| 00:53.14 | carrar | mmmm....Sounds |
| 00:53.36 | WIMPy | beeps |
| 00:54.03 | carrar | You mean, Background(beep) |
| 00:54.36 | carrar | Hows your holiday? |
| 00:54.42 | WIMPy | No, I don't accept extensions. |
| 00:55.00 | WIMPy | What holiday do you have? |
| 00:56.47 | carrar | I have no bitcoinsm sorry |
| 00:56.52 | carrar | m=, |
| 00:59.33 | carrar | I have labor day for $20 |
| 00:59.59 | carrar | What do you gots? |
| 01:00.21 | carrar | The first big Labor Day in the United States was observed on September 5, 1882 |
| 01:00.50 | carrar | Oregon was the first state to make it a holiday in 1887 |
| 01:00.53 | carrar | FYI |
| 01:01.48 | WIMPy | That's on may 1st here. |
| 01:02.32 | carrar | in the land of beer? |
| 01:03.00 | carrar | lived in Berlin a year |
| 01:03.11 | WIMPy | Hmm. I guess there are more of them. |
| 01:03.24 | carrar | not nevessary more, just better |
| 01:03.32 | WIMPy | I think Belgium has a lot to offer for beer lovers. |
| 01:05.06 | carrar | http://pics.osburn.com/photo/7511/original |
| 01:05.08 | carrar | heh |
| 01:06.03 | WIMPy | Is that what is displayed on an Air Show? |
| 01:06.19 | carrar | haha |
| 01:06.27 | carrar | just something I wanted to display |
| 01:06.30 | carrar | (tempelhoff) |
| 01:06.56 | carrar | back when you could ask someone for their gun and they would give it too you |
| 01:07.00 | carrar | hahah |
| 01:07.08 | WIMPy | I heard someone is collecting money to make Temelhof a big party area. |
| 01:07.37 | carrar | thats one hella large party area then |
| 01:07.49 | carrar | That was a showcase airport in it'sday |
| 01:07.59 | carrar | very impressive still |
| 01:08.00 | p3nguin | its day |
| 01:08.12 | WIMPy | I haven't seen it. |
| 01:08.14 | carrar | so much history there |
| 01:08.55 | WIMPy | Not sure there's much left since Berlin has been rebuilt quite massively the last 20 years. |
| 01:09.15 | WIMPy | They called it Europes largest building site. |
| 01:09.56 | carrar | well it was a major airport |
| 01:09.59 | carrar | lots of land |
| 01:11.03 | *** join/#asterisk adeeln (~adeel@184.175.36.92) |
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| 01:17.08 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:33.47 | *** join/#asterisk adolfomaltez (~taro@190.62.232.249) |
| 01:43.28 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 01:49.25 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 01:53.35 | *** join/#asterisk james_zhu (~Administr@183.16.205.99) |
| 01:55.47 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279412163.dsl.bell.ca) |
| 02:00.22 | dijib | p3nguin, found this finally.. http://ibot.rikers.org/%23asterisk/ |
| 02:00.37 | p3nguin | Took you a long time. |
| 02:00.48 | dijib | yeh well .. took me some time to get around to |
| 02:00.55 | dijib | and ive been looking at your working dialplan |
| 02:01.08 | dijib | callcentric & ipkall of any use to me? |
| 02:01.22 | dijib | i also still need to patch that COUNT thing |
| 02:01.26 | dijib | you found |
| 02:02.28 | p3nguin | CallCentric has some services, but they kind of suck. IPkall offers free DIDs (Washington state only). |
| 02:03.15 | dijib | LD services? |
| 02:03.36 | p3nguin | callcentric.com |
| 02:03.39 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 02:04.15 | dijib | ive got this dead air issue on dialout. 1 out of 4 calls the call connects but the call is silent on both ends. any idea? |
| 02:05.00 | p3nguin | Did you say you do see 180 Ringing in the sip messages? |
| 02:06.37 | dijib | yes |
| 02:07.28 | p3nguin | Show me your incoming extension. |
| 02:07.46 | dijib | is there a password protected pastebin like site? |
| 02:08.22 | ChannelZ | Yes. it's called Your Asterisk Doesn't Work |
| 02:08.24 | p3nguin | You can paste in pastebin.com and mark it as private, and use an expiration. |
| 02:08.46 | dijib | private oonly ppl who have the link can see? |
| 02:08.52 | p3nguin | correct |
| 02:09.02 | dijib | k then im just going to give you the whole dialplan. |
| 02:09.27 | dijib | i really need to clean it up those, i have ; exten => all over the place |
| 02:10.19 | p3nguin | It's probably okay for a new dialplan. You'll clean up unused lines when you get it sorted. |
| 02:11.15 | p3nguin | You could also remove all the lines starting with comments for the pasting, if you wanted. |
| 02:11.55 | dijib | incoming pvt msg p3nguin |
| 02:12.12 | dijib | too late also |
| 02:12.17 | dijib | :/ |
| 02:12.34 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 02:14.07 | p3nguin | grep -v ^\; /etc/asterisk/extensions.conf > extensions-no-comments.conf |
| 02:16.16 | dijib | wow |
| 02:16.19 | dijib | what else done i know |
| 02:18.35 | p3nguin | I'm not sure why your phones context includes the fax-in context. You don't plan to fax yourself from one of your phones, do you? |
| 02:19.35 | dijib | nope |
| 02:19.53 | dijib | thats prolly why, it was in or around the same times i added that |
| 02:21.32 | p3nguin | Are you willing to make some changes to your inbound context/extension? |
| 02:23.36 | dijib | sure yeh whatever you suggest i bet would be better then not |
| 02:23.42 | dijib | are you talking about the 100's? |
| 02:23.57 | p3nguin | Before we do that, I've got an automotive question to run past you. |
| 02:24.38 | dijib | sure i might not know ti though :s |
| 02:26.14 | p3nguin | I have a '98 Blazer 4x4, with rear disc brakes that are giving me a problem. The driver side rear brake keeps wearing down the inner pad like the caliper isn't releasing. I put a clamp on the caliper to compress it, and it seems to move freely but stiffly. Do you think the hose could be collapsed inside, preventing the caliper from letting go, or do you think the caliper needs replaced? |
| 02:27.45 | dijib | hmmm |
| 02:28.01 | dijib | i had a similar issue with my front passanger on that vehicle |
| 02:28.26 | p3nguin | The slide pins move freely, and I cleaned them well and put new caliper lube on them to see if that helps any. |
| 02:28.48 | dijib | i usually excersise the piston when i have it off |
| 02:29.05 | dijib | use the c-clamp and compress. then press break then repeat |
| 02:29.32 | p3nguin | The pins don't have much travel on this model, so even when the pins get frozen the pads don't usually wear down because of it. |
| 02:30.57 | p3nguin | I guess they actually can move up to about half an inch, but there is never that much movement in them during normal operation. |
| 02:31.19 | dijib | true.. what are you calling slide pins though? |
| 02:31.27 | dijib | yes im a nuub with cars too |
| 02:31.28 | p3nguin | the slide pins |
| 02:31.53 | dijib | i know the boot. the hydrolic line, the piston..... |
| 02:32.03 | dijib | whats the slide pin? |
| 02:32.20 | p3nguin | I thought you were more mechanically inclined than asterisk inclined. That's why I was giving you this question. |
| 02:32.35 | dijib | its like this { O }_ |
| 02:34.13 | p3nguin | The caliper bracket bolts to the axle tube end, and the caliper bolts to the bracket. The bolts actually attach to pins which slide in holes in the bracket rather than bolting solid, which allows side-to-side movement as the caliper compresses and releases. |
| 02:34.22 | dijib | ahh k i know |
| 02:34.29 | *** join/#asterisk adeel (~adeel@184.175.36.92) |
| 02:34.35 | dijib | i broke one of those once on the jimmy and had to replace the caliper or something |
| 02:34.48 | p3nguin | I broke one just three hours ago. |
| 02:35.04 | dijib | overtightening? |
| 02:35.09 | p3nguin | The pin on the passenger side was frozen up, and I used a big wrench trying to free it. |
| 02:35.19 | dijib | 3/4inch drive :D |
| 02:35.22 | dijib | kidding. |
| 02:35.27 | dijib | johnson bar |
| 02:35.29 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 02:35.29 | p3nguin | I twisted off the slide pin in the bracket. |
| 02:35.40 | p3nguin | I just used a wrench to break it off. |
| 02:36.05 | p3nguin | So I had to go get a new bracket and new pins. |
| 02:36.19 | dijib | crappy |
| 02:36.28 | raden | is there a particular linux os prefered for asterisk installs ? |
| 02:36.36 | p3nguin | Not really. |
| 02:36.38 | dijib | i need to fix my battery. the led terminal melton when i shorted it the other day |
| 02:36.48 | dijib | CentOS minimal |
| 02:36.50 | dijib | :D |
| 02:37.05 | p3nguin | CentOS and Debian-related distros have official packages available. |
| 02:37.35 | p3nguin | So we'll skip the car talk and get back to your dial plan. |
| 02:37.53 | dijib | i need a 3.43 tranny |
| 02:38.05 | p3nguin | Never heard of it. |
| 02:38.23 | dijib | the 4l60e |
| 02:38.26 | dijib | whatever thats called |
| 02:38.33 | dijib | doesnt it have to match gear ratio? |
| 02:38.56 | p3nguin | They don't work like that. |
| 02:39.23 | dijib | see im a car nuub |
| 02:40.05 | p3nguin | If you need a trans, and you're sure it's a 4L60E, just tell the people at the junk yard that you need a 4L60E for your '98 Suburban, or whatever year and vehicle you have. |
| 02:40.27 | dijib | so a trans is a trans across the board |
| 02:40.40 | p3nguin | If they don't ask if it is 4x4, be sure to tell them. |
| 02:40.53 | p3nguin | For the most part, a 4L60E is a 4L60E. |
| 02:41.02 | dijib | i didnt know that |
| 02:41.12 | dijib | i thought they were gear ratio specific |
| 02:41.19 | p3nguin | It's the year and engine size that makes the difference. |
| 02:41.43 | dijib | ye but 94-00 is good @ 5.7 |
| 02:41.46 | dijib | L |
| 02:42.16 | p3nguin | A 700R4 that was put behind a small V6 will be built with lighter weight parts than one put behind a 5.7L V8 4x4 full size truck. |
| 02:42.33 | p3nguin | The gear ratio is the same, though. |
| 02:43.16 | p3nguin | The rear end (or rear and front) ratio is where they make the differences for how the trans performs for a given car or truck. |
| 02:43.52 | p3nguin | Like for a Camaro, you'll often see a 3.23 rear end, but in a 4x4 full size truck, you might see 3.42, 3.73, or even 4.11 gears. |
| 02:44.35 | p3nguin | Anyway, back to your dial plan... |
| 02:44.44 | p3nguin | If I call your number, do you want me to hear any ringing before your system answers? |
| 02:45.09 | *** join/#asterisk adeel (~adeel@184.175.36.92) |
| 02:46.37 | dijib | no, i want to to answer right away and ask to enter a responce |
| 02:46.47 | dijib | response |
| 02:46.59 | dijib | sorry just had to the wife |
| 02:49.44 | dijib | your not rewriting everythingon me again r ya |
| 02:49.46 | dijib | ? |
| 02:50.28 | p3nguin | I'm going to help you build a dial plan rather than paste pieces together from the net. |
| 02:50.31 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:51.15 | dijib | i wrote some of that. other i got from you |
| 02:52.15 | *** join/#asterisk moy (~moy@bas5-toronto47-1168012481.dsl.bell.ca) |
| 02:52.56 | dijib | hey with your brake, have you excersised that piston much? i found if i cleaned it up & wd40 it helped |
| 02:52.57 | p3nguin | I usually like my first line to be a NoOp() so I don't ever have to renumber my priorities ever. I can change any single line after that and never have to worry about numbering the priorities. So lets start with your first line in your inbound context. I'll use s for my examples, but I expect you to use your actual phone number. |
| 02:53.19 | p3nguin | I didn't take it apart, but I worked it in and out a few times, and it didn't change the way it felt. |
| 02:53.34 | p3nguin | exten => s,1,NoOp() |
| 02:53.50 | p3nguin | Now you said you wanted it to answer and start playing your message right away... |
| 02:54.00 | p3nguin | exten => s,n,BackGround(IVR) |
| 02:54.19 | p3nguin | Then you'll want to have some time for someone to enter an extension... |
| 02:54.27 | p3nguin | exten => s,n,WaitExten(5) |
| 02:54.30 | dijib | yeah but i need an Answer() for fax detect |
| 02:54.40 | dijib | i said 15 in mine |
| 02:54.52 | p3nguin | How does the fax detection work? |
| 02:55.10 | p3nguin | BackGround() answers the line, so you do not need an explicit Answer() before it. |
| 02:55.10 | dijib | any fax with be detected on answer() |
| 02:55.46 | dijib | but i dont know if the fax detect needs an answer... i think thats what i read |
| 02:55.48 | p3nguin | Would the fax tones be detected while your message is playing? |
| 02:56.18 | p3nguin | If so, you've got your first three lines of dial plan written right up there. |
| 02:56.43 | dijib | yep |
| 02:59.46 | p3nguin | If you need to set some variables or write to some functions, you can do it before the BackGround(). That's where the beauty of the NoOp() comes in. |
| 03:00.23 | dijib | it just detects and sends to f |
| 03:02.00 | p3nguin | If you wanted to use the loop with limit, that can be incorporated here, too. |
| 03:02.17 | p3nguin | I'll put this in a pastebin. |
| 03:02.32 | dijib | i have your dialplan |
| 03:02.36 | dijib | dont worry about it |
| 03:02.45 | dijib | ill get that in tonight or tomorrow |
| 03:03.00 | *** join/#asterisk radic (~radic@dslb-178-007-128-228.pools.arcor-ip.net) |
| 03:03.04 | raden | Ubuntu have a asterisk package ? |
| 03:04.45 | carrar | compile from source |
| 03:04.48 | carrar | CFS |
| 03:04.57 | dijib | svn |
| 03:07.04 | dijib | hey p3nguin change your dot4 |
| 03:07.23 | p3nguin | I have no idea what that means. |
| 03:07.30 | dijib | break oil |
| 03:07.41 | dijib | hydrolic oil |
| 03:07.42 | p3nguin | Oh, I probably have DOT3 break fluid. |
| 03:07.54 | p3nguin | It isn't too bad, though. |
| 03:08.01 | p3nguin | I probably won't change it. |
| 03:08.02 | dijib | thought it was 4 4 that truck |
| 03:08.23 | p3nguin | I'd have to look at the cap for the reservoir. |
| 03:08.42 | dijib | ya but could be crap in the lines |
| 03:08.44 | p3nguin | I'm thinking about taking the caliper apart. |
| 03:08.56 | dijib | buy a repair kit with boot then |
| 03:09.00 | dijib | if you dot hat |
| 03:10.37 | p3nguin | I kind of need a new boot. The heat from the break hanging has burned the one on it now, so it's a little brittle around the edge. |
| 03:10.46 | p3nguin | brake, that is. |
| 03:11.16 | dijib | ya ull rip if u try to repair] |
| 03:12.54 | p3nguin | When I call your number, and I get your outgoing message, what extensions do you want me to be able to dial? Should I be able to dial all your phones if I know the extens? |
| 03:14.59 | dijib | any but outbound-voipms |
| 03:15.18 | dijib | rb |
| 03:15.28 | p3nguin | We won't include outbound in the internal context, so that shouldn't be a problem. |
| 03:22.00 | p3nguin | dijib: This should be pretty good for your inbound for starters: http://pastebin.com/xEv6XuEf |
| 03:23.20 | p3nguin | Just replace the s with your real number. |
| 03:24.13 | p3nguin | or add an exten => your-real-number,1,Goto(s,1); |
| 03:25.40 | p3nguin | That's actually how I handle inbound numbers. I accept them and then send them off to their own contexts where 's' is used rather than the phone numbers. That allows multiple phone numbers to all go to the same dialplan and do the same thing easily. |
| 03:28.16 | p3nguin | I need to get to work on my BBQ sauce for tomorrow. |
| 03:28.24 | dijib | lol |
| 03:28.28 | dijib | JD man JD |
| 03:28.43 | dijib | already :wq! it |
| 03:29.04 | dijib | go do your thing, ill ask my questions later about this thing |
| 03:29.08 | dijib | like i? |
| 03:29.13 | p3nguin | I don't think I'll put in any Jack into my sauce I'm making. I might, though, now that you've given me the idea. |
| 03:29.32 | dijib | or guness ive had work well for me |
| 03:29.34 | p3nguin | Go ahead and ask. The wife is taking up the counter space making slaw right now, anyway. |
| 03:29.40 | dijib | mmmm beer can chicken.... |
| 03:29.42 | ChannelZ | Anyone who says they don't do it is a liar |
| 03:29.43 | dijib | now i wanna BBQ |
| 03:30.34 | dijib | k i think its time to smoe a doobie and then look at FAX |
| 03:35.18 | dijib | ~book |
| 03:35.18 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 03:40.36 | p3nguin | So what part did you have trouble understanding? |
| 03:45.30 | dijib | nothing really ive just tested that bit and seems to be working well |
| 03:45.39 | dijib | i like the invalid |
| 03:45.49 | dijib | thinking that should also have a count eh? |
| 03:46.51 | dijib | hey back in 10m |
| 03:47.46 | p3nguin | exten i is included in the count. |
| 03:53.50 | *** join/#asterisk IPNixon (~IPNixon@unaffiliated/ipnixon) |
| 03:54.21 | IPNixon | hey all. is there any way to specify that a Flash() in extensions_custom.conf be done on a zap channel rather than on sip? |
| 03:59.25 | ChannelZ | It operates on whatever channel it's called from |
| 03:59.33 | ChannelZ | (it actually makes no sense to use on a SIP channel) |
| 04:01.01 | IPNixon | gotcha |
| 04:01.14 | IPNixon | i have two channels; 1 sip and 1 zap |
| 04:01.40 | IPNixon | and every time i try (calling the custom ext from a spa3102), it tries doing it on sip |
| 04:04.47 | ChannelZ | well if you were doing it from the zap/dahdi channel side it should work |
| 04:05.30 | ChannelZ | don't think there's a way to say "do a hook-flash on this other channel" |
| 04:05.58 | raden | we went from asterisk 1.8 to 10 ? |
| 04:07.12 | p3nguin | ~asterisk10 |
| 04:07.13 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 04:07.15 | ChannelZ | Yes. Think if it like 1.10, but not as a decimal number, and stop worrying |
| 04:07.23 | ChannelZ | s/if/of/ |
| 04:07.50 | dijib | are yo running *10? |
| 04:07.54 | dijib | . |
| 04:08.25 | p3nguin | Whom are you asking? |
| 04:11.01 | dijib | anyone |
| 04:11.04 | dijib | * |
| 04:11.13 | p3nguin | I run 1.4 |
| 04:11.23 | ChannelZ | 1.8 |
| 04:11.29 | dijib | 1.8.5 |
| 04:11.37 | dijib | i was thinking 10x |
| 04:11.52 | dijib | but chickHENed out |
| 04:17.01 | p3nguin | 10 times the fun, I guess. |
| 04:20.52 | WIMPy | The good fun or the bad fun? |
| 04:25.32 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:51.19 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 04:57.17 | *** join/#asterisk adeel (~adeel@184.175.36.92) |
| 05:07.52 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 05:18.20 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 05:23.14 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:f27b:cbff:fe9d:f3e4) |
| 05:41.11 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279412163.dsl.bell.ca) |
| 05:44.34 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 05:46.23 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 05:50.38 | dijib | p3nguin, still up? |
| 05:50.53 | dijib | i hope not seeing its monday 2am. |
| 05:50.59 | dijib | i mean 12. |
| 05:51.01 | dijib | go to bed. |
| 05:51.12 | p3nguin | Yes, I'm up. |
| 05:51.49 | dijib | exten => i,n,waitexten(2) means XX = invalid do. |
| 05:51.57 | dijib | i think i need 3 characters |
| 05:51.59 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 05:52.03 | dijib | i mean digits |
| 05:52.08 | p3nguin | I have no idea what you're saying. |
| 05:52.18 | p3nguin | But the dial plan I created for you is COPY AND PASTE. |
| 05:52.48 | dijib | that line says if any 2 numbers dont match say invalid |
| 05:53.07 | p3nguin | Uh, no. |
| 05:53.09 | dijib | oh nvmd is that wait time |
| 05:53.32 | p3nguin | That says when you've entered an invalid extension, wait 2 more seconds for a valid one before going to the next line. |
| 05:53.47 | p3nguin | And the next line takes you back to the previous waitexten. |
| 05:53.58 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:54.54 | dijib | n(variable) is a neat way to move the call around eh.. im still wrapping my head around it |
| 05:55.16 | irroot | morning good folk top 'o the morning ... .happy labor(less) day to the merkans |
| 05:55.52 | p3nguin | Those are priority labels, not variables. |
| 05:55.57 | *** join/#asterisk james_zhu (~Administr@183.16.205.99) |
| 05:56.06 | dijib | whats 't' extension |
| 05:56.12 | p3nguin | timeout |
| 05:56.33 | dijib | i should really read into it |
| 05:56.42 | dijib | with your ~book |
| 05:56.45 | dijib | ~book |
| 05:56.45 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 06:00.16 | dijib | why would asterisk & safe_asterisk be running but im not able to log into console |
| 06:00.31 | dijib | since sys build. |
| 06:00.56 | dijib | i need to run safe_asterisk to be able to connect |
| 06:01.25 | irroot | dijib did you edit safe_asterisk or run "asterisk -c" |
| 06:01.29 | p3nguin | I'm not going to help you run asterisk as root. |
| 06:02.06 | dijib | asterisk -c runs it. |
| 06:02.17 | dijib | where do i chmod that? |
| 06:02.23 | dijib | to run as? asterisk |
| 06:02.34 | dijib | or edit cfg/ |
| 06:02.42 | irroot | you then running it manually ?? rather use safe_asterisk |
| 06:03.07 | dijib | nevermind http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-13-SECT-4.html |
| 06:03.38 | dijib | i would rather have safe_asterisk start it, as it monitors if asterisk is running and respawns if not |
| 06:05.26 | dijib | i have 2 instances of each running apparently |
| 06:05.28 | dijib | huh |
| 06:05.56 | ChannelZ | probably not really, if you are using asterisk -r |
| 06:07.03 | dijib | huh? |
| 06:07.24 | ChannelZ | right |
| 06:07.30 | p3nguin | What does "ps -C asterisk u" show you? |
| 06:07.54 | dijib | ive got 2x safe asterisk & 2x asterisk -c |
| 06:08.04 | dijib | so how do i connect? |
| 06:08.09 | dijib | i use the -r switch |
| 06:08.16 | p3nguin | Use fire. |
| 06:08.25 | p3nguin | big torch |
| 06:08.26 | dijib | tah hek is fire |
| 06:08.29 | ChannelZ | oh. thats's probably a problem. |
| 06:08.38 | dijib | ive only got a little canaster |
| 06:09.24 | dijib | where is /k when you need them |
| 06:09.51 | p3nguin | fire /k ? |
| 06:09.57 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:12.45 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 06:12.48 | p3nguin | There I went. |
| 06:15.17 | dijib | you went where? |
| 06:15.34 | dijib | and im vi'ing safe_asterisk now |
| 06:15.38 | dijib | damn im slow |
| 06:19.31 | dijib | how did the BBQ sauce turn out? |
| 06:20.20 | *** join/#asterisk qjb (~qjb@2001:980:4cc7:1:2455:6c84:9a1:32d7) |
| 06:20.59 | dijib | and i was thinking about your break lines and you thinking they have callapsed. if front then i doubt it if back then trace line and look for kinks, could be a break in pressure, ive had the fittings go on me before |
| 06:21.44 | dijib | /usr/sbin/asterisk -f -U asterisk -vvvg -c |
| 06:21.53 | dijib | thats whats runnings now |
| 06:35.33 | dijib | cepstral? |
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| 06:47.06 | dijib | p3nguin, you are a beautiful man. fax is working |
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| 07:58.19 | GreatSUN | hi all |
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| 08:16.36 | kwk | Hi. I have an asterisk with realtime queues. When I fire an AMI "queuestatus" event for say queue 1650, the AMI returns no info for the queue. When I do "queue show 1650" on the CLI and fire the same AMI event from before afterwards, I get the correct queue status. Here's the output: https://gist.github.com/1194380 |
| 08:16.51 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:17.40 | aberrios | ffs I beginning to really dislike ZoiPer |
| 08:18.04 | kwk | aberrios: I'm quite happy with jitsi btw. |
| 08:18.30 | aberrios | kwk, ta, might give it a go |
| 08:18.44 | aberrios | zoiper seems to be randomly crashing |
| 08:18.45 | catphish | kwk: asterisk doesn't know about realtime objects until they're used |
| 08:18.49 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:20.17 | kwk | catphish: ok, but when what's the difference between "queue show 1650" and "action: queuestatus\n queues:1650" ? I mean, both want to know ("use") the queue. Why is one treated differently to the other? |
| 08:20.59 | catphish | queuestatus probably lists all known queues then filters it (though that's just a guess) |
| 08:21.10 | catphish | queue show 1650 obviously looks it up |
| 08:21.38 | kwk | catphish: that's odd |
| 08:21.43 | catphish | how so? |
| 08:22.15 | catphish | did you actually give the queuestatus the id? |
| 08:22.25 | irroot | realtime queues are only loaded when a event occurs |
| 08:22.50 | kwk | irroot: ok, can I create such an event via the AMI? |
| 08:23.07 | irroot | a call is a event |
| 08:23.27 | kwk | irroot: one sec. want to try it out |
| 08:23.31 | irroot | so is logging in / out a memmber or setting member penalty / paused |
| 08:23.31 | catphish | can't you just assume that if it's not listed, its not in use? |
| 08:23.54 | irroot | but the latter is asterisk 10 |
| 08:24.10 | irroot | there been some changes i commited recently |
| 08:24.45 | kwk | irroot: i use 1.8.5 |
| 08:24.57 | irroot | ok then a call is best bet |
| 08:25.08 | catphish | i agree it would be nice if asterisk could do a "select all" on a realtime database when doing things like "sip show peers" |
| 08:25.37 | catphish | but i'm happy to assume if something isn't listed, its never been used |
| 08:27.47 | irroot | catphish the problem with that is it will load all the entries into ram and not have the bennifit of freeing it up when not needed |
| 08:28.33 | kwk | irroot: But there has to be a way to figure out what agents are logged in. An event is not enough by the way. When I login to 1650 an event "Newstate" and "Hangup" is fired but the queue status is still empty. When I have a call event though, the queue is loaded via AMI correctly. And then it'll show all it's logged in agents. |
| 08:28.37 | catphish | irroot: why would it not free them? it would only need to load them when the request was made, they could be immediately dropped again |
| 08:29.18 | catphish | wouldn't a queue be loaded when someone is logged into it? |
| 08:29.37 | kwk | catphish: no, not in my case |
| 08:29.45 | irroot | kwk i use realtime and look at the queue_members table to see who is in or out not via AMI i query the DB directly |
| 08:30.04 | kwk | irroot: i see. |
| 08:30.21 | catphish | that's a much saner approach |
| 08:30.36 | irroot | you could check the interface in ami to test there status |
| 08:32.17 | kwk | irroot: ok, this works. if there're callers waiting in the queue i get their status correctly. I'll query the agents via queue_members tables then. |
| 08:32.51 | irroot | kwk cool as long as it works for you |
| 08:33.43 | GreatSUN | irroot: hi |
| 08:33.47 | kwk | irroot: i mean, it would have been nice if the agents where available via the AMI though. I my opinion asterisk shouldn't differe in it's behaviour if queues are stored realtime or static. |
| 08:33.59 | irroot | hi there greatsun |
| 08:34.07 | GreatSUN | irroot: do you think you could help me with asterisk + dahdi and cidtransfer? |
| 08:34.17 | GreatSUN | afaik asterisk seams to set cid correctly |
| 08:34.40 | GreatSUN | but it doesnt seam to be set correctly to/from dahdi |
| 08:34.41 | irroot | kwk ill look into the ami issue and rework it want to open a bug for this and give me the id so i can work it |
| 08:34.59 | kwk | irroot: will open a bug |
| 08:35.52 | irroot | GreatSUN mmm not something i work with much what you expecting ?? the cid coming in on dahdi to go to sip ?? |
| 08:36.43 | irroot | kwk i have a way in my mind on how to do it quite easy im working with app_queue code atm there is a deadlock im trying to kill |
| 08:37.12 | GreatSUN | irroot: sip -> asterisk -> dahdi-chan -> dial through isdn |
| 08:37.45 | kwk | irroot: cool |
| 08:38.26 | catphish | irroot: do you work on realtime mysql at all? |
| 08:38.50 | irroot | no use odbc with pgsql mostly |
| 08:39.13 | kwk | irroot: i use odbc with mysql. so it shouldn't be a problem :) |
| 08:39.23 | catphish | ok, i'm looking for some feedback on a reasonably nasty hack i'm using |
| 08:39.24 | irroot | GreatSUN you want to set the number going out on dahdi ?? |
| 08:40.31 | GreatSUN | yeah |
| 08:42.24 | catphish | irroot: http://paste.codebasehq.com/pastes/599 |
| 08:44.09 | *** join/#asterisk greenwolf (~root@cpe-74-77-221-5.buffalo.res.rr.com) |
| 08:45.23 | greenwolf | good morning anyone around ? |
| 08:53.36 | *** join/#asterisk porche (~kursad@78.182.67.144) |
| 08:53.52 | porche | Hi |
| 08:54.22 | porche | I have got a question about AGI dial command, |
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| 08:57.43 | kaldemar | ~ask |
| 08:57.43 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 08:58.17 | kwk | irroot: https://issues.asterisk.org/jira/browse/ASTERISK-18416 |
| 08:58.38 | kwk | irroot: thank you for working on this!!! |
| 09:00.29 | irroot | catphish that hack is only for mysql i will need to see the code what is the motivation |
| 09:00.51 | catphish | irroot: please see the jira ticket linked to it |
| 09:01.33 | porche | Hi again sorry |
| 09:01.45 | porche | I have got an AGI application |
| 09:01.51 | porche | that dials a number, |
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| 09:02.02 | porche | if I do AGI->exec(Dial |
| 09:02.08 | porche | it loses callerid |
| 09:02.21 | porche | I use asterisk 1.6 |
| 09:02.39 | porche | I tried setting caller id on dial plan |
| 09:02.46 | porche | with a channel variable |
| 09:03.44 | ChannelZ | how |
| 09:06.21 | porche | _1NXXNXXXXXX,1,Set(CALLERID(all)=${callername} <${caller}>); |
| 09:08.03 | kaldemar | porche: how did you verify that the caller id is lost? where are you setting variables callername and caller? |
| 09:08.40 | porche | from cdr |
| 09:08.40 | *** join/#asterisk Gambith (~me@201.170.214.194.dsl.dyn.telnor.net) |
| 09:08.47 | porche | and the number I am getting the call |
| 09:08.53 | porche | I mean |
| 09:08.57 | porche | phone I am getting the call |
| 09:09.05 | ChannelZ | well assuming those variables are right, it should work - though only the number matters. It also assumes your ITSP allows you to set the CID to whatever you want |
| 09:09.30 | kaldemar | what phone is getting the call? |
| 09:09.35 | porche | yes normally it's accepted |
| 09:09.41 | ChannelZ | try just Set(CALLERID(num)=${caller}) for fun |
| 09:09.55 | porche | the issue is actually |
| 09:10.06 | ChannelZ | (again, assuming ${caller} is 1112223333 or whatever) |
| 09:10.07 | porche | 1. a call is done to 1st number |
| 09:10.23 | catphish | irroot: https://issues.asterisk.org/jira/browse/ASTERISK-18271 that was what i was trying to address, i assume the same problem exists in ODBC though I haven't tested it |
| 09:10.24 | porche | 2. when call is connected, another call is conducted, it's like call bridging |
| 09:10.25 | porche | but |
| 09:10.42 | porche | I have to do it over AGI as some actions required, |
| 09:10.52 | porche | I set the caller and callername from AGI |
| 09:10.54 | porche | and execute |
| 09:11.13 | Gambith | that sounds like a callcenter process... |
| 09:11.15 | kaldemar | AGI is not the problem, your issue is elsewhere. |
| 09:11.15 | porche | AGI->exec(Dial |
| 09:11.21 | porche | yes Gambith |
| 09:11.31 | irroot | catphish yeah it will work the basic idea is there but i dont see it been adopted into asterisk as multiple possible matches in dialplan is not supported and discouraged |
| 09:12.00 | porche | true Kaldemar, it's interesting |
| 09:12.09 | porche | I can set the accountcode |
| 09:12.12 | porche | from the same AGI |
| 09:12.26 | catphish | irroot: really? i was under the impression that there was an entire algorithm to select the correct match from in-memory dialplans |
| 09:12.47 | Gambith | porche, is vicidial in your solution ? |
| 09:13.04 | porche | no Gambith, it's custom |
| 09:13.08 | Gambith | oh.. IC |
| 09:13.14 | catphish | overlapping dialplan entries seem essential for anyone that wants to do cost-based routing |
| 09:13.36 | irroot | catphish there is a algo to do it AFAIK and we can check how official it is |
| 09:14.13 | porche | interesting, seems like |
| 09:14.13 | irroot | catphish i use wildcard / look up to do cost based routing in a DB |
| 09:14.18 | porche | Dial command on AGI |
| 09:14.34 | porche | which produces a new channel, does not copy existing variables |
| 09:14.47 | catphish | so you just match all outgoing numbers then odbc the route? that makes sense |
| 09:15.36 | catphish | the main problem is that in the UK, we have 0[1-9]. for a national rate number and 00. for international |
| 09:16.08 | catphish | although actually they don't overlap, so ignore me |
| 09:16.20 | kaldemar | porche: you better describe the whole scenario properly if you want someone to help you. |
| 09:17.39 | Gambith | Has anyone tried securing a line by implementing a code to allow calls to go out from an xt ? meaning.. the xt wil ring and accept calls, but in order to make a call to the pstn u need to key in a code |
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| 09:19.40 | catphish | Gambith: sounds fairly easy, just let someone dial 9, then ask them for the code |
| 09:19.48 | catphish | then allow them do dial a number |
| 09:20.22 | kaldemar | Gambith: core show application Authenticate |
| 09:20.30 | Gambith | tnx.. will read |
| 09:20.36 | catphish | even better :) |
| 09:27.29 | Gambith | great.. auth(file,options) in the file the structure is xt:md5hash |
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| 11:23.08 | puzzled | hi |
| 11:24.38 | stix_ | hi |
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| 11:45.41 | jkroon | hi guys, i'm seeing a potential issue with 1.8.5.0 and the http manager: WARNING[1447] manager.c: HTTP Manager, fdopen failed: Bad file descriptor! |
| 11:47.24 | jkroon | there only seems to be two locations in manager.c that performs an fdopen - how can i figure out which one is triggering the error, and more specifically, what's the cause? |
| 11:51.49 | irroot | jkroon either in auth_callback or generic_callback was it during auth ?? |
| 11:52.02 | jkroon | irroot, very good question ... |
| 11:52.18 | jkroon | no, after auth, when request SIPPeers |
| 11:52.40 | jkroon | remember this is manager via http. |
| 11:52.55 | jkroon | not sure how that interaction works. |
| 11:52.57 | irroot | there you have your answer young padowin ... use the source luke :P and have a good week |
| 11:54.35 | jkroon | irroot, i know about the source :p. what i don't know is where to start reading. |
| 11:58.08 | *** join/#asterisk af_ (~getsmart@78.134.21.126) |
| 11:59.09 | irroot | jkroon yeah only teasing |
| 11:59.36 | irroot | is it hapening often ?? |
| 11:59.56 | irroot | maybe change the message to include auth in the one so you will know |
| 12:03.07 | jkroon | well, a full backtrace would be useful. |
| 12:03.20 | jkroon | so can I make it dump core before continueing on it's merry? |
| 12:06.11 | irroot | dont think so unless you catch it while its breaking |
| 12:06.47 | irroot | need a breakpoint |
| 12:10.49 | jkroon | so there is no way to ask glibc/the kernel to perform a coredump of the running process and then continue running? |
| 12:10.52 | jkroon | that sucks. |
| 12:12.05 | jkroon | backtrace + backtrace_symbols_fd() perhaps ... |
| 12:13.15 | irroot | gdb |
| 12:14.01 | jkroon | i would have preferred a core dump as I can load that into gdb yes, but I don't want to run production systems inside of gdb. |
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| 12:31.25 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 12:31.31 | Dovid | how is fax deteciton done in Asterisk 1.8 |
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| 12:34.57 | wdoekes2 | jkroon: you can do a bt and detach |
| 12:34.59 | wdoekes2 | see -ex |
| 12:35.34 | jkroon | wdoekes2, I want to add code inside asterisk to when it happens immediately generate a core dump, or what do you suggest? |
| 12:35.49 | jkroon | keep in mind this is a production system where I am seeing this, I seem unable to reproduce elsewhere. |
| 12:36.27 | wdoekes2 | I didn't read much backwards.. I simply wanted to say that you can get a backtrace without having to "stop" the process (prolonged) |
| 12:36.40 | jkroon | yea, that's the problem :( |
| 12:37.03 | wdoekes2 | ? |
| 12:37.46 | Dovid | how is fax deteciton done in Asterisk 1.8 |
| 12:38.17 | wdoekes2 | Dovid: if we don't answer in 6 minutes, does not mean that you need to spam |
| 12:38.27 | Dovid | spam ? wow. thats harsh |
| 12:38.45 | wdoekes2 | no it's not.. your message hasn't even scrolled 10 lines yet |
| 12:42.36 | Dovid | lol |
| 12:42.42 | Dovid | looking at the default confs nw |
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| 13:01.51 | *** join/#asterisk greenwolf (~root@cpe-74-77-221-5.buffalo.res.rr.com) |
| 13:02.01 | greenwolf | asterisk is still hold the rtp media |
| 13:03.30 | greenwolf | asterisk is suppose to send a new INVITE and connected both callers with rtp media on their end |
| 13:03.46 | greenwolf | any reason why asterisk is hosting these rtp media streams still after the call is setup? |
| 13:03.53 | greenwolf | i did directmedia=yes |
| 13:05.08 | kaldemar | greenwolf: maybe you have some other option enabled that requires asterisk to stay on the media path, for example option t or T in app Dial. |
| 13:05.20 | *** join/#asterisk serafie (~erin@nat/digium/x-kgrvzgvaezmeoesa) |
| 13:05.59 | greenwolf | i cant tell because i send the calls directly to a2billing.php script |
| 13:06.06 | greenwolf | to process the calling card numbers |
| 13:06.23 | kaldemar | then see what the script does. |
| 13:07.12 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 13:07.14 | greenwolf | so take both the r and R out of dial |
| 13:09.02 | as001 | Hello does anyone know response codes of Asterisk manager interface ? I have got 4 but sometimes 12 24 72 etc... I use perl module Asterisk::Manager and its sendcommand method on Asterisk 1.6.2.16. |
| 13:09.15 | Dovid | if using T/t and INFO for DTMF shouldn't rtp go direct? |
| 13:09.53 | greenwolf | im having problems getting rtp to go direct from openser to asterisk to outbound |
| 13:12.51 | as001 | I meant event is always the same and $ok from $ok = $astman->sendcommand ... is sometimes 4 sometimes 12 or 24 or 72 etc... |
| 13:13.36 | as001 | Where can I found meaning of those codes |
| 13:13.43 | *** part/#asterisk as001 (~uros@82.117.198.142) |
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| 13:18.26 | greenwolf | s |
| 13:20.03 | atan | Silly question, but using a PAP2 type adapter (forget the correct name, but the one that you run into your regular POTS line not your phone) can you connect your SIP phone directly somehow or _must_ there be a middle man SIP proxy? |
| 13:26.06 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 13:28.52 | irroot | atan its possible dpending on the devices |
| 13:31.57 | atan | irroot, it would be that linksys adapter, I think it says SPA something or order. |
| 13:32.11 | irroot | SPA-3102 |
| 13:32.22 | irroot | it will send calls to and from its self |
| 13:32.33 | irroot | or from it |
| 13:32.56 | atan | I'll need to hunt mine down to get the exact model. My interest in this would be to use a polycom VOIP phone without needed to put a small Asterisk box locally. |
| 13:33.00 | irroot | to a sip device as long as the sip device will accept calls it should be fine |
| 13:33.03 | atan | I will if I must, but... was totall just curious. |
| 13:33.34 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 13:33.43 | irroot | same would apply to a tennor but that has more intelegence and will deffinatly work |
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| 13:35.16 | *** join/#asterisk AviMarcus (~avi@bzq-79-182-184-28.red.bezeqint.net) |
| 13:36.10 | AviMarcus | Hi. Does anyone have any suggestions for reasonable quality, good priced a-z routes? With a low minimum commitment? I'm currently looking for australia, the price I have now is 1.3c/minute.. |
| 13:38.52 | *** join/#asterisk SirDekar (be22cebd@gateway/web/freenode/ip.190.34.206.189) |
| 13:39.38 | SirDekar | hi, question, why voicemail fails sometime? it starts to say "you have..." then suddently hangup |
| 13:47.17 | kaldemar | SirDekar: what do you see in CLI with verbosity and core debug enabled? sounds like the problem is with digit sounds. |
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| 13:57.05 | irroot | i may be biased but t38gateway rox |
| 14:01.46 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 14:02.46 | Dovid | lol |
| 14:02.59 | Dovid | iroot: Of course yo are ;) Where is it holding? |
| 14:03.26 | Dovid | AviMarcus: You can't get cheap and quality. you get what you pay for ;) |
| 14:03.50 | AviMarcus | Hey Dovid. I said reasonable :) |
| 14:03.58 | AviMarcus | and apparently I get 1.1c to the normal places. |
| 14:04.01 | Dovid | you can try voipjet but you need a carriers liscence to use them. you can also try teliax but they arent the cheapest. no one will give you cheap and no commit. it's how we roll |
| 14:04.07 | Dovid | good luck 1.1 |
| 14:04.10 | Dovid | with no commit |
| 14:18.56 | *** join/#asterisk jkroon (~jkroon@dsl-241-232-76.telkomadsl.co.za) |
| 14:19.50 | Dovid | anyone here have luck with the digium fax driver? |
| 14:21.43 | *** part/#asterisk AviMarcus (~avi@bzq-79-182-184-28.red.bezeqint.net) |
| 14:26.27 | atan | Dovid, how hard is it to get a carriers license? |
| 14:29.28 | *** join/#asterisk last1 (~last1@modemcable238.94-200-24.mc.videotron.ca) |
| 14:29.48 | atan | goes to check out the website |
| 14:29.59 | atan | Their price list causes firefox great pain. |
| 14:30.00 | Maliuta | atan: why do you want one? What country are you in? |
| 14:30.27 | Maliuta | atan: oh, and why are you still using FF? |
| 14:30.29 | atan | Maliuta, I don't? Oh who knows. I just resell voip to a few friends as a novelty service. |
| 14:30.39 | atan | Maliuta, you going to go all chrome on me? |
| 14:31.09 | atan | Hmm... is Voipjet == Voip.ms value or == voip.ms premium? |
| 14:31.10 | Maliuta | atan: there are several options, chrome is just one |
| 14:31.37 | last1 | any of you compiled dahdi succesfully on virtual ( xen ) debian ? |
| 14:31.56 | Maliuta | last1: VM or not shouldn't matter |
| 14:32.05 | last1 | I'm trying to do that but it fails with: error in zaphfc/base.c : error: 'modes' undeclared (first use in this function) |
| 14:32.11 | Maliuta | last1: you have your debian foo screwed up ;P |
| 14:32.25 | WIMPy | Too old version? |
| 14:32.29 | Maliuta | last1: looks like you have a missing .h |
| 14:32.34 | *** join/#asterisk marl_scot (~matt@office.unk.com) |
| 14:32.43 | Dovid | atan: It depends where |
| 14:32.55 | last1 | I run debian 2.6.35-4 |
| 14:33.01 | last1 | and the error is in base.c:1687 |
| 14:33.04 | Maliuta | last1: is this raw source? or are you using a source package |
| 14:33.22 | Maliuta | last1: that is a kernel version, not a debian version |
| 14:33.32 | last1 | sorry, debian 6.0 |
| 14:33.52 | last1 | I am building the dahdi package manually |
| 14:34.04 | Maliuta | last1: from what source? |
| 14:34.15 | marl_scot | can anyone tell me if the following should work to reset the device state (as shown by 'core show hints') : exten 5210 => Set(DEVICE_STATE(210@internal=NOT_INUSE)) |
| 14:34.17 | last1 | let me check. whichever apt-get install dahdi-source got me |
| 14:34.32 | last1 | 2.3.0.1 |
| 14:34.34 | Maliuta | last1: apt-get build depends |
| 14:34.46 | Maliuta | or build-depends |
| 14:35.07 | Maliuta | or build-dep |
| 14:35.08 | atan | Interesting. Why do voip providers not like "call center traffic" ? |
| 14:35.24 | atan | I'm looking at voipjet right now and they make it very clear they want nothing to do with a call center. |
| 14:35.32 | marl_scot | (* 1.8.5.0) |
| 14:35.33 | last1 | 514MB to be installed.. great |
| 14:35.34 | last1 | lol |
| 14:35.47 | atan | Are they just trying to avoid the telemarketing scams or would inbound customer service not be allowed? |
| 14:35.51 | Maliuta | atan: because there is a smeg load of it and the bandwidth cost for them would outweigh the income it generated? |
| 14:36.21 | atan | Maliuta, if they bill per minute I don't understand why they would not want it though. |
| 14:36.42 | Maliuta | atan: see my above question |
| 14:36.54 | Maliuta | bandwidth != free |
| 14:37.36 | atan | Yes of course but voip is all sold per minute right, for the most part, so why would you not want more minutes being used =\ |
| 14:38.01 | Maliuta | actually not all VoIP is sold per minute |
| 14:38.21 | atan | Well fair enough but their website shows termination rates per minute/second |
| 14:38.22 | Maliuta | I call my parents in .ca for $0.08 untimed |
| 14:38.43 | atan | Woah. Hold up. 8 cents and no call limit? |
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| 14:39.18 | atan | Any chance your provider is seeking new customers if this is the case? :) |
| 14:40.41 | Maliuta | atan: if you don't mind your packets coming all the way to .au and then back. $0.08AU untimed to US, UK, and a whole bunch more |
| 14:40.52 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 14:41.04 | atan | Maliuta I would love to know more. Please share! |
| 14:41.21 | Maliuta | atan: www.pennytel.com.au |
| 14:42.37 | last1 | ok, I built all the deps |
| 14:42.46 | last1 | and it's still failing on that line |
| 14:42.50 | last1 | how can I see what is missing ? |
| 14:43.38 | Maliuta | last1: search on packages.debian.org |
| 14:45.12 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 14:46.19 | Maliuta | last1: I'd say, from a cursory look that you're not using debian packages (i.e. packages from the debian repositories, or that are in a debian release) |
| 14:46.39 | Maliuta | last1: in which case I'm going to say "you're up shit creek ..." |
| 14:49.49 | Faustov | atan: http://www.txrxcomms.co.uk/ - it's run by a guy who often sits here, good quality and prices |
| 14:50.48 | atan | takes a look |
| 14:51.43 | last1 | well, I did: apt-get install dahdi ( comes with dahdi-linux ) |
| 14:51.50 | last1 | then I installed dahdi-source |
| 14:52.21 | last1 | the thing is that I run a Xen kernel and I have to run m-a with -k /usr/src/kernels/2.6.34-5.kernel |
| 14:52.28 | last1 | which m-a does not like |
| 14:52.29 | Maliuta | last1: that all depends on what your apt sources are |
| 14:52.52 | last1 | apt-get install dahdi-source puts a dahdi.tar.bz2 file in /usr/src |
| 14:52.57 | last1 | so that's what I work with, doing it manually |
| 14:53.05 | Maliuta | is going to bed before he becomes a pumpkin |
| 15:13.31 | marl_scot | anyone know how to reset * hints for an extension? i have a problem sometimes with one of my * boxes, where the hints get confused and an extension whos as inuse when it isnt, have tried using set device_state, but it doesnt seem to reset the state :( |
| 15:13.51 | marl_scot | s/whos/shows/ |
| 15:22.05 | irroot | its beer 'o clock |
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| 15:33.19 | sunfone | mmm beer |
| 15:33.34 | *** join/#asterisk Fritz09 (~Adium@pop1-3802.catv.wtnet.de) |
| 15:33.43 | sunfone | anyone have a .cnf file for Cisco 7940 they might share? |
| 15:34.42 | atan | sunfone, sure |
| 15:35.17 | sunfone | cool! |
| 15:35.31 | atan | http://pastebin.com/4wMAF0Ku |
| 15:35.33 | irroot | ~beer sunfone |
| 15:35.33 | infobot | ACTION deftly decants a fine Piraat for sunfone |
| 15:36.48 | sunfone | cheers irroot! |
| 15:38.24 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 15:40.35 | tzafrir_laptop | last1, don't you have a symlink from /lib/modules/`uname -r`/build to /usr/src/kernels/2.6.34-5.kernel ? |
| 15:41.31 | tzafrir_laptop | m-a a-i dahdi # should work in that case |
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| 15:46.39 | gnabn | Hi. Is there a way / function I can use to check RTP packets and detect when an IVR waiting ring stops and the call is answered with voice? |
| 15:46.58 | navaismo | with rtp set debug on |
| 15:47.53 | p3nguin | dovid: Yes, many of us use fax for asterisk successfully. |
| 15:48.20 | gnabn | thanks navaismo, you are talking about production envirnment, not testing? |
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| 15:50.56 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279412163.dsl.bell.ca) |
| 15:52.40 | gnabn | navaismo: where you talking about production environment or just for debugging? |
| 15:53.01 | dijib | who is a txfax pro? |
| 15:53.10 | navaismo | gnabn that command wil show you a flood of rtp packets received and send, it works in the asterisk cli, so yes for both enviroments can use it |
| 15:53.10 | p3nguin | I use SendFAX(). |
| 15:53.28 | p3nguin | Since that's what fax for asterisk uses, and all. |
| 15:53.32 | navaismo | but it will show a lot information in the cli |
| 15:54.12 | Dovid | anyone use rxfax with Audiocodes? |
| 15:54.28 | gnabn | navaismo: I see, I'll need to parse maybe, if I need it for production purposes. I'll try to dig from here. Thanks |
| 15:55.04 | dijib | ok that script i have whats it using. ? im trying to get email2fax working |
| 15:55.12 | dijib | and in that script it calls for txfax |
| 15:55.29 | p3nguin | I don't know anything about such script. |
| 15:55.38 | dijib | so i tried to compile and install spandsp is it? |
| 15:55.41 | p3nguin | I've seen it mentioned, but I've never looked at it. |
| 15:56.03 | dijib | ive almost got it working to the point where it calls for txfax |
| 15:56.06 | dijib | to transmit |
| 15:56.36 | p3nguin | I don't see any reason SendFAX wouldn't work just as well. |
| 15:57.07 | dijib | you think> |
| 15:57.08 | dijib | ? |
| 15:57.09 | p3nguin | Where did you get email2fax? I'll take a quick look at it. |
| 15:57.24 | dijib | im trying to have an email address i email and have it send a fax through that |
| 15:57.30 | p3nguin | I know. |
| 15:57.34 | dijib | im just about to pastbin it for you |
| 15:57.51 | p3nguin | You could just give me the link to the page where you get email2fax. |
| 15:58.49 | dijib | http://pastebin.com/8kuxXLhG |
| 15:59.00 | dijib | now ive got to find the link |
| 15:59.21 | dijib | http://wpkg.org/email2fax/index.php/Installation |
| 16:01.36 | p3nguin | Maybe it'll work. |
| 16:01.52 | p3nguin | For now, I'll stick with my dail-a-fax method instead of email2fax. |
| 16:02.02 | dijib | ive got it to the point where it reads the email, and tries to send the fax |
| 16:02.12 | dijib | i need email2fax |
| 16:02.21 | dijib | makes things easy |
| 16:02.25 | dijib | wanna see the output? |
| 16:02.33 | dijib | and i smell leftover lobster in the garbage |
| 16:03.13 | dijib | yikes i need to skip to the lue ill be right back |
| 16:03.24 | p3nguin | I don't have any trouble picking up the phone and dialing some numbers to send a fax, but email2fax certainly isn't any more difficult. |
| 16:06.22 | *** join/#asterisk ollii (~risker@port-87-193-161-154.static.qsc.de) |
| 16:20.20 | *** join/#asterisk irroot (~irroot@197.169.76.237) |
| 16:31.27 | dijib | i can send to a list of numbers in a subject line with a pdf attached with the fax contents, or tiff. |
| 16:31.57 | dijib | i can edit the document on computer then save as pdf. |
| 16:32.28 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 16:32.49 | p3nguin | I need to see if SendFAX() can send a PDF or if it has to be a TIFF. |
| 16:33.06 | Bipul | p3nguin, There is problem.. |
| 16:33.13 | p3nguin | Again? |
| 16:33.23 | dijib | sendfax submits one or more facsimile transmission requests to a Hyla FAX facsimile server. |
| 16:33.39 | dijib | what if you dont want to run hyla? |
| 16:33.42 | p3nguin | Not here, it doesn't. |
| 16:33.47 | p3nguin | I don't use hylafax. |
| 16:33.47 | dijib | ok |
| 16:33.53 | Bipul | When i have updated my system.... all old files vanish.... |
| 16:34.01 | p3nguin | Restore them from backups. |
| 16:34.09 | p3nguin | The backups you made before you upgraded. |
| 16:34.29 | dijib | asterfax is this then? |
| 16:34.52 | p3nguin | No clue, I use res_fax and res_fax_digium, aka fax for asterisk. |
| 16:34.58 | dijib | backups :) |
| 16:36.20 | p3nguin | I keep a thumb drive attached to my asterisk box and do a backup via cron every day, plus I do a periodic image of the entire thing to external hard drive. |
| 16:36.55 | *** join/#asterisk Godfather_ (~estanteri@89.131.93.52) |
| 16:37.53 | dijib | hardcore |
| 16:38.17 | dijib | if your hardware has fire. how long do you need to restore? |
| 16:38.17 | p3nguin | smrt |
| 16:38.22 | dijib | have ready built images? |
| 16:38.36 | p3nguin | It would take me about 15 minutes to restore. |
| 16:38.42 | dijib | thats hot. |
| 16:38.45 | irroot | p3nguin you upgrading to 1.8/10 using the res_fax_spandsp.c |
| 16:39.08 | Bipul | nops i have not made any backup |
| 16:39.36 | p3nguin | Running an embedded appliance with only a 4G flash module greatly reduces the time of backup and restore via image. |
| 16:40.13 | p3nguin | It would probably take me longer to dig out another appliance than it would to restore the image. |
| 16:40.39 | p3nguin | In worst case, I could write the image to a hard drive and put in a regular PC. |
| 16:40.54 | p3nguin | or to a thumb drive. |
| 16:40.58 | p3nguin | or to CF card. |
| 16:47.27 | dijib | so i guess you need to run a light install, low mem? |
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| 17:06.23 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 17:06.23 | ChannelZ | ian |
| 17:06.33 | ChannelZ | grrrph |
| 17:15.21 | StaRetji | folks, I have problem with IVR, I press # but asterisk console says -- User entered nothing. |
| 17:15.45 | WIMPy | You use Read()? |
| 17:15.51 | StaRetji | I'm using exten => 1,4,Read(digits,,1) to move around contexts |
| 17:15.51 | StaRetji | yes |
| 17:16.07 | StaRetji | I have them many, but it seems only # doesn't work |
| 17:16.12 | StaRetji | so far, any other number is okay |
| 17:16.19 | WIMPy | That's the way Read() works. |
| 17:16.29 | StaRetji | you mean # wont work? |
| 17:16.48 | WIMPy | It reads until the time is up, the maximum number of digits have been entered, or # is pressed. |
| 17:17.21 | WIMPy | You could check READSTATUS, I guess. |
| 17:17.24 | StaRetji | I have this command exten => 1,8,GotoIf($["${digits:0:1}" = "#"]?ivr777777,2,msgtaraba1) |
| 17:17.48 | WIMPy | Or you do it with extensions. |
| 17:17.53 | StaRetji | so, I thought by pressing # it will take me to context ivr777777 to msg called msgtaraba1 |
| 17:18.04 | StaRetji | as it does when I enter 0 to 9 |
| 17:18.15 | WIMPy | Or * |
| 17:18.19 | WIMPy | But not # |
| 17:18.29 | WIMPy | # terminates input. |
| 17:18.32 | StaRetji | oh |
| 17:18.36 | StaRetji | I understand |
| 17:18.55 | StaRetji | so I'm mixing things here which can't be mixed |
| 17:19.06 | StaRetji | so how do I do then? I need to give option, 1,2,3 or # |
| 17:19.07 | StaRetji | ? |
| 17:19.08 | StaRetji | thx |
| 17:19.33 | StaRetji | 4 options all together |
| 17:19.52 | WIMPy | Either check READSTATUS or use extensions instead of read. |
| 17:20.14 | StaRetji | btw, option 1 leads to contex1 , 2 context2 and so on |
| 17:20.17 | WIMPy | The easiest would be to replace # with another digit, off course. |
| 17:20.31 | StaRetji | they are all in different context except # is in mainmother context |
| 17:20.44 | StaRetji | problem is I have recorded voice saying # |
| 17:20.45 | StaRetji | :/ |
| 17:21.14 | StaRetji | I mean, I received that voice |
| 17:21.34 | *** join/#asterisk nix8n82-phone (~AndChat@24.143.28.16) |
| 17:21.58 | WIMPy | Or you add an option to Read to allow # to be read. |
| 17:22.10 | StaRetji | is that possible? How? |
| 17:22.25 | StaRetji | sorry for so many question, I'm kinda in trouble if I don't fix it |
| 17:22.52 | WIMPy | You'd have to do some programming. |
| 17:23.22 | StaRetji | oh, to way out of my league :) |
| 17:23.53 | WIMPy | There's a good chance that wuld be really easy. |
| 17:24.06 | WIMPy | Maybe I should check that after lunch. |
| 17:24.28 | StaRetji | I have old config where I'm sure it worked |
| 17:24.32 | StaRetji | I will try to load it now |
| 17:24.38 | StaRetji | btw, bon apetite :) |
| 17:25.28 | WIMPy | Thanks |
| 17:25.54 | WIMPy | Although I never have trouble with my appetite... |
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| 17:28.41 | StaRetji | lamo |
| 17:28.42 | StaRetji | lmao |
| 17:28.45 | StaRetji | nice one ;) |
| 17:35.18 | WIMPy | I think I like the idea of modifying Read(). Have you already filed a feature request? :-) |
| 17:40.08 | StaRetji | nope, I'm about to get fired |
| 17:40.18 | StaRetji | lol |
| 17:40.23 | StaRetji | but seriously |
| 17:40.44 | StaRetji | I was suppose to put in in production and now I can't because of damn# |
| 17:40.49 | WIMPy | was serious |
| 17:41.15 | WIMPy | You can always use extensions instead of read. |
| 17:42.07 | WIMPy | But I would also find it useful, if Read could return #. |
| 17:47.23 | raden | is there a way to terminate asterisk to the normal Phone company and send callerid ? |
| 17:47.54 | WIMPy | raden: Sounds unavoidable. Maybe you should rephrase the question. |
| 17:48.47 | raden | I need a normal phone line connected to my asterisk box, is there a type of connection I can get from a phone company that allows me to send callerid ? |
| 17:49.02 | Kobaz | raden: pri |
| 17:49.11 | *** join/#asterisk gokulnath (~gokulnath@59.93.39.219) |
| 17:49.27 | Kobaz | raden: anywhere from 300 to 700 a month depending where you are |
| 17:49.42 | *** join/#asterisk ack_syn (~Jedi@unaffiliated/ackz0r) |
| 17:49.53 | WIMPy | s/pri/isdn/ |
| 17:49.57 | Kobaz | many sip providers will allow you to send callerid as well |
| 17:50.14 | WIMPy | So that starst at 24/month depending on where you are. |
| 17:50.15 | Kobaz | WIMPy: well yeah, bri will let you do it too |
| 17:50.26 | raden | Kobaz, I need a hardline |
| 17:50.32 | Kobaz | bri/pri |
| 17:50.37 | WIMPy | Where? |
| 17:50.38 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:50.45 | raden | wisconsin |
| 17:50.58 | WIMPy | Probably bad luck. |
| 17:51.00 | Kobaz | near any metro areas? |
| 17:51.04 | raden | nope |
| 17:51.07 | raden | middle of no where |
| 17:51.14 | WIMPy | Check if you can get BRI there. |
| 17:51.28 | Kobaz | expect to pay way more than you should |
| 17:51.40 | ack_syn | hi guys do you know a good solution for gsm gateway? I think asterisk isng a good idea, I am talking about 50 simultaneous channels. Sorry if it is not the right place do ask it, if you can help me it would be great |
| 17:52.08 | *** join/#asterisk irroot (~irroot@197.169.76.237) |
| 17:52.17 | WIMPy | ack_syn: 2N stargate or Teles i.pbx |
| 17:52.44 | ack_syn | WIMPy: oh sorry, I mean an opensource solution |
| 17:53.07 | WIMPy | ack_syn: How is that supposed to work without hardware? |
| 17:53.48 | WIMPy | You could try 50 USB sticks or 50 mobile phones via Bluetooth. But I'm not sure how good that would work. |
| 17:54.16 | ack_syn | WIMPy: some gsm cards, opensips + openbts for example |
| 17:54.28 | ack_syn | but I dont know if it's a well known solution |
| 17:54.53 | WIMPy | I haven't seen PCI GSM with more than 4 channels so you'd need 13 PCI slots. |
| 17:55.34 | WIMPy | You can get cheap 4 channel SIP gateways. |
| 17:55.54 | ack_syn | WIMPy: 4 channel is too low, I'd like to have at least 20 channels |
| 17:56.01 | StaRetji | WIMPy: sorry mate, how can I give option to choose between 1,2,3 with read and # with extension? |
| 17:56.02 | *** join/#asterisk jblack (~jblack@pool-71-173-14-99.sctnpa.east.verizon.net) |
| 17:56.06 | WIMPy | You said 50. |
| 17:56.10 | StaRetji | really need your help |
| 17:56.12 | StaRetji | thx |
| 17:56.15 | ack_syn | WIMPy: I said now /at least/ |
| 17:56.23 | ack_syn | WIMPy: the appliance solutions are usually too expensive |
| 17:56.54 | WIMPy | I think PCI would be more expensive. |
| 17:57.20 | ack_syn | WIMPy: ok, you said stargate or teles, do you know any portal with the devices and their prices? |
| 17:57.37 | WIMPy | StaRetji: Change your dialplan to use extensions or (or and) file a feature request for Read(). |
| 17:58.34 | WIMPy | ack_syn: When I checked some years ago, both where in the region of 20-30k EUR. |
| 17:58.46 | ack_syn | wow, as I said, too expensive |
| 17:59.20 | WIMPy | You can find 4 channel SIP gateways for <200 on ebay. |
| 17:59.27 | ack_syn | WIMPy: it seems in my country I can buy one by USD 10k |
| 17:59.32 | WIMPy | Or I think they also have 8 ch. |
| 17:59.38 | ack_syn | WIMPy: ok, I will check them |
| 17:59.44 | ack_syn | ty |
| 18:07.03 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 18:15.01 | StaRetji | WIMPy: easy said then done, I can mix Read 1,2,3 and Extension #? |
| 18:15.16 | WIMPy | no |
| 18:15.42 | StaRetji | I could bet # was working with Read, meh, weird |
| 18:15.43 | StaRetji | :_ |
| 18:15.45 | StaRetji | :) |
| 18:15.55 | WIMPy | I don't think so. |
| 18:16.00 | WIMPy | But it might soon. |
| 18:20.44 | anonymouz666 | irroot: that issue I talked to you last week is giving me lots of problems. the app_queues likes to delivery two different calls to same the same agent in the same second (as noticed by log full)! |
| 18:20.45 | *** join/#asterisk kam187osx (~kam187osx@78-105-127-53.zone3.bethere.co.uk) |
| 18:20.48 | kam187osx | hey guys |
| 18:21.19 | kam187osx | is there an easy way to make sure users can only call a number once.. eg if 5 users dial 11111 |
| 18:21.20 | irroot | anonymouz666 mmm |
| 18:21.29 | irroot | you got the bug ref for it ?? |
| 18:21.51 | anonymouz666 | nope, because it's happening in version 1.4 under heavy load |
| 18:22.02 | irroot | ah yes i remember |
| 18:22.56 | irroot | you can change the code ?? |
| 18:23.16 | anonymouz666 | sure, I just need to know what to change |
| 18:23.19 | anonymouz666 | :) |
| 18:23.37 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:23.53 | p3nguin | dijib: No, I have 256M RAM in that box, and I have room to upgrade if needed. |
| 18:24.15 | dijib | this email2fax is itching my brain |
| 18:24.27 | *** join/#asterisk Janos (~cramos@201.198.38.138) |
| 18:25.43 | dijib | p3nguin, http://pastebin.com/1ZRJpLQM what ya think? |
| 18:25.48 | p3nguin | staretji: If you're wanting callers to enter extensions, DON'T USE READ(). Read() is not really meant for that. |
| 18:25.52 | WIMPy | StaRetji: Is your request there? |
| 18:25.53 | dijib | i know SIP/voipms context wrong |
| 18:26.55 | p3nguin | You also don't have Zap channels, but you're trying to use them. |
| 18:27.12 | dijib | line 362-370 in previous post... have it still? |
| 18:27.21 | p3nguin | no idea |
| 18:27.23 | dijib | and dont worry about that. i can get that to work. |
| 18:27.30 | dijib | this post http://pastebin.com/8kuxXLhG |
| 18:27.39 | dijib | it calls for tx-fax |
| 18:27.42 | dijib | txfax |
| 18:27.50 | dijib | should i try to use fax-tx? |
| 18:27.57 | dijib | is that a part of SendFax? |
| 18:28.16 | Janos | hello, got a 4 port fxo digium card, it's a bit old, everything works but the problem is that when i call from the outside and hangup on the outside before anyone picks up, the call never hangs up on the asterisk, with a regular phone this does not happen, with i do with all incoming calls is send them to a queue and they stay there until somebody answers, is this a bug ? "feature" ? misconfiguration ? |
| 18:28.48 | irroot | anonymouz666 you want to try code it ? |
| 18:28.51 | p3nguin | No, the txfax() application is not part of the SendFAX() application. |
| 18:29.18 | dijib | is it case sensetive? no right? |
| 18:29.33 | p3nguin | nah, I don't think it is. |
| 18:29.52 | p3nguin | But if you don't have txfax, it will certainly be sensitive to that. |
| 18:29.57 | dijib | i should have sendfax right |
| 18:29.58 | dijib | ? |
| 18:30.02 | p3nguin | yes |
| 18:30.04 | dijib | fax show capabilities? |
| 18:30.10 | p3nguin | sure |
| 18:30.22 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 18:30.35 | dijib | how do i see the arguments Digium FAX Driver handles? |
| 18:30.53 | p3nguin | I don't understand your question. |
| 18:31.18 | p3nguin | The fax driver does not support arguments. Perhaps you mean that you want to see the usage of SendFAX()? |
| 18:31.49 | *** join/#asterisk Diffen (~diffen@c-f477e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 18:31.51 | dijib | i change txfax in this script to sendfax |
| 18:31.55 | p3nguin | If yes, core show application SendFAX |
| 18:31.58 | dijib | and i get this |
| 18:32.10 | dijib | sed: -e expression #1, char 2: unterminated `s' command |
| 18:32.11 | dijib | sendmail: No recipients specified although -t option used |
| 18:32.17 | p3nguin | (just like every other registered application) |
| 18:32.20 | dijib | the script errors out |
| 18:32.34 | dijib | nevermind not * |
| 18:32.36 | dijib | its GM |
| 18:32.41 | gokulnath | hey, i am managing a server with huge number of incomings, the asterisk used to hang in that. Then someone told me about ulimit and i increased that value and it's working fine now. But what is this real issue |
| 18:32.44 | p3nguin | unterminated s command means someone forgot to close their sed expression. |
| 18:32.54 | p3nguin | s/old/new <--- fail |
| 18:33.02 | p3nguin | s/old/new/ <--- win |
| 18:33.22 | p3nguin | s/win/winner!/ |
| 18:34.18 | anonymouz666 | irroot: yes ! I want to test the code |
| 18:36.03 | irroot | https://reviewboard.asterisk.org/r/1119/ <- look here for the ignorebusy stuff you can do something similar |
| 18:37.26 | *** join/#asterisk dandate2 (~dan@124.6.157.210) |
| 18:37.43 | dandate2 | is there a cli command to see what inbound route an active caller had dialed? |
| 18:37.55 | p3nguin | core show channels |
| 18:38.27 | dandate2 | awesome |
| 18:41.55 | Diffen | Evning. I have just connected to my Asteriskserver via: telnet address 5038. I wonder if its possible to save all the info thats floating by to a mysql database so I can build a web interface to view the data. Its nice to be able to see if a phone are lagged and so on. |
| 18:41.55 | Nugget | telnet is eeeeeeevil! |
| 18:44.25 | WIMPy | Diffen: You should take a look at AMI for that. |
| 18:45.15 | *** join/#asterisk iscario (~div@ppp-sei11-46-193-132.103.wb.wifirst.net) |
| 18:45.52 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 18:47.13 | Diffen | Wimpy: Yes, i am connected now just to see how it looks, and it looks good... But I guess i need some program to save it all to a db? |
| 18:47.37 | irroot | im shuting down .... |
| 18:49.39 | dandate2 | is there a simple way to bypass banned voip ports in those ghetto 3rd world countries that block sip |
| 18:49.50 | p3nguin | Yes. Use different ports. |
| 18:50.10 | anonymouz666 | irroot: hey |
| 18:50.12 | p3nguin | Some ITSPs even offer non-standard ports for exactly that reason. |
| 18:50.12 | dandate2 | does that have to be configured on the server end or can the end user in say bangladesh on xlite do that themselves? |
| 18:50.18 | irroot | .... |
| 18:50.23 | anonymouz666 | irroot: bad crash from app_queue :/ |
| 18:50.27 | anonymouz666 | ast_hangup |
| 18:50.37 | irroot | you got some info on it ?? |
| 18:50.41 | anonymouz666 | yes |
| 18:50.43 | p3nguin | You'll have to configure asterisk to use a different port for the phone in question, as well as configure the phone to use that port. |
| 18:50.50 | irroot | id really move to SVN 1.8 |
| 18:51.05 | p3nguin | It should be as easy as adding port=5080 to the phone's entry to change it to 5080. |
| 18:51.10 | irroot | im running backported app_queue from trunk |
| 18:51.52 | WIMPy | Diffen: Yes, you have to do that part yourself. |
| 18:51.52 | anonymouz666 | irroot: http://pastebin.com/Eybgj4ka |
| 18:52.27 | anonymouz666 | irroot: but from version 1.4? |
| 18:52.35 | anonymouz666 | I mean, in version 1.4 ? |
| 18:53.06 | irroot | anonymouz666 yeah i know makes it hard to support been about a year sunce we used it |
| 18:53.24 | anonymouz666 | 377 active channels 217 active calls |
| 18:54.12 | anonymouz666 | that's too much to asterisk |
| 18:54.17 | anonymouz666 | very weird things happens |
| 18:54.18 | Diffen | wimpy no problems :) my plan is to save all the peerstatus info and then be able to ask eventquestions from a web interface. i guess that should be doable? |
| 18:55.19 | irroot | anonymouz666 ok got a full bt ?? |
| 18:55.23 | WIMPy | sure |
| 18:55.31 | anonymouz666 | irroot: yes |
| 18:55.37 | anonymouz666 | but the value is optimized |
| 18:55.51 | anonymouz666 | even optimized will help you ? |
| 18:56.04 | irroot | no not much |
| 18:56.07 | *** join/#asterisk cyford (~cyford@c-24-30-15-39.hsd1.ga.comcast.net) |
| 18:56.41 | anonymouz666 | :/ |
| 18:56.53 | irroot | could be a double free |
| 18:57.08 | irroot | its libc barf ing |
| 18:57.21 | anonymouz666 | double free bad things happens |
| 18:57.38 | anonymouz666 | never seen this behaviour before, except when load is high |
| 18:57.52 | anonymouz666 | and i have many many setup using this version |
| 18:58.04 | irroot | yeah |
| 18:58.43 | kam187osx | hmm i'm using a channel driver eg oh323 to dial out, how do i perform an operation when that driver hangs up? |
| 18:58.55 | kam187osx | it doesnt seem to move onto the next line in extensions |
| 18:59.06 | irroot | unfortunately we have moved on im running 10 beta and will be rolling it out to customers when rc hits optionally |
| 18:59.28 | irroot | kam187osx dial option g and h extension |
| 18:59.50 | kam187osx | ahh |
| 18:59.52 | kam187osx | great :) |
| 18:59.53 | kam187osx | thanks |
| 19:00.19 | anonymouz666 | irroot: do you think that running the latest 1.8 will be more stable regarding queues than this 1.4.42 version? |
| 19:01.13 | irroot | anonymouz666 indeed the SVN version has some important fixes for pickup leaving orphan channels |
| 19:01.46 | irroot | 1.8.7 will be best 1.8 yet and will ensure we can help you |
| 19:01.56 | kam187osx | hmm g: When the called party hangs up, continue to execute commands in the current context at the next priority. |
| 19:02.05 | kam187osx | thats perfect, but what about if the caller hangs up? |
| 19:02.14 | irroot | the h extension |
| 19:02.24 | irroot | the call goes to the h exten |
| 19:02.28 | kam187osx | ah h is an extension? |
| 19:02.41 | anonymouz666 | irroot: you know, if I update to the 1.8, running 250 active calls, any problem will be impossible to solve |
| 19:02.48 | irroot | exten h,1,NoOp(im hung up) |
| 19:03.40 | irroot | anonymouz666 one option is to use a backup server or install modules into modules-1.8 |
| 19:04.02 | anonymouz666 | yes |
| 19:04.05 | irroot | this is what i have done have a dir for all versions |
| 19:04.09 | anonymouz666 | that's the only thing I can do |
| 19:04.26 | kam187osx | ahh |
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| 19:13.54 | WIMPy | StaRetji: Where is your feature request? |
| 19:14.49 | WIMPy | The patch is ready. |
| 19:17.51 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 19:18.47 | StaRetji | WIMPy: sorry mate, I'm having some problem with my boss |
| 19:18.55 | StaRetji | calling me every 5 minutes |
| 19:19.05 | StaRetji | I hate that guy |
| 19:19.29 | StaRetji | Btw, is patch requiring upgrading of asterisk? |
| 19:19.45 | StaRetji | I will post feature request asap |
| 19:20.09 | WIMPy | Ask him if he wnats you to get something done or if you're cheaper than his psychologist. |
| 19:20.26 | WIMPy | I did it for 1.8 SVN. |
| 19:21.03 | StaRetji | lol good idea |
| 19:21.37 | StaRetji | great job, I will have to upgrade, but I have to be careful, have a2billing and lots of stuff, now on 1.4 |
| 19:21.58 | StaRetji | thx WIMPy, you're good man |
| 19:30.35 | *** join/#asterisk cyford (~cyford@c-24-30-15-39.hsd1.ga.comcast.net) |
| 19:46.01 | kam187osx | finally, that did it |
| 19:46.09 | kam187osx | thanks guys |
| 19:46.33 | kam187osx | got this wierd issue where the customer starts flooding me with calls to the same number :/ |
| 19:49.30 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:51.07 | kam187osx | hopefully i got the right approach... on incoming call, check DB for test/$EXTEN if it exists, hangup as a call to that number already exists... if not, store key $EXTEN in the db and dial |
| 19:51.14 | kam187osx | on hangup (either direction) delete the key |
| 19:51.27 | dijib | anybody have a working email to fax solution? |
| 19:54.07 | eduzimrs | dijib have u listened about iaxmodem? |
| 19:54.23 | dijib | nope what about it? |
| 19:54.51 | dijib | doesnt it use hylafax? |
| 19:55.07 | eduzimrs | it probably does what u want |
| 19:56.43 | eduzimrs | dijib http://iaxmodem.sourceforge.net/howto.php |
| 19:57.33 | eduzimrs | there`s a problem if u are using redfone, caue it doesnt support iax protocol |
| 20:09.00 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:15.05 | *** part/#asterisk kam187osx (~kam187osx@78-105-127-53.zone3.bethere.co.uk) |
| 20:23.45 | Janos | looks like busydetect=yes does the trick when you don't have cpc on the land line |
| 20:30.37 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 20:31.10 | navaismo | <PROTECTED> |
| 20:34.33 | Janos | now for another question, one of my pbx is giving me problems, sometimes the asterisk process starts sucking up my cpu and at least all sip communications die, sip show channels show calls that are not actually taking place, asterisk cli does work as long as i don't use autocomplete since this freeze the session, restarting the asterisk process solves the problem but i would like to find a more permanent solution, i'm using asterisk 1.6.2.9 which comes wi |
| 20:34.33 | Janos | th current debian stable 'squeeze', any idea on how to diagnose this further ? |
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| 21:30.27 | p3nguin | navaismo: What is your definition of secure in that context? |
| 21:31.22 | navaismo | humm ... |
| 21:31.58 | p3nguin | Click to call should only do one thing: call you and another party. |
| 21:32.31 | *** join/#asterisk Micc (~Micc@c-98-232-46-178.hsd1.wa.comcast.net) |
| 21:32.34 | navaismo | yep via webpage, maybe the correct name is clicktocallback |
| 21:32.44 | navaismo | but remember the hacked server? |
| 21:32.59 | navaismo | i think im not secure that button |
| 21:33.12 | navaismo | permission in the manager |
| 21:36.57 | *** part/#asterisk Micc (~Micc@c-98-232-46-178.hsd1.wa.comcast.net) |
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| 22:14.10 | CaptWho | what is the name of the configuration file that I tell asterisk where the location of the mysql database is? |
| 22:14.37 | navaismo | res_config_mysql I think |
| 22:14.49 | CaptWho | thanks navaismo |
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| 23:55.56 | gushi | Hey there all. Is there anyone that could help me with a codec issue? I get no audio on calls with an error logged (constantly) about a dropped frame because the native format has changed. |
| 23:56.33 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 23:57.16 | navaismo | what codecs do you enable for the involved peers?? |
| 23:58.23 | gushi | All. |
| 23:58.38 | gushi | Although I don't have g723 or 729 support enabled in asterisk. |
| 23:59.36 | navaismo | and the log what codec show? |