in that group. If you don't want to ti use that line the go remove it from the group. |
| 05:40.18 | Kernel_Core | [TK]D-Fender: is there any channellimit option in iax2 ? |
| 05:40.36 | [TK]D-Fender | Kernel_Core, Go look on the WIKI page for it. |
| 05:40.47 | Kernel_Core | :D there isn't |
| 05:41.00 | Kernel_Core | even in iax.conf isnot available... |
| 05:43.02 | [TK]D-Fender | Kernel_Core, then maybe the answer is "no". |
| 05:47.50 | adeel | JT, walked 50 km in 2 days? where do you live? africa? |
| 05:48.37 | JT | australia, i'm interstate and don't have my car with me, and you can't drive to all places, if you know what i mean |
| 05:49.49 | adeel | that'll explain it |
| 05:50.30 | JT | went exploring to a few places that don't have car access anyway |
| 05:50.52 | JT | and the structure of the walking surface also caused additional pain in legs :) |
| 05:51.09 | adeel | having an ATV or dirt bike would probably be the best form of transportation in the bush |
| 05:53.31 | JT | most of this was less than 10km from the CBD |
| 05:53.35 | JT | think down |
| 05:57.32 | *** part/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 06:02.40 | jameswf-home | 3 games of bowling this wii thing is more adicting than the crackberry |
| 06:03.19 | JT | doesn't the crackberry correct adicting to addictive? ;) |
| 06:03.43 | jameswf-home | yes but I am on the korean laptop |
| 06:04.06 | JT | ah |
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| 06:24.57 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 06:25.58 | Kernel_Core | how do I check dialtone before dialing in asterisk ( Zap channels ) ? |
| 06:29.56 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 06:34.05 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 06:39.37 | *** join/#asterisk nobesnickr (n=pmccaffr@ip72-201-157-30.ph.ph.cox.net) |
| 06:53.24 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 07:00.26 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 07:05.23 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 07:10.41 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:12.55 | *** join/#asterisk kamh (n=q@host-81-190-236-85.wroclaw.mm.pl) |
| 07:19.39 | *** join/#asterisk xnixan (n=xnixan@196.218.222.117) |
| 07:20.00 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 07:36.30 | *** join/#asterisk ikaRus1 (n=none@80.179.36.48.static.012.net.il) |
| 07:36.51 | ikaRus1 | i just installed asterisk 1.6 latest beta, i dont see my machine listening on port 5038, asterisk -vvvvR works like expected |
| 07:36.58 | ikaRus1 | what am i missing?? |
| 07:45.09 | jblack | 5038? |
| 07:45.49 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 07:46.31 | jblack | Oh, there's 5038 |
| 07:47.18 | jblack | 5038 is the call manager. You should be able to set it up in /etc/asterisk/manager.conf |
| 07:57.57 | *** join/#asterisk dachary (n=loic@magenta.dachary.org) |
| 07:59.50 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 08:01.26 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 08:03.29 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 08:13.26 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:19.40 | ikaRus1 | it is set up there. |
| 08:20.08 | ikaRus1 | ; |
| 08:20.08 | ikaRus1 | ; Asterisk Call Management support |
| 08:20.08 | ikaRus1 | ; |
| 08:20.08 | ikaRus1 | [general] |
| 08:20.08 | ikaRus1 | enabled = yes |
| 08:20.09 | ikaRus1 | port = 5038 |
| 08:20.11 | ikaRus1 | bindaddr = 0.0.0.0 |
| 08:20.22 | ikaRus1 | from /etc/asterisk/manager.conf |
| 08:20.48 | ikaRus1 | netstat -napt does now show it listening |
| 08:22.25 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 08:25.10 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 08:33.39 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 09:12.47 | *** join/#asterisk stuffcorpse (n=pickle@60-234-225-122.bitstream.orcon.net.nz) |
| 09:15.47 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 09:29.58 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 09:43.14 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 09:47.05 | *** part/#asterisk stuffcorpse (n=pickle@60-234-225-122.bitstream.orcon.net.nz) |
| 09:50.39 | dachary | http://www.zultys.com/index.jsp?tab=productdetail&product=wip2&detail=datasheet-wip2&type=phones |
| 09:51.28 | dachary | anyone managed to get the sources for this Linux kernel ? They don't seem to offer downloading or sending them by mail. |
| 09:53.24 | Strom_C | (a) what about the "software" link at the bottom, and (b) have you tried asking them? |
| 09:53.48 | coppice | zultys, and the other company started by the same people, don't seem to play nicely |
| 10:00.55 | tzafrir_laptop | dachary, kernel.org? |
| 10:01.07 | *** join/#asterisk Ipkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 10:01.15 | Ipkaf | hi |
| 10:02.01 | drmessano | Ipkaf |
| 10:02.06 | drmessano | Download this please: |
| 10:02.09 | drmessano | ~book |
| 10:02.10 | 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 |
| 10:02.14 | drmessano | and do some reading |
| 10:02.49 | Ipkaf | thx a lot drmessano |
| 10:03.09 | drmessano | Don't thank me, just do it.. |
| 10:03.53 | tzafrir_laptop | dachary, though if you asked this seriously, kernel.org is not the place you should download from |
| 10:04.22 | Ipkaf | i just have a stupid question |
| 10:04.39 | Ipkaf | i got sipura 3000 |
| 10:04.46 | drmessano | Ipkaf: You've been asking stupid questions for a month.. Please do some reading |
| 10:04.57 | Ipkaf | ok |
| 10:05.02 | Ipkaf | i will |
| 10:05.12 | Ipkaf | i got sipura 3000 |
| 10:05.53 | Ipkaf | and also i got asterisk server and analog telephone line |
| 10:05.55 | drmessano | I guess not.. |
| 10:06.11 | Ipkaf | i got |
| 10:06.29 | coppice | dachary: I've heard various complaints about zultys and software, both before the went bankrupt and after. i guess the new owners didn't change anything. zed-3 is a new company started by the founder of zultys. they seems quite similar |
| 10:06.32 | dachary | tzafrir_laptop: the kernel is most certainly modified. I want to distribute the phone but I'm not allowed to do so if I don't distribute the corresponding sources (not a random linux kernel source, the *corresponding* source). See the bad spot I'm in ? This is very frustrating. |
| 10:06.33 | drmessano | You're also running FreePBX, which isn't supported here |
| 10:06.36 | *** join/#asterisk LuisTorres (n=chatzill@bl6-197-187.dsl.telepac.pt) |
| 10:07.10 | drmessano | and no one is gonna help you in FreePBX if you don't start reading and learning and stop sucking the life out of everyone that is dumb enough to start helping you |
| 10:07.51 | Ipkaf | listen one thing |
| 10:07.57 | dachary | coppice: if they just did not pay attention it may be easy to get the sources from them. If they are deliberately hiding the sources it may be more difficult. Any article about this ? |
| 10:08.11 | Ipkaf | if u don't want to help me u can it's ur liberty |
| 10:08.14 | tzafrir_laptop | dachary, ok, sorry for misreading your question :-) |
| 10:08.26 | Ipkaf | okay ?? so stop saying something dirty |
| 10:08.56 | dachary | tzafrir_laptop: I was not spefic enough. I guess a simple phone call to the right person could solve it all. |
| 10:08.57 | dachary | bbl |
| 10:09.08 | coppice | dachary: why not just distribute something else? |
| 10:10.28 | dachary | because this one is known to work |
| 10:11.04 | tzafrir_laptop | dachary, search for "gpl-violations.org" |
| 10:11.14 | dachary | Hitachi-Cable Wireless IPC-5000AE |
| 10:11.24 | dachary | might be an alternative but is not out yet |
| 10:11.32 | tzafrir_laptop | It has been mentioned in the press often enough if you're looking for articles |
| 10:11.34 | dachary | tzafrir_laptop: asked already |
| 10:12.12 | dachary | that's not good news. mentionned often tend to point to : intentional lack of distribution :-( |
| 10:12.15 | dachary | bbl |
| 10:27.35 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:34.46 | *** join/#asterisk s0lid (n=s0lid@58.69.1.43) |
| 10:40.13 | dachary | tzafrir_laptop: coppice do you have specific links to articles explaining the status of the GPL with regard to the Zultys phones ? |
| 10:40.57 | tzafrir_laptop | dachary, How can you tell it uses Linux (the kernel)? |
| 10:41.02 | coppice | not right now. a while ago I saw various complaints while looking for stuff about zed-3 |
| 10:41.07 | Kernel_Core | hi all |
| 10:41.09 | dachary | *they* tell on the page :-) |
| 10:41.13 | tzafrir_laptop | What about some userspace programs? (e.g: busybox)? |
| 10:41.31 | coppice | both zultys and zed-3 make a big thing about them being linux based |
| 10:42.11 | tzafrir_laptop | Kernel_Core is linux-based as well |
| 10:42.14 | dachary | hopes it's just a matter of asking them and they will say : "oh, sorry, we did not realize we had to do this, here it's done!" :-) |
| 10:43.27 | Kernel_Core | is there anyway to check dialtone before ZAP dials ? sometimes my lines arenot connected to TDM400 , and when I want to dialout , it tries to dial over the ZAP channel which isnot coonected to TDM400 ! |
| 10:43.56 | tzafrir_laptop | Kernel_Core, definetly maybe |
| 10:44.11 | Strom_L | Kernel_Core: why are you disconnecting lines from your tdm400? |
| 10:44.20 | tzafrir_laptop | someone wrote a patch to do that. It should work. At least in some cases |
| 10:44.30 | dachary | zultys.com: searching for linux : 49 pages, searching for gpl : 0 pages |
| 10:44.34 | Kernel_Core | Strom_L: sometimes happens, |
| 10:44.52 | Kernel_Core | tzafrir_laptop: where do I get that patch ? |
| 10:45.11 | tzafrir_laptop | Kernel_Core, hmm.. not connected at all? (FXO port) |
| 10:45.50 | tzafrir_laptop | wctdm in recent Zaptel will set the channel to be in alarm, and then you won't be able to dial through it |
| 10:46.02 | tzafrir_laptop | Or, if it is in a group, it will be skipped |
| 10:46.10 | Kernel_Core | tzafrir_laptop: port 1-3 isnot connected but 2-4 are connected , I am looking for a solution , when Line 1 isnot connected in Dial Application , then use the second line and//// |
| 10:46.30 | Kernel_Core | I am useing 1.4.10 |
| 10:46.47 | Kernel_Core | with 1.4.20.1 asterisk |
| 10:48.32 | tzafrir_laptop | 1.4.10 should have it |
| 10:48.42 | Kernel_Core | but how do I enable it ?! |
| 10:48.52 | Kernel_Core | by default it seems , it isnot enabled |
| 10:49.31 | tzafrir_laptop | when you disconnect a line the channel should have "RED" written in /proc/zaptel/1 and in 'zap show channel N' you should see 'InAlarm: yes' |
| 10:50.18 | Kernel_Core | # cat /proc/zaptel/1 |
| 10:50.18 | Kernel_Core | Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1" (MASTER) |
| 10:50.18 | Kernel_Core | <PROTECTED> |
| 10:50.18 | Kernel_Core | <PROTECTED> |
| 10:50.19 | Kernel_Core | <PROTECTED> |
| 10:50.19 | Kernel_Core | <PROTECTED> |
| 10:50.22 | Kernel_Core | here is the result |
| 10:50.30 | Kernel_Core | now all of the lines are connected |
| 10:50.40 | Kernel_Core | but I don't see any Alarm here |
| 10:50.42 | Strom_L | ~pb |
| 10:50.43 | 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 |
| 10:51.10 | Kernel_Core | :D |
| 10:51.14 | Kernel_Core | I know pb |
| 10:51.17 | matnel | 2C |
| 10:52.19 | Kernel_Core | tzafrir_laptop: when I issue zaptel show channel 1 , I have InAlarm: 0 |
| 10:52.37 | Kernel_Core | how do I enable Alarm ? |
| 10:53.02 | tzafrir_laptop | hmmm.. so maybe this only actually works in 1.4.11? shouldn't be |
| 10:53.26 | tzafrir_laptop | are you sure 1.4.10 is running? cat /sys/module/zaptel/version |
| 10:53.47 | Kernel_Core | tzafrir_laptop: I have another problem too, when I restart the server , I have to replug the lines ! |
| 10:54.14 | Kernel_Core | # cat /sys/module/zaptel/version |
| 10:54.15 | Kernel_Core | 1.4.10 |
| 10:55.12 | tzafrir_laptop | I faintly recall something like that resolved on 1.4.11 . But I'm not sure |
| 10:57.27 | Kernel_Core | Okey |
| 10:57.57 | Kernel_Core | I use oslec for echo cancellation |
| 10:58.01 | Kernel_Core | does it affect ? |
| 10:59.20 | tzafrir_laptop | No |
| 10:59.50 | Kernel_Core | tzafrir_laptop: why when I restart the machine I have to replug the lines ? |
| 11:00.15 | tzafrir_laptop | A bug in the driver? |
| 11:00.30 | Kernel_Core | I remember in older version of zaptel , zaptel 1.2 ...there was no need |
| 11:00.35 | Kernel_Core | I don't know |
| 11:00.36 | Kernel_Core | maybe |
| 11:01.39 | tzafrir_laptop | anyway, zaptel now has a script called "live_zap" intended to simplify testing newer versions without actually installing them |
| 11:02.02 | tzafrir_laptop | So if you can afford a short downtime, maybe it would be worth testing 1.4.11 |
| 11:03.15 | Kernel_Core | I am going to install 1.4.11 |
| 11:04.17 | coppice | anyone know why zaptel 1.4.11 might build OK, but then give |
| 11:04.19 | coppice | <PROTECTED> |
| 11:04.21 | coppice | when I try to make install? |
| 11:17.53 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-181-rrdg-esr-2.dynamic.isadsl.co.za) |
| 11:22.32 | *** part/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 11:26.27 | coppice | aha. you must do the entire build as root. I am used to doing "make" as me, and "make install" as root. It seems this now fails |
| 11:27.07 | Kernel_Core | tzafrir_laptop:damen .... I installed 1.4.11 nothing changed , there is no Alarm at all ... |
| 11:28.47 | Kernel_Core | tzafrir_laptop: and one more problem , I have to replug my phone lines! |
| 11:37.43 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 11:50.36 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 11:54.21 | tzafrir_laptop | coppice, hmm... the missing variable there is $(PWD) |
| 11:55.13 | coppice | well, it works OK doing the whole thing as root |
| 11:55.41 | tzafrir_laptop | http://bugs.digium.com/12750 "Can't install zaptel trough sudo" |
| 11:55.52 | tzafrir_laptop | Workaround from there: su - |
| 11:56.13 | tzafrir_laptop | I'm not sure exactly what's the specific problem |
| 11:57.19 | coppice | well, whatever it is, I think its recent. I don't think I used to have that issue |
| 11:57.43 | tzafrir_laptop | (and no: no need to ruin 'make' as root) |
| 11:59.02 | coppice | probably |
| 12:09.44 | Kernel_Core | tzafrir_laptop: do you have any solution for my restart ? :D when I restart I have to replug my phone lines ! |
| 12:30.47 | *** join/#asterisk techie (n=techie@adsl-76-214-14-114.dsl.lsan03.sbcglobal.net) |
| 12:33.08 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 12:40.16 | tzafrir_laptop | Kernel_Core, what happens if you don't replug your phone lines? |
| 12:42.22 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 12:49.53 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 12:50.50 | Kernel_Core | tzafrir_laptop: zap show channel 1-4 become onhook |
| 12:51.13 | Kernel_Core | tzafrir_laptop: and zttool shows that lines arenot connected 4/4/0 |
| 12:51.17 | Kernel_Core | instead of 4/4/4 |
| 12:51.36 | tzafrir_laptop | But are there any actual problems with calls? |
| 12:52.02 | Kernel_Core | no |
| 12:52.28 | Kernel_Core | when the state of zap is onhook , then calls willnot trunk to out or in |
| 12:57.50 | *** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled) |
| 12:58.52 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 13:01.22 | Kernel_Core | tzafrir_laptop: I added new country ringtone to Asterisk :P |
| 13:01.33 | Kernel_Core | tzafrir_laptop: I added new country ringtone to Zaptel :P |
| 13:01.47 | Kernel_Core | but is it enough ? or should I modify asterisk ? |
| 13:01.59 | tzafrir_laptop | What country? How did you add it? |
| 13:03.13 | Kernel_Core | tzafrir_laptop: Iran |
| 13:16.25 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 13:16.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:24.32 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 13:29.28 | *** join/#asterisk bmg505 (n=leon@196-209-78-202-tbnb-esr-2.dynamic.isadsl.co.za) |
| 13:42.58 | hsv-al | thats right...im going to pick up a decked out breakfest from crackerbarrel, and picking up starbucks and bringing it back to fuel another morning of irc/net addiction, while you all starve |
| 13:45.46 | *** join/#asterisk disposable (i=disposab@blackhole.sk) |
| 13:47.21 | disposable | when i define a conference room as conf => 500,1234,4321 and the call it via Meetme(500,i,4444), which password will be needed? the 4444 or 1234? can i omit the password in meetme()? |
| 13:47.47 | disposable | s/the/then |
| 13:53.41 | russellb | yes, you don't need it in extensions.conf if you have it in the configuration |
| 13:53.55 | russellb | personally, i consider meetme.conf as not very useful. |
| 13:54.08 | russellb | if you use the 'd' option, you don't need meetme.conf at all |
| 13:54.10 | hsv-al | russellb, up early eh? |
| 13:54.13 | hsv-al | not sleeping still :) |
| 13:54.14 | russellb | nods |
| 13:54.28 | hsv-al | review irc logs |
| 13:54.37 | hsv-al | i almost murdered someone last night [in the literal sense] |
| 13:54.44 | disposable | russellb: thank you |
| 13:54.55 | russellb | that's not acceptable behavior in #asterisk. |
| 13:55.07 | hsv-al | well, I witnessed a stranger molesting a girl in the forest |
| 13:55.12 | hsv-al | when i was running my 5mile run |
| 13:55.15 | hsv-al | called da cops on him |
| 13:56.23 | hsv-al | If I had a knife, I probably would of cut him |
| 13:56.44 | russellb | O.O |
| 13:57.07 | hsv-al | saw where he walked home too |
| 13:57.16 | hsv-al | lives in a 700,000$+ home in providence |
| 13:58.03 | hsv-al | i wont even follow up with it, but i Love hearing how people would react themselves |
| 13:58.23 | hsv-al | if you saw some douchebag, who was bigger then you doing something like that, would you call the cops, try to attack him, stop it, or what? |
| 13:58.53 | disposable | hsv-al: a five mile run??? wow :) |
| 13:59.04 | russellb | well ... i would hope that you gave the police all of the information that you had. |
| 13:59.23 | hsv-al | ya, they questioned me for about 30min |
| 13:59.28 | hsv-al | and i showed them the house i saw the guy walk in |
| 14:00.00 | disposable | after running for 5 miles i would have died of stroke or something |
| 14:00.05 | hsv-al | in that situation they can't do anything he said |
| 14:00.25 | hsv-al | because nothing is documented, unless repeat complaints, who knows , not a state law expert. |
| 14:04.09 | hsv-al | well, im just hooked into it now because i melted like 60 lbs in 3-4 months |
| 14:04.22 | stevie_ramjet | hsv-al, congrats |
| 14:04.48 | hsv-al | thx, i used to be 240, 183 now |
| 14:04.52 | hsv-al | but yall wouldnt like my diet |
| 14:04.57 | hsv-al | its tough, but it works :) |
| 14:05.38 | hsv-al | heh, another native putnopvu@c-71-228-178-34.hsd1.al.comcast.net |
| 14:05.44 | stevie_ramjet | Yeah, losing that amount of weight in that short period pretty much means you've got to be3 working like a dog. |
| 14:05.44 | hsv-al | where are you in town? |
| 14:05.52 | *** join/#asterisk mfournier (n=marc@142-109-204-62-pool.cable.fcom.ch) |
| 14:05.59 | stevie_ramjet | East on 72 over Chapman Mountain. |
| 14:06.15 | hsv-al | never heard of that area |
| 14:06.26 | hsv-al | i've only been here 4 years, so not that plugged in still. |
| 14:06.55 | stevie_ramjet | If you take 565 east, it eventually ends and becomes highway 72. I live in a neighborhood shortly after that transition. |
| 14:07.23 | hsv-al | :) |
| 14:07.30 | hsv-al | disposable, if you are interested in losing weight |
| 14:07.57 | russellb | stevie_ramjet: you're encouraging stalkers by narrowing down where you live :) |
| 14:08.05 | hsv-al | just eat tunafish for lunch(plain), plain spinach leaves.........and drink water for lunch......egg beaters for lunch.......and chicken breast or burger patties for dinner w/ no bread |
| 14:08.08 | stevie_ramjet | russellb, heh |
| 14:08.13 | hsv-al | coupled with 3-5 miles a day, and the weight falls off |
| 14:08.22 | hsv-al | oh , and dont drink soda, that was my biggie |
| 14:08.24 | stevie_ramjet | hsv-al, yikes, that is pretty hard. |
| 14:08.41 | hsv-al | eggbeaters for breakfest rather |
| 14:09.00 | hsv-al | well, its cool when i look profile view in mirror |
| 14:09.06 | hsv-al | it doesnt look like i goto a lamaz class anymore HEH |
| 14:10.39 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 14:11.43 | hsv-al | stevie, if you play golf, u should come to hampton cove sometime, i go there about once a week with friends |
| 14:11.47 | hsv-al | usually sat morning, or sun |
| 14:12.03 | stevie_ramjet | hsv-al, ah unfortunately I am not a golfer. |
| 14:12.15 | russellb | i own clubs but have not played in years |
| 14:12.20 | hsv-al | you russel? |
| 14:12.30 | hsv-al | well my friends are like 120+ golfers |
| 14:12.35 | stevie_ramjet | lol |
| 14:12.37 | hsv-al | +50 handicap, more then rookies |
| 14:12.40 | russellb | i might fit in, heh |
| 14:13.05 | stevie_ramjet | Gotta go cut grass. Later! |
| 14:13.08 | hsv-al | hamptoncove is 50+ i think for a round, includes a cart |
| 14:13.13 | hsv-al | airport road = 35 |
| 14:13.22 | russellb | cheaper is better. |
| 14:13.37 | russellb | i've got to run, too ... time to get ready for the day |
| 14:13.38 | russellb | cya |
| 14:13.42 | hsv-al | laterz |
| 14:15.41 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 14:16.11 | hsv-al | hello d-fender |
| 14:16.51 | coppice | oh dear. zaptel-1.4.11 is causing oopses that were not there before |
| 14:17.04 | hsv-al | im starving now, , you know what |
| 14:17.12 | hsv-al | im going to cave in, and get a "mcskillet burrito" from mcdonalds |
| 14:17.13 | hsv-al | HEH |
| 14:17.15 | hsv-al | brb |
| 14:18.19 | coppice | ah, the wonderful regional flavours of McD :-) |
| 14:24.28 | *** join/#asterisk darnoq (n=d@chello089076192243.chello.pl) |
| 14:24.53 | lmadsen | lol |
| 14:24.59 | lmadsen | McDicks is just gross |
| 14:25.24 | errr | McChoke & Puke |
| 14:25.52 | lmadsen | pretty much |
| 14:30.38 | coppice | when most asians think of american food, they think of MacDonalds :-) |
| 14:31.20 | errr | wow do they have the wrong idea |
| 14:32.14 | Maliuta | well, lets just say americans aren't known for exporting cullinary delights |
| 14:32.52 | lmadsen | indeed |
| 14:33.16 | lmadsen | unless it involves red meat :) |
| 14:33.45 | Maliuta | no, it has to involve fat and be cheap. not good, just cheap |
| 14:34.42 | coppice | america gave the world some of its worst food, but also gave it those New York cheesecakes and cinnamon rolls. :-) |
| 14:37.53 | errr | and fajitas |
| 14:38.13 | *** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 14:41.22 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:14.04 | *** join/#asterisk rdgr (n=rich@jwad-resnet-31341.d.port.ac.uk) |
| 15:22.48 | *** join/#asterisk macros73 (n=cs@c-24-131-77-140.hsd1.pa.comcast.net) |
| 15:22.52 | *** join/#asterisk asderisk (n=ethemc@78.162.1.136) |
| 15:23.47 | asderisk | hi all, "[Jun 1 14:15:18] WARNING[9683] loader.c: Error loading module 'func_odbc.so': /usr/lib/asterisk/modules/func_odbc.so: undefined symbol: ast_odbc_request_obj" can anybody help me . Thanks ... |
| 15:30.19 | *** join/#asterisk Ntr0P (n=nnp@84-203-11-30.mysmart.ie) |
| 15:31.37 | Ntr0P | hey guys, any of ye understand the way in which the HTTP Digest style response is constructed? I've read the SIP and HTTP rfc's but I the way I'm calculating it must be incorrect as asterisk is giving me a 503 Server Error |
| 15:31.48 | Ntr0P | The code currently used by my app is here http://rafb.net/p/xdHIFn64.html and the sequence of requests is here http://rafb.net/p/9DIKjV41.html |
| 15:33.47 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 15:33.49 | Ntr0P | I'm assuming it is the response value that is incorrect. The rest of the request seems fairly straight forward |
| 15:46.43 | *** join/#asterisk marlow (n=marlow@loke.sca.airwire.ie) |
| 15:50.15 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 15:51.31 | *** join/#asterisk nicox (n=nicox@213-33-15-212.adsl.highway.telekom.at) |
| 16:02.19 | *** join/#asterisk toresbe (n=toresbe@89.10.27.54) |
| 16:02.21 | toresbe | Hello :) |
| 16:02.33 | toresbe | Any Norwegian asterisk users here with experiences on PSTN connection providers? |
| 16:04.13 | marlow | toresbe : pstn is the same anywhere .. it's crap :) |
| 16:05.01 | marlow | toresbe : beyond that .. the PSTN in Norway is pretty much the same as in Denmark .. |
| 16:05.03 | [TK]D-Fender | toresbe, we call them "the telephone company" here... |
| 16:05.19 | marlow | toresbe : it should work .. on and off |
| 16:05.26 | [TK]D-Fender | toresbe, perhaps you could be a a little, maybe even a LOT more specific about what you'd like to know |
| 16:14.17 | *** part/#asterisk asderisk (n=ethemc@78.162.1.136) |
| 16:15.56 | tzafrir | marlow, if that's your opinion, just call it POTS. But aren't there some minor variations between those countries? |
| 16:17.10 | marlow | tzafir: not on POTS |
| 16:17.17 | marlow | tzafir: on ISDN major |
| 16:18.34 | marlow | tzafir: at least the difference on POTS in the nordic is marginal, if at all .. versus in Ireland, where you just have to move from one County to the other to break your system |
| 16:18.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:20.23 | marlow | tzafir: personally, i don't think it's worth the money even to bother with POTS .. a digital circuit will work reliable and every time .. but that's personal preference |
| 16:24.03 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 16:25.02 | *** join/#asterisk CCFL_Man2 (i=e1ad51b9@70.105.221.163) |
| 16:25.55 | jks | marlow, do you know if normal digium cards work without any special configuration/firmware on Danish PRI lines? |
| 16:28.11 | toresbe | I mean in the sense of "people who will provide you with a SIP/IAX connection to the PSTN" |
| 16:28.26 | toresbe | which I'd say should be relatively clear from the context, but oh well. |
| 16:29.45 | marlow | jks: yep .. should be no problem .. if you have a E1 card that is |
| 16:30.06 | jks | marlow, ofcourse |
| 16:30.30 | marlow | jks: configuration depends a bit on the telco, but in general, it should be straight forward |
| 16:30.37 | jks | marlow, I just thought you meant there were major things to be aware of on ISDN |
| 16:31.23 | jks | trying to figure the best way to get a fax service up and running |
| 16:31.45 | marlow | jks: well .. if you want to do that on ISDN |
| 16:31.55 | jks | marlow, what are my alternatives? |
| 16:31.56 | marlow | jks : you'd be better off with CAPI based hardware |
| 16:32.10 | jks | marlow, can yo recommend anything? |
| 16:32.37 | marlow | jks: ISDN30 or ISDN2 ? |
| 16:32.50 | marlow | jks: ISDN2 you'd just need a AVM Fritz! card |
| 16:32.53 | jks | I think the only ones I saw with CAPI support were some Eicon Diva cards |
| 16:32.55 | jks | marlow, ISDN30 |
| 16:33.12 | marlow | jks: yep .. Eicon Diva is definatly not a bad choice |
| 16:33.16 | marlow | jks: with hardware CAPI |
| 16:33.42 | marlow | jks: the thing there is, that it'll work pretty much with any fax machine |
| 16:34.04 | jks | so hardware CAPI means that they have fax dsp hardware builtin? |
| 16:34.28 | marlow | jks: no .. there's better software fax solutions for CAPI |
| 16:34.36 | marlow | jks: and yes, there is a DSP on the board |
| 16:34.41 | marlow | jks: that'll work for you |
| 16:34.46 | jks | marlow, how better if its software? |
| 16:35.15 | jks | I mean, would be better than the other commercial fax "drivers" available for Digium and other cards? |
| 16:35.21 | jks | would it* |
| 16:35.34 | marlow | jks: i've not seen the commercial drivers .. or testet them |
| 16:36.04 | jks | okay, I have looked at a few of them - but it's very difficult to find any independent comparisons |
| 16:36.09 | marlow | jks: but the open source fax bits for asterisk used to have problems with a lot of fax machine |
| 16:36.13 | marlow | jks : +s |
| 16:36.27 | jks | the thing is that the Eicon cards are very expensive |
| 16:36.28 | marlow | jks: i found that capi-based fax nearly always works |
| 16:36.42 | marlow | jks : correct .. because it's an active card |
| 16:36.46 | marlow | jks : with it's own dsp |
| 16:37.05 | marlow | jks : the only passive card, that does CAPI is AVM Fritz! for ISDN2 |
| 16:37.12 | jks | for example a PIKA card costs 699 USD including license for the fax software... a digium card is 300 USD or similar, and then add a few hundred dollars for the fax license |
| 16:37.15 | *** join/#asterisk ariel_ (n=ariel_@c-66-176-41-202.hsd1.fl.comcast.net) |
| 16:37.27 | ariel_ | hello folks |
| 16:37.31 | jks | the Diva card with 8 fax channels costs 5500 USD |
| 16:38.04 | marlow | jks: as i said .. i can only speak from my own experience .. and the asterisk integrated fax solutions weren't a great success in my experience |
| 16:38.12 | marlow | jks : if they worked, fine .. but often they wouldn't |
| 16:38.53 | jks | marlow, yeah okay... I would like to find out where the problem lies |
| 16:38.53 | marlow | jks : with CAPI based solutions i've never had troubles |
| 16:38.53 | jks | marlow, I know of the problems related to spandsp, but I wouldn't be using that |
| 16:38.53 | marlow | jks: often it's timing .. things like that |
| 16:38.53 | marlow | jks: this is where the DSP on the active cards helps you |
| 16:39.23 | jks | yeah... it's just a bit to order an ISDN-30 connection, and only be able to use 8 of the channels |
| 16:39.37 | marlow | jks: well .. depends |
| 16:39.46 | marlow | jks: what you can do, is to use a dual-pri card |
| 16:39.48 | jks | hehe ofcourse - depends on how important it is for me |
| 16:39.59 | marlow | jks: crosswire one port to the fax-card |
| 16:40.08 | marlow | jks: and use the other channels for telephony |
| 16:40.17 | jks | marlow, I haven't got a use for voice channels, sadly |
| 16:40.32 | marlow | jks: ehehe .. well .. that's an entire different story then :-D |
| 16:40.43 | marlow | jks: but you can order a FRA |
| 16:40.49 | marlow | jks: fractional ISDN30 |
| 16:41.01 | jks | It would just be nice with some "real world" benchmarks from someone who tried both types of systems... it seems logic that an active card can do better at achieving 100% compatiblity with old fax machines |
| 16:41.17 | marlow | jks: dunno if you can get 8 channels, but definatly 10 channels on ISDN30 presentation |
| 16:41.22 | jks | but is 95% comp. vers 100%... or is it 40% vers. 100% |
| 16:41.47 | jks | marlow, yeah, but the costs involved are actually the same for a fractional ISDN in my case |
| 16:41.52 | marlow | jks: i think that depends very much on how many different brands of fax machines you encounter |
| 16:42.05 | marlow | jks: the installation, yes .. but not the monthly fee |
| 16:42.22 | jks | marlow, also the monthly fee |
| 16:42.37 | marlow | jks: thats weird ? who are you ordering from ? |
| 16:42.50 | jks | marlow, I haven't decided yet... got offers from Telia, TDC and Colt |
| 16:43.06 | marlow | jks : TDC definatly differs on FRA and PRI |
| 16:43.13 | jks | marlow, yes, normally |
| 16:43.50 | marlow | jks : so what's special in your case ? |
| 16:44.13 | jks | marlow, I'm not buying through the normal channel so to speak |
| 16:44.42 | jks | marlow, they have a new product that involves flatrate fees, etc... bit complicated |
| 16:44.59 | marlow | jks: well .. |
| 16:45.04 | jks | but I'm most probably not choosing TDC as they are a bit more expensive than Telia, and quite a bit more expensive than Colt |
| 16:45.07 | marlow | jks : besides .. you've got another issue |
| 16:45.12 | jks | marlow, okay? |
| 16:45.28 | marlow | jks : eicon 4 port (8 channels card) is 4BRI |
| 16:45.49 | jks | marlow, they have a PRI card with 8 DSPs for fax |
| 16:46.02 | marlow | jks : ah yes .. just saw that |
| 16:46.07 | marlow | jks : that's PRI alright |
| 16:46.29 | marlow | jks : but it's a T1 |
| 16:47.00 | jks | marlow, they have an E1 version :-) |
| 16:47.09 | marlow | jks: well .. that should do then |
| 16:47.16 | jks | Eicon Diva Server PRI E1-8 Single-Span |
| 16:47.41 | marlow | jks : anyhow .. where i think you've got to calculate |
| 16:47.52 | marlow | jks : how important compability is for your business |
| 16:48.14 | marlow | jks : and what it'll cost you in the long run in support/stress etc., if you choose a solution, that gives you hassle |
| 16:49.13 | jks | yeah, that's right... right now it's just to solve a "legacy problem".. i.e I don't expect that I will have to support fax for many years to come |
| 16:49.27 | jks | but on the other hand... I don't want something that only works half of the time |
| 16:49.53 | marlow | jks: exactly .. |
| 16:50.05 | jks | I have also considered buying Digium cards and using the new T.38 thing in Asterisk and then buying T.38 compatible adapters for the old fax-machines, but I haven't examined further if that is a good idea |
| 16:50.05 | marlow | jks : with some dealers, you can actually make a deal .. to test a product |
| 16:50.10 | marlow | jks : before you buy |
| 16:50.31 | jks | marlow, hmm, yes that would be nice - but I haven't found a Danish reseller of these cards :- |
| 16:50.42 | marlow | jks : ehehe . no need |
| 16:50.59 | jks | doing the T.38 thing would eliminate the need for hardware DSPs and asterisk fax software, etc. |
| 16:51.03 | marlow | jks : give you an example .. i bought wireless kit in Taiwan .. that nobody here in Ireland has |
| 16:51.13 | marlow | jks : i have 45 days to return it, if i'm not happy |
| 16:51.32 | jks | okay, quite nice... I'll have to find someone who will do that on these PRI cards then |
| 16:51.33 | marlow | jks : we're talking 5k EUR kit though |
| 16:51.46 | marlow | jks : just a matter of negotiation |
| 16:51.53 | marlow | jks : if you don't ask, you won't get an answer |
| 16:52.01 | jks | I normally import stuff so that part is not a problem for (recently purchased GSM gateways in Taiwan, SSD drives in Korea, tokens from Israel, etc.) |
| 16:52.55 | marlow | jks : exactly .. they might want the money upfront .. but often they'll be happy enough to take the product back, if it didn't do what they promised |
| 16:52.57 | jks | ofcourse, but it doesn't seem so likely that a US dealer will give me 45 days of testing time if I buy a 299 USD digium card :) |
| 16:53.06 | marlow | jks : as long as you agree that upfront |
| 16:53.27 | marlow | jks : nope .. in that range you are talking peanuts |
| 16:53.43 | marlow | jks : so you'd bite the bullet, bite the card and sell it on ebay, if it doesn't do the job:) |
| 16:55.02 | marlow | jks : honestly .. $299 isn't even worth arguing over if it's for a business |
| 16:55.29 | marlow | jks : i can see the argument, if it was for a private person |
| 16:55.47 | jks | marlow, well, it's not _just_ 299 USD... it's 299 USD for the card for example |
| 16:56.03 | jks | marlow, but then I would probably need the echo cancellation module... and then a number of fax licenses |
| 16:56.16 | jks | marlow, so it would probably run into 2000 USD total |
| 16:56.35 | marlow | jks : you only need the echo cancellation for voice |
| 16:56.43 | marlow | jks : i've not seen much issue with fax there |
| 16:56.50 | jks | so I just though it would be worth asking for other people's experience before I burn 2000 USD finding out that I should have bought the 5000 USD card in the first place :-) |
| 16:57.05 | *** join/#asterisk af_ (n=getsmart@88-149-241-109.dynamic.ngi.it) |
| 16:57.06 | jks | marlow, oh, okay - that's what I thought too - but I was told here yesterday that I needed that for faxing too |
| 16:57.28 | marlow | jks : if people have issue with echo on their fax |
| 16:57.35 | marlow | jks : they wouldn't be able to fax to anybody anyhow |
| 16:57.37 | jks | perhaps doing the T.38 thing isn't a bad solution... they have fax machines already |
| 16:58.21 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:59.18 | marlow | jks : how many faxes do you need to receive simulatnously? |
| 16:59.43 | jks | marlow, I'm not really sure... I'm aiming for 8 at the start, but I want to be able to expand if demand is there |
| 16:59.44 | *** join/#asterisk ManxPower (n=manxpowe@86.sub-70-221-48.myvzw.com) |
| 16:59.44 | marlow | jks : that's the first question . because if you need to use 8 lines all the time, the money for the Diva card is really spend well |
| 16:59.55 | marlow | jks : and is it inbound or outbound ? |
| 17:00.01 | jks | marlow, mostly inbound |
| 17:00.07 | jks | probably 90% inbound |
| 17:00.36 | marlow | jks : well .. the issue here is simple .. if you go for a software solution, you'd need the CPU power to handle that, too |
| 17:01.15 | jks | yep, but a quad-core server doesn't cost anything these days |
| 17:01.34 | marlow | jks : so when you look at a software solution, you'd always have to ask'em how many faxes it can process at the same time |
| 17:02.10 | jks | I have tried benchmarking spandsp on the server... it doesn't seem unrealistic to me that I will be able to do it without any problems |
| 17:02.35 | ManxPower | jks: It won't be quite as reliable as a real fax machine, but it is close. |
| 17:02.35 | marlow | jks: eh .. just a question on how many fax machines it won't like :) |
| 17:03.01 | jks | ManxPower, do you have any experience on how close it is? - is it like 95% of the time it works, or is it 50%? :-) |
| 17:03.03 | marlow | ManxPower : i found, that it wouln't talk to at least 30% of the fax machines we encountered |
| 17:03.20 | ManxPower | Older spandsp versions had significant issues with Brother / Cannon fax machines |
| 17:03.31 | jks | marlow, but that was spandsp, right? - and not one of the commercial offerings |
| 17:03.38 | marlow | jks : correct |
| 17:03.41 | *** part/#asterisk ikaRus1 (n=none@80.179.36.48.static.012.net.il) |
| 17:03.51 | marlow | jks : and the CAPI based solutions would work every time |
| 17:03.52 | ManxPower | jks: we have not had compat complaints since we switched to the latest (at the time) spandsp. |
| 17:04.03 | jks | the commercial ones all claims better compatibility and also support for higher speeds than spandsp |
| 17:04.29 | ManxPower | But we also have a fax machine in a POTs line for outging faxes, and incoming faxes that have problem with the spandsp DIDs |
| 17:04.37 | jks | ManxPower, okay, that sounds great... I tried just for the experiment to use spandsp on a IAX-connection... but it wasn't reliable |
| 17:05.11 | ManxPower | jks: I would not expect it to be reliable if you are trying to fax over voice over ip over internet |
| 17:05.26 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 17:05.26 | marlow | jks : i'd say your problem is that you're trying fax over voip |
| 17:05.28 | ManxPower | All of our systems run spandsp on the same server as the PRI lines |
| 17:06.24 | jks | ManxPower, ofcourse - it was just an experiement.. but it actually worked some of the time :) |
| 17:06.26 | ManxPower | Once the fax is accepted, I wrote a script to convert it to PDF and e-mail the user. |
| 17:06.52 | ManxPower | jks: Yes, it is just reliable enough it gives you hope. |
| 17:07.17 | jks | ManxPower, exactly, because if it works one time with a fax machine, I assume it means that spandsp is compatible with that fax |
| 17:07.32 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 17:07.34 | jks | ManxPower, and the times that it didn't work with that fax machine, I'll blame that on the network |
| 17:13.17 | jks | ManxPower, which PRI interface card do you use, by the way? |
| 17:13.50 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 17:23.19 | ManxPower | jks: Digium and Sangoma 2-port T-1 cards |
| 17:23.42 | *** part/#asterisk mfournier (n=marc@142-109-204-62-pool.cable.fcom.ch) |
| 17:23.49 | jaytee | I'm real happy with the performance of my TE-212P card. |
| 17:25.32 | ariel_ | anyone setup a multi-box-location Skill based routing for a call center? |
| 17:26.37 | jks | ManxPower, which of the digium cards do you use? - I'm having a hard time finding out the actual differences between the series? (except the obvious ones like single-span, dual-span, etc.) |
| 17:26.57 | ManxPower | jks: The TDM210P, I think. |
| 17:27.07 | jks | ManxPower, okay, thanks! |
| 17:27.14 | ManxPower | sorry, TE210P |
| 17:32.21 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:32.21 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:47.21 | LuisTorres | Hi..., after a restart of asterisk the mem still occupied. only after a reboot of the server it cleans. Anyone knows why this happen? |
| 17:56.21 | ManxPower | LuisTorres: It cannot happen in normal operation. Are you using AGI? Did you do a restart or a reload in the CLI |
| 17:57.20 | LuisTorres | ManxPower: I did a stop now.., but the main mem on the 'top' didnt free |
| 17:57.43 | LuisTorres | no AGI |
| 17:57.53 | ariel_ | LuisTorres, did you try a kill -9 pid |
| 17:57.55 | LuisTorres | simple sip calls |
| 17:58.06 | LuisTorres | no kill -9 |
| 17:58.11 | LuisTorres | I will try |
| 17:58.35 | ariel_ | LuisTorres, does this happen allot? |
| 17:58.57 | ManxPower | LuisTorres: Once Asterisk exits the operating system is supposed to free the memory. |
| 17:59.12 | ManxPower | Did you confirm asterisk is not running by doing a ps -axwww | grep asterisk |
| 17:59.27 | ManxPower | Because if you kill -9 Asterisk the startupscript will normally relaunch Asterisk |
| 17:59.40 | ManxPower | you would want "service asterisk stop" or do a "stop now" in the CLI |
| 18:00.06 | LuisTorres | yes I confirm. Asterisk as been shutdown |
| 18:00.18 | LuisTorres | but on total memory , it keeps there |
| 18:00.32 | LuisTorres | going to try then with kill -9 |
| 18:00.39 | ManxPower | I do not know what else to suggest |
| 18:00.57 | drmessano | Total memory or the memory asterisk is using? |
| 18:01.11 | LuisTorres | total mem |
| 18:01.14 | drmessano | Are you expecting your totoal OS memory used to have gone down? |
| 18:01.40 | LuisTorres | at least the amount of that asterisk is occupieing |
| 18:01.42 | drmessano | Total OS memory will not go down |
| 18:01.48 | drmessano | It's linux |
| 18:02.05 | drmessano | You won't see memory just disappear when you close an app like in Windows |
| 18:02.33 | florz | drmessano: hu? |
| 18:02.47 | LuisTorres | will stay resident until linux die? |
| 18:02.51 | florz | LuisTorres: how do you know how much asterisk is occupying? |
| 18:02.59 | drmessano | He does, florz |
| 18:03.02 | drmessano | Doesn't |
| 18:03.05 | LuisTorres | Im checking with Top command |
| 18:03.12 | drmessano | He is looking at total OS memory, he said |
| 18:03.18 | drmessano | and is expecting it to go down |
| 18:03.21 | drmessano | Which it will not |
| 18:03.52 | florz | drmessano: if "total OS memory" means "the amount of RAM not being unused" - of course it will |
| 18:04.07 | LuisTorres | with kill -9 it frees up |
| 18:04.36 | drmessano | florz: No, it will not |
| 18:04.45 | florz | drmessano: why not? |
| 18:05.43 | drmessano | Because linux memory management isn't windows memory management.. It isn't about "what's open" |
| 18:06.17 | LuisTorres | so what happen when an app as been closed? |
| 18:06.23 | LuisTorres | stays resident? |
| 18:07.09 | florz | drmessano: Well, that's not exactly a reason. But it's really easy to test that linux _does_ free any userspace pages for which no process does have any reference anymore ... |
| 18:07.12 | drmessano | LuisTorres: The best way I know how to explain it to you since you obviously don't know how it works, is that the OS "caches" memory and allocated as needed |
| 18:07.25 | *** join/#asterisk xenonex (n=xenonex@88.204.197.192) |
| 18:07.38 | LuisTorres | drmessano: ok thanks |
| 18:08.04 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 18:08.19 | florz | drmessano: just create a small programm that does a brk() for a big piece of memory, write some data to the alloced mem, and then exit() |
| 18:08.29 | drmessano | florz: I am really not into a 5 hour discussion about linux memory management.. google has some great hits for it.. If you want to go on thinking the way you're thinking, I certainly will not stop you, but I can't fix everyones misconceptions. |
| 18:09.55 | LuisTorres | dow you know any issues with sipp stress test ? |
| 18:10.32 | florz | drmessano: nope, but you might want to fix your own misconceptions ;-) |
| 18:10.51 | florz | drmessano: this might help you with that: perl '-e$x="x"x(50*1024*1024);sleep(10)' |
| 18:11.45 | LuisTorres | getting loads of this , after a load of calls : bytes allocated in __ast_device_state_changed_literal at line 308 of devicestate.c |
| 18:12.47 | drmessano | florz: I actually know how it works.. but I appreciate your effort.. Like I said, Google. |
| 18:14.24 | florz | drmessano: now, you do claim that when that perl process exits, the amount of free memory reported by top doesn't increase, don't you? |
| 18:15.38 | drmessano | I don't claim anything.. I told you, I am not spending 5 hours arguing with you about this like I did the RF thing.. You seem to have a lot of time on your hands which is better spent google, IMO |
| 18:16.00 | LuisTorres | bytes allocated in __ast_device_state_changed_literal at line 308 of devicestate.c |
| 18:16.11 | LuisTorres | and how about this? |
| 18:16.28 | LuisTorres | is it normal? sorry to bother you guys with this |
| 18:17.34 | JT | in linux, if you have lots of memory listed as "free", either the system hasn't been doing much, or something is wrong :) |
| 18:18.29 | florz | drmessano: given that I know pretty well how the linux VM subsystem works - I don't think so ;-) |
| 18:18.53 | florz | JT: ... or a process that had a lot of private memory alloced just exited ... |
| 18:20.27 | *** join/#asterisk c|oneman (n=none@207-80-252-216-dsl.enter-net.com) |
| 18:27.17 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:27.41 | c|oneman | I love tomato |
| 18:28.06 | drmessano | Sorry, we don't support 3rd party apps here |
| 18:28.52 | c|oneman | I didn't ask for help.... |
| 18:29.25 | drmessano | Oh, I forgot.. it's Sunday.. |
| 18:29.28 | drmessano | ~sunday |
| 18:29.28 | jbot | Sunday is the day all trolls swarm to #debian, avoid at all cost to remain sane. |
| 18:29.35 | drmessano | It was a JOKE |
| 18:29.47 | drmessano | You know, words said to invoke a humorous response |
| 18:30.07 | LiNeTuX | drmessano: Like "Vista" ? |
| 18:30.29 | drmessano | No, LiNeTuX, there is nothing funny about Vista... you sicko |
| 18:30.35 | LiNeTuX | heh |
| 18:32.36 | c|oneman | the vista theme for iphone is funny |
| 18:32.59 | hsv-al | got sick and tired of being confined |
| 18:33.03 | hsv-al | to my computer to read the PDF |
| 18:33.09 | hsv-al | so I had to spend my lifes savings to buy the book :( |
| 18:33.16 | drmessano | c|oneman: We don't support the iphone here either |
| 18:33.23 | drmessano | hsv-al: poor thing.. |
| 18:33.37 | c|oneman | why didn't you just print it |
| 18:33.44 | hsv-al | to thick |
| 18:33.52 | hsv-al | to many , $45 was worth it i decided |
| 18:34.02 | c|oneman | too |
| 18:34.17 | hsv-al | klonemen, u think soo? |
| 18:34.20 | hsv-al | .... |
| 18:34.20 | c|oneman | drmessano: do you support grammar corection? |
| 18:34.30 | hsv-al | :) |
| 18:34.46 | hsv-al | koneman, you going is do they're work? |
| 18:34.48 | drmessano | "correction", hyes |
| 18:34.49 | drmessano | -h |
| 18:34.58 | hsv-al | heh |
| 18:34.59 | c|oneman | lol |
| 18:35.20 | drmessano | hsv-al: You paid $45? |
| 18:35.26 | *** join/#asterisk techie (n=techie@adsl-76-214-14-114.dsl.lsan03.sbcglobal.net) |
| 18:35.30 | drmessano | hsv-al: I got mine from Amazon for $29, I think |
| 18:35.35 | c|oneman | kloneman? did you take kde pills? |
| 18:35.37 | hsv-al | its ok, convenience |
| 18:36.17 | c|oneman | buying stuff you need from time to time is ok :) |
| 18:36.22 | ariel_ | wonders what book cost so much? |
| 18:36.33 | c|oneman | 45$ is a lot? |
| 18:36.37 | hsv-al | The Asterisk 1.6 book that came out in B&Nt oday |
| 18:36.49 | drmessano | There's a 1.6 book? |
| 18:36.55 | hsv-al | covers IAX 3, and AGI revision 2 |
| 18:36.59 | *** join/#asterisk `Asterisk (n=Asterisk@AC827FE3.ipt.aol.com) |
| 18:37.01 | c|oneman | when I walk in to a bookstore most computer books are 60$+ |
| 18:37.04 | drmessano | oh |
| 18:37.08 | ariel_ | iax3 |
| 18:37.08 | `Asterisk | hi there |
| 18:37.10 | ariel_ | ??? |
| 18:37.11 | drmessano | lol |
| 18:37.22 | drmessano | `Asterisk: Change your nick please |
| 18:37.34 | `Asterisk | k |
| 18:37.47 | drmessano | ty |
| 18:38.02 | Obelix | i have a cuestion about asterisk |
| 18:38.06 | Obelix | who can help me ? |
| 18:38.11 | Strom_L | just ask your question |
| 18:38.15 | drmessano | Someone may have an eanser |
| 18:38.21 | Obelix | thanks |
| 18:38.36 | Obelix | i have ubuntu |
| 18:38.39 | Obelix | on my PC |
| 18:39.01 | Strom_L | can you please ask your question on one line and not press enter after every phrase? |
| 18:39.02 | Obelix | and when i put the sip on asterisk |
| 18:39.07 | Obelix | ok |
| 18:39.55 | ariel_ | wonders how can you put sip on asterisk when it's built into the basic release.... |
| 18:40.26 | drmessano | I got finished adding the command line to my gentoo box, so STFU ariel_ |
| 18:40.36 | drmessano | :P |
| 18:40.43 | hsv-al | drmessano, the 1.6 book covers how iax3 allows 40% optimized communication using 1 UDP port |
| 18:40.46 | hsv-al | but over 28.8 speeds |
| 18:41.08 | drmessano | hsv-al: Does it tell you how to get 1.6 to not crash? |
| 18:41.11 | hsv-al | HD audio over dialup, no jitter, requires 35 minutes of buffering. |
| 18:41.50 | hsv-al | yes, but it's required to be installed on Fedora Core 2 |
| 18:42.31 | *** join/#asterisk CrashSys (i=Kumba@azrael.crashsys.com) |
| 18:42.39 | drmessano | I was testing 1.6 at one time.. but decided it wasn't worth it.. |
| 18:43.10 | hsv-al | ariel, iax3 required a Core 2 Quad, and 36gigs of ram |
| 18:43.22 | hsv-al | 73gig sas 15k rpm drives |
| 18:44.05 | ariel_ | hsv-al, wow, all that to pump all it's calls over one port. |
| 18:44.11 | CrashSys | Anyone ever experimented with trying to centralize voicemail on 1.4? |
| 18:44.37 | *** join/#asterisk LuisTorres (n=chatzill@bl6-200-239.dsl.telepac.pt) |
| 18:44.45 | drmessano | CrashSys: What ever do you mean? |
| 18:44.54 | ariel_ | CrashSys, what do you call Centralize voicemail? |
| 18:44.59 | CrashSys | well basically, being able to have one server keep the voicemail, but trigger the WMI on other server connected to it |
| 18:45.02 | LuisTorres | srry just git disconnected |
| 18:46.26 | CrashSys | the rest of it I can do through creative dialplan contexts |
| 18:46.56 | ariel_ | CrashSys, why? |
| 18:47.17 | CrashSys | ariel: Because if someone has voicemail, they'd probably like the red light to blink on their polycom |
| 18:47.31 | ariel_ | why store them on one box? |
| 18:47.37 | CrashSys | or why to centralized voicemail? Because that's what the customer has specified |
| 18:47.38 | drmessano | You can store it on a SAN |
| 18:48.18 | CrashSys | drmessano: So, all the asterisk boxes can share an NFS mount for /var/spool/asterisk/voicemail? |
| 18:48.22 | ariel_ | We have many boxes storing all there recordings to one box. We just mount that drive on those and point the conf files to that mount. |
| 18:48.22 | JT | i assume hsv-al is taking the piss |
| 18:48.56 | CrashSys | that sounds easy enough |
| 18:49.15 | drmessano | JT: yes |
| 18:50.06 | mgdm | ~book |
| 18:50.06 | 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 |
| 18:50.31 | hsv-al | exten => s,1,Dial (SIP/jon:doe@guaranamoron.tld) |
| 18:50.34 | hsv-al | why isnt this working? |
| 18:52.16 | hsv-al | Dial[space]( |
| 18:52.18 | hsv-al | ahhh oops |
| 18:56.18 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 18:56.52 | c|oneman | If I'm doing QoS rules, in theory I would want the bulk bandwith classes to be one of the first on the list - right? |
| 18:57.03 | c|oneman | so as to sort them out quickly |
| 18:57.42 | c|oneman | or else wouldn't the router have to take (a rather large amount of packets) and test a number of conditions for each one |
| 18:57.47 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:58.07 | c|oneman | please someone understand what I'm saying... lol |
| 18:58.16 | hsv-al | cloneman |
| 18:58.31 | hsv-al | what kind of htb rates are you working with |
| 18:59.13 | c|oneman | I'm using QoS for half-dummies... |
| 18:59.38 | hsv-al | I usually do about 3x d/l rate |
| 18:59.42 | hsv-al | in qos.conf |
| 18:59.55 | hsv-al | think about download shaping |
| 19:00.37 | hsv-al | i have this good pdf linked in favorites |
| 19:00.39 | hsv-al | read this |
| 19:00.40 | hsv-al | http://trash.net/~kaber/hfsc/SIGCOM97.pdf |
| 19:01.33 | c|oneman | thats intense |
| 19:02.51 | hsv-al | its really not hard math to understand the background of the document |
| 19:03.02 | hsv-al | maybe some upper level calculus, and statistics thrown in |
| 19:03.25 | c|oneman | you know theres a red flag when they use mathemetical notation to give their email addresses |
| 19:04.11 | hsv-al | its a trifecta |
| 19:04.15 | hsv-al | 3some email address |
| 19:04.31 | c|oneman | heh |
| 19:26.44 | jks | anyone have experience with the new AudioCodes Mediant 600 gateway and asterisk? |
| 19:38.11 | Guggemand | ive only tried the Mediant 1000 and 2000 |
| 19:45.13 | *** join/#asterisk XnOSX (i=4de20eec@gateway/web/ajax/mibbit.com/x-bfb1d8f302f5031b) |
| 19:45.42 | XnOSX | anybody cant tellme if exist any howto about de Asterisk Stats V2? |
| 19:45.51 | XnOSX | i need install it |
| 19:52.02 | [TK]D-Fender | XnOSX, Go to their site |
| 19:54.48 | XnOSX | yap but in this site are not any howto friend |
| 19:56.50 | jks | Guggemand, hmm, it seems the Mediant 1000 runs the same software as the Mediant 600 |
| 19:58.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:00.05 | *** join/#asterisk asdx (n=diego@adsl-141-8.click.com.py) |
| 20:02.08 | jks | Guggemand, have you tried the T.38 support in it? |
| 20:02.25 | Guggemand | nope, not yet |
| 20:03.50 | jks | Guggemand, I'm just trying to ensure that it will work before cashing out for the gateway, ISDN30 setup, etc. :) |
| 20:05.07 | Guggemand | sounds like a good idea :) |
| 20:05.22 | Guggemand | but does t.38 even work with asterisk ? |
| 20:05.44 | jks | Guggemand, it has t.38 pass-through in 1.4.x |
| 20:05.52 | Guggemand | ahh okay |
| 20:06.07 | jks | Guggemand, but I was actually considering CallWeaver until t.38 support is improved in asterisk |
| 20:06.35 | jks | Guggemand, have you tried using spandsp (or similar) with the mediant box? (i.e. just using regular g.711 and not t.38) |
| 20:07.00 | drmessano | T.38 isn't the solution to the problem with FAX |
| 20:07.07 | drmessano | It's an attempt to make it better |
| 20:07.28 | jks | the only solution is to stop faxing |
| 20:07.32 | Guggemand | no, i havent tried anything with faxing :) |
| 20:07.37 | Guggemand | i hate faxing :) |
| 20:07.40 | drmessano | But a perfect T.38 implementation doesn't mean perfect fax |
| 20:07.43 | jks | Guggemand, me too |
| 20:08.07 | drmessano | So switching to something with better T.38 support or using that as a purchase point is just non-sensical |
| 20:08.42 | jks | drmessano, well, asterisk only support pass-through... so it makes sense... |
| 20:09.21 | ManxPower | We just never have problems with fax and Asterisk -- but that might be because we have the fax machines in dedicated analog POTS lines |
| 20:09.53 | drmessano | You're assuming T.38 will fix something, TKS |
| 20:09.55 | jks | ManxPower, exactly |
| 20:10.06 | jks | drmessano, not really, no |
| 20:10.13 | drmessano | <jks> drmessano, well, asterisk only support pass-through... so it makes sense... |
| 20:10.17 | jks | drmessano, it's just a different transport mechanism |
| 20:10.26 | jks | drmessano, well, if you can do both termination and origination you have more possibilities |
| 20:10.41 | drmessano | Alrighty then |
| 20:10.48 | drmessano | Good luck with it |
| 20:11.00 | jks | drmessano, thanks |
| 20:11.47 | ManxPower | The nice things about T.38 is you don't have to worry about jitter |
| 20:11.53 | jks | ManxPower, exactly |
| 20:12.06 | jks | ManxPower, as far as I understand, even on a controlled LAN it can be a problem |
| 20:12.20 | ManxPower | The not nice thing is every T.38 device seems to do it slightly different |
| 20:12.23 | jks | ManxPower, so when you have the media converter in a seperate box (like the mediant 600) - it would be beneficial to do t.38 |
| 20:13.27 | *** join/#asterisk jeev (i=jeev@naptime.net) |
| 20:13.30 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 20:22.23 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 20:28.00 | *** join/#asterisk s0lid (n=s0lid@58.69.1.43) |
| 20:37.13 | *** join/#asterisk xzcvczx (n=simon@gentoo/user/xzcvczx) |
| 20:37.33 | *** join/#asterisk Amun (n=Jonsi@c-71-205-221-174.hsd1.mi.comcast.net) |
| 20:37.36 | xzcvczx | whats the command from the console that will dial an extension then connect it to a supplied number? |
| 20:37.55 | mgdm | Originate? |
| 20:39.30 | xzcvczx | ah thanks.... |
| 20:40.42 | *** join/#asterisk bootc (n=bootc@arcadia.prv.bootc.net) |
| 20:40.56 | bootc | hey folks |
| 20:41.44 | bootc | I have a Dial() statement with multiple phones in it, and want to set the CDR(accountcode) to whichever phone answered the call |
| 20:42.02 | bootc | how can I achieve that? I tried using M() but it doesn't appear to actually set the CDR properly |
| 20:42.41 | *** join/#asterisk crud92822 (n=crud9282@76.248.75.147) |
| 20:44.15 | *** part/#asterisk crud92822 (n=crud9282@76.248.75.147) |
| 20:44.19 | *** join/#asterisk crud92822 (n=crud9282@76.248.75.147) |
| 20:44.34 | crud92822 | So who is around this lovely sunday? |
| 20:44.47 | Strom_L | no one |
| 20:44.49 | Strom_L | we're all dead |
| 20:44.56 | Strom_L | can't we decompose in peace ?!?? |
| 20:44.59 | Strom_L | jesus |
| 20:45.10 | crud92822 | awesome |
| 20:46.29 | crud92822 | well I cannot get my SIP to work. I see it listed in "sip show peers" but it won't show up in the registry. when i call the #, the call never makes it to asterisk. |
| 20:47.26 | Strom_L | i assume you're talking about a SIP account with an ITSP |
| 20:47.47 | crud92822 | www.inphonex.com to be specific |
| 20:49.28 | Strom_L | ok |
| 20:49.42 | Strom_L | so, obvious question: did you triple-check everything? |
| 20:49.51 | crud92822 | I had it working at one point. Then I took the server to the datacenter, couldnt get it to work there, so I brought it back to the office and now it doesnt work here either. |
| 20:50.21 | Strom_L | is the account active? |
| 20:50.23 | ariel_ | crud92822, inbetween the move did you change any settings? |
| 20:50.33 | *** part/#asterisk crud92822 (n=crud9282@76.248.75.147) |
| 20:50.52 | Strom_L | PROBLEM SOLVED PEOPLE |
| 20:50.54 | Strom_L | LET'S HAVE LUNCH |
| 20:51.00 | jaytee | lol |
| 20:51.17 | *** join/#asterisk crud92822 (n=crud9282@76.248.75.147) |
| 20:52.24 | ariel_ | seems it's closer to dinner time here |
| 20:52.28 | crud92822 | I keep getting "SIP/2.0 405 method not allowed" |
| 20:52.47 | crud92822 | I will paypal someone $50 if they can get this stupid thing to work. |
| 20:53.11 | jeev | what's the problem |
| 20:53.24 | crud92822 | it's not accepting incoming SIP calls |
| 20:53.32 | crud92822 | shows up in sip peers |
| 20:53.34 | crud92822 | not in registry |
| 20:53.40 | Strom_L | crud92822: did you check whether the account is active? |
| 20:53.54 | Strom_L | did you check whether your method follows the provider's recommended settings? |
| 20:54.04 | crud92822 | with my SIP provider? yes its active |
| 20:54.16 | crud92822 | im using the exact settings they recommend on their site |
| 20:54.20 | ariel_ | 406 means your sending the info incorrect to them as your registry statement |
| 20:54.32 | crud92822 | im getting 405 |
| 20:54.34 | Strom_L | ariel_: 405, not 406 |
| 20:54.48 | ariel_ | oh yes but there basic same |
| 20:54.57 | ariel_ | what is your regerty line? |
| 20:55.01 | ariel_ | do you have nat setup? |
| 20:55.09 | crud92822 | yes i have NAT |
| 20:55.23 | Strom_L | is the asterisk server behind NAT? |
| 20:55.37 | crud92822 | register => username:password@sip.inphonex.com:5060/500 |
| 20:55.51 | drmessano | uh |
| 20:55.51 | crud92822 | Yes. I've also tried forwarding ports in my router. |
| 20:56.11 | ariel_ | so your also sending the registry to exten 500 on your box |
| 20:56.17 | crud92822 | yes. |
| 20:56.21 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-238.hsd1.ut.comcast.net) |
| 20:56.23 | Strom_L | i assume you substituted your actual username/password for "username:password" |
| 20:56.24 | crud92822 | but the call doesnt even hit the box |
| 20:56.27 | crud92822 | yes. |
| 20:56.54 | Strom_L | crud92822: not even when doing a sip debug? |
| 20:57.14 | crud92822 | no |
| 20:57.17 | [T]ank | i have gone to a 100% sip provider solution in my call centers. I have a stack of T1 cards that I am looking to unload. Is anyone interested? Selling for probably 50% of retail price. |
| 20:57.22 | XnOSX | how i can connect asterisk with the mysql for set the cdr in database? |
| 20:57.39 | crud92822 | I only get the transmit message and then the SIP/2.0 405 error |
| 20:57.53 | Strom_L | crud92822: that's funny...inphonex recommends the registration string be "My-number:my-password@sip.inphonex.com " |
| 20:57.56 | [T]ank | XnOSX: asterisk addons provides the ability with cdr_mysql |
| 20:58.01 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:58.06 | ariel_ | [T]ank, stack of T1 cards? type and why not keep some as backups |
| 20:58.15 | XnOSX | i have a database and table created and the cdr_mysql.conf configured but the activity not set in the db |
| 20:58.16 | crud92822 | mynumber is the username |
| 20:58.33 | [T]ank | ariel_: I will be keeping some... mostly sangoma a104d |
| 20:58.48 | ariel_ | ebay |
| 20:58.49 | [T]ank | i have a few digium cards i would like to keep |
| 20:59.04 | [T]ank | i will be using ebay, but wanted to offer them here first |
| 20:59.19 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177581983.dsl.bell.ca) |
| 20:59.31 | Strom_L | crud92822: do SIP calls work across the same network connection? |
| 20:59.39 | ariel_ | I actually ahave 2 spare TE412P as it is. Which I am also keeping as spare for my call centers |
| 20:59.41 | [T]ank | i could part with 1 or 2 digiums if anyone wants them |
| 20:59.47 | crud92822 | this server will only be receiving SIP calls, not making |
| 20:59.57 | Strom_L | crud92822: yes, but that's not what I asked |
| 21:00.09 | ariel_ | crud92822, remove the last part :5060/500 |
| 21:00.11 | crud92822 | when I call the sip #, I get the standard SIP provider "this extension is unavailable". |
| 21:00.16 | crud92822 | ok, hold on ariel |
| 21:00.18 | ariel_ | and on the cli it should say registry sent |
| 21:01.16 | crud92822 | removed and reloaded sip |
| 21:01.28 | ariel_ | [T]ank, what price for the sangoma a104d I might want one as a test card for some of our testing boxes |
| 21:01.35 | crud92822 | I got the "reliably transmitting NAT to ______" etc etc |
| 21:01.46 | crud92822 | and then the sip read from inphonex with the 405 error |
| 21:01.59 | asdx | can asterisk be used for more than a pbx now? |
| 21:02.05 | Strom_L | crud92822: do they have an option to regenerate your account credentials? |
| 21:02.16 | asdx | ~book |
| 21:02.16 | 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 |
| 21:02.20 | ariel_ | asdx, it always was able to be used for multi companies |
| 21:02.33 | crud92822 | i can manually login to the site and reset my password |
| 21:02.39 | Strom_L | crud92822: give that a try |
| 21:03.38 | asdx | ariel_: i mean, can it scale up to carrier grade and stuff |
| 21:04.35 | ariel_ | scale to carrier grade, wow. Well I run a 22 box mult call center with 3 full ds3 and a oc-12.... maybe |
| 21:04.46 | crud92822 | tried that. same thing. |
| 21:05.04 | crud92822 | shows up in peers as "OK" |
| 21:05.09 | crud92822 | just not in registry |
| 21:05.45 | asdx | ariel_: cool |
| 21:05.55 | ariel_ | asdx, there are a few carrier's using asterisk |
| 21:06.04 | asdx | ariel_: i deployed a single asterisk box with 33+ users |
| 21:06.05 | asdx | :) |
| 21:06.17 | asdx | ariel_: yeah, teliax is one of them i think |
| 21:06.20 | Strom_L | crud92822: ignore peers -- it's irrelevant to your registration issue |
| 21:06.26 | xzcvczx | anyone know any good sip.conf documentation for 1.6? |
| 21:07.05 | asdx | asterisk rocks |
| 21:08.51 | crud92822 | ok |
| 21:09.26 | ariel_ | crud92822, where are you seeing the 405 error? |
| 21:10.48 | crud92822 | in the CLI |
| 21:10.49 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 21:11.44 | ariel_ | crud92822, so when a call come into your box, your asterisk is displaying that message |
| 21:11.54 | crud92822 | no |
| 21:12.03 | crud92822 | every few seconds |
| 21:12.10 | crud92822 | or when i do a sip reload |
| 21:12.17 | ariel_ | ok so it's a message when your box is trying to register |
| 21:12.37 | asdx | is there a release date for 1.6.0 (or it's when it's done) :p |
| 21:12.39 | ariel_ | can you post your sip.conf on pastebin.ca remove your password. |
| 21:13.07 | ariel_ | will not be using 1.6 until it's almost ready for 1.8 to come out |
| 21:13.08 | crud92822 | ok |
| 21:13.34 | asdx | i will use 1.6 |
| 21:13.48 | ariel_ | actually I will be moving to Call Weaver soon |
| 21:13.57 | asdx | why? |
| 21:14.08 | ariel_ | sip stack is far better |
| 21:14.23 | ariel_ | less memory leaks |
| 21:14.42 | asdx | didn't they do improvements in chan_sip with 1.6 or latest 1.4 releases? |
| 21:14.46 | ariel_ | but I will always still for some of my providers have 1 or 2 asterisk gw |
| 21:14.49 | crud92822 | http://pastebin.com/d3d5f165b |
| 21:15.33 | Strom_L | crud92822: register line goes in the "general" section |
| 21:15.50 | asdx | ariel_: i think they added tcp/tls support, etc, in 1.6 |
| 21:16.00 | asdx | stun, and stuff like that |
| 21:16.18 | [TK]D-Fender | crud92822, Indeed it should br right before your first peer entry |
| 21:16.34 | asdx | btw, my telco free'ed SIP again |
| 21:16.35 | asdx | xD |
| 21:16.47 | asdx | sighs |
| 21:16.47 | [TK]D-Fender | crud92822, 15& 16 belong only under [general] as well |
| 21:17.05 | [TK]D-Fender | asdx, Doesn't make it any more legal does it? can't you hear the sirens? RUN!!!!! |
| 21:17.13 | asdx | :D |
| 21:17.20 | crud92822 | ok hold on |
| 21:18.58 | crud92822 | ok we're getting somewhere |
| 21:19.10 | crud92822 | it registers now. now im just getting a blank call |
| 21:19.49 | ariel_ | blank call? |
| 21:20.00 | crud92822 | silent |
| 21:20.11 | crud92822 | ahhh |
| 21:20.12 | crud92822 | hold on |
| 21:21.06 | crud92822 | I needed to add back in the /500 |
| 21:21.41 | crud92822 | i hate when crap is that simple... HA! |
| 21:22.09 | Strom_L | hah |
| 21:22.14 | jeev | FENDER! |
| 21:22.23 | Strom_L | now you owe $50 to me, ariel_, and [TK]D-Fender |
| 21:22.38 | ariel_ | lol |
| 21:22.58 | crud92822 | lol, i will split it ;-p |
| 21:23.22 | Strom_L | sounds like a plan to me |
| 21:23.37 | jeev | fenderrrrrrrrrrrrrrrrrr |
| 21:24.05 | Strom_L | lame |
| 21:24.26 | Strom_L | not like i actually expected him to pay, but still |
| 21:25.32 | jeev | hahaha |
| 21:26.05 | jeev | Fender, the girl is getting static on the line with the polycom :( could it be the upstream is 768 and it's being used? |
| 21:26.10 | jeev | i can move her over to the 2mbit up.. |
| 21:26.46 | ariel_ | static on line could me many things |
| 21:27.51 | jeev | she said that the tones aren't working sometimes either |
| 21:27.59 | jeev | and the operator on the other line when they call the court system doesn't understand them sometimes |
| 21:28.24 | jeev | i'm going from the polycom to the switch, switch through dsl to the datacenter, 100mbit non-bullshit one wilshire datacenter to <2 ms wholesaler |
| 21:29.58 | Strom_L | what kind of "static" is this? |
| 21:30.08 | [TK]D-Fender | jeev, anything possible |
| 21:30.11 | ariel_ | jeev, static, dtmf, and no sound. you need to check things as it sounds like nat/network issues. Have they worked before of is this first setup? T/S should look at all possible issues. But if your able to move them to faster connect that is a start. |
| 21:30.56 | ariel_ | Polycom to Switch (Where is Switch) |
| 21:35.50 | jeev | the switch is like probably 25 feet of wiring |
| 21:36.07 | jeev | but it's like 10 feet away |
| 21:36.12 | jeev | it goes up the wall, ceiling to the center of the office |
| 21:36.51 | *** join/#asterisk CVirus (n=GoD@196.205.192.192) |
| 21:37.51 | jeev | fender, before i order 19 more polycom's... i wanna make sure it functions |
| 21:38.19 | ariel_ | the polycom's are on the same lan/network as the asterisk ? |
| 21:40.07 | jeev | nope |
| 21:40.21 | jeev | the asterisk is <10ms, ~15 miles away at www.onewilshire.com |
| 21:40.37 | *** part/#asterisk xzcvczx (n=simon@gentoo/user/xzcvczx) |
| 21:41.01 | ariel_ | Polycom's are great phones but don't do well with nat to it's registration points. |
| 21:41.11 | jeev | registration is fine... i dont know what is causing it |
| 21:41.18 | jeev | she calls the courts with a regular phone, presses 4 3 1 |
| 21:41.20 | jeev | and it goes through immediatel |
| 21:41.23 | jeev | this one doesn't, brb |
| 21:41.29 | Strom_L | dtmfmode issues perhaps? |
| 21:41.59 | ariel_ | regular phones. Are you using ulaw/ inband/info or rfc ? |
| 21:46.21 | hsv-al | Ask Jeev.com |
| 21:49.40 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 21:52.03 | jeev | :> |
| 21:52.07 | jeev | i'm using rfc |
| 21:52.11 | jeev | 2833 |
| 21:52.18 | Obelix | Jun 1 17:45:17 WARNING[14842]: res_musiconhold.c:426 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3' |
| 21:52.18 | Obelix | Jun 1 17:45:17 WARNING[14842]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player |
| 21:52.28 | Obelix | i need help :( |
| 21:53.01 | Obelix | who can help me |
| 21:53.01 | Obelix | ? |
| 21:53.15 | *** join/#asterisk flosch (n=flosch@unaffiliated/flosch) |
| 21:53.38 | Strom_L | Obelix: read the message |
| 21:53.45 | Strom_L | there are no files in the directory |
| 21:54.47 | flosch | hi all |
| 21:57.07 | Obelix | what kind of file i most put there/ |
| 21:57.07 | Obelix | ? |
| 21:57.50 | Guggemand | what kind of files does an "mp3player" normally require ? |
| 21:58.12 | Obelix | mp3 |
| 21:58.25 | Guggemand | what kind of files do you think you should use then? |
| 21:58.30 | Obelix | yea |
| 21:58.32 | Obelix | thx |
| 21:58.58 | Strom_C | where the hell do people get the idea that they can just stop thinking? |
| 21:59.38 | Obelix | and i get an error |
| 21:59.41 | Obelix | if i want to make a call |
| 21:59.46 | Obelix | why ? |
| 21:59.51 | Obelix | reson 8 |
| 21:59.51 | Guggemand | that |
| 21:59.53 | Guggemand | depends |
| 21:59.54 | Guggemand | on |
| 21:59.55 | Guggemand | the |
| 21:59.56 | Guggemand | error |
| 22:00.24 | Strom_C | Obelix: seriously -- stop pressing enter after every phrase. it's fucking annoying. |
| 22:01.25 | Obelix | ok man sorry |
| 22:02.46 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 22:03.05 | ariel_ | ok I am out of here... See you all around... |
| 22:11.26 | drmessano | Hmm |
| 22:11.27 | drmessano | That |
| 22:11.29 | drmessano | Is.... |
| 22:11.35 | drmessano | Quite interesting.... |
| 22:11.36 | drmessano | No? |
| 22:11.55 | drmessano | DAMN YOU SHATNER |
| 22:12.43 | Strom_C | haha |
| 22:13.42 | Strom_C | damnit, i've been out of the loop with north american ITSPs...is there a pay-as-you-go ITSP that offers T.38? The only one i've found is bandwidth.com, and they're not pay-as-you-go |
| 22:17.40 | *** join/#asterisk Larisa (n=Larisa@AC827FE3.ipt.aol.com) |
| 22:26.45 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 22:31.48 | *** join/#asterisk FrankPiffleHeart (n=boffvert@81-178-80-240.dsl.pipex.com) |
| 22:33.16 | *** part/#asterisk nicox (n=nicox@213-33-15-212.adsl.highway.telekom.at) |
| 22:34.14 | FrankPiffleHeart | er newB to irc and asterisk, I was wondering if anyone could point to documentation on setting up a tdm410 in the uk? |
| 22:39.13 | jaytee | http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf |
| 22:40.38 | jaytee | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf |
| 22:41.33 | FrankPiffleHeart | appreciated, but i am looking for something specifically for the cards operation in the uk. |
| 22:41.39 | jaytee | that should help you setup and check your /etc/zaptel.conf and /etc/asterisk/zapata.conf files for the tdm410. just make sure loadzone=uk |
| 22:41.41 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 22:42.38 | jaytee | and defaultzone=uk in should both be in your zaptel.conf file. |
| 22:42.59 | FrankPiffleHeart | yep uk setup |
| 22:43.09 | FrankPiffleHeart | it rings fine |
| 22:44.20 | FrankPiffleHeart | doesnt pick up wether it be through automation or passing it through to an fxs on the same card |
| 22:45.48 | jaytee | meaning you can call the number and it rings but it doesn't answer the call? |
| 22:46.17 | FrankPiffleHeart | yep. |
| 22:47.04 | jaytee | how are you routing the incoming call? |
| 22:48.26 | FrankPiffleHeart | tried with the standard Asterisk Future of Teleph... settings so in that case it was sent to an er extension that just picked up and then echo, but it never actually picked up |
| 22:49.15 | FrankPiffleHeart | so i install asterisknow and routed the call through i think all unmatched straight throught to an user assigned to the fxs |
| 22:49.43 | FrankPiffleHeart | but it seems to be the same problem in either configuration |
| 22:50.34 | *** join/#asterisk grandpapadot (n=no@74.185.89.46) |
| 23:12.26 | jaytee | ~book |
| 23:12.27 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
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