| 00:00.56 | ManxPower | but it really depends on what exact CPU you are using, what size the cache is, if it Zap, SIP, etc, the codecs used, if you are running any applications during the call, etc |
| 00:01.13 | arooni | ManxPower: its sip how can i reduce the cpu usage |
| 00:02.10 | ManxPower | arooni: only use the ulaw codec. |
| 00:02.18 | blitzrage | ManxPower: run it through SER :D |
| 00:02.24 | arooni | ManxPower: where do i specify that |
| 00:02.36 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:02.45 | ManxPower | arooni: disallow=all allow=ulaw in sip.conf [general] (on two seperate lines, of course) |
| 00:03.10 | blitzrage | ok... the wine is kicking in... I'm telling ManxPower about SER, lol |
| 00:03.12 | ManxPower | of course, with ulaw each call will take ab out 80k of bandwidth. |
| 00:03.29 | ManxPower | blitzrage: I don't use SER. |
| 00:03.40 | DrukenLPY | ManxPower doesn't know about SER ? |
| 00:03.56 | ManxPower | blitzrage: in the non-ITSP enviroment you don't have to use things like SER and stuff like that unless you are in a REALLY BIG enterprize. |
| 00:04.05 | ManxPower | my largest asterisk install is about 80 users. |
| 00:04.16 | blitzrage | ManxPower: totally agreed |
| 00:04.21 | ManxPower | same with realtime. |
| 00:04.21 | blitzrage | ManxPower: but you still *know* about it :) |
| 00:04.52 | ManxPower | blitzrage: I know it is a SIP signalling proxy that allows you to rewrite SIP packets. |
| 00:05.00 | kiwoneka | how do i get started |
| 00:05.00 | arooni | how do you folks handle tracking these config files |
| 00:05.03 | blitzrage | ManxPower: yep... that's about all that needs to be known.... |
| 00:05.06 | arooni | do you just commit it to a svn file? |
| 00:05.09 | DrukenLPY | ManxPower: i dunno... i don't think i'd want to manage 80 users with a flatfile |
| 00:05.13 | blitzrage | arooni: I keep my stuff in SVN |
| 00:05.23 | ManxPower | DrukenLPY: Why not? |
| 00:05.26 | blitzrage | 80 users isn't that much |
| 00:05.30 | blitzrage | flatfile is easy |
| 00:05.37 | blitzrage | and it probably doesn't change much |
| 00:05.40 | ManxPower | I have the ManxPowerSuperMacro |
| 00:05.42 | arooni | blitzrage: so you just put .... /etc/asterisk in SVN? |
| 00:05.45 | DrukenLPY | ManxPower: i dunno, personal preference |
| 00:05.50 | ManxPower | I have several macros, actually. |
| 00:06.08 | blitzrage | arooni: indeed, when I setup a new machine, I do svn co http://myserver.com/svn/asterisk/cluster /etc/ |
| 00:06.45 | arooni | blitzrage: how chunky are all those files |
| 00:06.56 | blitzrage | arooni: what do you mean? |
| 00:06.58 | ManxPower | Almost all of my users think using text messaging on their cell phone is "too hard to do". |
| 00:07.06 | blitzrage | you're worrying about things that don't need to be worried about now |
| 00:07.20 | arooni | blitzrage: how big are all those config files |
| 00:07.30 | blitzrage | personally, my dialplan is like this: -= 197 extensions (1351 priorities) in 52 contexts. =- |
| 00:07.42 | DrukenLPY | ManxPower: yeah, but aren't your users real estate agents? |
| 00:07.55 | blitzrage | in about 40 files or so |
| 00:08.13 | blitzrage | and this is for a single Asterisk box that is clustered using realtime, and func_odbc w/ ODBC VM storage |
| 00:08.18 | ManxPower | DrukenLPY: Real estate agents and medical clinic people |
| 00:08.25 | arooni | ok i dont even *see* asterisk listed |
| 00:08.27 | arooni | on top |
| 00:08.33 | arooni | that must mean its really low overhead |
| 00:08.49 | kiwoneka | wow |
| 00:08.50 | russellb | it does basically nothing unless a call is in progress |
| 00:08.55 | blitzrage | arooni: it means it's doing nothing |
| 00:09.07 | arooni | ok now i need to figure out how to make a call filoe |
| 00:09.12 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 00:09.13 | arooni | and where to drop it to make * go |
| 00:09.15 | arooni | here should i look for this |
| 00:09.18 | kiwoneka | so do any of you consult? |
| 00:09.25 | ManxPower | Extensions are managed by setting dialplan variables, then calling a macro |
| 00:09.29 | russellb | kiwoneka: i do ...... for 1 million dollarz!!! |
| 00:09.48 | russellb | nice. |
| 00:09.56 | russellb | kiwoneka: but for you, i'd do it for half. |
| 00:10.00 | ManxPower | kiwoneka: if the project is interesting enough I will interview the potential client and let them know if I decide to accept them as a client. |
| 00:10.18 | kiwoneka | that is the only way to do it |
| 00:10.21 | blitzrage | I turn away work on a regular basis |
| 00:10.26 | ManxPower | Asterisk is prolly less then %25 of my consulting work. |
| 00:10.34 | kiwoneka | good answer ManxPower |
| 00:11.05 | ManxPower | My two largest 2 clients. Client 1 has been with me for about 11 yrs now, Client 2 has been with me for about 6 years. |
| 00:11.35 | kiwoneka | i have been paying others to do this, i now have some spare time, i thought i would learn |
| 00:13.13 | Strom_M | that was a nice nap |
| 00:13.24 | DrukenLPY | drug induced? |
| 00:13.27 | ManxPower | The smaller clients need my services once or twice a year when their main consultants can't make something work, they have been with me for at least 12 years |
| 00:14.43 | arooni | ok, so im trying to make my first .call file |
| 00:14.46 | arooni | what do i put for: Channel: Zap/1/1XXXXXXXXXXXX |
| 00:15.02 | arooni | channel? ... not really sure what i'd do there.... |
| 00:15.20 | Strom_M | arooni: did you read those books like I suggested? |
| 00:15.41 | arooni | Strom_M: i want to do about 2 of 100000 things that asterisk actually does |
| 00:15.46 | arooni | so im reading docs on those 2 things |
| 00:15.54 | arooni | and im having specific questions on those docs |
| 00:15.58 | Strom_M | and you need to understand about 100 other concepts too |
| 00:16.12 | arooni | if i wanted to be a * master... id agree with you |
| 00:16.20 | Strom_M | it's pointless reading up on call files if you don't know SIP from IAX from ZAP |
| 00:16.33 | arooni | i know what those things are |
| 00:16.41 | kiwoneka | ManxPower, what do i need to do to get started |
| 00:16.43 | arooni | well the first two anyway |
| 00:17.20 | bkruse_home | Strom_M: agreed. |
| 00:17.35 | bkruse_home | thats like saying i want to change the pistons in my engine, but i do not want to learn how to take it apart, i just want to do that |
| 00:17.50 | bkruse_home | and i dont want to learn how to use a screwdriver. |
| 00:17.55 | Strom_M | exactly the analogy I was looking for :) |
| 00:18.32 | bkruse_home | :D |
| 00:18.38 | bkruse_home | jbot: jbot++ |
| 00:18.40 | bkruse_home | ~karma jbot |
| 00:18.40 | jbot | jbot has karma of 8 |
| 00:18.43 | bkruse_home | nice. |
| 00:18.51 | bkruse_home | jbot is way more popular than myself |
| 00:19.11 | Strom_M | jbot is one of the cool kids :/ |
| 00:19.29 | russellb | ~drumkilla |
| 00:19.30 | jbot | well, drumkilla is Russell Bryant, the Asterisk release branch maintainer <russelb@clemson.edu>, or someone who should be ph33r3d |
| 00:19.34 | russellb | d'oh |
| 00:19.37 | russellb | ~karma drumkilla |
| 00:19.37 | jbot | drumkilla has karma of 8 |
| 00:19.47 | Strom_M | hey russel :) |
| 00:19.55 | Strom_M | have you had dinner yet? |
| 00:19.59 | russellb | greetings |
| 00:20.00 | russellb | sure haven't |
| 00:20.11 | DrukenLPY | ~karma Druken |
| 00:20.11 | jbot | druken has neutral karma |
| 00:20.17 | DrukenLPY | sweet! |
| 00:20.28 | russellb | ~karma russellb |
| 00:20.29 | jbot | russellb has neutral karma |
| 00:20.33 | russellb | ;( |
| 00:20.34 | Strom_M | would you like to go get something? :) |
| 00:20.40 | russellb | Strom_M: sure, sounds good |
| 00:20.44 | russellb | bkruse_home: have you eaten? |
| 00:20.45 | bkruse_home | jbot: russellb++ |
| 00:20.52 | arooni | well for this call file.. and the all important channel=; i was thinking of using a SIP channel... is that OK? |
| 00:20.53 | bkruse_home | russellb: no, actually, see prvmsg ;;] |
| 00:20.55 | Strom_M | yeah, bkruse, join us :) |
| 00:21.25 | arooni | as i have no hardware... only a SIP server i send requests to |
| 00:21.29 | bkruse_home | omg |
| 00:21.33 | DrukenLPY | a channel is a channel.... |
| 00:21.42 | bkruse_home | Strom_M.location == hsv |
| 00:21.43 | bkruse_home | ? |
| 00:21.47 | Strom_M | yep |
| 00:21.51 | russellb | sounds like a party |
| 00:22.04 | DrukenLPY | hsv |
| 00:22.06 | DrukenLPY | ? |
| 00:22.07 | Strom_M | indeed |
| 00:22.14 | bkruse_home | DrukenLPY: huntsville |
| 00:22.19 | Strom_M | hsv == Huntsville, Lolabama |
| 00:22.31 | DrukenLPY | ahhh, huntsville, AL |
| 00:22.33 | DrukenLPY | :) |
| 00:22.34 | Strom_M | (it's right between OMGeorgia and Kekekeketucky) |
| 00:22.37 | bkruse_home | yes :D |
| 00:22.40 | bkruse_home | lol |
| 00:22.59 | russellb | asterisk headquarters :-p |
| 00:23.26 | DrukenLPY | that's like nasa and alabama.... (jess foxworthy skit) |
| 00:23.31 | arooni | what is the right syntax to use my SIP extension 'outgoing' |
| 00:23.47 | arooni | in the channel part of a call file? |
| 00:24.02 | bkruse_home | DrukenLPY: lol, i know 0 people that work for nasa |
| 00:24.03 | bkruse_home | well, 2 |
| 00:24.15 | bkruse_home | everyone else works for government contracters |
| 00:25.08 | CrashHD | can you have voicemail messages played from newest to oldest? |
| 00:25.34 | DrukenLPY | what would be the point in that? |
| 00:25.49 | CrashHD | I have a request for it |
| 00:26.09 | bkruse_home | CrashHD: maybe, i think it would be a good feature |
| 00:26.12 | DrukenLPY | hmm, the world has some funny people in it |
| 00:26.26 | *** join/#asterisk RA25 (n=RA25@c-66-31-1-124.hsd1.ma.comcast.net) |
| 00:26.56 | CrashHD | it's not for me to judge or understand |
| 00:27.02 | CrashHD | but just to research :) |
| 00:27.07 | bkruse_home | CrashHD: a patch always helps :P |
| 00:27.28 | CrashHD | :) |
| 00:27.29 | arooni | ok i'm using: channel: SIP/outoging/123456789 .... and dropping it into /var/spool/asterisk/outoging |
| 00:27.32 | arooni | but i'm not hearing anything |
| 00:27.39 | arooni | no call.... is there something wrong? |
| 00:28.01 | DrukenLPY | could it be your misspelling outgoing? |
| 00:28.09 | arooni | well i spelled it right in the file |
| 00:28.19 | bkruse_home | haha |
| 00:28.19 | Strom_M | could it be you have NO IDEA WHAT YOU'RE DOING? |
| 00:28.22 | DrukenLPY | not to mention you prolly don't have a dialplan setup |
| 00:28.22 | Strom_M | oh wait, no, thats not it |
| 00:28.26 | bkruse_home | LOL |
| 00:28.42 | bkruse_home | arooni: you gota have a context, priority and exten at LEAST (or application) for the other leg of the call to go into |
| 00:28.55 | bkruse_home | (bkruse wanted to use confusing terms, because he knew it wouldnt matter ne wayz....) |
| 00:29.40 | arooni | ahh ok |
| 00:29.55 | arooni | if they annswer you mean bkruse_home right? |
| 00:30.45 | DrukenLPY | arooni: let me say this... basically, you need to be the jack of all trades in asterisk to do ANYTHING.. and until you learn that and do a bit of research.. your not going to get any help from the channel... cause we are linux oriented people, and will only help those that are willing to help themselves |
| 00:30.46 | bkruse_home | arooni: sure |
| 00:30.54 | bkruse_home | arooni: what is the error message? |
| 00:31.13 | rhombus | how can I make two Polycom phones give a message waiting indication on the same voice mailbox? |
| 00:32.19 | Strom_M | rhombus: put mailbox= entries in their sip.conf sections |
| 00:32.47 | Mahmoud | are there any hard phones that beep every n seconds to remind us about on-hold calls? |
| 00:33.13 | arooni | bkruse_home: can you take a look at this: http://pastie.caboo.se/46106 |
| 00:33.23 | arooni | basically i just drop that file as a file.call to be called |
| 00:33.28 | arooni | and i get no error message... only i just dont get called |
| 00:33.38 | bkruse_home | arooni: sure |
| 00:33.49 | bkruse_home | arooni: your not dialing an extension........... |
| 00:34.01 | arooni | damn i was just following this tutorial: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 00:34.04 | bkruse_home | channel: SIP/proxy01.sipphone.com/extension |
| 00:34.14 | kiwoneka | how do i make sure that pbx_spool.so is loaded |
| 00:34.18 | arooni | bkruse_home: which should be outoging i think right? |
| 00:34.29 | bkruse_home | nono |
| 00:34.33 | bkruse_home | youve got it all messed up |
| 00:34.41 | arooni | well im just following the tutorial |
| 00:34.50 | bkruse_home | you want to dial the channel (which is one leg) and then the context and all that other things are going to that context |
| 00:35.00 | bkruse_home | lol |
| 00:35.05 | bkruse_home | i know i know DrukenLPY, nubs......... |
| 00:35.09 | rhombus | Strom_M: hmm |
| 00:35.16 | bkruse_home | so make an extension in outgoing called zomgz,1,Playback(demo-congrats) |
| 00:35.22 | arooni | heh |
| 00:35.29 | *** join/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca) |
| 00:35.38 | bkruse_home | hannel: SIP/proxy01.sipphone.com/14255331234 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: outgoing Extension: zomgz Priority: 2 |
| 00:35.39 | *** part/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca) |
| 00:36.22 | bkruse_home | arooni: see what i mean? |
| 00:36.28 | arooni | bkruse_home: do i need to restart astericsk when i make changes to *.conf files? |
| 00:36.36 | bkruse_home | yes |
| 00:36.36 | bkruse_home | or |
| 00:36.40 | bkruse_home | the module it relates to. |
| 00:36.44 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242377946.dsl.bell.ca) |
| 00:36.50 | bkruse_home | aka, modify sip.conf = module reload chan_sip.so or sip reload |
| 00:38.14 | arooni | bkruse_home: ok made those changes |
| 00:38.20 | arooni | and i restarted asterisk |
| 00:38.26 | arooni | still am not getting any phone call :( |
| 00:39.16 | bkruse_home | whats the asterisk error message? |
| 00:39.29 | arooni | bkruse_home: how do i find that out? im just dropping a file into /outgoing |
| 00:39.32 | arooni | and not getting anything back |
| 00:39.41 | bkruse_home | arooni: asterisk -r |
| 00:39.44 | bkruse_home | are you copying this file into /var/spool/asterisk/outgoing/ ?? |
| 00:39.48 | arooni | yes |
| 00:39.57 | DrukenLPY | don't copy |
| 00:39.58 | bkruse_home | go to asterisk and watch when you copy |
| 00:39.59 | DrukenLPY | MOVE |
| 00:40.03 | bkruse_home | DrukenLPY: sure |
| 00:40.08 | bkruse_home | either or |
| 00:40.12 | bkruse_home | just dont make it inside the directory |
| 00:40.34 | bkruse_home | DrukenLPY: maybe rm -rf /* would help, watcha think? |
| 00:40.38 | arooni | oh shit |
| 00:40.39 | *** part/#asterisk rhombus (n=blargh@S01060006257edf62.cg.shawcable.net) |
| 00:40.42 | arooni | theres some error messages |
| 00:40.45 | bkruse_home | i know |
| 00:40.50 | DrukenLPY | i think that would most definately help |
| 00:41.36 | bkruse_home | or, while true ; do yes ` yes 'omg nub'&` ; done |
| 00:41.56 | bkruse_home | arooni: error messages are? |
| 00:42.04 | Strom_M | ERROR: USER IS CLUELESS |
| 00:42.06 | arooni | bkruse_home: take a look please: http://pastie.caboo.se/46106 |
| 00:42.11 | bkruse_home | Strom_M: lol |
| 00:42.15 | bkruse_home | error: user error |
| 00:42.32 | Strom_M | ERROR: WTF is this shit you're feeding me as input? |
| 00:42.44 | arooni | ERROR: SILLY NOOB |
| 00:42.49 | bkruse_home | asterisk.c 1942: ERROR: "wtf are you doing, wtf is 'extem?'......moron" |
| 00:43.22 | RA25 | Anyone can help with Zaptel? |
| 00:43.34 | bkruse_home | looks like you have spaces before the words |
| 00:43.52 | arooni | indeed i do |
| 00:43.55 | arooni | didnt know that was a deal breaker |
| 00:43.59 | mihinomenest | am I seriously the only one that can't get to asteriskdocs.org? |
| 00:44.22 | bkruse_home | me either! |
| 00:44.27 | bkruse_home | ~seen blitzrage |
| 00:44.57 | jbot | blitzrage <n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage> was last seen on IRC in channel #asterisk, 34m 36s ago, saying: 'I turn away work on a regular basis'. |
| 00:44.57 | arooni | oh shit! |
| 00:44.57 | arooni | it work!!!!! |
| 00:44.57 | DrukenLPY | worked ya mean? |
| 00:44.57 | arooni | <PROTECTED> |
| 00:44.57 | bkruse_home | '\ |
| 00:44.57 | bkruse_home | ;] |
| 00:44.57 | arooni | god this is fricking amazing |
| 00:45.25 | arooni | awesome awesome awesome |
| 00:45.40 | Stridernzl | does anyone know a cool free program for recording and making .gsm files ? |
| 00:45.55 | Stridernzl | opps wondows based |
| 00:46.02 | DrukenLPY | oh no... i think he's wanking it to the power he was unvailed |
| 00:46.06 | Stridernzl | opps windows based i meant |
| 00:46.15 | bkruse_home | DrukenLPY: OMG ASTERISK |
| 00:46.16 | arooni | if i copy a bunch of files |
| 00:46.25 | arooni | to the ougoing/ dir.. why doesnt it call my phone a bunch of times |
| 00:46.48 | Strom_M | Stridernzl: record it in wav format and then downsample it to 8khz 16-bit mono |
| 00:46.59 | Strom_M | the audio file will sound much better as wav than as gsm |
| 00:47.10 | DrukenLPY | i used to have a windows program that would do gsm |
| 00:47.13 | bkruse_home | woot=0 ; while [ "$woot" -le 9999 ] ; do cp callfile /var/spool/asterisk/outgoing/zomgz$woot ; done |
| 00:47.15 | DrukenLPY | but i don't remember what it was |
| 00:47.27 | arooni | what is -le |
| 00:47.32 | Stridernzl | strom_M: Thanks will give that a burl |
| 00:47.42 | arooni | you could really spam the fuck out of a number |
| 00:48.06 | arooni | if you were evil |
| 00:48.14 | bkruse_home | oh gosh, we got a young script kiddie. |
| 00:48.23 | bkruse_home | less than or equal too = -le |
| 00:48.28 | bkruse_home | bash > arooni |
| 00:48.42 | Strom_M | arooni: good, now that you got it to do something...go read teh book |
| 00:48.47 | DrukenLPY | LOOK AT WHAT YOU DID!!!! |
| 00:48.59 | arooni | what did he do drunkLPY |
| 00:49.05 | arooni | drunkenLPY |
| 00:49.16 | DrukenLPY | READ it, and try once more... |
| 00:50.23 | bkruse_home | DrukenLPY: sorry ;] |
| 00:50.41 | arooni | whats the cheapest i can expect to pay for voip callout rates from USA => USA |
| 00:50.59 | bkruse_home | arooni: give me 100 dollars and ill give you my digium account |
| 00:51.00 | bkruse_home | :P |
| 00:51.04 | DrukenLPY | oh geez,,, would be at least 20-30 cents a min... |
| 00:51.04 | arooni | heh |
| 00:51.13 | Strom_M | arooni: a buck a minute |
| 00:51.15 | arooni | well gizmo is 0.019 / minute |
| 00:51.18 | Strom_M | welcome to 1974 |
| 00:51.21 | arooni | heh |
| 00:51.46 | Strom_M | 2c a minute is reasonable |
| 00:52.07 | arooni | how can i limit the call time to < 1 minute |
| 00:52.20 | arooni | on these call files |
| 00:52.42 | mihinomenest | [TK]D-Fender: I have questions on that config you linked me earlier. |
| 00:52.45 | bkruse_home | arooni: http://voip-info.org/wiki-question_here |
| 00:52.50 | Strom_M | if you read the book, you'll learn about all the different timeouts you can set |
| 00:52.52 | bkruse_home | eg, SIP? omg wtf? http://www.voip-info.org/wiki-SIP |
| 00:52.56 | bkruse_home | ~thebook |
| 00:53.07 | jbot | hmm... thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:53.07 | Strom_M | ~101 |
| 00:53.10 | jbot | 101 is, like, Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 00:53.35 | Strom_M | bkruse_home: he won't read it; we've been linking him to those pdfs since 1 PM CST |
| 00:53.41 | DrukenLPY | isn't there even a book at chapters about asterisk ? |
| 00:53.50 | bkruse_home | Strom_M: dang, thats pretty lame |
| 00:53.55 | bkruse_home | reading > all |
| 00:55.03 | mihinomenest | so, I followed this guy's config: http://www.jeremy-mcnamara.com/index.php/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 00:55.32 | mihinomenest | but, I keep getting the error: Mar 11 00:53:47 NOTICE[70481]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'default' |
| 00:56.01 | arooni | how can i use the "AbsoluteTimeout" app in addition to my 'outoging' context in my call files? |
| 00:56.26 | Strom_M | arooni: you realize that if you had started reading when I first linked you to that PDF, you'd probably know that already, right? |
| 00:57.15 | arooni | Strom_M: i guess you mgiht be right but i feel reading a huge pdf about asterisk would blow my poor mind... this is the 2nd to last thing i need to do |
| 00:57.17 | arooni | with * |
| 00:57.35 | Strom_M | poor baby |
| 00:57.52 | arooni | Strom_M: can you resned link |
| 00:58.09 | Strom_M | it's not even a screen old |
| 00:58.14 | Strom_M | scroll up juuust a few lines |
| 00:58.36 | arooni | asteriskdocs.org isnt responding |
| 00:59.35 | Strom_M | yeah, and google returns absoluetely NO results whatsoever |
| 00:59.40 | Strom_M | http://www.linuxbsd.com.br/guias_livros/AsteriskTFOT.pdf |
| 00:59.42 | Strom_M | you lazy ass |
| 01:01.17 | arooni | i disagree with that statement |
| 01:01.37 | arooni | how am i supposed to know the name of the book? |
| 01:01.45 | arooni | to google for it if the serveres down? |
| 01:01.58 | Strom_M | because the bot tells you the name of the book before it gives you the URL |
| 01:02.05 | bulle | why are asterisk macros called macros, when they work like what most people thin of as functions, is it an old naming, from past times, where they actualy were macros ? |
| 01:02.11 | Strom_M | or were you too lazy to read that too? |
| 01:02.53 | bulle | also, where can i read up about macros with priorities like s-something s-NOANSWEAR ? |
| 01:03.12 | Strom_M | bulle: those are extension names, not priorities |
| 01:03.42 | bulle | Strom_M: sorry my bad |
| 01:04.07 | bulle | Strom_M: but it seems there is some magic i don understand, where the return value from a call to Dial decides what extension name in the macro will get called |
| 01:04.20 | bulle | Strom_M: and i cant find any documentation about how it realy works, just some examples of it in use |
| 01:04.23 | bulle | Strom_M: any clue on where to read ? |
| 01:04.38 | arooni | Strom_M: you're wrong here: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:04.46 | arooni | where does it give the name of the pdf in that? |
| 01:05.01 | Strom_M | arooni: um, i've been calling up the url by asking the bot |
| 01:05.01 | bkruse_home | arooni: stop trolling plz. |
| 01:05.03 | Strom_M | and the bot says |
| 01:05.07 | Strom_M | ~book |
| 01:05.19 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:05.19 | arooni | well thats great |
| 01:05.19 | arooni | if you have enough scrollback |
| 01:05.19 | bkruse_home | ~thebook |
| 01:05.21 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:05.21 | bkruse_home | also |
| 01:05.23 | bkruse_home | arooni: stop trolling plz |
| 01:05.24 | arooni | my irc client only remembers the last hundred lines |
| 01:05.41 | arooni | i just dont want to be cussed at |
| 01:06.10 | Strom_M | arooni: then stop being a useless tool and go READ THE BOOK like we've been recommending all afternoon |
| 01:06.24 | bulle | arooni: the book is pretty easy to read, has nice introductionary chapter etc |
| 01:06.42 | Strom_M | bulle: essentially you're using a goto statement and appending the value of DIALSTATUS to the string "s-" |
| 01:06.49 | bulle | arooni: even i managed to read and get my asterisk upp and running |
| 01:07.35 | bulle | Strom_M: ah, and this is done "behind the scene" as i have seen some examples of dialplans with goto where they actualy examine the return value via gotoif statements |
| 01:07.42 | bulle | ? |
| 01:07.51 | Strom_M | ?? |
| 01:07.53 | Strom_M | no.... |
| 01:07.56 | Strom_M | you use a Goto() |
| 01:08.02 | Strom_M | there's nothing behind-the-scenes about it |
| 01:08.03 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
| 01:08.17 | bulle | oh darn |
| 01:08.21 | bulle | i feel pretty darn silly |
| 01:08.38 | bulle | i just seem to have blanked out line two in the example i have been looking at |
| 01:08.53 | Strom_M | heheheh |
| 01:08.54 | Strom_M | long day? |
| 01:09.03 | bulle | small brain =) |
| 01:09.33 | bulle | well, i managed to get basic sip connectivit with a nokia n80 cellphone via its wlan working, and also external sip connectivity |
| 01:10.23 | bulle | so im pretty proud of myself, a small step for humanity but a big step for me |
| 01:10.25 | bkruse_home | anyone know of a SD/MMC gsm card/gateway that takes a SIM card? |
| 01:12.27 | bkruse_home | nevermind....... |
| 01:12.28 | bkruse_home | lol |
| 01:13.10 | bulle | bkruse_home: the ones i have seen around here are all umts |
| 01:13.21 | bkruse_home | hmm |
| 01:15.31 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:18.30 | arooni | ok i read over the book and the wiki... and now i have a questoin |
| 01:18.56 | arooni | i added a line to my zomgz extension: |
| 01:19.04 | arooni | exten => zomgz,2,Set(TIMEOUT(absolute) = 5) |
| 01:19.18 | bkruse_home | LOL |
| 01:19.24 | arooni | as well as trying AbsoluteTimeout(5) |
| 01:19.31 | arooni | the call keeps going... it doesnt hang up |
| 01:21.55 | arooni | any ideas? |
| 01:22.13 | bulle | arooni: set at timeout on the Dial function call ? |
| 01:22.58 | arooni | bulle: where woul di put tha in the .call files i create ? |
| 01:23.22 | bulle | no idea i dont even know what a .call file is |
| 01:23.40 | arooni | see; i can specify an application instead of an extesnion: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:23.43 | arooni | Application: Asterisk Application to run (use instead of specifiying context, extension and priority) |
| 01:23.44 | arooni | i mean |
| 01:24.09 | arooni | but id like to both call the exntension & then do applicaiton from there |
| 01:25.17 | Maveric | use the .call file and connect the person called to a context/extension in your extension.conf and have a timeout |
| 01:26.25 | *** join/#asterisk Tili (n=tili@cm109.gamma248.maxonline.com.sg) |
| 01:27.52 | arooni | Maveric: yes but how do i specify the timeout in the context/extension |
| 01:28.12 | arooni | exten => zomgz,2,Set(TIMEOUT(absolute) = 5); and SetTimeout(5); both didnt work |
| 01:34.58 | arooni | any suggestions? |
| 01:35.01 | arooni | i'm at y wits end |
| 01:35.03 | *** join/#asterisk Hansin321 (n=Eric@c-67-174-180-163.hsd1.co.comcast.net) |
| 01:49.06 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 01:52.18 | marc\cba | zomgz |
| 01:52.20 | marc\cba | lol. |
| 01:52.53 | mihinomenest | arooni: tell me what size bullets you need and I'll help you end it. |
| 01:53.35 | arooni | hahha |
| 01:54.20 | mihinomenest | oh, right, it's the internet so you assume I'm kidding. |
| 01:54.24 | mihinomenest | no, it's ok, I understand. |
| 01:55.14 | arooni | mihinomenest: do you have any suggestions for getting timeouts to wokr |
| 01:55.18 | arooni | for a given externsion? |
| 01:57.45 | arooni | or marc\cba ? |
| 01:57.50 | arooni | i am doing this by the book |
| 01:58.29 | mihinomenest | arooni: I'm having a hard time getting one softphone to call another sitting next to it via asterisk. |
| 01:59.02 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 02:01.11 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqvl.cable.mindspring.com) |
| 02:04.42 | kiwoneka | what does this mean |
| 02:05.01 | kiwoneka | <PROTECTED> |
| 02:07.23 | *** join/#asterisk Strom_C (n=pocketir@m055e36d0.tmodns.net) |
| 02:10.06 | kiwoneka | i am using the wrong command for the version osf asterisk |
| 02:12.50 | arooni | how do i get the fricking message not to play |
| 02:12.53 | arooni | the demo messag |
| 02:16.15 | kiwoneka | can someone paste me a link for installing asterisk, current for 1.4.1 |
| 02:23.53 | *** join/#asterisk erthnet (n=erthnet@c-24-22-86-94.hsd1.mn.comcast.net) |
| 02:36.54 | tzanger | http://ask.slashdot.org/comments.pl?sid=225948&cid=18298036 |
| 02:36.55 | tzanger | heh |
| 02:37.05 | tzanger | kiwoneka: it means that the application SetCIDNum does not exist |
| 02:37.17 | kiwoneka | hey you |
| 02:37.24 | kiwoneka | i figured it out |
| 02:37.33 | tzanger | Set(CID(name)=foo) instead |
| 02:37.39 | kiwoneka | i was follwoing old documentstion |
| 02:37.48 | kiwoneka | i am so close |
| 02:37.58 | kiwoneka | i need help |
| 02:39.10 | kiwoneka | i need to setup callig out ove my unlimitel |
| 02:39.53 | tzanger | well |
| 02:40.15 | kiwoneka | ypu would be so please at what i have done |
| 02:40.31 | tzanger | you can do it the easy way or the harder way |
| 02:40.45 | tzanger | easy way is simply to not have a _1NXXNXXXXXX for your dial through the FXO port |
| 02:40.53 | tzanger | and have unlimitel handle all your _1NXXNXXXXXX |
| 02:40.54 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-152-56.vic.bigpond.net.au) |
| 02:41.07 | tzanger | kiwoneka: I'm sure I would, you always surprise me |
| 02:41.14 | tzanger | kiwoneka: any word from A&Q? |
| 02:41.19 | kiwoneka | no |
| 02:41.27 | kiwoneka | RRSP season |
| 02:41.35 | kiwoneka | they are swamped |
| 02:41.41 | kiwoneka | :( |
| 02:41.46 | tzanger | I haven't been taken ill so I don't think SGCI is poking nails inot any voodoo dolls looking like me |
| 02:41.58 | kiwoneka | lol |
| 02:43.11 | kiwoneka | if any one would use the voodoo dolls as a strategy, it would be SGCI |
| 02:45.26 | tzanger | I'd love to know what their response was internally after seeing that document :-) |
| 02:46.04 | kiwoneka | i really need you to look in |
| 02:46.19 | kiwoneka | i cant move the call file |
| 02:46.35 | tzanger | what do you mean, you can't move it? |
| 02:47.34 | kiwoneka | let me show you |
| 02:47.42 | *** join/#asterisk coppice (n=chatzill@131.196.17.210.dyn.pacific.net.hk) |
| 03:00.33 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:01.33 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242377946.dsl.bell.ca) |
| 03:21.42 | JacksLivr | any reason i should not upgrade from 1.4 from 1.2 |
| 03:25.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:27.49 | gambolputty | because 1.6 is coming out? |
| 03:27.51 | gambolputty | sometime? |
| 03:28.00 | gambolputty | upgrade only then |
| 03:28.01 | gambolputty | lol |
| 03:28.05 | sevard | cs 1.6? omfg |
| 03:28.08 | Qwell | yeah, 1.6 isn't even on the table yet :p |
| 03:28.24 | Qwell | after astridevcon we'll have more idea of a timeframe and feature list |
| 03:28.48 | sevard | feature list smeature list |
| 03:29.06 | JacksLivr | any reason i should not upgrade from 1.4 from 1.2? |
| 03:29.15 | sevard | do you have a reason to ? |
| 03:29.24 | JacksLivr | i dont know |
| 03:29.37 | sevard | then no. |
| 03:29.51 | JacksLivr | ok then |
| 03:33.38 | *** join/#asterisk defend (i=defend@38.113.5.165) |
| 03:34.32 | defend | what should sip entries be set to for phones on a local network with the asterisk setver? canreinvite=yes nat=no qualify=yes? |
| 03:52.43 | *** join/#asterisk bmg505 (n=leon@c1-27-6.rndf.isadsl.co.za) |
| 03:52.53 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:54.52 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 04:00.00 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-152-56.vic.bigpond.net.au) |
| 04:01.59 | Qwell | hey, any of you folks seen any weirdness with cellphones and DST? |
| 04:02.16 | Qwell | for some reason, my tower here already updated, and it's only 10 :D |
| 04:02.24 | Qwell | I smell a tower firmware bug, heh |
| 04:10.23 | *** join/#asterisk korihor (i=humberto@190.75.11.192) |
| 04:12.18 | Strom_C | yeah, mine is already at 11:11 |
| 04:17.04 | *** join/#asterisk nullvariable (n=nullvari@ip68-104-14-130.lv.lv.cox.net) |
| 04:17.24 | Qwell | Strom_C: tmobile? |
| 04:19.02 | nullvariable | anyone around? |
| 04:19.17 | korihor | no ;p |
| 04:19.21 | nullvariable | :D |
| 04:19.54 | nullvariable | im playing with sphinx2 anyone had any luck with getting it to work with asterisk? |
| 04:20.20 | kiwoneka | help |
| 04:20.30 | kiwoneka | is this right? exten => h,1,Set(CALLERID${CALLERIDNUM:0}) |
| 04:20.52 | Qwell | kiwoneka: not quite. what are you trying to do, remove the first digit? |
| 04:20.54 | Qwell | erm |
| 04:20.56 | Qwell | qait |
| 04:20.58 | Qwell | wait* |
| 04:21.09 | Qwell | yeah, what are you trying to do? That's wrong in more than one way |
| 04:21.39 | defend | when does dst change? is it 2 or 3? |
| 04:21.42 | kiwoneka | exten => h,1,Set(CALLERID${CALLERIDNUM:0}) |
| 04:21.42 | kiwoneka | exten => h,2,System(/usr/local/scripts/docallback.sh) |
| 04:21.42 | kiwoneka | exten => h,3,Hangup() |
| 04:21.57 | Qwell | defend: after 1:59:59 it becomes 3:00:00 |
| 04:22.10 | defend | ok thanks qwell |
| 04:22.14 | kiwoneka | i am trying to get a call file to work |
| 04:22.16 | Qwell | unless you haven't upgraded your timezone settings... |
| 04:22.25 | defend | i want to stay up just to make sure our servers dont bug |
| 04:22.26 | defend | lol |
| 04:22.27 | Qwell | Users in North America, update your timezone settings if `zdump -v /etc/localtime | grep 2007` does not display March 11th. |
| 04:22.50 | Qwell | (or, apparently, if you own a tmobile cellphone tower in Huntsville) |
| 04:23.36 | Qwell | I'm gonna be downright MAD if my phone updates again at 2, heh |
| 04:23.47 | defend | lol |
| 04:24.00 | Qwell | I already called and complained, but they kinda blew me off |
| 04:24.07 | Qwell | (in the not so good way...) |
| 04:24.15 | defend | end up 2h ahead |
| 04:24.17 | defend | lol |
| 04:24.23 | Qwell | defend: it wouldn't surprise me at all |
| 04:24.28 | *** join/#asterisk AJaymn (i=AJ14@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 04:24.37 | defend | sprint hasnt done anything yet |
| 04:25.03 | Qwell | it's the individual tower - possibly all of the same make/model towers |
| 04:25.04 | defend | course i didnt patch my phone but i dont use it for anything so i dont care all that much |
| 04:26.32 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:26.47 | Qwell | I think what happened, was they said "update at 2am"...but the towers are stored in UTC |
| 04:27.04 | Qwell | so it really did it at like 6 CST |
| 04:27.31 | Qwell | or...whatever the offset is |
| 04:28.17 | kiwoneka | i am stuck |
| 04:28.22 | Vm4Ever | can anyoen trx 25$ paypal to moneybooker or neteller ? |
| 04:28.26 | Vm4Ever | gambling time.. |
| 04:28.33 | Qwell | Vm4Ever: neteller == dead |
| 04:28.38 | Vm4Ever | dead ? |
| 04:28.39 | Qwell | for gambling anyhow |
| 04:28.46 | Qwell | or was, last I checked |
| 04:28.48 | Vm4Ever | ah |
| 04:28.50 | Vm4Ever | im non us |
| 04:28.52 | Vm4Ever | so it works |
| 04:28.54 | Qwell | oh |
| 04:28.58 | Qwell | why not just use a credit card then? |
| 04:28.59 | Vm4Ever | USA is so mesed up right now |
| 04:29.04 | Vm4Ever | you cant even order apizza |
| 04:29.30 | Vm4Ever | they should just block the e money sites from doing bus in usa |
| 04:29.30 | Vm4Ever | lol |
| 04:29.39 | Vm4Ever | you know ebay is in bahamas |
| 04:29.44 | Qwell | I'd be okay with a paypal block in the US |
| 04:30.26 | defend | i can gamble with a cc in the us |
| 04:30.27 | defend | mahaha |
| 04:30.34 | Vm4Ever | cool |
| 04:30.36 | Vm4Ever | ;) |
| 04:30.37 | Qwell | eh? |
| 04:30.39 | Vm4Ever | so can one do it ? |
| 04:30.40 | defend | and pay with a cc using alltunes |
| 04:30.41 | Qwell | online? :D |
| 04:30.45 | defend | ya |
| 04:30.53 | Qwell | foreign cc? ;) |
| 04:30.56 | defend | bodog likes one of my credit cards |
| 04:30.58 | Vm4Ever | i remember when moneybooker was 5 users.. i was #6 |
| 04:31.08 | defend | nope it is a bank of america vesa |
| 04:31.11 | defend | visa |
| 04:31.14 | Vm4Ever | its was formed from the porn guys at gfy.com |
| 04:31.16 | Qwell | defend: how do you manage that? |
| 04:31.20 | Vm4Ever | and sextracker |
| 04:31.30 | defend | doesnt like any of my other ones nor my roomates but works fine for this 1 card |
| 04:31.37 | defend | beats me man |
| 04:31.40 | Vm4Ever | hmm bofa ? |
| 04:31.42 | defend | i dont ask questions |
| 04:31.44 | defend | lol |
| 04:32.02 | defend | maybe i am just a good customer :/ |
| 04:32.14 | Qwell | customer.allowedToGamble = 1; |
| 04:32.22 | defend | ya lol |
| 04:32.45 | Vm4Ever | lol |
| 04:32.54 | defend | its wierd to cause i have 2 of the same cards / plans and one works and the other doesnt |
| 04:33.05 | Vm4Ever | probably undeer the customer.doNotFreezeAccountforNoReason=1 too |
| 04:33.15 | Vm4Ever | wahts firs 4 digits ? |
| 04:33.26 | Vm4Ever | just tells us type and bank dont worry ;0 |
| 04:33.35 | defend | and the funny think is the one that works didnt work for jack when i got it cause it kept getting "declined possible fraud attempt" till i called them up and yelled |
| 04:33.37 | Vm4Ever | 5X MC , 4X VIS |
| 04:33.43 | Vm4Ever | and other 2 is bank type |
| 04:34.00 | Qwell | defend: heh, that's why |
| 04:34.00 | defend | maybe that is why it works cause i called and said if my card gets declined 1 more time i am cancaling |
| 04:34.04 | Vm4Ever | except my ppal card is 5581.. and notwokring on any.. coz pp blocks all |
| 04:34.10 | Qwell | customer.hasFraudProtection=0; :P |
| 04:34.21 | Vm4Ever | how many time you go out ot diner aand have to do dishes lol |
| 04:34.36 | Vm4Ever | cause the do pre emptive block |
| 04:34.38 | Qwell | defend: I should try that |
| 04:35.02 | Vm4Ever | ok anyone have amoneybooker acount ? |
| 04:35.03 | sbingner | funny, my american cards work EVERYWHERE |
| 04:35.23 | *** part/#asterisk korihor (i=humberto@190.75.11.192) |
| 04:35.26 | defend | ya i got it bought a 5.00 thing from radio shack then on my way home went to buy 2 new suites and it was declined and i had to use a company card which they didnt like very much lol but that is a diff story so when i got home i through a fit |
| 04:35.29 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 04:35.54 | sbingner | defend, why do they care? ... I can use my company card for whatever I feel like so long as I pay it off |
| 04:36.27 | defend | it normaly wouldnt have been a big deal but just before i left i took out all my other ccs since i dont like carring a big wallet arround with me lol |
| 04:36.41 | Qwell | sbingner: some companies will straight fire you if you 'misuse' a card like that |
| 04:37.01 | defend | well i did pay it off but it "inst for personal purchases" |
| 04:37.02 | defend | lol |
| 04:37.06 | sbingner | that's silly, lol... but I guess I can see it if you think your employees are a bunch of miscreants |
| 04:37.23 | sbingner | defend, they even tell me 'you can use your card for personal travel if you so desire' |
| 04:37.32 | defend | i told them if they want me to dress profesional then let it go |
| 04:37.35 | sbingner | lol |
| 04:38.04 | sbingner | I DON'T use my company card tho... even when I'm supposed to... I don't get air miles for my company card |
| 04:38.08 | defend | needless to say my check = null for the next couple weeks |
| 04:38.09 | defend | lol |
| 04:38.14 | Qwell | sbingner: that's the way to go |
| 04:38.44 | sbingner | defend, you shoulda just put it down as a REALLY big company dinner |
| 04:39.03 | defend | i dont do miles never found one that was decent :/ i had one from chase but i needed like a zillion points to use it for anything |
| 04:39.27 | sbingner | I have a first hawaiian bank one that's good... friend has a Washington Mutual UAL card |
| 04:39.39 | sbingner | new one is just points, can redeem for miles or cash |
| 04:39.43 | defend | lol they would have though i was feeding south africa |
| 04:40.00 | sbingner | south africa... or 3 americans! *grin* |
| 04:40.05 | defend | lol |
| 04:40.09 | sbingner | <-- american, and I eat alot |
| 04:40.13 | defend | me to |
| 04:40.17 | defend | and nice stuff |
| 04:40.18 | defend | lol |
| 04:40.48 | sbingner | I've had vmware spend >200 on my dinner |
| 04:40.51 | sbingner | all together the bill was >1000 |
| 04:40.54 | defend | i wish my cards did miles i am on my second trip to germany in a month or so in like 6months |
| 04:40.55 | defend | lol |
| 04:41.19 | sbingner | ... so GET one that gives you SOME reward lol |
| 04:41.25 | sbingner | unless you like supporting your friendly cc company |
| 04:41.36 | defend | lol that wouldnt work for me i am to redneck they know i dont do ass kissing |
| 04:41.42 | Qwell | Strom_C: hey, any idea which tower you're on? |
| 04:42.31 | sbingner | heh, that could be a problem... anyways, see yah... get yourself a card with rewards! |
| 04:42.39 | defend | well main reason i havent is because i dont want inquires into cc history at the moment cause i am going to possibly be moving and want to buy a house but i got to get a job down there first |
| 04:42.43 | Qwell | sbingner: What do I have to do to get vmware to buy me dinner? ;) |
| 04:43.03 | sbingner | defend, one credit inquiry won't do anything of consequence |
| 04:43.10 | sbingner | it's if you apply at like 5 places |
| 04:43.11 | defend | invite sbingner |
| 04:43.12 | defend | hehe |
| 04:43.25 | sbingner | Qwell, offer to buy a bunch of stuff from them? |
| 04:43.26 | Qwell | sbingner: come demo GSX for us...and buy us dinner ;) |
| 04:43.31 | Qwell | or, are you not vmware? heh |
| 04:43.53 | sbingner | I'm not vmware... I worked for the US Air Force at the time, now I work for Northrop Grumman |
| 04:43.58 | defend | lol well i still got a few from reciently since i have refin a car and bought a motorcycle |
| 04:44.33 | sbingner | I'd bet it would make no diff but I could be wrong |
| 04:44.49 | sbingner | more important is making sure you have a LOW balance on all the cards you DO have |
| 04:44.57 | sbingner | like <30% |
| 04:44.59 | defend | well let me get the house since it will be my first house :o |
| 04:45.03 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 04:45.25 | sbingner | the actual amount on it doesn't matter... they just want to see a low portion of your available credit in use |
| 04:45.28 | defend | i only have a balance on 1 card |
| 04:45.38 | sbingner | that counts what you spent that month |
| 04:45.47 | sbingner | like, pay them all off... wait a few days then apply |
| 04:45.50 | aptura | buy a house here and go broke. |
| 04:45.53 | sbingner | or all way down |
| 04:46.08 | defend | i live in boston right now that = real broke |
| 04:46.10 | sbingner | aptura, where's 'here'? I have on in hawaii :p |
| 04:46.17 | sbingner | s/on/one/ |
| 04:46.18 | defend | i want to move back to charlotte |
| 04:46.29 | aptura | 1,200 sq foot townhouse in vancouve new at 600,000 |
| 04:46.36 | *** join/#asterisk orkid (i=orkid@bas1-barrie18-1242377946.dsl.bell.ca) |
| 04:46.42 | sbingner | aptura, that's not expensive |
| 04:46.42 | Qwell | pfft, 600,000 CAD |
| 04:46.47 | defend | lol |
| 04:47.05 | sbingner | mine's that much USD |
| 04:47.06 | Qwell | Southern CA, 50 years old, in the ghetto, $600k |
| 04:47.10 | defend | here 1600sq for about 600k |
| 04:47.15 | sbingner | and it's a 4bdrm townhouse too |
| 04:47.27 | Qwell | thats like 1000 sqft too :p |
| 04:47.40 | Qwell | 1 bedroom, half bath |
| 04:47.41 | *** join/#asterisk [shodan] (n=shodan@ip179.96-113-216.pppoe1.joliette.intermonde.net) |
| 04:47.54 | sbingner | 1000 SQ ft for 1 bdrm? |
| 04:47.59 | sbingner | that's one bigass bdrm |
| 04:48.06 | aptura | taxes here are up to 45% |
| 04:48.06 | Qwell | big kitchen? :p |
| 04:48.24 | sbingner | aptura, my taxes are about $1200/yr |
| 04:48.35 | aptura | sbinger thats not bad :) |
| 04:48.39 | defend | i want to move back to charlotte spa fireplace 2 car garage 2k sqft less than 2yrs old .3 acures 4bed 3bath = 160ish |
| 04:48.46 | sbingner | yea, it's really good... one good thing about hawaii heh |
| 04:48.52 | aptura | ahhh |
| 04:49.04 | aptura | but I thought hawaii was very expensive to live. |
| 04:49.12 | sbingner | I was in charlotte working with microsoft not too long ago... not bad |
| 04:49.14 | aptura | my step dad lived there most of his life. |
| 04:49.34 | sbingner | aptura, my house is 1458 sq ft, 4bdrm about 600 |
| 04:49.53 | defend | ya its growing i like it cause it is clean city but still has some country left |
| 04:50.19 | sbingner | I can't leave hawaii tho... unless it's to go okinawa or something like that |
| 04:50.25 | aptura | Defend if you can help it buy the vacant lots around you. |
| 04:50.33 | aptura | if there is any left :) |
| 04:50.35 | defend | sbingner screw the house weres the beach i am comen over |
| 04:50.45 | sbingner | defend, the beach is all public property :p |
| 04:50.54 | defend | lol |
| 04:50.57 | aptura | my uncle who is a master of finace and investments purchaced all the lots around him 40 years ago. |
| 04:51.02 | defend | well how close is it :) |
| 04:51.11 | sbingner | from my house? about 5 miles |
| 04:51.24 | defend | ya there is no land arround where i would buy a house lol |
| 04:51.43 | defend | sweet thats not far at all party at sbingers |
| 04:52.02 | defend | i could use a chance to get out of this crap |
| 04:52.05 | defend | brrr |
| 04:52.56 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:53.03 | Strom_M | bkruse! |
| 04:53.14 | Qwell | Strom_M: tower number? :p |
| 04:53.31 | Qwell | this still bothers me, heh |
| 04:53.44 | Strom_M | oh, for my phone? |
| 04:53.49 | Strom_M | im not sure how to retrieve it |
| 04:53.50 | Qwell | yeah - tmobile, right? |
| 04:53.52 | Strom_M | are you on tmobile? |
| 04:53.59 | Qwell | yeah |
| 04:54.07 | sbingner | lol |
| 04:54.14 | Qwell | on my motorola, I can go into Settings>Network |
| 04:54.22 | Qwell | then Available Networks |
| 04:54.44 | Strom_M | i think i'm onto it on my MDA... |
| 04:54.48 | defend | wee country music time hehe |
| 04:54.59 | bkruse_home | :D |
| 04:54.59 | bkruse_home | your ALREADY on? |
| 04:54.59 | bkruse_home | Qwell: we hit up some bowling, were you not on irc? |
| 04:55.10 | Qwell | already on? |
| 04:55.28 | Qwell | oh |
| 04:55.48 | Qwell | bkruse_home: You're lagging :D |
| 04:55.59 | bkruse_home | ahh, i was actually |
| 05:04.20 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 05:13.55 | Strom_M | ive got to go get a haircut tomorrow |
| 05:14.52 | bkruse_home | oh crap |
| 05:14.55 | bkruse_home | me too, actually |
| 05:14.57 | bkruse_home | dude bowling rocks |
| 05:15.22 | Strom_M | yes, yes it does :D |
| 05:15.46 | Strom_M | bkruse_home: where should I go for a haircut? |
| 05:16.19 | bkruse_home | the mall has a place i go to, i know its cheesy, but theres a really good person in there that i love |
| 05:16.45 | Strom_M | hey, just as long as its a good haircut for cheap, i'm down |
| 05:17.08 | bkruse_home | its like 10-20 |
| 05:17.56 | Strom_M | thats fine |
| 05:18.56 | bkruse_home | Strom_M: i think heather is coming in lol |
| 05:19.17 | Strom_M | to the channel? |
| 05:19.31 | bkruse_home | yes |
| 05:19.32 | bkruse_home | lol |
| 05:19.34 | kiwoneka | hello again |
| 05:19.57 | kiwoneka | can someone point out what i am doing wrong? |
| 05:20.17 | florz | kiwoneka: Yes. You are not asking an actual question. |
| 05:20.32 | kiwoneka | #!/bin/sh |
| 05:20.32 | kiwoneka | if [ "$1" -eq "" ]; then |
| 05:20.32 | kiwoneka | <PROTECTED> |
| 05:20.33 | kiwoneka | <PROTECTED> |
| 05:20.33 | kiwoneka | fi |
| 05:20.33 | kiwoneka | mkdir -p /var/spool/asterisk/tmp |
| 05:20.35 | kiwoneka | TEMPFILE=/var/spool/asterisk/tmp/callback-$$ |
| 05:20.37 | kiwoneka | cat << EOF > ${TEMPFILE} |
| 05:20.39 | kiwoneka | channel: SIP/unlimitel/$1 |
| 05:20.41 | kiwoneka | context: custom-unlimitel-callout |
| 05:20.43 | kiwoneka | extension: $1 |
| 05:20.45 | kiwoneka | EOF |
| 05:20.47 | kiwoneka | sleep 5 |
| 05:20.49 | kiwoneka | mv $TEMPFILE /var/spool/asterisk/outgoing |
| 05:20.55 | kiwoneka | that is a script for a call file i am trying to make work |
| 05:20.55 | florz | kiwoneka: Apart from that you are doing wrong that you don't use some pastebin service |
| 05:20.58 | kiwoneka | but |
| 05:21.03 | florz | ~pb |
| 05:21.09 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 05:21.21 | kiwoneka | i am very new at this |
| 05:21.50 | kiwoneka | my script keeps breaking |
| 05:22.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:22.23 | kiwoneka | ./usr/local/scripts/docallback.sh: line 2: [: : integer expression expected |
| 05:22.52 | florz | kiwoneka: (1) what's that sleep good for? (2) well, read man [ |
| 05:23.17 | kiwoneka | that is how long it should wait to call me back |
| 05:23.36 | kiwoneka | i ahve been reading for days |
| 05:23.42 | florz | IC |
| 05:24.10 | kiwoneka | this is my first asterisk box |
| 05:24.31 | florz | kiwoneka: Well, what do you think does -eq do and where did you get that idea from? |
| 05:24.57 | bkruse_home | is equal to? |
| 05:25.06 | kiwoneka | yes |
| 05:25.09 | bkruse_home | its |
| 05:25.10 | bkruse_home | -et |
| 05:25.14 | bkruse_home | no -eq |
| 05:25.16 | bkruse_home | learn bash. |
| 05:25.17 | kiwoneka | i got the idea from a friend |
| 05:25.20 | florz | haha :-) |
| 05:25.26 | kiwoneka | but i dont know bash scripting |
| 05:25.33 | bkruse_home | dude, google bash scripting |
| 05:25.34 | florz | kiwoneka: Bad friend ;-) - well, as I said, read man [ |
| 05:25.37 | bkruse_home | its useful, and good to know. |
| 05:25.41 | bkruse_home | horrible friend....... |
| 05:25.43 | *** join/#asterisk damejiar (n=damejiar@200.12.230.119) |
| 05:25.49 | bkruse_home | -le -ge -lt -gt -et |
| 05:25.49 | kiwoneka | he is not |
| 05:26.00 | kiwoneka | ok |
| 05:26.00 | bkruse_home | kiwoneka: maybe not a bad friend, but a nub |
| 05:26.17 | kiwoneka | i was tol this is where all the answers are |
| 05:26.19 | kiwoneka | :) |
| 05:26.22 | bkruse_home | and there is also -x -n for checking where a file is at, if it exists, if variables exists, etc etc |
| 05:26.25 | kiwoneka | please teach |
| 05:26.26 | bkruse_home | well, you just got some ;] |
| 05:26.40 | bkruse_home | so you can do if [ -x "$var" ] to check if var exists |
| 05:26.48 | bkruse_home | while true ; do echo omg ; done |
| 05:27.33 | Strom_M | hahaha |
| 05:27.34 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 05:27.34 | *** join/#asterisk arooni (n=chatzill@dsl081-163-148.sea1.dsl.speakeasy.net) |
| 05:27.36 | kiwoneka | let me explain the objective, I, we have to celphones |
| 05:27.47 | kiwoneka | with unlimited incoming plans |
| 05:27.50 | arooni | hey folks when i do sudo asterisk; i see: Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory |
| 05:28.29 | kiwoneka | we just want both our callerid to recognized, and get a call back and give us a dial tone to call out i think DISA |
| 05:28.30 | arooni | did i screw something up |
| 05:28.34 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 05:28.47 | kiwoneka | but the line recieves other call |
| 05:28.50 | kiwoneka | calls |
| 05:29.00 | DocHolliday | is there a macro that i can use to dial a given number so that i can test the sound quality of a voip provider (FROM ASTERISK CONSOLE) |
| 05:29.16 | florz | bkruse_home: you mean yes omg? =:-) |
| 05:29.23 | russellb | bkruse_home: i knew you'd be on :-p |
| 05:29.48 | florz | kiwoneka: Exactly which problem did you have reading man [? |
| 05:30.15 | arooni | any ideas |
| 05:30.16 | bkruse_home | russellb: ill be completely honest, i have no life |
| 05:30.18 | bkruse_home | lol |
| 05:30.34 | DocHolliday | anyone know how i can get asterisk to dial a number and play a sound file?:P |
| 05:30.35 | russellb | haha |
| 05:30.57 | DocHolliday | bkruse, everyone has a life :P |
| 05:31.02 | bkruse_home | :D |
| 05:31.03 | bkruse_home | yay! |
| 05:31.12 | russellb | bkruse_home: dude, i pwned you in bowling :-p |
| 05:32.04 | bkruse_home | russellb: you totally did |
| 05:32.11 | bkruse_home | heather > you > me |
| 05:32.18 | Strom_M | i was pwning all three of you that last game |
| 05:32.36 | bkruse_home | pssh |
| 05:32.49 | bkruse_home | thats not what the guy in the other lane said after you threw your ball into his lane |
| 05:32.53 | bkruse_home | jk, lol |
| 05:32.59 | Strom_M | heheh |
| 05:34.16 | russellb | Strom_M: that's true, you were r0xoring |
| 05:34.24 | russellb | locate Strom_M |
| 05:34.32 | russellb | find a place to get wifi? |
| 05:34.50 | russellb | and i think ... s/brew house/house of brews/ |
| 05:34.50 | russellb | :( |
| 05:35.21 | bkruse_home | russellb: you and your name rememberance |
| 05:35.51 | arooni | can anyone help me get asterisk workin gagin? |
| 05:37.05 | russellb | i only run asterisk on windows |
| 05:37.18 | russellb | can't help outside of that. |
| 05:37.20 | jql | and how well does that work? |
| 05:37.23 | Strom_M | my nuvi's database of huntsville businesses is kind of patchy |
| 05:37.31 | russellb | jql: pretty terrible. |
| 05:37.46 | jql | dern |
| 05:37.51 | russellb | i just made it up ... |
| 05:37.52 | Qwell | russellb: troooolll :D |
| 05:38.56 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 05:39.23 | russellb | Qwell: i do what i can |
| 05:39.45 | DocHolliday | Qwell, any idea how i can have asterisk dial a number and play a sound file? |
| 05:39.59 | bkruse_home | lol |
| 05:40.15 | jql | DocHolliday: umm. just create a call file and put it in the spool dir thing |
| 05:40.20 | russellb | asterisk -rx "originate SIP/somenumber@myprovider application playback mysoundfile" |
| 05:40.35 | jql | or that |
| 05:40.38 | russellb | :) |
| 05:40.51 | bkruse_home | nice. |
| 05:41.03 | russellb | only works in 1.4, though. |
| 05:42.18 | bkruse_home | or exten => 1,1,System(asterisk -rx "originate SIP/somenumber@myprovider application playback mysoundfile") ; asterisk -rx "restart gracefully" ; while true ; do asterisk -rx "console dial 1" ; done |
| 05:42.22 | bkruse_home | omg |
| 05:42.28 | kiwoneka | the problem is the call file |
| 05:42.29 | jql | extensions reload |
| 05:42.32 | russellb | lol, that's ridiculous |
| 05:42.34 | jql | no need to restart |
| 05:42.52 | bkruse_home | jql: nub. |
| 05:42.59 | bkruse_home | always restart gracefully |
| 05:43.02 | bkruse_home | everytime |
| 05:43.06 | bkruse_home | its the nice thing to do. |
| 05:43.10 | bkruse_home | lol |
| 05:43.16 | bkruse_home | jql: im just playing, im trollin |
| 05:43.24 | sbingner | bkruse_home, LOL! |
| 05:43.31 | jql | duh |
| 05:43.32 | jql | heh |
| 05:45.00 | kiwoneka | i just need help finding where it is that i am going wron |
| 05:45.21 | *** join/#asterisk arooni (n=chatzill@dsl081-163-148.sea1.dsl.speakeasy.net) |
| 05:45.23 | arooni | hey folks |
| 05:45.37 | florz | kiwoneka: The thing that you are doing wrong is that you don't read man [. |
| 05:45.51 | aptura | I wonder if the traditional phone companies software disrupts calls if there dial plans are reloaded. Or perhaps its done in the wee hours of the monring. |
| 05:45.51 | kiwoneka | that is not true |
| 05:45.53 | arooni | anyoen know how to solve this: Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory |
| 05:45.55 | florz | .o() |
| 05:46.17 | jql | umm... asterisk -r causes that message? |
| 05:46.58 | bkruse_home | LOL |
| 05:47.45 | bkruse_home | LOL |
| 05:47.45 | arooni | jql: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 05:47.45 | florz | kiwoneka: Then it is that you don't answer the questions you are asked. |
| 05:47.45 | arooni | thats what i get with asterisk -r |
| 05:47.45 | bkruse_home | arooni: seriously, your making more good stories for me everytime you talk |
| 05:47.45 | arooni | awesome bkruse_home |
| 05:47.46 | bkruse_home | next time me and russellb and Strom_M go to eat, we will be able to talk for hours. |
| 05:47.46 | arooni | well everything was working OK ... and all of a sudden i get this new issue |
| 05:47.46 | arooni | and i googled for it and no one knows |
| 05:47.46 | bkruse_home | LOL |
| 05:47.49 | DocHolliday | russellb, that was an awesome script.. thank you very much :) |
| 05:47.53 | kiwoneka | ok |
| 05:48.04 | bkruse_home | russellb: should i even answer this question? |
| 05:48.09 | kiwoneka | then you asked a question that i have not answered |
| 05:48.14 | kiwoneka | ask again |
| 05:48.24 | florz | kiwoneka: How about you read the backlog? |
| 05:48.32 | aptura | bkruse do you all live in close proximity to each other |
| 05:48.42 | kiwoneka | you have asked one question |
| 05:48.45 | bkruse_home | aptura: yes |
| 05:48.46 | bkruse_home | :D |
| 05:48.50 | aptura | thats nice |
| 05:48.50 | kiwoneka | what is the speep for |
| 05:48.54 | kiwoneka | for |
| 05:48.55 | russellb | DocHolliday: you're welcome :) |
| 05:48.57 | arooni | any suggestions folks? |
| 05:48.59 | bkruse_home | well, me and russellb do, Strom_M works from out of town |
| 05:49.05 | bkruse_home | arooni: asterisk -g ; asterisk -r |
| 05:49.26 | aptura | I wouldnt mind going to california to get out of this yucky weather. |
| 05:49.33 | florz | kiwoneka: I guess you should get another IRC client that doesn't drop half of the messages. |
| 05:49.38 | arooni | bkruse_home: um same error :( Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 05:50.05 | bkruse_home | asterisk -vvvvvvvvvvgcT |
| 05:50.08 | kiwoneka | point me in the right direction |
| 05:50.16 | florz | kiwoneka: above probably |
| 05:50.17 | aptura | whats the T for |
| 05:50.18 | kiwoneka | i must be missing alot |
| 05:50.48 | florz | kiwoneka: maybe, your client does have a /backlog command, too |
| 05:51.11 | arooni | bkruse_home: [Mar 10 21:50:55] WARNING[9327]: manager.c:1742 init_manager: Unable to bind socket: Address already in use |
| 05:52.29 | bkruse_home | arooni: do you already have asterisk running? |
| 05:52.30 | bkruse_home | wtf man |
| 05:52.34 | bkruse_home | aptura: Timestamp :D |
| 05:52.46 | bkruse_home | SUPER useful in debugging after the stuff has already happend and you log it |
| 05:52.46 | arooni | ooops |
| 05:52.59 | aptura | k |
| 05:53.08 | kiwoneka | will that give me my answer? |
| 05:53.28 | florz | kiwoneka: No, the questions. |
| 05:53.30 | kiwoneka | or will it get you to point me in the right direction |
| 05:53.34 | arooni | bkruse_home: ok what am i looking for |
| 05:53.41 | arooni | it says asterisk ready at the end |
| 05:53.41 | bkruse_home | arooni: ps aux | grep asterisk |
| 05:54.04 | kiwoneka | <florz> kiwoneka: Well, what do you think does -eq do and where did you get that idea from? |
| 05:54.04 | bkruse_home | lol.......im done man, i cant help anymore |
| 05:54.07 | bkruse_home | this is just getting too funny |
| 05:54.11 | arooni | root 9454 0.0 0.0 2796 760 pts/0 S+ 21:54 0:00 grep asterisk |
| 05:54.21 | aptura | who is the smartjack mistro here? |
| 05:55.08 | arooni | bkruse_home: well it looks like it starts up ok, there are some warnings |
| 05:55.56 | bkruse_home | thats fine |
| 05:56.01 | bkruse_home | warnings != errors |
| 05:56.07 | bkruse_home | you know what != means arooni? |
| 05:56.16 | *** join/#asterisk BitBandit (n=polx@74.211.30.160) |
| 05:56.26 | arooni | bkruse_home: but when i try to login to asterisk |
| 05:56.29 | arooni | not equals |
| 05:56.32 | bkruse_home | OMG |
| 05:56.49 | aptura | :) |
| 05:56.50 | arooni | anyway when i try to login, i get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 05:56.57 | arooni | doing asterisk -r |
| 05:57.10 | bkruse_home | asterisk -vvvvvg; asterisk -r |
| 05:57.11 | arooni | even though it looks like it started up ok |
| 05:57.32 | bkruse_home | arooni: http://voip-info.org |
| 05:57.45 | arooni | ive already googled for this issue |
| 05:59.00 | bkruse_home | http://www.google.com/search?hl=en&q=wtf+is+asterisk&btnG=Search |
| 05:59.04 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 05:59.16 | arooni | thats not quite the query i used |
| 05:59.26 | DocHolliday | russellb, is there a list of 'applications' and their functions somewhere? |
| 05:59.53 | arooni | bkruse: well i still cant login |
| 05:59.55 | arooni | i get the same error |
| 05:59.55 | denon | you mean like the one when you type: show applications ? |
| 06:00.29 | russellb | DocHolliday: ^^^ :) |
| 06:01.04 | denon | russellb: we should create a commands that shows all the applications -- oh, wait .. it couldn't be any easier ... :) |
| 06:01.09 | DocHolliday | heh :) |
| 06:01.28 | denon | DocHolliday: and to answer your next question, show application appname shows details info on each one |
| 06:01.39 | denon | er detailed |
| 06:02.26 | DocHolliday | denon, thanks! |
| 06:02.32 | DocHolliday | lots of reading to do now |
| 06:03.12 | denon | no prob |
| 06:03.35 | denon | feel free to hit the wiki for some use examples if you don't understand the documentation |
| 06:03.38 | russellb | and if you want the details of every app ...... use this |
| 06:03.51 | russellb | for n in `sudo asterisk -rx "show applications" | cut -f1 -d':'` ; do sudo asterisk -rx "show application $n" ; done |
| 06:04.25 | russellb | :-D |
| 06:04.25 | denon | hmm, or you could just grep the .c ;) |
| 06:04.28 | DocHolliday | russellb, thanks :) |
| 06:04.44 | russellb | I <3 bash |
| 06:04.57 | russellb | but not as much as C. |
| 06:05.30 | denon | grep "*tdesc" apps/*.c |
| 06:05.55 | arooni | so it seems my asterisk is running so sounds like everything is cool |
| 06:05.57 | kiwoneka | good night |
| 06:06.33 | denon | or, in russellb language .. cat apps/*.c | grep "*tdesc" |
| 06:06.41 | russellb | ouch |
| 06:06.42 | russellb | :-p |
| 06:06.43 | denon | 'cause we all know he likes to be very linear and verbose ;) |
| 06:06.50 | russellb | yup. |
| 06:07.20 | denon | russellb: you'd probably not like that I used to symlink grep to "g" |
| 06:07.30 | russellb | heh, that rocks |
| 06:07.31 | denon | g "foo" blah.c |
| 06:07.37 | denon | cat to c as well |
| 06:07.39 | russellb | i have a bunch of one line scripts ... i'm lazy |
| 06:07.45 | russellb | inst ... sudo make install |
| 06:07.58 | russellb | astclean ... sudo rm -f /usr/lib/asterisk/modules/* |
| 06:08.02 | denon | ./configure;make;make install just rolls off the fingers for me .. |
| 06:08.07 | denon | 'cept when I'm doing asterisk |
| 06:08.14 | denon | then I have to be ... non-standard |
| 06:08.24 | russellb | cfg iax.conf ... sudo vim /etc/asterisk/$1 |
| 06:08.32 | russellb | asterisk works like that! |
| 06:08.35 | russellb | 1.4, anyway |
| 06:08.39 | denon | oh |
| 06:08.41 | denon | well .. |
| 06:08.43 | russellb | well! |
| 06:08.47 | denon | well .. um erm .. |
| 06:08.50 | russellb | pwned! |
| 06:08.53 | denon | yea see here's the deal .. |
| 06:08.56 | denon | I'm allergic to 1.4 |
| 06:08.59 | denon | I break out in hives |
| 06:09.03 | russellb | fair enough |
| 06:09.04 | denon | sorry, I'd love to run it .. |
| 06:09.09 | denon | but I cant .. doctor's orders |
| 06:09.12 | russellb | i don't mind :) |
| 06:09.26 | russellb | some people say it's working great ... others have problems |
| 06:09.28 | russellb | same old story |
| 06:09.31 | denon | I've begged and pleaded .. "please doc, I gotta run it .. russell will be mad .. " |
| 06:09.34 | denon | but he won't budge |
| 06:10.03 | denon | hehe nod .. it's the same with any major architectural shift .. you have to break a few things for the good of the structure |
| 06:10.16 | denon | something x86 should have done long ago (ie: alpha style) |
| 06:10.23 | DocHolliday | lol |
| 06:15.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:35.43 | arooni | anyonek now how to install the latest version |
| 06:35.45 | arooni | of asterisk? |
| 06:35.57 | antlers | https://www.riven.net/~moose/gallery/Asterisk_Nortel-i2004 |
| 06:36.07 | antlers | that brings me so much joy... anyone else using Nortel unistim based phones? |
| 06:38.17 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-188-233-116.hsd1.ca.comcast.net) |
| 06:38.37 | *** join/#asterisk lorinc (n=ang@pool-2896.adsl.interware.hu) |
| 06:41.37 | arooni | is this a problem: asterisk -vvvvvvvvvvgcT |
| 06:41.48 | arooni | Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory |
| 06:42.18 | arooni | ls |
| 06:42.21 | denon | asterisk isnt running |
| 06:42.25 | suma | have a problem with zaptel compilation |
| 06:42.25 | denon | fire up safe_asterisk first |
| 06:42.37 | arooni | denon: are you talking to me? |
| 06:42.37 | denon | oh, no -r in there |
| 06:42.43 | suma | zaptel: version magic '2.6.19-1.2911.6.5.fc6 SMP mod_unload 686 REGPARM 4KSTACKS ' should be '2.6.19-1.2911.6.5.fc6 SMP mod_unload 586 REGPARM 4KSTACKS ' |
| 06:42.49 | denon | well I was till I realized you weren't doing a 0r |
| 06:42.50 | denon | er -r |
| 06:43.15 | arooni | denon: well if i do asterisk -r i get: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 06:43.29 | arooni | denon: asterisk is running when i just do 'asterisk' |
| 06:43.39 | arooni | but i cant login into the daemon |
| 06:44.31 | denon | arooni: for that you do need -r |
| 06:44.38 | arooni | denon: um so what should i do? |
| 06:44.39 | denon | so, fire up asterisk however you like (preferrably with safe_asterisk), then run asterisk -r |
| 06:44.41 | arooni | it was working great befoer |
| 06:45.18 | denon | heh |
| 06:49.45 | *** part/#asterisk ggilbert (n=ggilbert@tinman.treke.net) |
| 06:54.01 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 06:55.33 | *** join/#asterisk Strom_M (n=strom@m055e36d0.tmodns.net) |
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| 07:02.18 | DocHolliday | anyone here used voipjet? |
| 07:03.20 | Strom_M | no, no one has ever used voipjet ever |
| 07:03.38 | DocHolliday | heh |
| 07:04.11 | DocHolliday | strom carlson? |
| 07:04.27 | Strom_M | thats me |
| 07:04.49 | DocHolliday | i have been on your site a lot :) always enjoyed the various recordings. |
| 07:04.57 | Strom_M | thanks :) |
| 07:05.30 | DocHolliday | its like a trip back through time.. your nick just made everything click. |
| 07:07.47 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:08.10 | DocHolliday | Strom_M, i know a lot of people in the Canadian HP scene |
| 07:14.13 | creature1 | morning |
| 07:16.02 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
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| 07:41.18 | thekidrio | anyone here use the 501 from polycom? I can make out going calls connect atleast at first, no voice gets transmitted. Sounds like a NAT problem but I am new to these phones |
| 07:43.03 | [TK]D-Fender | thekidrio: If its the only phone behind that NAT, you don't need to do anything |
| 07:46.08 | DocHolliday | [TK]D-Fender, have you used voipjet |
| 07:46.16 | [TK]D-Fender | DocHolliday: nope |
| 07:46.32 | DocHolliday | :( who do you use for north america termination? |
| 07:47.51 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 07:50.50 | [TK]D-Fender | DocHolliday: My employer :) |
| 07:50.59 | [TK]D-Fender | DocHolliday: And on fallback : All of my clients :) |
| 07:51.42 | DocHolliday | [TK]D-Fender, may I ask who your employer is? happy to bring him some business. |
| 07:52.18 | [TK]D-Fender | DocHolliday: They're a plumbing distribution company.... |
| 07:52.42 | DocHolliday | ahh.. well i don't have any problems with my toilets at the moment |
| 07:53.18 | DocHolliday | and when i do have a problem the plunger works best :> |
| 07:55.19 | thekidrio | [TK]D-Fender, yeah only the phone behind |
| 07:55.39 | [TK]D-Fender | thekidrio: Then do the usual * NAT things, and you should be fine |
| 07:56.36 | DocHolliday | [TK]D-Fender, do you make house calls? |
| 07:57.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:57.16 | [TK]D-Fender | DocHolliday: Only for my more... intimiate... "services" ;) |
| 07:58.06 | DocHolliday | ahhh, i don't swing that way |
| 08:00.58 | [TK]D-Fender | nor do I :) |
| 08:07.59 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 08:14.00 | *** join/#asterisk arooni (n=chatzill@dsl081-163-148.sea1.dsl.speakeasy.net) |
| 08:14.11 | arooni | are there any *free* SIP providers? |
| 08:14.49 | JT | free for what? |
| 08:15.12 | arooni | JT free for connecting calls to the PSTN |
| 08:17.13 | tzafrir | not really |
| 08:17.51 | tzafrir | there are those who will publish themselves as "free", but you'll notice it is for calls of up to a minute, or whatever |
| 08:19.04 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 08:19.05 | arooni | tzafrir: actuallyh |
| 08:19.10 | arooni | i only need to make calls < 1 minute |
| 08:19.14 | arooni | so that would work out great! |
| 08:19.38 | tzafrir | but be prepared to read some small prints |
| 08:20.04 | arooni | well |
| 08:20.09 | arooni | if they let me make all the calls i want < 1 minute |
| 08:20.12 | arooni | i'm a happy camper |
| 08:20.51 | tzafrir | here is one example: http://www.voipcheap.co.uk/ |
| 08:20.55 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 08:22.16 | arooni | i need access from asterisk |
| 08:22.20 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 08:22.24 | arooni | programtic access to their sip gateway thingy |
| 08:23.47 | tzafrir | they use sip |
| 08:24.06 | tzafrir | look around and you'll find instructions on how to connect with Asterisk to their service |
| 08:26.30 | tzafrir | http://www.voipcheap.co.uk/en/sipp.html |
| 08:27.25 | tzafrir | "GSMFR" is gsm, G.711 is either alaw or ulaw (or both), etc. |
| 08:30.37 | *** join/#asterisk foxxtrot (n=craig@c-67-185-241-244.hsd1.wa.comcast.net) |
| 08:31.14 | coppice | both? alternate samples with alternate compression? :-) |
| 08:31.45 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 08:31.55 | tzafrir | they write that they support "G.711" |
| 08:33.01 | tzafrir | BTW: I was looking into rebuilding oslec yesterday. I couldn't find mmx.h |
| 08:33.16 | tzafrir | should it be part of spandsp? |
| 08:33.47 | coppice | i threw it out, because there was a question mark over its GPL ancestry |
| 08:34.08 | coppice | you can grab a copy from older spandsp versions |
| 08:37.05 | yonahw-work | ~101 |
| 08:37.16 | jbot | rumour has it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
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| 10:36.15 | hellojoe | hi guys... |
| 10:36.21 | hellojoe | anyone out there? |
| 10:38.46 | yonahw-work | hellojoe: what's up |
| 10:39.16 | hellojoe | hey Yonahw-work... Any ideas/suggestions on how to implement a jajah like script? |
| 10:39.30 | hellojoe | enter 2 numbers through a web interface and connect the parties. |
| 10:39.48 | yonahw-work | sorry I have no knowledge, I am quite the newbie |
| 10:40.06 | hellojoe | okay... welcome to the club |
| 10:40.10 | hellojoe | enjoy asterisk |
| 10:40.13 | yonahw-work | thanks |
| 10:40.20 | yonahw-work | im loving it, lots to learn |
| 10:40.31 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 10:40.32 | hellojoe | yup |
| 10:42.23 | tzafrir | hellojoe, that's what Originate in the manager is for, or even plain call files |
| 10:43.25 | hellojoe | tzatrir: thanks a lot. I have not used the manager yet. Let me check it out. So far stayed away from AGI and stuck to the realtime apps |
| 10:43.46 | hellojoe | if you could point to samples to get started, it would help me burn the night |
| 10:43.57 | hellojoe | in a productive sorta way... |
| 10:44.11 | hellojoe | not sure where you are, I am in California and its 344AM :-( |
| 10:52.51 | yonahw-work | hellojoe: where in CA are you? I'm in Israel but used to live in Los Angeles |
| 10:53.15 | hellojoe | ahh... I am travelling to Tel Aviv next week.. I am in the bay area |
| 10:53.57 | yonahw-work | right on, I spent a couple of months in Berkely around 10 years ago |
| 10:54.51 | hellojoe | i am in Fremont if you still remember the neighborhood. Not that far from Berkley (south of oakland) |
| 10:54.59 | hellojoe | where in Israel are you ? |
| 10:55.09 | hellojoe | we could hang out over a beer |
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| 11:01.09 | adde | Whats the differens between extensions.ael and extensions.conf - which one is the one to edit? |
| 11:04.44 | jql | conf |
| 11:06.04 | adde | I stripped down the conf took a way lots of sample contexts... when i reload the asterisk i received lots of warnings....so i checked the ael and there where loads of contexts i removed from conf |
| 11:06.10 | adde | any idea why? |
| 11:07.38 | jql | It still loads the ael until you tell it not to |
| 11:07.58 | *** join/#asterisk friedrich| (n=friedric@e177243159.adsl.alicedsl.de) |
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| 11:13.25 | adde | ok... noload in modules... i get iut... thanks |
| 11:13.31 | adde | *get it |
| 11:25.42 | adde | I am using my gizmo callin as incoming... When i dial it i see the call in Asterisk debug. But it then drops it...In sip ive linked it to context incoming in extensions.conf and in the context i have Answer, Playback, Hangup.... Any ideas? |
| 11:26.41 | adde | exten => s,1,Answer() |
| 11:26.49 | adde | I suppose thats right? |
| 11:29.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:44.37 | reber | i got a problem with conference calls (more than 2 persons calling in the same room). Is it exten => 2010,1,Conference(54321) or exten => 2010,1,MeetMe(54321) ? |
| 11:49.08 | creature1 | reber: here you will find some useful examples : http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference |
| 11:50.37 | creature1 | adde: what number do you dial in your client? |
| 11:51.05 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 11:58.21 | *** join/#asterisk PhilMcKracken (n=chatzill@adsl-146-71-225.mia.bellsouth.net) |
| 11:59.11 | PhilMcKracken | Anyone here use kphone? |
| 12:01.05 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:02.49 | *** join/#asterisk LakeSolon (n=blake@64-83-205-22.dhcp.stcd.mn.charter.com) |
| 12:02.50 | reber | creativx, i got a problem with conference module : Both softphones connects ok, but nothing can be heard. Here is the configuration : meetme.conf : [rooms] conf => 54321 and in extensions : exten => 2010,1,Conference(54321) Any ideas ? |
| 12:04.43 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
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| 12:06.36 | reber | we are sunday aren't we ;) |
| 12:06.53 | PhilMcKracken | :-| |
| 12:09.38 | PhilMcKracken | :-O |
| 12:12.15 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 12:12.21 | PakiPenguin | hi |
| 12:12.46 | PhilMcKracken | :-( |
| 12:12.46 | PhilMcKracken | >-( |
| 12:12.46 | PhilMcKracken | :-[ |
| 12:12.46 | PhilMcKracken | :-/ |
| 12:12.47 | PhilMcKracken | ;-\ |
| 12:12.58 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
| 12:13.03 | PhilMcKracken | hi |
| 12:13.14 | PhilMcKracken | :-< |
| 12:13.29 | PhilMcKracken | :-$ |
| 12:13.44 | chrisknight | Hello |
| 12:13.50 | PhilMcKracken | do u have question? |
| 12:13.58 | reber | wtf |
| 12:14.16 | chrisknight | Not really, your spunky this morning |
| 12:15.49 | DrukenLPY | did everyone remember to change their clocks last night? |
| 12:16.52 | PhilMcKracken | yawn |
| 12:16.52 | PhilMcKracken | somebody ask questions |
| 12:16.52 | PhilMcKracken | I am needing some person who maybe has used kphone |
| 12:16.58 | *** join/#asterisk geejay101 (i=Tannenba@87.110.169.119) |
| 12:17.34 | PhilMcKracken | oh no. I'm soo late for temple |
| 12:17.46 | chrisknight | Actually I do, I have 1 'ring all' ring group (200). There are 2 phones in the group, 100, & 101. I have my DID set up to ring right into that group. If I call with my cell phone, both phones ring like they both should. If I call from a land line, only 101 rings... Whats up with that? |
| 12:18.26 | geejay101 | good afternoon |
| 12:18.29 | chrisknight | Clocks?!? Oh yes, thanks.. |
| 12:18.35 | DrukenLPY | are you calling from a foreign landline? |
| 12:18.38 | PhilMcKracken | good evening |
| 12:18.40 | *** join/#asterisk sm7xab (n=sm7xab@81-232-158-20-no28.tbcn.telia.com) |
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| 12:18.50 | chrisknight | nope, in ohio. |
| 12:19.11 | DrukenLPY | i mean a foreign landline from the asterisk system :) |
| 12:19.30 | PhilMcKracken | is the landline a rotary phone? |
| 12:19.30 | DrukenLPY | in otherwords, your not calling from a phone connected to asterisk |
| 12:20.01 | sm7xab | Hi! A lot of people complain that the outgoing sound during a telephone call is very low. Is there any possibility to increase the volume with Asterisk? |
| 12:20.12 | chrisknight | forrest, I went next door & used their land line (verizon) to test it |
| 12:20.30 | geejay101 | I am looking for a possibility to play an announcement to the CALLED party after the caller has hung up ? I cant find that. I only see possibility to play announcement in dial command after a call is connected. |
| 12:20.31 | DrukenLPY | sm7xab: what hardware are you using? |
| 12:21.08 | sm7xab | DrunkenLPY: Asterisk and Linksys PAP2. Haven't found anything in the PAP that I could decode as sound settings. |
| 12:21.09 | PhilMcKracken | I am needing some person who maybe has used kphone |
| 12:21.39 | geejay101 | sm7 look into zapata.conf rxgain txgain |
| 12:21.43 | DrukenLPY | sm7xab: well a pap2 is for the phones, i'm talking to interface with the pstn |
| 12:22.18 | sm7xab | DrunkenLP: No PSTN interface. Pure SIP. Sorry. Didn't understand the question... |
| 12:22.26 | DrukenLPY | chrisknight: i'd be checking your dialplan, probably an error in there |
| 12:22.51 | chrisknight | dialplan? For incoming calls? ok... |
| 12:22.54 | DrukenLPY | sm7xab: pure voip, and your getting complaints your too quiet?? speak up :) |
| 12:23.21 | geejay101 | sm7 - for pure sip adjust sound levels in voip adapater or softphone |
| 12:23.58 | sm7xab | DrunkenLPY: Well, that's one solution. The difference is very noticeable if I call my SIP line and my Cell-phone or someone else. |
| 12:25.30 | geejay101 | sm7 - your cellphone also has a volume adjustment |
| 12:25.45 | PhilMcKracken | KPhone's DMTF doesn't work. I hear the tones, but they don't get sent to the other guy. WTF? |
| 12:26.08 | DrukenLPY | PhilMcKracken: codec |
| 12:26.19 | *** join/#asterisk Witwolf (n=carel@c3-26-1.eno.dial.mweb.co.za) |
| 12:26.21 | sm7xab | geejay101: I've been trying to adjust the levels in my PAP adapters. Haven't found it though. That's why I'm trying to adjust it using Asterisk. |
| 12:26.36 | PhilMcKracken | I tried all the options, some of which are suppsed to be supported |
| 12:26.47 | Witwolf | Hi, I am experiencing problems with on hold music only on my FXO when I make outgoing or get incoming calls. It is allmost like static or noise which come thru now and then. I do not experience problems when staying with TCP/IP, only on the FXO, the FXS does not have the problem. Could anyone help? |
| 12:27.11 | sm7xab | geejay101: Using one phone I've tried calling different places and only my SIP phone is low-volume :-( |
| 12:28.38 | PhilMcKracken | I don't think I can add codecs to KPhone |
| 12:28.41 | geejay101 | the pap adapters can be adjusted for gain. Somewhere in the extended menues if I recall correctly |
| 12:28.52 | DrukenLPY | let me browse me pap2's config, i'm not sure if there's a volume thing in it or not.... |
| 12:29.12 | PhilMcKracken | And the thing is, I've tried Ekiga, which doesn't work (I get |
| 12:29.47 | PhilMcKracken | 'security check failed') |
| 12:29.56 | PhilMcKracken | And I've tried twinkle which has a DTMF timeout known bug |
| 12:29.57 | geejay101 | The pap2 adapters have an amazing variety of options - make sure to check also they are configured for you country / line impedance |
| 12:30.03 | PhilMcKracken | So I'm running out of options |
| 12:30.35 | PhilMcKracken | :-[ |
| 12:30.56 | geejay101 | I am looking for a possibility to play an announcement to the CALLED party after the caller has hung up ? |
| 12:31.20 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 12:31.28 | sm7xab | Hmmm.. The last part is interesting. Think I've forgotten to configure the PAP correctly. I'll check that. |
| 12:32.45 | DrukenLPY | geejay101: wouldn't the pstn gateway change that? oh... for the phone that's connected to it... whoops |
| 12:34.39 | geejay101 | Druken, I dont understand what you mean. the two legs of my call are both SIP. I want that when the caller hangs up that I play a message to the called party. Is there any way to "catch" the hangup of the caller in the dialplan ? |
| 12:35.53 | geejay101 | The dial command has the g option - g: When the called party hangs up, exit to execute more commands in the current context. |
| 12:36.00 | sm7xab | Hmmm. Just found "FXS port input/output gain". Changed it from -3 to 0. Should make stuff a bit louder I guess.. |
| 12:36.53 | PhilMcKracken | What do people use Asterisk for if they still have to pay for a PSTN gateway service? |
| 12:37.02 | geejay101 | Basically I want something similar like the g dial option - execute some action for the callee |
| 12:37.11 | geejay101 | congrats sm7 |
| 12:37.45 | *** join/#asterisk Arth_72 (n=Arturo@88-149-232-233.f5.ngi.it) |
| 12:37.47 | Arth_72 | hi |
| 12:37.59 | sm7xab | geejay101: Stuff is very nicely hidden. I've been looking at this some time. Could also be that I'm getting blind :-( |
| 12:38.05 | Witwolf | Does anyone know what might cause a staggered & noisy MOH only on a FXO port? |
| 12:38.25 | DrukenLPY | geejay101: i dunno... afaik there's no way to hang up and still have the callee into the system |
| 12:39.07 | geejay101 | sm7 - I discovered only after some months that my paps were configured for the wrong country - PAP2 EU version was factory set to US line impedance. |
| 12:39.21 | PhilMcKracken | :-S |
| 12:39.59 | geejay101 | Druken, that sounds discouraging. |
| 12:40.38 | DrukenLPY | sorry... :) |
| 12:41.28 | DrukenLPY | you could do the call reverse, by using a call file..... |
| 12:41.46 | DrukenLPY | so that YOUR the callee... |
| 12:42.08 | DrukenLPY | er.. called |
| 12:42.17 | PhilMcKracken | DrukenLPY: grammar fine! |
| 12:42.28 | *** join/#asterisk lvnatic (i=lvnatic@84.236.21.135) |
| 12:42.39 | lvnatic | hi |
| 12:42.55 | PhilMcKracken | bye |
| 12:48.03 | PhilMcKracken | B-) |
| 12:49.22 | PhilMcKracken | :o) |
| 12:50.56 | DrukenLPY | oh sweet shit, supposed to be +12 on tuesday |
| 12:51.07 | PhilMcKracken | :-D |
| 12:51.17 | DrukenLPY | that's driven with the window open weather |
| 12:51.30 | PhilMcKracken | What do people use Asterisk for if they still have to pay for a PSTN gateway service? |
| 12:51.55 | DrukenLPY | meaning? |
| 12:52.21 | PhilMcKracken | THe only reason I'd want a PBX is for PSTN service |
| 12:52.33 | PhilMcKracken | considering that there's tons of free stuff out there |
| 12:52.47 | PhilMcKracken | if all you want is VoIP |
| 12:53.11 | masked | get a tdm or somethin? |
| 12:53.12 | DrukenLPY | well, i use asterisk for my telephone network.. and basically to connect all my phones together |
| 12:53.42 | DrukenLPY | i can have phones ring in several diffrent places for my business |
| 12:54.37 | PhilMcKracken | I see |
| 12:54.59 | PhilMcKracken | So it's for managing one's own network of phones |
| 12:55.23 | JT | PhilMcKracken: it allows you to build a pbx in software |
| 12:55.32 | JT | as opposed to buying a proprietary one |
| 12:55.44 | DrukenLPY | which would cost a fucken fortune |
| 12:56.48 | DrukenLPY | my local hospital just went to a voip telephone system |
| 12:57.00 | PhilMcKracken | Has anyone ever had trouble with libsndfile? |
| 12:57.10 | PhilMcKracken | cool |
| 12:57.48 | DrukenLPY | i'm waiting to see a doctor with a wifi cordless :) |
| 12:58.40 | Arth_72 | hi jt ... i got asterisk working and running .... with my isdn hfc card (thanks to tzafrir that help me out yesterday with zapata and zaptel config) |
| 12:58.50 | PhilMcKracken | Well I'm waiting to see a softphone that isn't broken |
| 13:00.00 | Arth_72 | phil: which softphones r u using ? |
| 13:01.12 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 13:01.39 | DrukenLPY | anyone know if i can split a pri, half voice and half data and still use it with asterisk?? hehehe |
| 13:01.51 | PhilMcKracken | Ekiga, kphone, twinkle |
| 13:02.36 | Arth_72 | anyone know if i can compile spandsp 0.0.3 with asterisk 1.2.14 ? or do i have to use 0.0.2pre26 ? |
| 13:02.54 | Arth_72 | and which kind of problem do u have phil ? |
| 13:02.55 | JT | PhilMcKracken: yeah softphones seem to suck |
| 13:03.25 | JT | DrukenLPY: afaik, yes |
| 13:03.41 | JerJer | wholy daylight savings time mess batman |
| 13:03.56 | DrukenLPY | seems the te110p will do it... |
| 13:04.29 | DrukenLPY | JerJer: why a mess? all my systems changed by themselves |
| 13:05.00 | PhilMcKracken | They shoulda called it tinkle |
| 13:05.34 | JerJer | DrukenLPY: i am already getting pestered about IP Phones being one hour off |
| 13:05.42 | JerJer | grand stream |
| 13:05.47 | JerJer | Sipuras |
| 13:05.49 | PhilMcKracken | Arth_72: I With Ekiga, I can't even place a call. I get 'security check failed' |
| 13:05.57 | DrukenLPY | np ntpd? |
| 13:06.02 | Arth_72 | phil: ouch |
| 13:06.08 | DrukenLPY | s/np/no |
| 13:06.30 | PhilMcKracken | With Kphone, others I call can't get the DTMF phone |
| 13:06.51 | PhilMcKracken | With Kphone, others I call can't get the DTMF -tone- |
| 13:07.01 | adde | Ive got a call in number. Ive registered it in sip.conf. So now when i dial it asterisk receives the call. Ive created an softphone extension. but i cant get it to dial the extension. |
| 13:08.06 | PhilMcKracken | with twinkle, DTMF times out after like 30 seconds and you can't send tones after that. It's weird |
| 13:08.56 | PhilMcKracken | I'm trying to compile a newer version of twinkle and it's whining about not having libsndfile even though I do |
| 13:10.05 | Arth_72 | phil: did u tried linphone ? |
| 13:11.29 | PhilMcKracken | no |
| 13:13.57 | PhilMcKracken | anyone familiar with 'asteriskguru.com ? |
| 13:14.10 | PhilMcKracken | idefisk? |
| 13:17.52 | masked | yep |
| 13:17.55 | masked | works a treat |
| 13:18.04 | masked | cool filter options |
| 13:20.15 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:23.13 | zoa | masked: in max a week we will release a complete new version |
| 13:23.18 | zoa | a looooot better |
| 13:23.26 | zoa | with sip support too |
| 13:24.20 | masked | orright |
| 13:24.22 | masked | more lines? |
| 13:25.18 | zoa | you want more lines ? |
| 13:25.28 | zoa | 6 is not enough ? |
| 13:25.41 | masked | hehe |
| 13:25.59 | masked | nah i don't care |
| 13:26.23 | masked | a pocketpc port would be awesome tho ;) |
| 13:26.29 | masked | with good iax support |
| 13:26.34 | PhilMcKracken | open source? |
| 13:26.38 | zoa | nopez |
| 13:26.55 | masked | video support |
| 13:26.59 | zoa | not yet |
| 13:27.03 | PhilMcKracken | then nevermind |
| 13:27.04 | masked | 100% serious about that. |
| 13:27.09 | zoa | but encryped iax |
| 13:27.10 | zoa | :) |
| 13:27.14 | masked | mmm |
| 13:27.14 | zoa | i dont like video |
| 13:27.17 | masked | yeah |
| 13:27.18 | masked | but |
| 13:27.21 | masked | unlike you |
| 13:27.22 | zoa | but we will add it afterwards |
| 13:27.24 | zoa | first messaging |
| 13:27.27 | masked | the rest of the world does |
| 13:27.33 | masked | yeah sure |
| 13:27.41 | masked | im would be cool |
| 13:27.52 | masked | especially if it were compatible with freshtel's firefly IM. |
| 13:27.54 | zoa | i first want to support the "hint" in asterisk |
| 13:28.08 | zoa | freshtel's firefly is something proprietary |
| 13:28.11 | PhilMcKracken | I feel the victim of astroturfing |
| 13:29.37 | masked | zoa: what then... xmpp? |
| 13:30.39 | zoa | well, for the IM on iax, i will try to find a way to be compatible with the main asterisk (meaning i will have to talk to the guy maintaining iax in asterisk) |
| 13:30.42 | zoa | becuase its not there yet |
| 13:30.56 | zoa | for sip, i guess there is some standard for it, in the future we should do xmmp too i guess |
| 13:31.48 | masked | yeah |
| 13:31.48 | PhilMcKracken | What happened to everyone? |
| 13:31.52 | masked | i think jabber is the way to go |
| 13:32.01 | zoa | yeah, well if its only messaging yes |
| 13:32.10 | zoa | the problem is that you want it to work with the ITSP's too |
| 13:32.23 | zoa | so that the customer only needs 1 account with them |
| 13:32.28 | zoa | for voip + messaging |
| 13:33.22 | PhilMcKracken | Arth_72: hello? |
| 13:34.27 | masked | not a big deal for the itsp to have a jabber server |
| 13:34.39 | masked | especially if they don't already have another solution |
| 13:34.44 | masked | saves coding another up |
| 13:34.57 | masked | less of a burden too |
| 13:35.26 | zoa | true |
| 13:35.38 | zoa | but ideally it should be possible to combine it with asterisk |
| 13:35.51 | zoa | for asterisk to handle all types of channels |
| 13:35.56 | zoa | they are working towards it already |
| 13:36.10 | masked | there is xmpp/asterisk work going on already. |
| 13:36.13 | Arth_72 | phil:sorry .. was afk .. what do u need ? |
| 13:36.19 | zoa | yeah i know |
| 13:37.00 | masked | and jabber is already available, makes for an easier migration if another itsp wants to join in |
| 13:37.05 | PhilMcKracken | I good softphone |
| 13:42.50 | *** join/#asterisk Corydon76-home (i=red@pdpc/supporter/sustaining/Corydon76-home) |
| 13:42.50 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:52.47 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 13:57.25 | gambolputty | TZ_TIMEZONE=EST+5EDT,M3.2.0/2,M11.1.0/2 |
| 14:01.29 | *** join/#asterisk darken_darken (n=marco@97.137.77.83.cust.bluewin.ch) |
| 14:02.35 | zoa | olle! |
| 14:02.54 | oej | ...sorry, busy |
| 14:09.19 | Arth_72 | in spandsp dl dir there is a codec_g726.c file and a patch for the makefile .. do i have to use them ? in the web pages there r no info about these files .... |
| 14:15.45 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
| 14:18.32 | PhilMcKracken | I'm building a softphone for you. No ordinary softphone will do |
| 14:18.48 | digime | CAN anyone recommend a toll free DID provider (USA)? |
| 14:19.34 | PhilMcKracken | By which u mean a provider of toll free DID's |
| 14:20.06 | coppice | Arth_72: in which dl directory? |
| 14:20.17 | zoa | yo yo coppice |
| 14:20.17 | PhilMcKracken | that u pay for |
| 14:21.38 | digime | Yes I mean an 800# DID per minute |
| 14:21.49 | Arth_72 | in spandsp web site |
| 14:22.21 | PhilMcKracken | I got some junk in my trunk |
| 14:22.35 | coppice | Arth_72: where exactly do you mean? |
| 14:23.18 | PhilMcKracken | digime: SIP? |
| 14:24.03 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:25.58 | Arth_72 | in the spandsp web site ... in the 0.0.2 version there r thosefiles ... |
| 14:26.18 | Arth_72 | i would like to know if i have to use them in asterisk compilation |
| 14:26.39 | Arth_72 | here to be exact, http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/ |
| 14:26.47 | coppice | Arth_72: if you won't answer the question, you won't get any assistance |
| 14:27.02 | Arth_72 | i answered :) |
| 14:27.14 | *** join/#asterisk Corydon76-home (i=red@pdpc/supporter/sustaining/Corydon76-home) |
| 14:27.14 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 14:27.25 | coppice | ah, OK. Ignore that. I forgot that was left there are a test version |
| 14:27.39 | Arth_72 | oh, ok, thank you :) |
| 14:27.53 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:29.44 | zoa | coppice, how does g726 sound like ? |
| 14:29.53 | Strom_C | good morning |
| 14:29.55 | zoa | i mean compared to other codecs |
| 14:30.05 | Strom_C | zoa: g726 sounds almost the same as g711 |
| 14:30.12 | *** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com) |
| 14:30.25 | zoa | and it uses half the bandwitdh, correct ? |
| 14:30.25 | coppice | its easy to try for yourself. its better than things like G.729, but its quite a high bit rate |
| 14:30.45 | Arth_72 | coppice .. do i have to use 0.0.2pre26 or can i use 0.0.3 ? does the 0.0.3 compile with asterisk 1.2.14 ? |
| 14:30.45 | coppice | where almost the same == nothing like :-) |
| 14:30.50 | zoa | and cpu wise, is it heave on cpu ? |
| 14:31.09 | zoa | heavy |
| 14:31.13 | coppice | its light on CPU usage compared to the lower bit rate codecs |
| 14:31.22 | zoa | and compared to gsm ? |
| 14:31.30 | Strom_C | i dunno...my philosophy on codecs is that you should use the highest bit rate codec you can get away with |
| 14:31.33 | coppice | less than GSM |
| 14:31.38 | zoa | could i easily do 120 of them on a standard p4 ? |
| 14:31.46 | zoa | 120 enc/dec's |
| 14:31.55 | Strom_C | zoa: type "show translation" at the CLI |
| 14:32.06 | zoa | i dont have g726 here :) |
| 14:32.09 | coppice | Strom_C: but at 32kbps you can get near Hi-Fi results with other codecs |
| 14:32.25 | zoa | isnt speex better at 32kbps ? |
| 14:32.34 | coppice | a lot better |
| 14:32.56 | coppice | G.726 sounds like G.711 with a sock stuck in the earpiece :-) |
| 14:33.02 | Strom_C | coppice: this is telephony, not music distribution |
| 14:33.25 | coppice | and people are just too used to crappy phone calls |
| 14:33.37 | Strom_C | that I agree with...g.711 for the win :) |
| 14:33.40 | coppice | really good voice is actually more demanding than music |
| 14:33.52 | *** join/#asterisk pat_lehem (i=lehem@bzq-88-154-176-93.red.bezeqint.net) |
| 14:34.00 | coppice | G.711 sounds lousy |
| 14:34.36 | Strom_C | yeah, but if you're going to use the PSTN at any point in the call, you're going to be using G.711 anyway |
| 14:34.52 | coppice | true, but the PSTN is lousy |
| 14:35.14 | coppice | one of the key goals of ISDN was to dump G.711 and go wideband. it never took off |
| 14:35.52 | Arth_72 | <PROTECTED> |
| 14:36.22 | pat_lehem | hi all. Does anyone have an idea about how to improve IAX call quality? I have excellent quality on my Zap and SIP channels, but really bad and jittery audio on IAX. I've tried both jitterbuffers (old and new), and even turned off the jitterbuffer completely. |
| 14:36.35 | pat_lehem | also, I turned off trunking and iax provisioning |
| 14:36.42 | pat_lehem | but still, nothing improves the quality |
| 14:36.43 | PhilMcKracken | use sip |
| 14:36.49 | zoa | how much delay do you have on the line ? |
| 14:36.52 | Strom_C | pat_lehem: which version of asterisk? |
| 14:37.17 | pat_lehem | It worked really good in 1.0.X, the problem started when I upgraded to 1.2.16 |
| 14:37.26 | coppice | Arth_72: I think 0.0.2pre26 works with 1.2.14 if you use the test versions of app_rxfax and app_txfax in the snapshots directory. I haven't done anything to make 0.0.3prexx work with * |
| 14:37.35 | zoa | a1.2.16 does not have the new jitter buffer i think |
| 14:37.36 | pat_lehem | well, to 1.2.14, but I later upgraded again to 1.2.16 |
| 14:37.43 | pat_lehem | really?? |
| 14:37.50 | pat_lehem | it says in the code that it does |
| 14:37.50 | coppice | sorry, works with 1.4.x |
| 14:38.46 | zoa | grep the code for attractel, or scx_jb |
| 14:38.55 | zoa | if those strings are not there, the jb is not there |
| 14:39.21 | pat_lehem | is there no jb at all? or just the new one? |
| 14:39.49 | pat_lehem | 1.0.x had one, I was using it until a month ago |
| 14:40.03 | zoa | just not the new one |
| 14:40.21 | Arth_72 | ok, coppice, thatks for the infos |
| 14:40.53 | pat_lehem | in the code there is an option to compile with new or old jb, and I tried both. So there must be some difference between them, there are also all sorts of new configuration options |
| 14:41.09 | Arth_72 | coppice: anyway i use 1.2.14 actually .. bristuff doesn't work with 1.4.x or i didn't find the patch for it |
| 14:44.46 | Arth_72 | bye |
| 14:45.22 | pat_lehem | so, basically you're saying that in order to use IAX with good audio quality, I need to either use 1.0.x or 1.4.x? |
| 14:45.45 | PhilMcKracken | dont leave |
| 14:46.03 | PhilMcKracken | or use sip |
| 14:46.39 | pat_lehem | it means moving 20K users to SIP clients instead of their already installed IAX clients |
| 14:46.48 | pat_lehem | this is a rather problematic scenario |
| 14:48.30 | zoa | if you dont need a jb on sip, you shouldnt have to use one on iax either |
| 14:48.34 | zoa | it has to be something else |
| 14:48.45 | zoa | do you see anything on the CLI ? |
| 14:49.03 | Strom_C | pat_lehem: ive not had problems with IAX on 1.2 ever |
| 14:49.31 | zoa | are you using realtime ? |
| 14:54.11 | pat_lehem | I think the jb in SIP is part of the RTP impelmentation |
| 14:54.17 | pat_lehem | yes, I am using realtime |
| 14:54.23 | pat_lehem | but for sip as well |
| 14:54.29 | zoa | ok, get rid of realtime for iax |
| 14:54.34 | zoa | it might fuck things up |
| 14:54.38 | zoa | it used to be single threaded |
| 14:54.45 | zoa | and then they made some kind of multithreading |
| 14:54.49 | zoa | but i dont think its very reliable |
| 14:55.07 | zoa | so if one of the guys is registering, and the db doesnt reply within 20ms |
| 14:55.13 | adde | Please can somone help me, after 8 hours and still no luck im getting desperate. All i wich to achieve is for the asterisk to divert the incoming call (incoming worksfine i see the connection) to a Gizmo user... exten => 2,2,Dial(SIP/gizmouser@proxy01.sipphone.com,130,t) Iv added that extension, but Asterisk just drops the call... :S |
| 14:55.16 | zoa | all audio packets from other streams could be on hold |
| 14:55.35 | pat_lehem | actually, we started without realtime at all, we just added realtime because we thought the iax2 reload was killing the audio |
| 14:55.36 | zoa | and lost |
| 14:55.45 | zoa | aha |
| 14:55.51 | zoa | we made that new jitter buffer |
| 14:55.56 | zoa | its not just for RTP |
| 14:55.57 | pat_lehem | the system is a realtime system, so it's either using realtime, or a reload every minute |
| 14:56.00 | zoa | its in channel.c |
| 14:56.51 | pat_lehem | oh, the one I saw (in 1.2.16) is in jitterbuf.c and chan_iax2.c is using it |
| 14:57.15 | zoa | look at the header, does it say copyright attractel ? |
| 14:58.12 | zoa | thats the old jitter buffer |
| 14:58.14 | zoa | not the newo ne |
| 14:58.34 | *** join/#asterisk silvertip257 (n=chatzill@c-71-230-114-9.hsd1.pa.comcast.net) |
| 14:59.49 | pat_lehem | no reference to attractel anywhere in the code. |
| 15:00.10 | pat_lehem | It's the one from 1.2, there was a different one before, in 1.0.x |
| 15:00.12 | zoa | so its the old jitter buffer |
| 15:00.13 | zoa | yes |
| 15:00.23 | zoa | well in version 1.4 ours is included |
| 15:00.28 | zoa | but thats not going to help you |
| 15:00.30 | coppice | the jb in 1.2.x is an IAX2 specific one that Steve Kann donated |
| 15:00.31 | zoa | with your problem |
| 15:00.36 | zoa | its not a jitter problem |
| 15:00.40 | zoa | im sure |
| 15:01.01 | pat_lehem | I thought so at first, but after trying SIP, I don't know |
| 15:01.11 | zoa | do you hear problems with sip ? |
| 15:01.14 | pat_lehem | no |
| 15:01.26 | zoa | try my realtime suggestion |
| 15:01.30 | zoa | disable it and do not reload |
| 15:01.33 | zoa | see if you have problems |
| 15:01.37 | zoa | if you dont, you know where the problem is |
| 15:02.02 | pat_lehem | I was looking at bandwidth and cpu and stuff, and then I just tried SIP for fun, and it surprisingly worked. |
| 15:02.07 | pat_lehem | I'll try that now. |
| 15:02.33 | pat_lehem | should I disable RT only for iax users/peers, or for the dialplan as well?? |
| 15:02.43 | zoa | just for the iax i think |
| 15:02.54 | pat_lehem | that's easy, I'll try now. |
| 15:03.07 | zoa | if you have iax.conf in the RT, it would hit the database for every authentication / Register |
| 15:03.37 | zoa | also, you might want to check how much registers a second you have |
| 15:03.45 | pat_lehem | I know, that's what I wanted it to do. I also use realtime in SIP |
| 15:04.07 | zoa | there used to be a bug in iaxclient that would create an avalanche of registers |
| 15:04.10 | zoa | you could sniff those |
| 15:04.14 | pat_lehem | yes, |
| 15:04.22 | zoa | iaxtel used to have a similar problem |
| 15:04.27 | pat_lehem | I'll just turn on IAX debug. |
| 15:04.58 | zoa | im not sure iax debug will show you that many registers, it might ignore them if they come too fast |
| 15:05.10 | pat_lehem | or tcpdump them. but what bothers me most is the fact it worked great in 1.0.x, I'll try turning off RT |
| 15:05.27 | zoa | i think it will be RT |
| 15:06.03 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 15:06.39 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 15:07.06 | reber | i got a problem with app_conference module : Both softphones connects ok, but nothing can be heard. Here is the configuration : meetme.conf : [rooms] conf => 54321 and in extensions : exten => 2010,1,Conference(54321) Any ideas ? |
| 15:08.46 | zoa | its not app_meetme ? |
| 15:09.21 | reber | zoa, there are 2 modules. Seems like app_meetme need special hardware. |
| 15:09.45 | zoa | or ztdummy or so |
| 15:10.01 | zoa | app_conference is not included with asterisk, is it ? |
| 15:10.14 | zoa | if its not it might not be updated for the asterisk version you are using |
| 15:10.21 | Strom_C | you can run app_meetme with the ztdummy module |
| 15:10.25 | zoa | if you dial from 1 softphone to another, is it working ? |
| 15:11.02 | reber | Strom_C, ok. Do you use it yourself ? |
| 15:11.18 | Strom_C | yes |
| 15:11.40 | reber | Strom_C, and does it work between 3 or more softphones ? |
| 15:11.59 | zoa | yes it does |
| 15:12.10 | Strom_C | yes |
| 15:13.05 | silvertip257 | I'm doing a research project for one of my networking classes on Asterisk VoIP PBX, anybody have any good resources (besides asterisk.org and their wiki) ? |
| 15:13.07 | silvertip257 | thanks |
| 15:13.17 | zoa | www.asteriskguru.com (my site) |
| 15:13.25 | zoa | www.nerdvittles.org (or .com) |
| 15:13.28 | zoa | euhm |
| 15:13.28 | silvertip257 | your site! for real |
| 15:13.30 | zoa | there are some more |
| 15:13.46 | zoa | asteriskdocs.org |
| 15:13.50 | zoa | voip-info.org |
| 15:13.56 | zoa | soft-switch.org for fax info |
| 15:13.58 | yonahw-work | zoa: thanks for all the work you did on your site it has been quite informative at times |
| 15:14.05 | zoa | thanks! |
| 15:14.07 | silvertip257 | faxing with asterisk?!? |
| 15:14.26 | *** join/#asterisk thansen|laptop (n=thansen@160.7.243.19) |
| 15:14.32 | Sweeper | ?!?!? |
| 15:14.36 | silvertip257 | lol |
| 15:14.41 | silvertip257 | hah ha |
| 15:14.49 | silvertip257 | ok thank you zoa |
| 15:14.51 | zoa | asterisk can (be made to) fax, sms, text to speech, speech recognition, radio, etc |
| 15:14.52 | Sweeper | ~book |
| 15:15.01 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:15.01 | creature1 | silvertip257: it's possible |
| 15:15.01 | silvertip257 | wow |
| 15:15.02 | Sweeper | jbot: ~book |
| 15:15.09 | Sweeper | arg, the lag~ |
| 15:15.11 | zoa | ah yes, there are several books |
| 15:15.21 | silvertip257 | I'm not going to buy a book just yet |
| 15:15.26 | zoa | you can download one |
| 15:15.31 | Sweeper | it's free \o\ |
| 15:15.40 | silvertip257 | oh where do i get it? |
| 15:15.45 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 15:15.47 | zoa | http://asteriskguru.com/downloads/AsteriskTFOT.zip |
| 15:15.53 | silvertip257 | oo thanks |
| 15:15.55 | zoa | dunno if that is the latest version |
| 15:15.59 | silvertip257 | ok |
| 15:16.05 | zoa | you should be able to find it on oreilly or asteriskdocs too |
| 15:16.11 | silvertip257 | all right |
| 15:16.22 | silvertip257 | what distros do you guys use for desktop OSes? |
| 15:16.25 | silvertip257 | just wondering |
| 15:16.29 | zoa | windows :) |
| 15:16.40 | silvertip257 | awww :( |
| 15:16.42 | coppice | CP/M |
| 15:16.47 | zoa | some here use ubuntu too |
| 15:17.01 | silvertip257 | zoa - not going ubuntu or redhat ? |
| 15:17.02 | adde | If i have set up teh sip.conf correctly so incoming sip calls are sent to Default context, shouldnt this work or am i totally wrong: [Default] |
| 15:17.03 | adde | exten => 1,s,Dial(SIP/gizmouser@proxy01.sipphone.com,130,t) |
| 15:17.06 | zoa | me, no |
| 15:17.08 | silvertip257 | :( |
| 15:17.11 | zoa | too much stuff i need on windows |
| 15:17.13 | silvertip257 | poor zoa |
| 15:17.17 | silvertip257 | oh I see |
| 15:17.18 | adde | last lines in extensions.conf |
| 15:17.48 | reber | does anyone here use app_conference module ? Is it buggy ? |
| 15:17.59 | adde | Shouldnt that divert the call to the Gizmo user |
| 15:18.04 | zoa | it should work, (if its for the right asterisk) |
| 15:18.10 | Strom_C | adde: "s" is not a priority |
| 15:18.12 | zoa | but i think it will scale less than the app_meetme |
| 15:18.25 | zoa | but could be completely wrong with that :) |
| 15:18.41 | thansen|laptop | can someone give some direction on some good voip PSTN providers? |
| 15:18.46 | creature1 | adde: S is just things to do on 'startup' |
| 15:18.51 | thansen|laptop | compatible with asterisk |
| 15:18.51 | creature1 | s even. |
| 15:19.00 | *** join/#asterisk Dimik (n=Dimik@unaffiliated/dimik) |
| 15:19.09 | adde | so how shall i write the line 1,1 ? |
| 15:19.30 | creature1 | adde: you could f.ex write 333 => 1,1,Dial... |
| 15:19.30 | Strom_C | adde: well, you said asterisk is just dropping the call, right? |
| 15:19.39 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 15:19.46 | adde | yes |
| 15:19.48 | creature1 | adde: and when dialing 333 that one will be executed |
| 15:19.51 | Strom_C | what does the CLI say when you place a call through it? |
| 15:20.08 | Strom_C | creature1: no, the syntax is exten => 333,1,something |
| 15:20.14 | creature1 | adde: if the user dialing 333 uses that extension where the context is specified |
| 15:20.15 | adde | strom: a lot |
| 15:20.27 | creature1 | Strom_C: ah, my misstake. too used to AEL :) |
| 15:20.33 | Strom_C | adde: pastebin it |
| 15:20.35 | Strom_C | ~pb |
| 15:20.39 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:20.40 | adde | strom: i pmsg you |
| 15:20.41 | Strom_C | no |
| 15:20.43 | Strom_C | use pastebin |
| 15:20.51 | creature1 | adde: let everyone help instead, will give you more feedback. |
| 15:21.02 | adde | Where do i paste? |
| 15:21.09 | adde | whats paste bin? |
| 15:21.14 | creature1 | adde: at the url jbot msged |
| 15:21.14 | Strom_C | i just had the bot give you a link |
| 15:21.48 | adde | Didnt get anything from bot :S |
| 15:21.52 | creature1 | adde: it's a way to publish text, so you can show us your config file without having to paste all of the lines in this chan. |
| 15:21.56 | creature1 | adde: http://pastebin.ca |
| 15:22.26 | *** join/#asterisk PhilMcKracken (n=chatzill@adsl-144-156-25.mia.bellsouth.net) |
| 15:23.57 | adde | http://pastebin.ca/390555 |
| 15:24.23 | Strom_C | no no no |
| 15:24.30 | Strom_C | dont turn on sip debug |
| 15:24.30 | PhilMcKracken | what about my needs |
| 15:24.30 | creature1 | adde: the config files please |
| 15:24.56 | creature1 | adde: sip.conf and extensions.conf |
| 15:25.32 | creature1 | adde: and remember to remove sensitive passwords |
| 15:28.39 | adde | http://pastebin.ca/390564 |
| 15:30.02 | creature1 | adde: what happens if you dial 100 with your gizmo? |
| 15:30.42 | adde | i dont have a gizmo |
| 15:31.18 | creature1 | adde: ok, what happens if you dial 100 logged in as the user mygizmo? |
| 15:32.58 | adde | call failed: not found |
| 15:33.31 | creature1 | adde: type dialplan show in console |
| 15:33.49 | creature1 | adde: have you forgot to reload the config after updating it? |
| 15:34.24 | adde | no |
| 15:34.54 | creature1 | adde: try this.. instead of dial.. exten => 100,1,Answer, exten => 100,2,Playback(tt-monkeys), exten => 100,3,Hangup |
| 15:34.59 | adde | http://pastebin.ca/390576 <- DIAPLAN SHOW |
| 15:36.14 | *** join/#asterisk friedrich| (n=friedric@e177243159.adsl.alicedsl.de) |
| 15:37.52 | *** join/#asterisk newmedian (n=newmedia@TOROON63-1168100004.sdsl.bell.ca) |
| 15:38.35 | *** join/#asterisk darken_darken (n=marco@97.137.77.83.cust.bluewin.ch) |
| 15:39.42 | adde | still not working... |
| 15:39.43 | adde | :S |
| 15:39.59 | *** join/#asterisk saftsack (n=oliver@p54A7EB49.dip.t-dialin.net) |
| 15:40.42 | *** join/#asterisk heart (n=zippetto@lugbari/people/heart) |
| 15:47.50 | adde | Hmm no i got the monkeys... weird... the only thing i did was take away all contexts except default - and removed the password from [mygizmo] |
| 15:48.59 | creature1 | hmm maybe because (but i dont thinks that's why) you had exten 100 both in default and gizmo, never tried that. No idea if that works. |
| 15:49.07 | creature1 | though it should mao :) |
| 15:49.18 | creature1 | imo even |
| 15:51.07 | adde | Thanks a lot for your help... |
| 15:51.18 | adde | Now i got through :) |
| 15:51.21 | creature1 | great :) |
| 15:52.09 | adde | Do you know a simple example of a short menu with 3 options? all ive found are so advanced i wanna take one step at a time so i really learn... |
| 15:56.20 | *** part/#asterisk pat_lehem (i=lehem@bzq-88-154-176-93.red.bezeqint.net) |
| 15:58.02 | russellb | adde: http://pastebin.ca/390600 |
| 16:01.24 | adde | thanks m8 |
| 16:03.34 | russellb | you're welcome |
| 16:03.42 | russellb | i took it out of configs/extensions.conf.sample |
| 16:03.59 | zoa | yo yo russel |
| 16:04.20 | zoa | working sundays again ? :) |
| 16:04.22 | russellb | hey zoa ! |
| 16:04.28 | russellb | hehe, you know it |
| 16:04.30 | russellb | i'm addicted ... |
| 16:04.37 | zoa | yeah same here |
| 16:06.11 | bulle | is there some way to kill "zombie" sip connections ? |
| 16:06.40 | bulle | if i phone some sip url, that doesnt answear, the connection stays forever, holding a rtp port etc |
| 16:07.10 | bulle | core show channels show 0 channels |
| 16:07.48 | bulle | sip show channels show active channel |
| 16:07.56 | russellb | bulle: what version are you running? |
| 16:08.18 | bulle | russellb: 1.4.1 |
| 16:10.21 | *** join/#asterisk RoyK (n=roy@ti211310a080-6495.bb.online.no) |
| 16:10.40 | bulle | russellb: http://pastebin.ca/390616 |
| 16:12.00 | russellb | alright, well if that is happening, it is a bug. turn on "sip history" and "sip debug", get the SIP messages from the call that makes it happen, and then the history for that call "sip show history" |
| 16:12.05 | russellb | and then post it to bugs.digium.com |
| 16:12.39 | bkruse | omg russell, im at work... |
| 16:13.10 | bkruse | waiting for mogorman, this is SO lame. |
| 16:13.16 | russellb | bkruse: that does sound lame. |
| 16:13.24 | bkruse | it rly is |
| 16:13.27 | Qwell | that sounds uber-leet! |
| 16:13.31 | bulle | russellb: aye, will do |
| 16:13.37 | bkruse | :D |
| 16:13.52 | russellb | bulle: not much else i can do ... sorry |
| 16:13.53 | bkruse | because i cant do anything until mogorman gets here..... |
| 16:13.59 | file | bkruse: never! |
| 16:14.07 | russellb | bkruse: play some joke on him |
| 16:14.09 | zoa | yo yo file |
| 16:14.20 | bulle | russellb: no worries, its no big show stopper for me, at this point |
| 16:14.23 | RoyK | wow. zoa? you alive? |
| 16:14.23 | bkruse | russellb: i need to, because i have to wait on him before i can work on this project |
| 16:14.24 | bkruse | super lame |
| 16:14.25 | russellb | bkruse: move his computer into a closet, and say "sorry man, they moved you to a different office" |
| 16:14.33 | zoa | yes yes |
| 16:14.35 | zoa | alive and kicking |
| 16:14.35 | bkruse | or "sorry man, i fired you" |
| 16:14.36 | bulle | russellb: i was just wondering if it was a known issue, and if so, what the solution was |
| 16:15.01 | russellb | bulle: i don't think so. it should time out eventually |
| 16:15.17 | russellb | if not, then, it's broken |
| 16:15.30 | bulle | russellb: well, this particular one have been staying there for like 12 hours or so |
| 16:15.38 | russellb | heh, yeah, that's not right :) |
| 16:15.42 | zoa | how have you been royk ? |
| 16:17.18 | russellb | file: i fixed menuselect so it automatically enables stuff when you install their dependencies again ... |
| 16:17.33 | russellb | file: i think it was you i was talking to about that.. |
| 16:20.15 | silvertip257 | g'day I must go |
| 16:26.25 | Qwell | hmm, where's Strom |
| 16:27.30 | adde | When i divert the call through gizmo it looks allright but no sound in any direction gos through but the connection is up... Is that codec problems...where do i sort it if so? |
| 16:28.08 | bkruse | Qwell: in a hotel, chillin |
| 16:28.20 | *** join/#asterisk mog (i=ejabberd@71.207.215.93) |
| 16:28.21 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:28.37 | creature1 | adde: are you calling inside your network? |
| 16:28.39 | Qwell | I know that as soon as I eat lunch, he's gonna say we should go out... so I'm refraining, heh |
| 16:28.56 | adde | no the gizmo user im redirecting to is outside |
| 16:29.11 | creature1 | adde: then it might be a NAT-problem actually |
| 16:29.39 | adde | hmmm... I have no possibility to redirect any ports... is there a solution? |
| 16:29.55 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 16:29.56 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 16:30.55 | creature1 | adde: not sure but STUN might work for you, http://www.voip-info.org/wiki/view/STUN or |
| 16:32.30 | adde | my problem is im on a crappy adsl modem which works as a hub... but no portforwarding or anything. |
| 16:32.41 | *** join/#asterisk hijoe (n=hijoe@c-67-160-249-95.hsd1.ca.comcast.net) |
| 16:33.39 | creature1 | adde: that's a wicked solution |
| 16:34.08 | creature1 | is it your isp that decided to send you such a modem? |
| 16:34.24 | *** join/#asterisk kiwoneka (n=kiwi@KTNRON06-1168103823.sdsl.bell.ca) |
| 16:34.42 | *** join/#asterisk newmedian (n=newmedia@TOROON63-1168100004.sdsl.bell.ca) |
| 16:34.53 | kiwoneka | good afternoon to all |
| 16:35.48 | adde | Yes... and i gave them hell... so know they are going to reinstall the whole adsl service...give me new modem, statix ip and drop the price :D |
| 16:36.32 | creature1 | adde: sounds great :) |
| 16:38.23 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 16:38.42 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 16:48.58 | JacksLivr | anybody got a 7910 working with *? |
| 16:49.00 | *** join/#asterisk ozant (n=ozan@88.239.25.10) |
| 16:49.03 | Qwell | JacksLivr: yep |
| 16:49.21 | JacksLivr | hi Qwell! |
| 16:51.04 | JacksLivr | i have it pulling an IP from the dhcp server on the asterisk system. but although it pulls an IP and is pingable, it sticks on configure IP |
| 16:51.42 | Qwell | http://lists.digium.com/pipermail/svn-commits/2007-March/022271.html |
| 16:52.05 | Qwell | ^^ Best feature EVER |
| 16:53.13 | JacksLivr | lol |
| 16:53.14 | DrukenLPY | Qwell: you scare me.... |
| 16:56.00 | ozant | hi all, i got a question about AGI, is't it possible to run asterisk functions ? like AGENT() in an agi script. Like $agi->exec('Set','agentinfo=${AGENT(10:name)}') |
| 16:56.29 | *** join/#asterisk Strom_M (n=strom@c-71-207-196-66.hsd1.al.comcast.net) |
| 16:56.57 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 16:57.47 | zoa | qwell, great commit |
| 16:57.52 | Qwell | ;) |
| 16:58.14 | Strom_M | ooh, i missed something good |
| 16:58.18 | Qwell | Strom_M: http://lists.digium.com/pipermail/svn-commits/2007-March/022271.html |
| 16:58.31 | creature1 | ozant: yeah you should be able to do like $ua->agent('AsteriskAGIQuery/$VERSION'); |
| 16:58.47 | creature1 | in a agi-script |
| 16:58.53 | Strom_M | hahah cool |
| 16:59.13 | Strom_M | Qwell: where should I go for a haircut? |
| 16:59.16 | creature1 | Qwell: great commit :) |
| 16:59.19 | Qwell | Strom_M: dunno |
| 16:59.47 | Strom_M | what, you dont get haircuts? |
| 16:59.49 | Qwell | Strom_M: done the lunch thing yet? |
| 16:59.53 | Qwell | nope, I do my own :p |
| 17:00.05 | DrukenLPY | army issue? |
| 17:00.20 | russellb | lunch?! |
| 17:00.25 | russellb | i'm such the hungry ... |
| 17:00.31 | Strom_M | Qwell: no, havent had lunch yer |
| 17:00.41 | Strom_M | you, me, and russelb should figure out lunch |
| 17:00.42 | Qwell | You guys wanna go somewhere? |
| 17:01.02 | russellb | yes! |
| 17:02.17 | zoa | yes! |
| 17:02.25 | zoa | come over here and i will cook |
| 17:02.26 | Qwell | zoa: hop on the next flight - lets go |
| 17:02.29 | JacksLivr | is this open to everyone??? |
| 17:02.29 | Qwell | or that |
| 17:02.31 | russellb | zoa: :D |
| 17:02.31 | Strom_M | zoa: you're in huntsville too? |
| 17:02.35 | zoa | no |
| 17:02.40 | zoa | but you said go somewhere |
| 17:02.47 | zoa | my place qualifies for somewhere |
| 17:02.52 | zoa | so you agreed already |
| 17:02.56 | zoa | i will start cooking now |
| 17:02.58 | zoa | bbye |
| 17:03.23 | zoa | bring a bottle of wine please |
| 17:03.45 | Qwell | American wine? why? |
| 17:03.48 | DrukenLPY | gonna wine and dine them eh zoa? |
| 17:03.58 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 17:05.06 | Qwell | JacksLivr: sure, join that channel :) |
| 17:05.52 | russellb | bkruse: mog: you guys should come to |
| 17:05.56 | zoa | no no |
| 17:06.04 | zoa | i will buy them a burger and drink the wine myself |
| 17:06.06 | zoa | great deal |
| 17:06.13 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 17:06.14 | mog | ? |
| 17:06.17 | Qwell | mog: lunch |
| 17:06.22 | JacksLivr | Qwell: join what channel? |
| 17:06.25 | mog | where? |
| 17:06.29 | Qwell | that one :D |
| 17:06.31 | Qwell | see invite |
| 17:07.07 | zoa | yeah yeah starve me to death again! |
| 17:07.13 | Qwell | zoa: k |
| 17:07.16 | zoa | :) |
| 17:09.27 | bkruse | zoa: ill get you a virtual milkshake |
| 17:09.32 | bkruse | i give them to file all the time |
| 17:09.46 | zoa | thanks! |
| 17:09.50 | zoa | they have to be good |
| 17:10.03 | bkruse | very good |
| 17:10.18 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-171-220.ny325.east.verizon.net) |
| 17:10.39 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:11.07 | zoa | yo yo olle |
| 17:13.24 | *** join/#asterisk boch (n=fran@190.48.231.134) |
| 17:13.31 | boch | hello all |
| 17:15.28 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 17:18.33 | *** join/#asterisk [TK]D-Fender (n=joe_blow@64.235.216.2) |
| 17:22.31 | *** join/#asterisk Mercestes (n=Merceste@208.252.223.195) |
| 17:31.30 | *** join/#asterisk Strom_C (n=strom@66.0.239.106) |
| 17:33.59 | Mercestes | Mr. S. *Salutes* |
| 17:34.51 | russellb | Strom_C was pwning me in bowling last night |
| 17:35.50 | Mercestes | yay, updating for that daylight savings time bs. |
| 17:36.35 | bkruse | anyone up for food thats in hsv? |
| 17:38.02 | Strom_C | its an asterisk lunch-o-rama |
| 17:38.49 | digime | who is everyone using these days for toll free DID? |
| 17:39.00 | Mercestes | well, must be gone. workign on sunday in California when I';m supposed to be spending it with my wife. |
| 17:39.03 | Mercestes | Top of ht eday |
| 17:39.16 | Mercestes | digime, : Heard ok things about Teliax. |
| 17:39.28 | digime | Teliax? okay will check it out |
| 17:39.41 | digime | I need something very, very reliable. |
| 17:40.58 | digime | voicepulse doesn't offer 800 did yet unfortunatley |
| 17:45.25 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 17:45.50 | PhilMcKracken | digime: SIP? |
| 17:46.21 | Sweeper | broadvox \o\ |
| 17:48.14 | *** part/#asterisk mog (i=ejabberd@71.207.215.93) |
| 17:51.36 | nays85 | digime : what do you mean? |
| 17:52.19 | nays85 | digime : voicepulse lists toll free availability on connect.voicepulse.com |
| 17:54.14 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 17:54.27 | *** join/#asterisk flying_Luck (n=melifaro@ppp85-141-155-9.pppoe.mtu-net.ru) |
| 17:55.50 | reber | did anyone here tried to send sms to french phones like described here : http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Sms |
| 17:56.18 | Qwell | J4k3: lemme find my quote from yesterday |
| 17:57.16 | Qwell | trixbox certification == "lolz, I can click teh buttons on teh gui" |
| 17:57.23 | J4k3 | haha yeah |
| 17:57.32 | welby | lol |
| 17:59.41 | J4k3 | heh the f1000g thinks its an hour behind... today is the day I stick it back in the box and return it :P |
| 18:00.07 | J4k3 | yay for Return Materials Authorization numbers :) |
| 18:02.35 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 18:07.39 | *** join/#asterisk null3d (n=xokappa@69.46.31.46) |
| 18:13.07 | Nivex | doh! my 7960 didn't pick up the new DST |
| 18:13.14 | J4k3 | Intel capital invests in fonality... "We make CPUs for button pushing, not actual work" |
| 18:14.14 | J4k3 | nivex: bah, real men use GMT. This daylights savings time zone nonsense is for the birds! |
| 18:14.15 | J4k3 | ;) |
| 18:14.46 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-188-233-116.hsd1.ca.comcast.net) |
| 18:15.48 | Nivex | and metric! |
| 18:17.32 | bkruse | Failed to authenticate on INVITE, omg i hate sip |
| 18:22.31 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 18:22.33 | Dovid | hi all |
| 18:22.43 | Dovid | can anyone tell me what this means? |
| 18:22.44 | Dovid | WARNING[2380]: chan_sip.c:3598 process_sdp: Insufficient information for SDP (m = '', c = '') |
| 18:32.38 | creature1 | Dovid: Yes, lack of SDP information |
| 18:35.01 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 18:35.40 | *** join/#asterisk ricko73 (n=dhartman@24-196-47-160.dhcp.fdul.wi.charter.com) |
| 18:36.24 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-188-233-116.hsd1.ca.comcast.net) |
| 18:37.45 | Dovid | crazy: whats that / |
| 18:37.47 | Dovid | ? |
| 18:38.07 | creature1 | Dovid: Most probably you have videosupport set to yes but the SDP information is sufficent. There are many gateways that do not handle SDP's which contain video. |
| 18:38.31 | creature1 | insufficent even.. |
| 18:39.08 | kiwoneka | hello everyone |
| 18:39.11 | ricko73 | I have a question regarding how asterisk gets local time from the system |
| 18:39.26 | creature1 | Dovid: SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. SDP is used from VOIP signalling protocols like SIP, H.323 and some minor VOIP protocols to transfer media setup information about a multi media client from A to B. |
| 18:39.36 | creature1 | (pasted from http://www.voip-info.org/wiki/view/SDP) |
| 18:39.38 | ricko73 | does it query the *libc library, or only look to /etc/localtime |
| 18:39.39 | kiwoneka | is there any specific standard for organizing extensions.conf? |
| 18:39.48 | *** join/#asterisk \lart (i=foobar@pool-72-73-230-114.cmdnnj.fios.verizon.net) |
| 18:39.53 | kiwoneka | keeping it neat |
| 18:39.59 | \lart | greetings all.. Anyone happen to have polycom bootloader 3.3.2b and firmware 2.1.0? the reseller i bought from has proven useless thus far. |
| 18:40.46 | kiwoneka | there are issues with that boot loader and firmware with 1.4 |
| 18:40.52 | creature1 | kiwoneka: [general] and [globals] are standards, you can organize it however you want in addition to that. |
| 18:40.58 | kiwoneka | prinamry difference is for the ip650 |
| 18:41.06 | kiwoneka | with hd voice |
| 18:41.26 | kiwoneka | what do you do |
| 18:41.31 | \lart | kiwoneka: i'm still on 1.2.16, so i presume i'd still be ok? i prefer to use freepbx, and i was under the impression that 1.4+fpbx wasn't quite cooked yet.. |
| 18:41.53 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 18:42.38 | creature1 | kiwoneka: i think it uses gettimeofday |
| 18:42.59 | [TK]D-Fender | FreePBX has always been a half-baked idea... |
| 18:43.24 | \lart | [TK]D-Fender: be that as it may, it's still my preference. :) |
| 18:44.03 | kiwoneka | \lart, you should upgrade to 1.6.7 |
| 18:44.22 | \lart | kiwoneka: 1.6.7? I'm already on 2.0.1 |
| 18:44.41 | [TK]D-Fender | \lart: What model(s) do you have? |
| 18:44.51 | \lart | [TK]D-Fender: ip601 |
| 18:44.53 | kiwoneka | well, all that is for is hd voice on the ip650 |
| 18:45.29 | \lart | no DST fixes in that release? |
| 18:45.36 | [TK]D-Fender | \lart: No, MUH more |
| 18:45.59 | kiwoneka | great phones |
| 18:46.09 | kiwoneka | 1.6.7 |
| 18:46.12 | [TK]D-Fender | \lart: 2.1.0 adds the MicroBrowser to IP 430/501 tweaks a LOT of gains, doubles the volume resolution on lower models, etc |
| 18:46.23 | [TK]D-Fender | Add tables support to the MB.... |
| 18:46.32 | creature1 | ricko73: i think it uses gettimeofday |
| 18:47.03 | \lart | [TK]D-Fender: ok, i'm sold. my reseller has been pretty useless thus far. any chance you've got the files and wouldn't mind mailing them or pointing me at a spot to grab them? |
| 18:47.47 | *** join/#asterisk yonahw (n=yonahw@84.229.141.58) |
| 18:48.16 | ricko73 | creature1: thanks. We've found on uclibc systems that don't include the zoneinfo files (they use the TZ variable) that the voicemail app is not able to get a localtime value |
| 18:48.18 | *** join/#asterisk \lart (i=foobar@pool-72-73-230-114.cmdnnj.fios.verizon.net) |
| 18:48.23 | ricko73 | it therefore defaults to GMT |
| 18:48.25 | *** part/#asterisk \lart__ (i=foobar@pool-72-73-230-114.cmdnnj.fios.verizon.net) |
| 18:48.29 | ricko73 | or rather UTC |
| 18:48.45 | ricko73 | seems like a flaw to me |
| 18:48.50 | creature1 | ricko73: ok, i found this on a website "Asterisk keeps internal time as epoch time (the number of seconds past midnight on January 1, 1970 GMT) and converts it to local time for a given time zone." |
| 18:49.03 | ricko73 | right, but looking at /etc/localtime |
| 18:49.10 | ricko73 | at least as far as I can tell |
| 18:49.35 | ricko73 | it should be trying to get that info from the *libc function |
| 18:49.50 | creature1 | ricko73: http://www.asterisk.org/doxygen/1.4/localtime_8h.html seems like.. |
| 18:50.14 | ricko73 | perhaps they fixed this in 1.4 |
| 18:50.38 | ricko73 | or not |
| 18:55.30 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 18:55.46 | *** join/#asterisk adde (n=adde@tn-84-218-53-11.dsl.bredbandsbolaget.se) |
| 18:55.55 | adde | Could somone please help me out with teh code to have directly under Default with a menu 1,2,3 which will play a different sound... I just need something really simple to understand it... |
| 18:58.44 | *** join/#asterisk [Mr_X] (n=mrx@88.118.89.102) |
| 19:00.06 | *** part/#asterisk ricko73 (n=dhartman@24-196-47-160.dhcp.fdul.wi.charter.com) |
| 19:05.15 | *** join/#asterisk Tili (n=tili@cm109.gamma248.maxonline.com.sg) |
| 19:06.46 | Tili | zaptel is writing udev rules for creating zap devices but i only see /dev/zap/ctl and rest 3. but nothing about /dev/zap/1 and so on |
| 19:06.52 | Tili | no channels are created |
| 19:06.55 | Tili | using FC4 |
| 19:07.33 | *** join/#asterisk welby (i=welby@gateway/tor/x-dc916fcf42f62f6e) |
| 19:09.30 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 19:14.11 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 19:17.23 | tzafrir | Tili, do you see anything in /dev/class/zaptel ? |
| 19:17.35 | tzafrir | If not, don't suspect udev . |
| 19:19.11 | \lart | Ahh.. Very good. My IP601 is now happy with its new bootrom and sip app. Thanks [TK]D-Fender. DST fixed as well. |
| 19:19.20 | Tili | there is no /dev/class |
| 19:19.25 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 19:19.30 | Tili | tzafrir: there is no /dev/class |
| 19:19.45 | Tili | i have only one line in zaptel.conf as fxsks=1,2,3,4 |
| 19:22.14 | [TK]D-Fender | Tili: And have you confirmed the modules loaded in "cat /proc/interrupts" ? |
| 19:24.18 | Tili | [TK]D-Fender: what do i have to configure there? secondly i have only 1 CPU |
| 19:24.48 | Tili | the problem is that on load zaptel is creating all other devices like channel/pseudo but not channels themselves like 1 2 3 4 |
| 19:25.59 | *** join/#asterisk Juggie (i=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 19:28.42 | Tili | ok sorry guys |
| 19:28.53 | Tili | i was loading wrong module. wcfxo. i had to load wctdm. |
| 19:30.21 | *** join/#asterisk KnowWhat (n=KnowWhat@125.209.64.75) |
| 19:31.17 | tzafrir | Tili, /sys/class , sorry |
| 19:31.24 | *** join/#asterisk Carlson (n=pocketir@m125e36d0.tmodns.net) |
| 19:35.48 | *** join/#asterisk Witwolf (n=carel@c3-26-1.eno.dial.mweb.co.za) |
| 19:36.06 | Carlson | how to be the next vonage? |
| 19:36.29 | geejay101 | I am looking for a possibility to play an announcement to the CALLED party after the caller has hung up ? I cant find that. I only see possibility to play announcement in dial command after a call is connected. Is there any way to announce something to the called party after the caller has hung up ? |
| 19:37.03 | Carlson | geejay: cant do that, since after the hangup, the channel is torn down |
| 19:39.03 | KnowWhat | geejay101: you want to impose somethin on user, when he decided to hung up, means he doesnt want to hear you more |
| 19:39.33 | thansen|laptop | anyone in here had experience with broadvoice? |
| 19:40.38 | *** join/#asterisk welby (i=welby@gateway/tor/x-650b505d15b9d3e9) |
| 19:40.39 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 19:41.01 | *** join/#asterisk RoyK (n=roy@ti211310a080-12322.bb.online.no) |
| 19:43.01 | Witwolf | Hi, could anyone confirm my finding. I have problems with MOH on a FXO when I put an incoming call on hold. I think it could be from an interrupt problem. I get jittering and noise when the music disappears for about half a second. Everything else works fine, it only MOH. |
| 19:45.30 | *** join/#asterisk elpropagandista (n=elpropag@208-106-57-5.dsl.dynamic.sonic.net) |
| 19:46.09 | *** join/#asterisk |kahless| (n=kvirc@i577AC577.versanet.de) |
| 19:46.13 | |kahless| | hi all |
| 19:49.39 | *** join/#asterisk Vec (n=Vec@dsl-241-222-88.telkomadsl.co.za) |
| 19:50.26 | kgx | geejay101: thats impossible but i've seen some support places having a message saying "To hang up, please press 0", and when the user presses 0, it says "Thank You! RTFM please" |
| 19:50.59 | kgx | imo its a little unneccessary but its done by our telecom company! |
| 19:51.14 | kgx | (btw, i was joking about rtfm :p ) |
| 19:53.38 | *** part/#asterisk elpropagandista (n=elpropag@208-106-57-5.dsl.dynamic.sonic.net) |
| 19:55.42 | *** join/#asterisk RA25 (n=RA25@c-66-31-1-124.hsd1.ma.comcast.net) |
| 19:59.35 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:59.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:01.30 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-188-233-116.hsd1.ca.comcast.net) |
| 20:04.05 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 20:06.37 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 20:06.38 | Vec | For some strange reason one of my SIP phones is registering with 2 usernames, I have no nidea why, it is a GT BT-100, and does not support multiple accounts, any ideas (It is reachable on both numbers)? |
| 20:07.10 | PhilMcKracken | It's because I haxxored joo |
| 20:08.50 | Vec | PhilMcKracken, hmm that makes sence |
| 20:10.35 | toot | hey folks. got a few emails in the last few days asking for the link to the project we are working on. if anyone wants to look its at http://adsldemo.tigercube.net/ user: demo pass: demodemo |
| 20:15.46 | Vec | I have just discovered the wierdest bug in a Grandstream BT-100 |
| 20:16.29 | JT | those things are full of bugs |
| 20:16.54 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 20:17.03 | Vec | The phone only supports one account, but somehow I was able to add to, so they both registerd but now I can't delete the 1. |
| 20:18.13 | Vec | even if I hard reset it the account is still there |
| 20:19.21 | KnowWhat | toot: can i msg you? |
| 20:20.51 | *** join/#asterisk topping (n=topping@209-204-141-95.dsl.static.sonic.net) |
| 20:21.16 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:21.16 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:21.17 | *** join/#asterisk Assid (n=assid@59.183.1.31) |
| 20:22.06 | russellb | eep |
| 20:22.11 | KnowWhat | toot: are you there? |
| 20:22.13 | Assid | heya |
| 20:22.18 | toot | hey |
| 20:22.19 | Assid | sup russellb |
| 20:22.22 | toot | plz msg away |
| 20:22.28 | russellb | not too much |
| 20:22.36 | russellb | hanging out in my office :-/ |
| 20:22.43 | Assid | today as well? |
| 20:22.51 | Assid | once a geek.. always a geek |
| 20:22.52 | Vec | how can I kick off (deregister) phones from asterisk cli ? |
| 20:22.54 | russellb | i'm a workaholic |
| 20:23.00 | Assid | apparently |
| 20:23.11 | russellb | but ... my job is to work in asterisk ... so ... |
| 20:23.20 | russellb | my line between work and fun is kind of vague :) |
| 20:23.25 | Assid | but then i dont qualify to say anything .. im working anytime any place i feel like |
| 20:23.31 | bkruse | russellb: same :D |
| 20:23.42 | Assid | i wouldnt mind a job working on asterisk |
| 20:23.45 | bkruse | boy i wish i could work at digium, that would be swell.... |
| 20:23.51 | *** join/#asterisk Stridernzl (n=neville@222-153-89-14.jetstream.xtra.co.nz) |
| 20:23.58 | Assid | get a decent pay.. im set |
| 20:24.04 | russellb | Assid: there is a job openings page on digium.com :) |
| 20:24.18 | Assid | i cant qualify |
| 20:24.30 | Assid | i can say that without even looking at the page |
| 20:24.36 | Assid | unless they want php developers |
| 20:24.49 | russellb | i think there is a job opening for a php dev, actually. |
| 20:25.00 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 20:25.07 | Assid | hrmm |
| 20:25.15 | Assid | + i am in india |
| 20:25.27 | russellb | oh, and you've have to move to the other side of the earth |
| 20:25.55 | Assid | oh like thats easy |
| 20:26.04 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 20:27.00 | russellb | :) |
| 20:28.16 | *** join/#asterisk orkid (i=orkid@bas1-barrie18-1242374109.dsl.bell.ca) |
| 20:33.55 | *** join/#asterisk welby (i=welby@gateway/tor/x-4dfff40b21c94a9c) |
| 20:36.35 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 20:37.00 | toot | i'd love a move to the states :) |
| 20:37.01 | EmleyMoor | Is there a way I can record the time of a call to later have Asterisk Say it back? |
| 20:37.59 | Assid | once the channel is closed.. the ip phone hangs up done see how oyu gonna do it |
| 20:38.09 | Assid | russellb: i got a polycom that refuses to login |
| 20:38.12 | Assid | wanna debug? |
| 20:38.30 | Assid | actually wait |
| 20:38.55 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 20:39.15 | Assid | russellb: pm? |
| 20:40.07 | Assid | err.. damn |
| 20:44.00 | Assid | yeay it works |
| 20:45.21 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 20:46.16 | RoyK | http://karlsbakk.net/fun/banned%20commercials%20-%20fuck%20you%20in%20the%20ass%20(funny%20danish%20commercial).mpeg |
| 20:46.21 | Assid | russellb: you wanna try that rtcp bug ? |
| 20:47.18 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 20:47.29 | EmleyMoor | As I was saying when I was so rudely interrupted... |
| 20:47.39 | *** join/#asterisk [kahless] (n=kvirc@i577ACEC2.versanet.de) |
| 20:47.39 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-3d3563ed0072abe0) |
| 20:47.55 | EmleyMoor | Is there any way I can record the time of a call in the database in such a way I can later have asterisk Say it back? |
| 20:47.57 | Mahmoud | http://photos1.blogger.com/blogger/2116/144/1600/fightingkeyboardistsFreedomDogs.0.jpg |
| 20:50.09 | EmleyMoor | I am already recording the last number to ring a given set of phones, and the last to be answered on a given channel, just not sure how I can handle time in a usable format |
| 20:51.01 | Assid | EmleyMoor: you can log the cdr to a database |
| 20:51.17 | Assid | and have it query the last record.. and have it say it back like "your last call was for .... seconds" |
| 20:51.32 | Assid | may have to use festival or somethig fro that |
| 20:51.37 | Assid | but never tried it myself |
| 20:51.53 | EmleyMoor | I want more "Telephone number x called at y" - not sure how to log the time |
| 20:52.08 | EmleyMoor | I have "The last call was from x" implemented already |
| 20:52.15 | Assid | cdr logs everything |
| 20:52.44 | EmleyMoor | Yes, it does, but I need something that can be called upon in the dialplan |
| 20:52.56 | EmleyMoor | I see what you mean, though |
| 20:56.27 | EmleyMoor | I am wondering if I can use ${TIMESTAN |
| 20:56.40 | EmleyMoor | I am wondering if I can use ${TIMESTAMP} into SayUnixTime somehow |
| 20:57.15 | *** join/#asterisk DragoraN (n=dragoran@217.67.19.74) |
| 20:57.16 | EmleyMoor | (or ${EPOCH}) |
| 20:57.17 | DragoraN | hi |
| 20:57.29 | bkruse | ${EPOCH} > all |
| 20:57.36 | DragoraN | is it possible to call from pc behing NAT to pc behind NAT ? |
| 20:57.59 | Strom_M | DragoraN: yes, with a lot of headache involved |
| 20:58.06 | EmleyMoor | Ah, yes, I see it now |
| 20:58.19 | DragoraN | Strom_M: is it possible with STUN ? |
| 20:58.23 | Assid | DragoraN: where is the asterisk box? |
| 20:58.31 | DragoraN | Strom_M: i cannot afford vpn |
| 20:58.32 | Assid | at the nat gateway ? |
| 20:58.44 | Assid | DragoraN: vpn is free |
| 20:58.45 | Strom_M | DragoraN: I don't recall if asterisk uses STUN |
| 20:58.47 | DragoraN | Assid: exactly, but only on one side |
| 20:58.54 | Strom_M | and there are free vpn programs IIRC |
| 20:58.59 | Assid | DragoraN: then no problem actually |
| 20:58.59 | DragoraN | Assid: i cannot afford, i cannot install it there |
| 20:59.05 | Assid | just port forward |
| 20:59.18 | Assid | and then have the one that doesnt have a direct access to register with the one that does |
| 20:59.27 | DragoraN | Assid: but multiple users can be on one side and multiple on other side |
| 20:59.31 | Assid | the host will automatically update and your good to go |
| 20:59.39 | Assid | DragoraN: as i said .. no problem |
| 20:59.45 | DragoraN | Assid: how ? |
| 20:59.48 | Assid | DragoraN: however, if i were you, i'd use IAX |
| 20:59.56 | DragoraN | Assid: why? |
| 20:59.58 | Assid | [02:29:13] <Assid> just port forward |
| 20:59.58 | Assid | [02:29:26] <Assid> and then have the one that doesnt have a direct access to register with the one that does |
| 21:00.01 | Assid | i just said how |
| 21:00.13 | Assid | and why ? because iax is more nat friendly than sip |
| 21:00.14 | [TK]D-Fender | yes you can work behind double-NAT if things are set right, and NO, STUN in not supported nor is it needed in * |
| 21:00.16 | Assid | bbiab |
| 21:00.16 | DragoraN | portforward for which one? |
| 21:00.30 | DragoraN | RTP packets just cannot pass. |
| 21:00.32 | DragoraN | singaling is ok |
| 21:01.48 | DragoraN | Assid: how do you mean port forwarding? |
| 21:04.32 | DragoraN | :( |
| 21:05.29 | Stridernzl | [TK]D-Fender - Ping :) |
| 21:05.45 | [TK]D-Fender | Stridernzl: pong |
| 21:05.50 | bkruse | aww, to late. |
| 21:08.02 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 21:08.49 | DragoraN | Assid: any idea? |
| 21:09.24 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:10.39 | JacksLivr | anybody got a 7910 that's working that would be willing lend a hand? |
| 21:11.23 | russellb | JacksLivr: do you have some 7970 firmware? :) |
| 21:11.28 | *** join/#asterisk svenna_ (n=svenna@p548D37DA.dip0.t-ipconnect.de) |
| 21:12.14 | JacksLivr | 7970? I can get it, i think |
| 21:13.19 | geejay101 | Apparently I havent made myself clear on what I am trying to accomplish. I want that AFTER the CALLER has hung up and the CALLED party is still in the line to play something to the CALLED PARTY. |
| 21:14.07 | bkruse | h,1,Playback(youranub?) |
| 21:14.53 | DrukenLPY | DragoraN: what's your problem ? |
| 21:15.18 | Strom_M | geejay101: the audio path is TORN DOWN. there is no way to do that. |
| 21:15.30 | [kahless] | memo to myself: never ever try to compile a kernel with a 660mhz cpu... (time: 146m6.197s) |
| 21:16.12 | geejay101 | Strom, HOW is the audio path torn down ? In the dial command ? |
| 21:16.17 | DragoraN | DrukenLPY: how to call between users behind NAT on both sides where asterisk is on one side |
| 21:16.52 | DrukenLPY | DragoraN: DON'T USE ASTERISK BEHIND A NAT!..... |
| 21:17.08 | DrukenLPY | SIP and nat don't mix well |
| 21:18.03 | geejay101 | Strom, is there only ONE audio path or TWO paths - one from the caller and one from the called party ? |
| 21:18.06 | bkruse | DrukenLPY: port masking and forwarding and network address translation doesnt work well with 10,001 ports to be serviced? pssh :P |
| 21:18.23 | DragoraN | DrukenLPY: asterisk is not behind nat, it has public IP, but users does not |
| 21:18.45 | EmleyMoor | I now have a "call return" function that gives both the number and the time of the call |
| 21:19.00 | geejay101 | Fully coned NAT works excellent with Asterisk - no problems even to connect two asterisks behind two different NATS. |
| 21:19.12 | bkruse | ............... |
| 21:19.19 | Strom_M | geejay101: perhaps the better question to ask is why you're trying to play audio after the calling party hangs up |
| 21:19.24 | DrukenLPY | DragoraN: and asterisk is in the path? |
| 21:19.40 | DragoraN | DrukenLPY: asterisk is sitting on one nat with public IP |
| 21:19.49 | EmleyMoor | It says "Telephone number", reads the number, "called" and then the time in format q at IMp |
| 21:19.50 | DragoraN | i want to have "external users" |
| 21:20.02 | EmleyMoor | (and "To return the call, press 3") |
| 21:20.31 | DrukenLPY | DragoraN: but you said asterisk wasen't behind the nat... so which is it? |
| 21:20.58 | DragoraN | DrukenLPY: asterisk is not BEHIND nat... users are |
| 21:20.58 | geejay101 | Strom, because I dont want to play it during the conversation. |
| 21:21.27 | Strom_M | but WHAT are you playing? |
| 21:22.15 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 21:22.47 | DrukenLPY | DragoraN: so where is the problem? i have users behind nat |
| 21:22.57 | geejay101 | Strom, just some audio file. |
| 21:23.29 | Strom_M | geejay101: you're not giving me the information i'm looking for |
| 21:23.35 | markit | hi, I've a BRI with a single port, mISDN. Is there a way to make calls use one specific channel? In outgoing chan_misdn let's you specify the port, not the channel, and in incoming I don't know how to "bind" msn with a channel |
| 21:23.47 | Strom_M | what are you trying to accomplish which leads you to believe you need to play audio after the calling party hangs up? |
| 21:23.53 | geejay101 | Strom, that is correct. Why is that relevant to the problem ? |
| 21:24.17 | Strom_M | because I'm trying to ascertain if there's a better way to accomplish your actual goal |
| 21:24.40 | Strom_M | now i'll ask again |
| 21:24.42 | Strom_M | what are you trying to accomplish which leads you to believe you need to play audio after the calling party hangs up? |
| 21:24.43 | DrukenLPY | most people hang up the phone after goodbye's.... |
| 21:24.53 | DrukenLPY | instead of waiting to see if there's more |
| 21:25.07 | geejay101 | Strom, the fact is that some audio has to be played - the question is when to do it. During or before the conversation is less desirable. |
| 21:25.17 | Strom_M | sigh, forget it |
| 21:25.35 | DrukenLPY | sounds like an asshole message to me... |
| 21:25.41 | DrukenLPY | collections or something |
| 21:25.47 | DragoraN | DrukenLPY: internal calls are ok, but when i want to connect external users which are behind their own nat, its a problem |
| 21:25.49 | Strom_M | if you're not willing to answer my questions, then I'm not willing to help you :) |
| 21:26.15 | JacksLivr | russellb: cmterm-7970_7971-sccp.8-0-4SR3A.exe - 7970/7971 SCCP IP Phone load ? |
| 21:26.30 | geejay101 | Fair enough. But still I dont see the relevance . Its a purely technical problem. |
| 21:26.58 | DrukenLPY | well, it's been explained to you, that "technically" it can't be done |
| 21:27.12 | russellb | JacksLivr: do you see .cop files on there? |
| 21:27.19 | russellb | and it's 7970 |
| 21:28.31 | DragoraN | DrukenLPY: any idea? |
| 21:28.46 | geejay101 | Technically everything is possible - the question is how much effort it takes to program that. The dial command provides already options to play something to the called party before the conversation and also provides options to return the controll to the caller after the called party has hung up. So this issue seems to be doable. |
| 21:28.50 | DrukenLPY | DragoraN: do you use nat=yes on your external users? |
| 21:28.54 | *** join/#asterisk sudhir492 (n=sudhir@c-71-63-59-45.hsd1.va.comcast.net) |
| 21:29.01 | DragoraN | DrukenLPY: yes |
| 21:29.03 | sudhir492 | Hi All |
| 21:29.15 | bkruse | sudhir492: hey |
| 21:29.30 | DragoraN | DrukenLPY: if i call them, i can hear them and they can hear me |
| 21:29.31 | bkruse | need help setting up the linx with the asterisk? |
| 21:29.31 | sudhir492 | how are you bkruse |
| 21:29.41 | bkruse | sudhir492: im alright, yourself? |
| 21:29.47 | sudhir492 | pretty good |
| 21:29.50 | DragoraN | DrukenLPY: bud when they call me, i cannot hear them but they can |
| 21:30.00 | sudhir492 | thanks |
| 21:30.46 | JacksLivr | cmterm-7970_7971-sccp.8-0-4SR3A.cop : 7970/7971 SCCP IP Phone load - Compatible CCM Versions: 5.0(1), 5.0(2), 5.0(3) |
| 21:31.10 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 21:31.34 | sudhir492 | For the first time I got couple of Cisco 7940 phones. Now the task is to program them for Asterisk. Cisco would not let me download any firmware without registering :-( |
| 21:32.20 | DragoraN | sudhir492: why to register? use sccp protocol |
| 21:32.50 | shido6 | sip works. |
| 21:32.55 | shido6 | u do need sip firmware. |
| 21:32.56 | DrukenLPY | DragoraN: sounds like a nat issue... |
| 21:32.59 | shido6 | how many phones? |
| 21:33.08 | shido6 | sudhir492: ? |
| 21:33.09 | DragoraN | DrukenLPY: so? |
| 21:33.10 | sudhir492 | Have never used that, does that work as good as SIP. |
| 21:33.25 | DragoraN | sudhir492: try |
| 21:33.59 | *** join/#asterisk uwe (n=uwe@a42-32.adsl.paltel.net) |
| 21:34.00 | DrukenLPY | DragoraN: so what is the config on the asterisk system |
| 21:34.02 | sudhir492 | Even in that case, I will need documentation, e.g. default files |
| 21:34.11 | DragoraN | DrukenLPY: want to post it? |
| 21:34.33 | *** join/#asterisk arooni (n=chatzill@dsl081-163-148.sea1.dsl.speakeasy.net) |
| 21:34.34 | DrukenLPY | DragoraN: sure, pastebin it |
| 21:34.52 | DragoraN | DrukenLPY: ok |
| 21:34.53 | JacksLivr | russellb: forgive the ignorance. is this what you are talking about? cmterm-7970_7971-sccp.8-0-4SR3A.cop : 7970/7971 SCCP IP Phone load - Compatible CCM Versions: 5.0(1), 5.0(2), 5.0(3) |
| 21:34.55 | arooni | are there any good *free* (or as close to possible to free) SIP providers that let me connect to PSTN? |
| 21:35.06 | russellb | JacksLivr: yeah, that looks like it! sorry for the delay |
| 21:35.20 | bkruse | russellb: /ignore time? |
| 21:35.20 | JacksLivr | heh, dont appologise |
| 21:37.17 | DragoraN | DrukenLPY: pastebin is laggy |
| 21:37.18 | JacksLivr | arooni: try www.totallyfreesip.org |
| 21:37.18 | DragoraN | DrukenLPY: :) |
| 21:37.18 | arooni | hmmm damn |
| 21:37.18 | arooni | not workin ;p |
| 21:38.11 | DragoraN | DrukenLPY: http://pastebin.ca/391059 |
| 21:38.12 | JacksLivr | they must have gone out of business for being totally free sip |
| 21:38.18 | arooni | haha |
| 21:38.42 | EmleyMoor | Has anyone come up with a way of producing a Strowger-like dialtone on Zap channels yet? |
| 21:39.01 | DrukenLPY | DragoraN: do an ipconfig on the asterisk machine for me... |
| 21:39.09 | DragoraN | DrukenLPY: ok |
| 21:39.30 | DrukenLPY | er.. ifconfig |
| 21:39.44 | DragoraN | DrukenLPY: i understand :) http://pastebin.ca/391061 |
| 21:40.17 | DrukenLPY | ok, so asterisk IS the nat |
| 21:40.43 | DragoraN | DrukenLPY: yes |
| 21:41.20 | JT | arooni: why would a totally free SIP termination provider be any good? |
| 21:41.33 | arooni | JT >> what do you mean by any good |
| 21:41.41 | JT | you'll probably only find 1800 termination that is not absolute rubbish |
| 21:41.42 | arooni | i just need the calls to happen ... i dont care about quality |
| 21:41.50 | JT | well it costs money to terminate to the pstn |
| 21:41.59 | JT | why would someone give it to you for free |
| 21:42.02 | JT | that's illogical |
| 21:42.05 | JacksLivr | they have pri's that they dont need and instead of turning them off, they let the community use it for free. makes sense? |
| 21:42.09 | EmleyMoor | arooni: Never mind the quality, feel the width, eh? |
| 21:42.27 | JT | JacksLivr: yeah how often does that happen? |
| 21:42.31 | JacksLivr | obviously im joking |
| 21:42.34 | [TK]D-Fender | DragoraN: "canreinvite=no" set this in [general] and for all of your SIP peers |
| 21:42.38 | JT | ah :) |
| 21:42.53 | DragoraN | [TK]D-Fender: why? |
| 21:43.06 | JacksLivr | i thought that was a silly question, i was just being stupid |
| 21:43.15 | [TK]D-Fender | DragoraN: To fix your no-audio issues |
| 21:43.33 | DragoraN | [TK]D-Fender: what it shoudl fix? |
| 21:43.41 | JT | arooni: you probably need to pay if you want to call more than toll free numbers |
| 21:43.53 | DrukenLPY | christ... as if i didn't notice that.... |
| 21:44.02 | [TK]D-Fender | DragoraN: You were saying your sip peers behind NAT don't get RTP. thats because they are trying to reinvite which will FAIL |
| 21:44.22 | [TK]D-Fender | DragoraN: You need to have all traffic flow through your server for it to work |
| 21:44.30 | DragoraN | [TK]D-Fender: yeah, problem solved, thanks! |
| 21:44.45 | DragoraN | DrukenLPY: solved, thanks much |
| 21:44.51 | DragoraN | i must go now, bye! |
| 21:44.52 | [TK]D-Fender | DragoraN: You're welcome |
| 21:45.08 | [TK]D-Fender | NEXT!!@@!!111101101 (c) BKW |
| 21:46.41 | Corydon76-home | Too short for copyright |
| 21:47.06 | Corydon76-home | Trademark, maybe, but long since abandoned |
| 21:47.52 | [TK]D-Fender | Corydon76-home: You = killjoy |
| 21:48.06 | Corydon76-home | Love you too |
| 21:52.09 | DrukenLPY | awee.... so no one has a free termination for arooni?? hehehehe :P |
| 21:52.28 | arooni | DrukenLPY: not so much :P |
| 21:53.58 | *** join/#asterisk emiquelito (n=evandro@c9150dcb.virtua.com.br) |
| 21:54.01 | [TK]D-Fender | arooni: Who said any of this is FREE? Anyone offering otherwise is likely selling something you think is sexy, but in the end will be lacking KY and your consider yourself thorourly %#@$ed. |
| 21:55.08 | Strom_M | [TK]D-Fender: please do not feed the troll :) |
| 21:55.42 | [TK]D-Fender | Strom_M: Sure... as long as the outcome resembles the outcome of that poor chap From Seven ;) |
| 21:55.54 | JacksLivr | heh |
| 21:58.21 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:01.52 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-188-233-116.hsd1.ca.comcast.net) |
| 22:10.15 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 22:10.16 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 22:10.47 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 22:10.55 | DocHolliday | anyone here used didww (did worldwide)? |
| 22:21.38 | *** part/#asterisk bkruse (i=bkruse@nat/digium/x-36938dc84960c636) |
| 22:21.38 | adde | what exactly does extension "s" mean ? |
| 22:21.38 | sudhir492 | shido6: Sorry, I had to step out. At this time, I got 3 7940 phones, and will get 4 more at a bargain price |
| 22:22.19 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 22:22.20 | kink0 | hello |
| 22:22.22 | sudhir492 | They all have MGCP, ver 6.4.00 |
| 22:22.57 | kink0 | I use zaptel and asterisk 1.4, but I got sometimes: Primary D-Channel on span 1 down, any idea ? |
| 22:23.31 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 22:24.23 | *** join/#asterisk topping (n=topping@ppp-68-122-72-235.dsl.pltn13.pacbell.net) |
| 22:24.53 | JT | kink0: layer 1 link problems |
| 22:25.06 | sudhir492 | kink0: there could be many reasons for primary D channel down |
| 22:25.24 | sudhir492 | after coming up, how long does it take for the channel to go down? |
| 22:25.25 | kink0 | I read a lot about bugs in digium that can cause these problems. |
| 22:25.43 | Qwell[] | kink0: Have you called support? |
| 22:25.48 | kink0 | I have the same in two separate spans, so I think are not wires or so |
| 22:26.05 | kink0 | Qwell[], no, I have not called yet. |
| 22:26.14 | sudhir492 | kink0: I have been running 4 port T1 card for past 3 years, extremely stable |
| 22:26.19 | kink0 | I am seeking about the problem before ask |
| 22:26.37 | mmartinn | Does anyone know why the Dial command would return ${DIALSTATUS} = NOANSWER even when no timeout was given -- the docs say it should try indefinitely? |
| 22:26.51 | kink0 | the case is I have other machine with 4 E1 up, connected to 2N GSM gateways , there not any problem, stable. |
| 22:26.54 | JT | kink0: is this a problem that has always existed or what? |
| 22:27.18 | kink0 | this one box has a TE410, 2 spans are connected to Telco, g703 ussing 75/120 balum |
| 22:27.22 | florz | mmartinn: It's more like it doesn't give up by itself. But someone downstream still can give up. |
| 22:27.54 | kink0 | JT: I had some unstability , I have detected the cause with lost of D channel two days ago |
| 22:28.13 | mmartinn | florz: Is that typical in North America? My analog experience has been that calls ring forever. |
| 22:28.22 | kink0 | JT: I have also try all hardware and so, PRI cables, connectors, etc. |
| 22:28.36 | mmartinn | kink0: The few times I've seen d-channels go down, it has been a hardware problem or the telco's fault. |
| 22:29.00 | florz | mmartinn: No clue. But you could try it, I guess? :-) |
| 22:29.09 | mmartinn | florz: weird... |
| 22:29.11 | JT | kink0: only 2 out of 4 spans in use on card? |
| 22:29.30 | kink0 | mmartinn, may be, anywise I did connect the telco PRI port to the stable machine ( where I have connected the 2N gateways ) and appears fine |
| 22:29.43 | kink0 | JT: yes, actually only 2 of 4 are in use in this box |
| 22:29.53 | JT | do both spans have problems? |
| 22:30.04 | kink0 | yes, both at same time |
| 22:30.17 | kink0 | when channel 16 is lost, also 47 is lost |
| 22:30.53 | Stridernzl | anyone know where i can tweak a setting that relates to detecting an incomming call and when that call ends .. ie our system does not seem to detect hangups very well |
| 22:31.07 | JT | kink0: ok, what are the zttest results like? |
| 22:31.47 | ltdwk | stridernzl: look at using hangup on polarity switch if available from your telco |
| 22:31.57 | JT | kink0: also put the output of cat /proc/interrupts in pastebin.ca |
| 22:32.29 | kink0 | 100.000000% 99.743652% 99.792480% 99.804688% 99.609375% 99.914551% 100.000000% |
| 22:32.42 | JT | ouch |
| 22:32.48 | JT | your zaptel timing is shit |
| 22:32.51 | JT | that is the cause |
| 22:33.05 | kink0 | JT do you think may be a timeout in interrupts ? |
| 22:33.10 | JT | it must remain above or equal to 99.97% at all time |
| 22:33.18 | JT | kink0: pastebin.ca the interrupts |
| 22:33.30 | Stridernzl | ltdwk: How would i find that out .. We use telecom New Zealand |
| 22:33.31 | kink0 | ok, give me a sec |
| 22:33.35 | ltdwk | look at zttool and see if you are missing any interrupts |
| 22:34.16 | JT | it's clear that his zaptel timing is bad from zttest and his symptoms alone :/ now the fun bit, fixing it |
| 22:34.19 | ltdwk | stridernzl: ring and ask for Reverse on Idle Condition |
| 22:34.46 | JT | they call it RoIC in .au, not sure about .nz |
| 22:35.02 | ltdwk | stridernzl: they may call it CPC or something else |
| 22:35.44 | ltdwk | on the telstra customnet lines i have here they do something different again, they send a wink directly after the call is hung up |
| 22:36.03 | JT | hmm |
| 22:36.28 | ltdwk | kewlstart will pick that up fine though |
| 22:36.34 | kink0 | one sec.. pastebin.com go very slowly now |
| 22:36.57 | ltdwk | when I had zaptel timing issues I ended up having to switch hardware to fix it |
| 22:37.00 | DocHolliday | anyone here used didww? |
| 22:37.16 | kink0 | uffff , I got an error from pastebin .. hmmmmm |
| 22:37.18 | kink0 | Warning: unlink(/home/pastebin/public_html/../cache/recent): No such file or directory in /home/pastebin/lib/pastebin/db.mysql.class.php on line 243 |
| 22:37.23 | JT | kink0: i specifically stated pastebin.ca |
| 22:37.29 | JT | pastebin.com is utter rubbish |
| 22:37.39 | kink0 | ok will try now .ca instead .com |
| 22:38.17 | *** join/#asterisk bsd_tech (n=bsdtech@ppp-69-238-219-226.dsl.irvnca.pacbell.net) |
| 22:38.19 | JT | i have no idea how they made such a bad pasting website, pastebin.COM |
| 22:38.27 | bsd_tech | ok in asterisk 1.4.1 menuconfig |
| 22:38.27 | JT | i could run a better site on a pentium 1 |
| 22:38.35 | kink0 | http://pastebin.ca/391153 |
| 22:38.47 | bsd_tech | what is asterisk module selection for ? |
| 22:39.02 | kink0 | JT: yes, really surprise me these errors from pastebin.com |
| 22:39.12 | kink0 | I pasted zttest and interrupts |
| 22:39.40 | JT | kink0: ok no irq sharing there |
| 22:40.06 | kink0 | no, no sharing. This is a dual Xeon 2.8, SCSI and I try enable and disable HT also |
| 22:40.18 | kink0 | HT appears does not alter the results |
| 22:40.18 | JT | what you're going to have to try is a few different things, best is to swap slots that the card is in |
| 22:40.42 | JT | also if there is any unused hardware, remove it |
| 22:41.24 | kink0 | I did yet, I used also ports 1,3 instead 1,2 , and also swap span1 and span2 |
| 22:41.24 | kink0 | but the same. |
| 22:41.24 | JT | using different ports will do nothing |
| 22:41.29 | kink0 | yes, did nothing. |
| 22:41.30 | JT | swapping spans will also do nothing |
| 22:41.36 | JT | unless your card is defective |
| 22:41.39 | JT | swapping pci slots |
| 22:41.43 | JT | that the card is in |
| 22:42.12 | kink0 | that is the case, did nothing. I have not try to swap pci slots, since is a 1U chassis, just one slot available for this card, but I can try in other machine |
| 22:42.21 | JT | hrm |
| 22:42.42 | kink0 | JT do you think about timming on the PCI bus ? |
| 22:42.54 | JT | you might have to try disabling apic etc |
| 22:43.05 | kink0 | I also try to force 33, 66 and auto , in BIOS setup, but same. |
| 22:43.18 | JT | kink0: i've solved timing issues by swapping pci slots and ensuring no irq sharing |
| 22:43.35 | JT | there are no other pci devices? |
| 22:43.37 | kink0 | yes, i read about sharing irq is not a good thing |
| 22:43.38 | ltdwk | also try assigning the interrupts for wct4xxp to CPU1 |
| 22:43.52 | ltdwk | i notice it is doing almost nothing |
| 22:44.00 | kink0 | no in this machine, I will need to move the TE410 to other computer |
| 22:44.20 | markit | anyone with a single port ISDN and mISDN? |
| 22:44.31 | JT | maybe disable any unused onboard hardware |
| 22:44.36 | JT | if you can |
| 22:45.06 | kink0 | there no any unnused hardware that takes cycles now, just there the mainboard and the TE410 card |
| 22:45.38 | kink0 | well, the on-board SCSI is active, because there SCSI hard drives ( there not any IDE/SATA on this machine ) |
| 22:46.06 | JT | you have 3 usb host controllers up |
| 22:46.25 | kink0 | yes, but there no any hardware connected there |
| 22:46.42 | JT | doesn't matter, the host controllers are hardware |
| 22:46.46 | kink0 | do you suggest to remove USB support at all ? |
| 22:46.47 | JT | taking up interrupts |
| 22:47.03 | JT | try not loading the modules, or disabling the hardware, if possible |
| 22:47.11 | ltdwk | best thing to do is just assign the te card to cpu1 that way the other stuff wont be causing any issues |
| 22:47.35 | JT | a lot of stuff doesn't let you assign irqs there days |
| 22:47.35 | kink0 | well, I can try to compile kernel without USB support ( is not as module ) |
| 22:47.56 | JT | kink0: is it built in to the kernel? |
| 22:48.12 | kink0 | yes |
| 22:48.35 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 22:48.39 | JT | damn |
| 22:48.41 | kink0 | I see the counters for interrupts, and the high ussage are from timer and eth0, of course I Can not dissable those |
| 22:48.51 | JT | i'm not sure it will actually do anything anyway |
| 22:48.58 | JT | i just thought it was worth a shot |
| 22:49.01 | kink0 | well, I can re-compile the kernel and reboot |
| 22:49.12 | JT | kink0: look at info on the wiki |
| 22:49.13 | kink0 | but I saw very little interrupts counts for USB |
| 22:49.17 | JT | on disabling apic, etc |
| 22:49.24 | JT | and even smp if you can |
| 22:49.30 | JT | using boot parameter |
| 22:49.32 | JT | s |
| 22:50.03 | ltdwk | It may simply come down to the fact that your system cannot process interrupts fast enough |
| 22:50.04 | *** join/#asterisk _sull (n=none@209-120-203-142.hosts.idv.net) |
| 22:50.07 | kink0 | then may be apic that "delays" interrups ? |
| 22:50.22 | ltdwk | no matter what you do that is a possibility |
| 22:50.40 | kink0 | ltdwk: hmmmm ... may be. Anywise is a Supermicro computer |
| 22:50.44 | JT | it must be a piecef of junk then ;) or just incompatible |
| 22:50.50 | JT | hrm supermicro is a good brand |
| 22:51.05 | mmartinn | Does /proc/interrupts show that the zap card is sharing interrupts with anything? |
| 22:51.08 | ltdwk | I had problems with the first box I put my te410p into |
| 22:51.10 | JT | mmartinn: no |
| 22:51.14 | mmartinn | hmm |
| 22:51.18 | mmartinn | =/ |
| 22:51.19 | kink0 | yes, I have other one supermicro, with other TE405 on it, and I have not any problems |
| 22:51.29 | JT | kink0: is it identical? |
| 22:51.41 | mmartinn | That sucks... |
| 22:51.43 | ltdwk | it was an intel-based box... I changed to an opteron based amd 8000 series chipset board and it worked like a dream |
| 22:51.46 | kink0 | ltdwk I discover some supermicro, need you set PCI to 66Mhz, for the TE cards |
| 22:51.52 | mmartinn | Boo on hardware ;( |
| 22:52.00 | kink0 | JT: no, are differents motherboards |
| 22:52.16 | mmartinn | Sounds like getting a new mobo is the answer |
| 22:52.32 | kink0 | both are 533 FSB, 2.8 Xeon CPU, SCSI 79xx controller |
| 22:52.48 | kink0 | but are differents models of motherboards, very simmilar, but not the same. |
| 22:52.48 | ltdwk | in my experience it's the chipset on the mobo that screws you over |
| 22:52.49 | mmartinn | We had a test box with 'not bad' hardware... once we switched to identical mobos that were more "server-class" our interrupt problems went away |
| 22:53.32 | ltdwk | mmartinn: that's dead on. desktop class boards had too high interrupt latency compared to server class boards |
| 22:53.48 | JT | i've seen just as many problems on consumer hardware as server grade |
| 22:54.13 | kink0 | ltdwk, chipset is the same, the differences on these motherboards are very little, I don't check in details, but may be one includes SATA and the other one no, or so. |
| 22:54.14 | JT | kink0: if you're recompiling a kernel anyway, set the timing hz to 1000 |
| 22:54.35 | JT | preempt may be worth a shot too |
| 22:54.38 | ltdwk | kink0: the northbridges may be the same, but the southbridges might be different? |
| 22:55.01 | mmartinn | I've heard people say that SATA with interrupts from DMA were a problem before |
| 22:55.27 | kink0 | I did not use for that reason any SATA, even no IDE with DMA, just SCSI |
| 22:55.50 | ltdwk | Looking at kink0's zttest output, sometimes his timing is consistently above 99.97, but infrequently not |
| 22:56.26 | ltdwk | Which tells you something that's being used on the system might be causing it |
| 22:56.28 | JT | yeah, it's inconsistant |
| 22:56.57 | ltdwk | the ethernet card is eating up a lot of interrupts.... perhaps that's the cause? |
| 22:57.05 | JT | kink0: try running zttest, then start up bonnie++ at the same time, and see if there is a dramatic difference |
| 22:57.15 | kink0 | do you observe variations on zttest result, or only may be over 99.97 ? |
| 22:57.39 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 22:57.48 | JT | yeah the variations show something of course |
| 22:57.58 | JT | bonnie++ is a hard drive benchmarker |
| 22:58.04 | kink0 | I means, if is more important stability and low desviation, or high output |
| 22:58.19 | JT | kink0: stability, and never below 99.97% |
| 22:58.26 | kink0 | I have not any disck benchamark, but I can stress in other ways the CPU and bus |
| 22:58.37 | JT | kink0: just install bonnie++ |
| 22:58.49 | ltdwk | http://pastebin.ca/391179 |
| 22:58.49 | kink0 | ok, I got sometimes under 99.97%, not much, but some meassures was under |
| 22:58.49 | JT | it's packaged for most distributions |
| 22:58.56 | ltdwk | check that one out. that's from one of my production systems |
| 22:59.13 | JT | ltdwk: looking at those results, it does under far too often |
| 22:59.17 | JT | goes under |
| 22:59.31 | mmartinn | hdparm can work the drive as well |
| 23:00.15 | JT | kink0: i meant, too |
| 23:00.17 | kink0 | ltdwk, I see you dedicated second CPU for the TE card interrupts |
| 23:00.33 | ltdwk | kink0: that's right, so I know that nothing on any other device will effect it |
| 23:00.36 | kink0 | I suposse you configured irq 16 to use only the second cpu, right ? |
| 23:00.57 | ltdwk | and my timing never drops below 99.98 |
| 23:01.29 | ltdwk | Best: 100.000000 -- Worst: 99.987793 -- Average: 99.991806 |
| 23:01.33 | kink0 | I saw in my box, all interrupts was done by cpu 0, while cpu 1 is doing near nothing interrups |
| 23:01.44 | kink0 | my worst goes as low as 99.95 |
| 23:02.11 | mmartinn | How do you guys feel about using the realtime stuff for Asterisk? Is running mysql or hitting a networked mysql machine too much for asterisk? |
| 23:02.17 | JT | i've found audio is heavily distorted at 99.96 |
| 23:02.22 | mmartinn | (since this is a performance discussion) |
| 23:02.26 | JT | and you get flame slips |
| 23:02.29 | JT | frame slips |
| 23:02.33 | ltdwk | mmartinn: i run realtime ldap |
| 23:02.48 | mmartinn | locally or over the network? |
| 23:02.53 | ltdwk | over the network |
| 23:03.08 | mmartinn | I'm nervous about load >.< |
| 23:03.23 | ltdwk | i use it for voicemail, and it feels instant from a user perspective |
| 23:03.26 | JT | load in what sense? |
| 23:03.33 | mmartinn | Over working the box |
| 23:03.44 | bkruse_home | kink0: thats not too low, BUT, the interrupt is wierd, they should switch every 1 second, between CPU's( about ) |
| 23:03.49 | JT | mysqld and asterisk on the one box? |
| 23:04.10 | JT | bkruse_home: imho his zttest results were terrible |
| 23:04.11 | mmartinn | We run * in a call center; these machines record every single call and handle a lot of PRI traffic. |
| 23:04.14 | JT | yes i've seen worse |
| 23:04.21 | JT | but they didn't look usable |
| 23:04.27 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 23:04.41 | bkruse_home | JT: he said 99.96 was his worst? |
| 23:04.48 | JT | no |
| 23:04.50 | JT | 99.95 |
| 23:04.55 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-114-223.cia.com) |
| 23:04.57 | ltdwk | bkruse: that's not true, most drivers won't switch cpus for interrupt handling unless manually configured to do so |
| 23:04.57 | bkruse_home | hmm, that is GETTING bad........fo sho |
| 23:05.17 | JT | bkruse_home: nah, in the pastebin, they go down to 99.49 |
| 23:05.28 | JT | and it goes up and down |
| 23:05.38 | bkruse_home | ltdwk: ill show you right now if you want to screen in.........trust me, i have to do it as my job, most times ...........and maybe your right with dual cpu's, but smp it does |
| 23:05.43 | bkruse_home | JT: eww |
| 23:06.14 | JT | the worst i've seen from someone was 60% :o |
| 23:06.31 | bkruse_home | JT: woah! ive never seen that bad! |
| 23:06.43 | JT | yeah it was attrocious |
| 23:06.45 | *** join/#asterisk orkid_ (i=orkid@bas1-barrie18-1242374356.dsl.bell.ca) |
| 23:06.56 | mmartinn | On our crappy hardware we managed to bring it to 60-70 sometimes while using hdparm at the same time ;( |
| 23:07.07 | JT | damn |
| 23:07.19 | bkruse_home | mmartinn: haha |
| 23:07.23 | bkruse_home | hdparm |
| 23:07.30 | bkruse_home | i do that on my asterisk server all the time................ |
| 23:07.33 | bkruse_home | .......... |
| 23:07.47 | mmartinn | It was a test; I didn't put it in cron or something :P |
| 23:07.59 | ltdwk | kink0: echo 02 > /proc/irq/23/smp_affinity |
| 23:08.08 | ltdwk | then see what the output of zttest is |
| 23:08.30 | bkruse_home | ltdwk: interesting........ |
| 23:14.09 | *** join/#asterisk Strom_M (n=strom@c-71-207-196-66.hsd1.al.comcast.net) |
| 23:15.32 | ltdwk | bkruse_home: i notice on my dual core here at work certain drivers do share interrupts on both cores |
| 23:15.44 | ltdwk | bkruse_home: others don't |
| 23:16.51 | kink0 | ltdwk, ok, I was trying with setpci to set latency on that card to 0, but that appears did nothing |
| 23:17.50 | ltdwk | kink0: did you set the smp affinity as i said above? |
| 23:18.11 | bkruse_home | ltdwk: interesting...... |
| 23:18.23 | kink0 | yes, I did now, and now I am checking zttest output, |
| 23:18.23 | bkruse_home | i would like to see how/when/why they do |
| 23:18.41 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 23:18.52 | kink0 | I see at this time ( moving IRQ to CPU 1 ) the performance from zttest is worst ... |
| 23:19.27 | ltdwk | that can't be a good thing! |
| 23:20.04 | ltdwk | i'm still not convinced you're actually missing interrupts, in which case that won't help you anyway... check out zttool |
| 23:20.20 | ltdwk | check for IRQ misses |
| 23:20.38 | JT | kink0: does it have 2 cpus or one? |
| 23:20.59 | kink0 | Best: 100.000000 -- Worst: 99.853516 -- Average: 99.979154 |
| 23:21.12 | kink0 | JT: two fisical CPU, 4 if I enable HT ( now is dissabled ) |
| 23:21.22 | ltdwk | that's better kink0 |
| 23:21.28 | ltdwk | your worse was much lower before |
| 23:21.42 | JT | hmm |
| 23:21.47 | kink0 | ltdwk, yes, just take fews second after I pass IRQ to the second CPU |
| 23:21.49 | JT | ltdwk: depends how long he runs zttets for |
| 23:21.59 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-188-233-116.hsd1.ca.comcast.net) |
| 23:22.00 | kink0 | JT I did 65 passes |
| 23:22.20 | ltdwk | average is within acceptable range now |
| 23:23.13 | JT | yeah but even 1/100 outside the range, especially by that much, is unacceptable |
| 23:23.17 | ltdwk | though i don't think any of this is going to change your d channel signalling not working |
| 23:23.28 | JT | you need consistant good results if you don't want problems |
| 23:24.00 | JT | my timing never ever goes below 99.97 |
| 23:24.06 | ltdwk | neither does mine |
| 23:24.08 | JT | it stays very stable |
| 23:24.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:24.34 | kink0 | ltdwk, then ? you think is not a timming problem ? |
| 23:24.37 | ltdwk | but that will only effect audio quality |
| 23:25.10 | kink0 | audio was really fine, not noticiable lost of any packet of audio |
| 23:25.13 | ltdwk | kink0: I'd say you need to try one of those spans in your working box and see if it's alright |
| 23:25.29 | ltdwk | kink0: if so, possible hardware problem |
| 23:25.43 | kink0 | the problem is I was unable to get over about 3000 seconds a call up, because some times all calls was droped when D channel was missed |
| 23:26.15 | ltdwk | Show us the exact messages zap/asterisk was spitting out |
| 23:27.27 | ltdwk | Ok, so the spans actually work, calls are just unstable? |
| 23:28.50 | JT | and the d channels go down every so often |
| 23:28.57 | kink0 | ltdwk, yes, I get: Primary D-Channel on span 1 down.No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 23:29.21 | kink0 | when there about 40 or 50 calls, I got the error about every 10 minutes |
| 23:29.48 | kink0 | when there less that 30 calls, I practically do not get the error, or may be one time per hour or less |
| 23:29.55 | ltdwk | i didn't ever get that when i had timing issues... I got framing errors |
| 23:29.56 | *** join/#asterisk Strom_C (n=strom@66.0.239.106) |
| 23:31.17 | ltdwk | I'd be interested to see how it goes now you've moved it's interrupts to CPU1 with the same call load |
| 23:31.22 | kink0 | I will observe results for fews hours, after moving IRQ to second CPU |
| 23:32.04 | kink0 | yes, I will need to observe for a while, since some times I was thinking I fixed the problem, again the problem was there, just I need to check enough time |
| 23:32.05 | ltdwk | seems like your box wasn't able to cope with the interrupt load on the TE card |
| 23:33.06 | ltdwk | your worst went up from 99.4 to 99.8 so you may very well be able to scrape by |
| 23:33.41 | JT | dude... no |
| 23:33.43 | JT | seriously |
| 23:33.54 | JT | they are not acceptable scores |
| 23:34.21 | ltdwk | ultimately i think the hardware has to be replaced |
| 23:34.29 | JT | i'd be worried if 1 in every 500 passes hit 99.96 |
| 23:34.44 | JT | too much risk of d channel data corruption |
| 23:35.22 | JT | it does very with hardware, yes |
| 23:35.27 | ltdwk | i tried a billion things on some systems to fix timing but ultimately i always had to replace the hardware |
| 23:35.29 | JT | vary |
| 23:35.42 | JT | thankfully i've never needed to do that so far |
| 23:36.11 | ltdwk | i've settled on using tyan AMD mainboards and not had any issues |
| 23:36.25 | JT | yeah they work well |
| 23:36.30 | JT | as does ibm xeon |
| 23:36.34 | kink0 | I have other box ready , I can move the TE card to this other box tomorrow, in the event our suspects about hardare are ok. |
| 23:37.05 | kink0 | Best: 100.000000 -- Worst: 99.951172 -- Average: 99.993070 |
| 23:37.12 | kink0 | well, that is not too bad ? no ? |
| 23:37.28 | ltdwk | a hell of a lot better..... but not perfect |
| 23:37.37 | kink0 | ( average 35 calls now ) |
| 23:37.48 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 23:38.15 | kink0 | I started these issue two weeks ago, whe I saw a lot of CONGESTION while there free channels |
| 23:38.40 | *** join/#asterisk saftsack (n=oliver@pD9E04E66.dip.t-dialin.net) |
| 23:39.14 | kink0 | but CONGESTION was returned as $DAILSTATUS while $HANGUPCAUSE was 27 !!! ( no existing dialed number ) |
| 23:39.29 | *** join/#asterisk Vec (n=Vec@dsl-244-215-172.telkomadsl.co.za) |
| 23:40.03 | kink0 | then I changed IF..($DIALSTATUS) by IF...($HANGUPCAUSE) in my extension.conf |
| 23:40.29 | kink0 | but that was other issue, I think are not related to lost of D channel. |
| 23:40.34 | Vec | Does anyone know about the Grandstream BT-102, giving trouble with DHCP, I 12 of them and they all seem erraticaly get an ip address ? |
| 23:41.19 | kink0 | ltdwk, I see now fews calls over 3000 seconds duration, that is good |
| 23:41.49 | *** part/#asterisk bsd_tech (n=bsdtech@ppp-69-238-219-226.dsl.irvnca.pacbell.net) |
| 23:41.58 | ltdwk | kink0: that's good. Ultimately you still have issues but they may not present as your d-channel going wacko and breaking things, it may just be some clicks in the audio now |
| 23:42.56 | kink0 | ltdwk, I had not noticiable clicks in audio, or not much, before. |
| 23:43.28 | kink0 | ltdwk, also, I did vmstat, and see the CPU ussage appears better ( about 40% idle with 34 calls ) |
| 23:44.10 | ltdwk | kink0: you're spreading the load over both cpus now |
| 23:44.47 | kink0 | yes, the CPU load goes too high ( 2% idle ) just when fews bytes are writen to disk ( even with just 80 bytes ) |
| 23:44.49 | ltdwk | kink0: if asterisk is using cpu0, your TE card and your ethernet card are also using cpu0, that will exasperate the problem |
| 23:45.42 | kink0 | I can try to pass SCSI irq also to second cpu, but I will leave as is now , at least fews hours. |
| 23:46.27 | ltdwk | i wouldn't do that |
| 23:46.30 | ltdwk | leaving the TE card on its own cpu by itself is the best thing you can do |
| 23:46.47 | ltdwk | very often it's cumbersome interrupt handlers for block devices that cause high interrupt latency |
| 23:48.56 | *** join/#asterisk thekidrio (n=dbzkai@24-205-76-13.dhcp.psdn.ca.charter.com) |
| 23:58.42 | JacksLivr | could someone look at this and suggest my aweful mistake? thanks. http://pastebin.ca/391246 |
| 23:59.31 | Strom_C | your awful mistake is misspelling awful |
