| 00:06.19 | *** join/#asterisk ming_zym (n=ming_zym@218.108.49.4) |
| 00:16.38 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 00:16.43 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 00:18.08 | *** join/#asterisk jack___ (n=jack@ip4da3b4f7.direct-adsl.nl) |
| 00:18.15 | jack___ | yo |
| 00:19.19 | jack___ | I would like to know if the following would work. I have an extension setup, so that any inbound calls made to a DID are being disconnected. After those line i have exten => h,1, Dial(..../${CALLERID(num)}) |
| 00:19.28 | jack___ | Is it possible to originate a call like this? |
| 00:20.01 | jack___ | or do i really need to use the outgoing asterisk spool or the follow me hack |
| 00:21.38 | p3nguin_ | jack___: What's the goal of your exercise? |
| 00:22.06 | jack___ | it's sort of like DISA |
| 00:22.10 | jack___ | I want a callback |
| 00:22.38 | jack___ | call a number, call get disconnected, and i should call me back and put me in some context |
| 00:22.52 | jack___ | s/i/it |
| 00:26.28 | jack___ | actually, i want the call to enter a context, have asterisk wait 30 seconds before picking up the call. Then i would disconnect myself (so it won't connect, and no charges are made) and then it should call me back |
| 00:26.58 | jack___ | So it should also resume the flow of the context while the disconnect is made by the caller |
| 00:27.02 | jblack | jack___: surely googling for callback got you somewhere |
| 00:27.18 | jack___ | yeah. true, most of the solutions are by passing around some files |
| 00:27.36 | jack___ | can't this be done from within the extension context directly? |
| 00:28.29 | p3nguin_ | jack___: http://pastebin.ca/1643242 |
| 00:28.46 | p3nguin_ | I threw this together real quick. See if it fits your needs. |
| 00:28.46 | jblack | callfiles really are the right way to do this sort of thing |
| 00:29.29 | jblack | p3nguin_: Oh, I see a nasty race condition in there |
| 00:29.32 | jack___ | p3nguin_: looks like a clean solution. going to try this |
| 00:29.53 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 00:29.57 | jack___ | jblack: you mean by using the DB function in conjunction with the call last? |
| 00:30.04 | p3nguin_ | This will assume that your outgoing context has a match for ${DB(call/last)}. |
| 00:30.24 | jack___ | that might screw up some routing between calls if the service is heavily used (which it's not) |
| 00:30.51 | p3nguin_ | You could easily make it any variable which is not stored in the DB. |
| 00:31.00 | p3nguin_ | I just figured the DB was a good way to do it. |
| 00:31.31 | jack___ | can't i just reset the CallerID? |
| 00:31.39 | jack___ | ;exten => h,1,Set(CALLERID(number)=${CALLERID(number)}) |
| 00:31.41 | jack___ | like this |
| 00:31.52 | p3nguin_ | What's that going to do? |
| 00:32.05 | jblack | It's not a bad concept p3ng, and a neat idea. There's just a race between the hangup and the dial. if anyone else hits that logic between the two, you're going to call the wrong person and drop an intended call |
| 00:32.32 | p3nguin_ | The caller ID num already equals the caller ID num, so that's pretty useless. |
| 00:32.34 | jack___ | p3nguin_: nothing actually, ignore that :) |
| 00:33.38 | jblack | jack___: There's a wakeup call agi out there that you could hack up to do a simple callback connect, that should be race free. |
| 00:33.45 | p3nguin_ | The problem I see is that there is nothing calling you. |
| 00:34.06 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 00:35.14 | jack___ | what do you mean by 'nothing call you'? |
| 00:35.59 | jack___ | does the 'h' flag also works when the caller hangs up? |
| 00:37.21 | p3nguin_ | I don't know about any flags, but the 'h' extension is supposed to execute when the call hangs up. |
| 00:37.22 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 00:37.40 | jack___ | i meant the extension, sorry |
| 00:39.12 | jack___ | one could solve the race condition by incorporating the id of the caller and the DID extension in the database variable |
| 00:39.30 | jack___ | this would limit the chance on collision |
| 00:40.09 | jblack | just make sure to clean it out.... |
| 00:40.20 | p3nguin_ | I don't see any race condition in what I created, but there is just that other problem of nothing calling you. |
| 00:40.54 | jblack | p3nguin_: The db is shared between calls, I believe. So between when the value is stored in the db, and when it's looked up, it could change. |
| 00:41.32 | p3nguin_ | Someone else would have to call in between the time he called and the time he hangs up. |
| 00:41.43 | jack___ | because? |
| 00:41.45 | jblack | yes. which is why it's called a "race" |
| 00:42.03 | p3nguin_ | I guess if it's a turtle race, it's still a race. |
| 00:42.16 | jack___ | yes, that would be solved by using unique family/ value keys |
| 00:42.32 | jack___ | p3nguin_: what's the other problem with the "nothing calling you" |
| 00:42.33 | jblack | Or just spool the damn thing, like you should this type of thing. :P |
| 00:42.44 | jack___ | it's the principle :) |
| 00:43.06 | jblack | whatever. |
| 00:43.11 | jack___ | i would spool it from an external application, not from within the pbx itself |
| 00:43.30 | p3nguin_ | Once you disconnect the call, h executes. There's nothing calling you. Asterisk would dial your number and then nothing would be there (if it even can dial you successfully). |
| 00:44.22 | *** join/#asterisk chendy (n=chatzill@219.133.68.177) |
| 00:44.56 | jack___ | isn't the asterisk calling me, putting me in a context which in turn can provide something like disa |
| 00:44.57 | p3nguin_ | Ordinarily, one channel gets bridged to the other channel forming a call between two devices. |
| 00:45.01 | jack___ | true |
| 00:45.24 | p3nguin_ | In this, there's just a single channel with nothing on the source side of it but asterisk itself. |
| 00:45.26 | jblack | that's why with a callfile you specify what extension gets hooked up to the number being called. |
| 00:46.10 | p3nguin_ | yeah, that would be a good idea. |
| 00:46.34 | jack___ | well, i'll use AGI then with some python script |
| 00:46.42 | p3nguin_ | When you use Originate, it would ring one device, then ring the other device, and ultimiately bridge the two. |
| 00:47.03 | *** join/#asterisk sudhir492 (n=sudhir@68.210.52.132) |
| 00:47.43 | sudhir492 | Is anyone using Cisco 7940/7960 phone with asterisk? I am having two problems with the phones. |
| 00:48.09 | p3nguin_ | sudhir492: Ask. |
| 00:48.13 | sudhir492 | Looks like some parts of SIPDefault.cnf is not taking effect on the phone |
| 00:48.28 | p3nguin_ | Be specific. |
| 00:48.45 | jack___ | does asterisk expand the plus sign in a extension rule? |
| 00:49.05 | sudhir492 | (Minor issue) preferred coded is set to ulaw even though I have the line: preferred_codec: "g729" |
| 00:49.12 | jblack | I believe it treats it just like any other character. |
| 00:49.34 | jack___ | aight |
| 00:49.44 | jack___ | Well. enough asterisk for today, thanks for the help |
| 00:49.46 | jblack | sudhir492: I doubt that. More likely is you didn't reload your config, or you're looking at the wrong leg of the call |
| 00:49.50 | *** join/#asterisk Tim_Toady (n=moi@adsl96-115.ath.forthnet.gr) |
| 00:50.07 | *** join/#asterisk Tim_Toady (n=moi@194.219.240.115) |
| 00:50.19 | sudhir492 | Another issue is that the phone is not reading dialplan.xml file, even though, I have the line: dial_template: "dialplan" |
| 00:50.50 | jblack | Your phone configuration thing, I'm sad to say, is not an asterisk issue at all. |
| 00:51.18 | jblack | I'm not trying to be mean, it's just that * doesn't know anything about your phone, and doesn't do any of the configuration. |
| 00:51.23 | sudhir492 | Actually, I checked, by change message_uri in SIPDefault.cnf file, and verifying that it has changed, but preferred codec did not change |
| 00:52.10 | p3nguin_ | Look in the phone menu or telnet "sh conf" and look at the codec. |
| 00:52.10 | Nugget | telnet is eeeeeeevil! |
| 00:52.16 | jblack | If you change the preferred codec in your sip.conf or your iax.conf (probably sip.conf since you're talking phone), then it will take effect as soon as you either restart asterisk, or do a sip reload. |
| 00:52.33 | jblack | if you're talking phone configuration, I couldn't tell you to save my life. Whatever phone you have, I don't have it. |
| 00:52.37 | sudhir492 | jblack: you are absolutely right that phone configuration is not an asterisk issue at all, but I got so frustrated, that thought someone here has definitely used those phones and can help me |
| 00:53.03 | p3nguin_ | That's what kind of phone I use, but I do not have this problem. |
| 00:53.17 | jblack | Ok, so you do understand that. You might be careful about the phrasing in the future, so that the right person speaks up to help you. :) |
| 00:53.43 | jblack | i.e. I know to keep my mouth shut when you're looking for help for your phone. ;) |
| 00:54.09 | sudhir492 | p3nguin, will you be able to share SIPDefault.cnf file with me (it does not have any asterisk specific information) |
| 00:54.37 | p3nguin_ | sudhir492: http://www.loligo.com/asterisk/Cisco/79xx/current/SIPDefault.cnf |
| 00:55.08 | sudhir492 | p3nguin: I have checked with that. Actually, I have used that as my baseline |
| 00:55.31 | p3nguin_ | Did you check the codec in telnet or in the menu yet? |
| 00:56.43 | sudhir492 | Yes, I checked in the phone menu |
| 00:57.38 | p3nguin_ | What did it say? |
| 00:58.12 | sudhir492 | preferred codec is ulaw on the phone |
| 00:58.55 | sudhir492 | but that is not the major issue. Major issue is that it does not read dialplan.xml |
| 00:59.46 | p3nguin_ | Did you happen to attempt overriding preferred_codec and/or dial_template in your SIP<mac_address>.cnf? |
| 00:59.50 | sudhir492 | Are you using SIPDefault.cnf file from jtodd too |
| 01:00.00 | p3nguin_ | They don't really belong in there, so if you did, take them out. |
| 01:00.11 | p3nguin_ | I'm using my own. |
| 01:00.19 | sudhir492 | no, I did not override them in SIP<mac> files |
| 01:00.44 | sudhir492 | will you be able to share yours? |
| 01:01.34 | p3nguin_ | Give me just a minute to get it for you. I'm trying to eat supper. :) |
| 01:02.14 | sudhir492 | oh sorry. take your time. |
| 01:02.40 | sudhir492 | and thanks a lot for taking out time |
| 01:05.51 | *** join/#asterisk ming_zym (n=ming_zym@121.0.29.237) |
| 01:05.55 | p3nguin_ | sudhir492: http://paste-bin.com/view/47fc93ed |
| 01:07.48 | p3nguin_ | I think everything I have in mine is on that one I listed earlier, plus that one has a lot more things defined that I don't even worry about. |
| 01:10.04 | sudhir492 | p3nguin: thanks for the file |
| 01:10.51 | sudhir492 | Yes, it looks like jtodd's and yours are almost the same as far as my needs are concerned. |
| 01:11.44 | sudhir492 | I have just 3 lines in my dialplan.xml file: <DIALTEMPLATE> |
| 01:11.44 | sudhir492 | <PROTECTED> |
| 01:11.44 | sudhir492 | </DIALTEMPLATE> |
| 01:12.10 | p3nguin_ | You like it that way? |
| 01:12.11 | sudhir492 | but I never see any request for that file coming to my tftp server |
| 01:12.22 | p3nguin_ | http://www.loligo.com/asterisk/Cisco/79xx/current/dialplan.xml |
| 01:12.48 | sudhir492 | Yes, I checked with that, and to me everything looks good |
| 01:13.59 | sudhir492 | I am giving up on both the issues, as they are not impacting the functionality in any major way :-( |
| 01:14.22 | sudhir492 | Have you used 3-way calling (cnfrn) feature in phone? |
| 01:14.46 | p3nguin_ | http://paste-bin.com/view/78c5e696 |
| 01:15.03 | p3nguin_ | I use conference on my phone. |
| 01:18.14 | sudhir492 | Let me factory reset the phone, and see if it is reading dialplan.xml then. |
| 01:18.48 | p3nguin_ | Show me your cnf files. |
| 01:19.09 | Katty | hi guys |
| 01:19.31 | p3nguin_ | hi |
| 01:20.00 | *** join/#asterisk broken-iPod (i=62f38cac@gateway/web/freenode/x-zemflsnmwwuedgiz) |
| 01:20.39 | broken-iPod | Can I set some music on hold classes to start at the beginning each time and others to play continuous? |
| 01:21.41 | jblack | good question |
| 01:22.31 | broken-iPod | Also, I'd like to change the outgoing music on hold on a per-user basis. |
| 01:22.34 | jblack | perhaps if you turn buffering off and use custom... |
| 01:23.24 | broken-iPod | So far, custom MOH seems to only work the opposite of how I want it to. |
| 01:24.33 | broken-iPod | If I make a call and the callee puts me on hold, I hear the custom class, but if I put the callee on hold, it reverts to the default. |
| 01:25.18 | jblack | so fiddle with the classes in your sip.conf, how you set it in your dialplan, etc |
| 01:25.23 | jblack | you'll get it |
| 01:25.46 | broken-iPod | I've been fiddling for the past day and a half. I only seem to get partial results regardless of what I do. |
| 01:26.32 | broken-iPod | Right now, I currently have it setup in sip.conf under each user musiconhold=blah. |
| 01:27.13 | broken-iPod | Yesterday I had in the dialplan SetMusicOnHold(${CALLERID(num)}) and set the classes according to my extensions. |
| 01:28.03 | broken-iPod | And I also had in an incoming context in my dialplan SetMusicOnHold(${EXTEN}) once again corresponding to the class for the extension/. |
| 01:28.15 | broken-iPod | That setup seemed to work except when using WaitExten(). |
| 01:29.32 | p3nguin_ | If you SetMusicOnHold(), then BackGround(somefile), then WaitExten(), it should work. |
| 01:29.35 | *** join/#asterisk chendy (n=chatzill@113.91.37.203) |
| 01:30.07 | broken-iPod | That's not what I meant, p3nguin_ |
| 01:30.18 | p3nguin_ | Well that's how you should be using it. |
| 01:30.23 | broken-iPod | Someone calls into my server and gets the WaitExten(). |
| 01:30.31 | broken-iPod | They dial my extension and it calls me. |
| 01:30.36 | p3nguin_ | That's a problem. |
| 01:30.56 | broken-iPod | What is? |
| 01:30.59 | p3nguin_ | If you play a file before WaitExten(), the problem will be corrected. |
| 01:31.09 | broken-iPod | I do, actually. |
| 01:31.30 | broken-iPod | My voice says "Please dial an extension." and then the WaitExten() happens. |
| 01:31.44 | sudhir492 | p3nguin_: http://pastebin.com/m415f1465 |
| 01:31.54 | broken-iPod | But when they dial my extension, SetMusicOnHold() didn't seem to work. |
| 01:32.42 | broken-iPod | If I put the caller on hold, it went to the default. |
| 01:33.00 | broken-iPod | However, if my extension was called directly from IPKall, it played the custom music class. |
| 01:35.42 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
| 01:37.15 | sudhir492 | p3nguin_: are you still there? |
| 01:37.27 | p3nguin_ | sudhir492: Yep. I was just looking over your paste. |
| 01:37.29 | sudhir492 | The problem was solved by factory resetting the phone |
| 01:37.50 | p3nguin_ | sudhir492: The SIP files all load up correctly, now? |
| 01:39.25 | sudhir492 | Yes. I think this is what happened: When I first converted the phone from Skinny to SIP, dialplan.xml file was not there on the server. The phone would not read dialplan.xml after that at all even though I power cycled the phone umpteen times. |
| 01:40.09 | p3nguin_ | strange |
| 01:40.15 | sudhir492 | Only when I factory reset the phone, it was forced to read all config files, and dialplan.xml file was read. |
| 01:40.37 | sudhir492 | Thats the only explanation I can think of. |
| 01:40.58 | p3nguin_ | I've seen reports of similar things with other files. |
| 01:41.12 | sudhir492 | Otherwise, I did not change any files at all while chatting with you. |
| 01:41.50 | p3nguin_ | You're using 8.12 firmware? |
| 01:41.57 | sudhir492 | Thats right |
| 01:42.00 | p3nguin_ | sudhir492: Did you ever use anything lower than that? |
| 01:42.22 | sudhir492 | No. It had skinny before this |
| 01:42.27 | p3nguin_ | okay |
| 01:42.39 | sudhir492 | I do not know what version |
| 01:44.39 | sudhir492 | Thanks for your help, and all the moral support! |
| 01:45.45 | p3nguin_ | Glad you got it solved. |
| 01:47.25 | *** join/#asterisk leon01 (n=leon01@ool-457c12b2.dyn.optonline.net) |
| 01:54.18 | Katty | kinda quiet tonight. |
| 01:54.43 | *** part/#asterisk leon01 (n=leon01@ool-457c12b2.dyn.optonline.net) |
| 01:54.55 | ChannelZ | makes armpit farts to liven things up |
| 01:55.02 | Katty | applauds |
| 01:55.08 | Katty | very talented. |
| 01:55.21 | Katty | how you doing otnight, ChannelZ? |
| 01:56.21 | jaytee | wow, y'know you're starved for entertainment when they start getting out the old junior high school armpit farts |
| 01:56.42 | Katty | nods sadly |
| 01:56.49 | Katty | but what else are we going to do before bed? |
| 01:57.35 | ChannelZ | hehe. I'm fine thanks, you? |
| 01:57.38 | jaytee | nuthin on but reruns of House and dumb movies |
| 01:57.51 | ChannelZ | yeah baseball ruined Sunday night TV |
| 01:57.55 | Katty | ChannelZ: you mean besides bored? ;) |
| 01:58.04 | Katty | i'm playing Resturant City on facebook. |
| 01:58.09 | Katty | that's how bored i am. |
| 01:58.15 | Katty | ryan's watching legend of the seeker. |
| 01:58.16 | ChannelZ | yikes |
| 01:58.28 | Katty | so i'm kinda listening to that in the background too |
| 01:58.32 | ChannelZ | I should fire up CitiesXL |
| 01:58.51 | ChannelZ | but I'm in this mood where I really don't feel like doing anything |
| 01:59.00 | Katty | yeah it'll be bedtime shortly anyway |
| 02:02.47 | jblack | bed is for girls. Real men never sleep |
| 02:02.58 | jblack | grins |
| 02:03.03 | Katty | mkay. |
| 02:03.09 | Katty | as long as i get to sleep |
| 02:03.43 | jblack | so, I'm plugging my blood glucose into this android app, and I'm noticing a pattern. |
| 02:03.51 | Katty | oh yeah? |
| 02:04.04 | jblack | yeah. good day/bad day/good day/bad day |
| 02:04.11 | Katty | interesting. |
| 02:04.19 | Katty | any thoughts on why that might be? |
| 02:04.20 | jblack | Like I planned it |
| 02:04.29 | jblack | obviously it's behaviour. |
| 02:04.42 | Katty | are you doing that on purpose? |
| 02:04.47 | jblack | Not sure exactly what type of behaviour. |
| 02:04.58 | Katty | hmm. |
| 02:04.58 | jblack | No. Didn't realize it was happening until I started graphing it. |
| 02:05.13 | Katty | hmmmm. |
| 02:05.33 | Katty | you should be able to graph something about glucose levels |
| 02:05.37 | Katty | like foods. |
| 02:05.39 | Katty | over time |
| 02:05.51 | jblack | I'm still trying to get glucose in. |
| 02:05.59 | Kobaz | and then have it automatically posted on twitter |
| 02:06.01 | Katty | well i mean the GL rating of food |
| 02:06.05 | jblack | 5-10 readings a day over 3 months takes a while to plug in. |
| 02:06.12 | Katty | ah. okay. |
| 02:06.21 | Katty | i was thinking more of tracking the levels of food you're eating each day |
| 02:06.27 | jblack | Yeah. I can track all that stuff too. Getting the old #'s in has been hard enough. |
| 02:06.29 | Katty | to see what seems to be spiking the levels. |
| 02:06.48 | Katty | is there something you seem to eat a lot of? |
| 02:06.51 | Katty | like..uhh.. every other day? |
| 02:07.03 | Katty | taco bell tacos, perhaps |
| 02:07.21 | jblack | Yeah. I might have a 48 hour splurge thing going on. |
| 02:07.36 | Katty | hmm. |
| 02:07.49 | jblack | a day and half of good numbers, so get something nice, and bonk. |
| 02:07.59 | Katty | nods |
| 02:08.21 | jblack | there's one nice thing though. |
| 02:08.42 | jblack | My range is slowly narrowing over time. My highs aren't as high, and my lows aren't as low. |
| 02:08.46 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 02:08.59 | Katty | that's wonderful! |
| 02:09.10 | Katty | (right?) |
| 02:09.31 | jblack | Well, used to be, OMGWTF was at 558. Now, it's at 230 or so. |
| 02:09.37 | Katty | nodsnods |
| 02:09.39 | Katty | that's most wonderful then |
| 02:09.44 | jblack | and I rarely go much under 90. |
| 02:09.56 | Katty | do you take uhh... |
| 02:09.57 | Katty | what's it called. |
| 02:10.00 | Katty | metaforim or something |
| 02:10.18 | jblack | Not yet. the nurse screwed up. |
| 02:10.22 | jblack | I'm sure I'll be getting it soon. |
| 02:10.29 | Katty | nods |
| 02:10.29 | jblack | though maybe not, since I'm doing so well with the shots. |
| 02:10.36 | Katty | my sister takes that, but she recently had to go off of it |
| 02:10.50 | Katty | she found when she stays below (i believe) 60g of carbs a day, it messes her up |
| 02:10.53 | jblack | poking myself has been a great inducement to treat this seriously. |
| 02:11.12 | Katty | she's not been taking it much lately. |
| 02:11.37 | Katty | i'm very happy you're taking it seriously |
| 02:11.45 | jblack | and one really nice thing... even when I eat bad, they don't go mad-crazy. |
| 02:12.09 | Katty | i'm guessing that's because you don't eat crazy all the time |
| 02:12.15 | jblack | I was getting 300+ levels for eating stuff most people thought healthy. Now it's 200+ for crap that's actually crap. |
| 02:12.16 | Katty | and your body can handle it better, now that you're in better health |
| 02:12.27 | jblack | I think it's the exercise. |
| 02:12.54 | Katty | yeah could be (= |
| 02:13.50 | jblack | weighs in |
| 02:14.34 | jblack | sigh |
| 02:14.48 | Katty | improvement? |
| 02:14.51 | Katty | even a teeny bit? |
| 02:15.05 | jblack | I gained 4 pounds today. |
| 02:15.22 | Katty | well your body weight fluctuates a lot |
| 02:15.33 | Kobaz | i think i lose 4 |
| 02:15.35 | Katty | i'm surprised you're not weighing in once a week |
| 02:15.36 | Kobaz | lost |
| 02:15.37 | jblack | yeah. I bounce around from 245 to 258 |
| 02:15.48 | Katty | i weigh in on tuesday mornings (= |
| 02:16.04 | jblack | I can vary 10 pounds in a day. :) |
| 02:16.11 | jblack | well, 8 is my record. |
| 02:16.19 | Katty | i seem to go between 2 or 3 |
| 02:16.28 | Katty | always less in the morning |
| 02:17.31 | jblack | I wish this app had rolling averages. |
| 02:17.50 | Katty | you could probably use open office |
| 02:17.57 | jblack | it would smooth this data out and give me something useful. |
| 02:18.19 | jblack | I would nothing more than to re-enter datasets again and again. :P |
| 02:18.50 | Katty | lol |
| 02:19.10 | Katty | monday i'm doing a writeup in excel. |
| 02:19.30 | Katty | comparing my DV% of friday with saturday and sunday |
| 02:19.37 | Katty | gonna see if i'm moving in the right direction |
| 02:19.49 | jblack | I bet you are. |
| 02:19.54 | Katty | probably. |
| 02:20.07 | Katty | especially since i've been eating healthier |
| 02:20.43 | Katty | after i've got my eating habits sorted out, i'll probably start taking suppliments on the things i'm still deficient in |
| 02:22.36 | [TK]D-Fender | Katty: "If you think you need to take all the pills the pharmaceutical compaines are telling you need to... then you're ALREADY on drugs" - Bill Maher |
| 02:22.50 | Kobaz | heh |
| 02:22.52 | [TK]D-Fender | Katty: the rest of the world has no problem simply "being". |
| 02:23.00 | Kobaz | i take a multivitamin |
| 02:23.04 | [TK]D-Fender | Katty: todays PSA :) |
| 02:23.11 | Katty | good thing i'm not on drugs anymore, eh? |
| 02:23.29 | [TK]D-Fender | Katty: Well there is the matter of your sister.... |
| 02:23.30 | Katty | Kobaz: yeah i probably will too, in the long run. |
| 02:23.41 | Katty | [TK]D-Fender: what my sister does is her own business. |
| 02:23.42 | [TK]D-Fender | Katty: You know.. this could be GENETIC! |
| 02:23.54 | [TK]D-Fender | Katty: Is she neurotic too? ;) |
| 02:24.04 | Katty | no, i'm the neurotic sister. |
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| 02:24.26 | Katty | but my immune system being stronger puts my mind at ease. |
| 02:26.21 | jblack | Bill Maher is a twit. He's the liberal version of Bill O'reilly |
| 02:26.21 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-212-10-157.acanac.net) |
| 02:26.51 | jblack | I must be a bill thing. |
| 02:28.16 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 02:28.39 | [TK]D-Fender | jblack: Sure he's goes off the deep end, but in term of frequency and relevency... considerably less :) |
| 02:30.10 | Katty | hmm. naptime. |
| 02:30.14 | Katty | later folks (= |
| 02:43.40 | jblack | I'm not sure. In his last show, he ranted for 40 minutes about the worthlessness of immunizations. |
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| 02:59.22 | ChannelZ | that reminds me I need to cancel my HBO |
| 03:01.46 | [TK]D-Fender | jblack: Well there its a piece of reality for you.... the common flu kills how many exactly? In what state? How does that relate to to immunized %? How much worse have flu's because as a result ot mutating to circumvent resistances? How mush have out attempts to use medication instead of building up a natural resistance cost us when this causes super-bug's to form? |
| 03:04.23 | [TK]D-Fender | jblack: Polio? Sure. Something that has a habit of wiping out nations? Sure.... the more you try to fight the common cold, the more likely it is to kill you later. |
| 03:05.18 | manxpower | 1990-91 through the 1998-99 flu seasons [10]. Statistical modeling was used to estimate how many flu-related deaths occurred among people whose underlying cause of death on their death certificate was listed as a respiratory or circulatory disease. During these years, the number of estimated deaths ranged from 17,000 to 52,000. |
| 03:05.35 | manxpower | so there we have your answer. |
| 03:06.13 | manxpower | I agree with the idea that immunizations are not critical for groups not at high risk of death from it. |
| 03:17.24 | [TK]D-Fender | manxpower: I love stats with a 400% margin of error :) |
| 03:17.53 | sudhir492 | how to find which codec is used in a call? |
| 03:18.38 | [TK]D-Fender | sudhir492: sip show channels <- (or IAX2, etc |
| 03:20.01 | sudhir492 | thx |
| 03:29.01 | mchou | anyone know how exactly round trip delay on pap2 is calculated? I'm getting totally absurd numbers (17hrs) |
| 03:29.50 | mchou | this is a pap2 hooked up on my lan to asterisk |
| 03:29.53 | *** join/#asterisk phuff (n=user@sourceforge/staff/luapffuh) |
| 03:30.50 | phuff | If I know a certain extension has a call in progress can I force transfer it to me from another extension? |
| 03:31.13 | phuff | Or join it in progress as a 3rd party? |
| 03:33.27 | ChannelZ | well you can use ChanSpy to do that |
| 03:33.40 | ChannelZ | not sure how to 'steal' the call |
| 03:34.25 | p3nguin_ | ChannelRedirect, maybe? |
| 03:34.53 | phuff | ChanSpy might work |
| 03:34.59 | phuff | I have a whitelist setup for incoming calls |
| 03:35.11 | phuff | But I'm trying to talk to google voice's phone verifier :) |
| 03:35.23 | phuff | So I need to be able to hop on a channel and dial some numbers |
| 03:35.59 | mchou | lol |
| 03:36.19 | mchou | talk about a use case :) |
| 03:36.20 | *** join/#asterisk lewellyn (n=lewellyn@greenviolet/lewellyn) |
| 03:36.30 | phuff | Yeah, seriously :) |
| 03:36.48 | phuff | It's a bit arcane/ |
| 03:37.00 | phuff | At our old house, our number used to belong to a stucco place |
| 03:37.15 | phuff | then we bought a new house, and it turns out the number they gave us belonged to an old chinese restaurant |
| 03:37.23 | phuff | We'd seriously get calls at like 3:00am for take out. |
| 03:37.28 | phuff | So I setup a whitelist |
| 03:37.28 | mchou | it's known as the "buried shovel" problem :) |
| 03:37.43 | mchou | hahah |
| 03:37.50 | mchou | that is classic |
| 03:38.09 | mchou | chinese restaurant from goggle voice |
| 03:38.30 | phuff | oh, no it was just fro qwest |
| 03:38.33 | phuff | from qwest |
| 03:38.42 | mchou | at this rate I'd just change my GV phone # :) |
| 03:39.10 | phuff | i'm just trying to get gv to work with this # |
| 03:39.13 | mchou | ah, then it's even easier |
| 03:39.20 | mchou | have qwest change it |
| 03:39.46 | mchou | assuming you arent using it for business etc |
| 03:39.50 | phuff | asterisk white list works just as well and kicks out most telemarketers too :) |
| 03:40.10 | mchou | astycrapper for teh win! |
| 03:40.56 | phuff | plus i finally got a chance to learn a bit about asterisk |
| 03:40.56 | phuff | :) |
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| 03:47.54 | mchou | you may be better off setting up a phone conference first.... |
| 03:48.19 | mchou | before GV calls you back with their phone verifier |
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| 03:49.24 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 03:49.25 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 03:49.52 | mchou | I wonder where dtmf tones really get sent to in that case |
| 03:52.14 | Kobaz | nvidia |
| 03:52.15 | Kobaz | er |
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| 04:00.11 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.6 (2009/09/03), 1.6.0.15 (2009/09/03), 1.4.26.2 (2009/09/03), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #switchvox #asterisk-bugs |
| 04:02.00 | [TK]D-Fender | Qwell: Astricon over? |
| 04:02.10 | Qwell | just a bit |
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| 04:08.48 | dlewis | anyone have any thoughts on this error: http://pastebin.com/d3bc5d9fb |
| 04:11.50 | ChannelZ | looks like you've got some includes with no actual files |
| 04:12.02 | ChannelZ | like it says 20 times |
| 04:17.47 | [TK]D-Fender | ChannelZ: Maybe you should look at your manager.conf <- |
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| 04:20.29 | [TK]D-Fender | ChannelZ: And in case you missed it... |
| 04:20.36 | [TK]D-Fender | [00:17]<[TK]D-Fender>ChannelZ: Maybe you should look at your manager.conf <- |
| 04:20.56 | ChannelZ | me? |
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| 04:27.08 | [TK]D-Fender | [00:08]<dlewis>anyone have any thoughts on this error: http://pastebin.com/d3bc5d9fb |
| 04:27.12 | [TK]D-Fender | dlewis: Yeah YOU :P |
| 04:28.47 | ChannelZ | yea.. not me |
| 04:31.20 | dlewis | [TK]D-Fender: thanks. |
| 04:32.00 | [TK]D-Fender | ChannelZ: AND DON'T YOU FORGET IT! |
| 04:32.15 | ChannelZ | Who what now? |
| 04:32.21 | ChannelZ | :) |
| 04:33.00 | TJNII | For a split second there I thought [TK]D-Fender was asking to analyize a pastebin from him! |
| 04:33.00 | TJNII | s/asking to/asking us to/ |
| 04:33.01 | TJNII | pokes infobot |
| 04:33.09 | TJNII | There he goes. |
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| 05:23.20 | DND | guys, im really having a hard time figuring out why the channel is not hanged up |
| 05:23.31 | DND | even the other party already hangs up |
| 05:24.58 | DND | during the IVP, i hanged up. but in CLI, it still goes directly to the VM |
| 05:27.56 | [TK]D-Fender | DND: And thus far do not see details like the kind of channel and things related to it would be important to share... |
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| 05:35.17 | DND | yes hold on im pasting it in pastebin |
| 05:37.24 | DND | im just finishing the VM |
| 05:38.25 | DND | here it ishttp://pastebin.ca/1643434 |
| 05:38.30 | DND | here it is http://pastebin.ca/1643434 |
| 05:38.53 | [TK]D-Fender | ~cds |
| 05:38.54 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 05:38.55 | [TK]D-Fender | DND: ^^^ Go call your telco |
| 05:38.56 | DND | i just had it rang once then i disconnect the external party |
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| 05:38.57 | [TK]D-Fender | DND: ^^^ Go call your telco |
| 05:39.31 | DND | what should i tell them? |
| 05:40.07 | [TK]D-Fender | DND: I want this sevice on my lines |
| 05:40.13 | [TK]D-Fender | DUH |
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| 06:30.21 | joelsolanki | Hi room |
| 06:30.40 | joelsolanki | i m trying to compile meetme in asterisk. |
| 06:30.45 | joelsolanki | i installed dahdi too |
| 06:31.06 | joelsolanki | but when i do make menuconfig still the meetme stuff shows XXX means not activated |
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| 06:31.10 | joelsolanki | any hints plz |
| 06:31.29 | joelsolanki | my asterisk version is 1.4.26.2 |
| 06:31.57 | joelsolanki | i installed dahdi 2.2.0 |
| 06:33.03 | joelsolanki | any help plz |
| 06:37.06 | russellb | re-run the asterisk configure script after installing dahdi |
| 06:40.05 | joelsolanki | oh ok |
| 06:40.08 | joelsolanki | let me do it |
| 06:41.23 | joelsolanki | ok cool that worked russellb |
| 06:41.24 | joelsolanki | :) |
| 06:41.45 | joelsolanki | thanks |
| 06:42.18 | russellb | you're welcome |
| 06:42.46 | russellb | and now, bed. |
| 06:43.17 | joelsolanki | :) |
| 06:43.19 | joelsolanki | good night |
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| 08:14.20 | DND | hi guys, |
| 08:14.41 | DND | why is it whenever i create a conference extension, the ivr doesnt work? |
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| 09:54.59 | Brooklyn | I have got an Asterisk box connected with a SIP trunk that functions fine with Aastra SIP phones. The Philips/NEC wireless phones have a problem with the audio volume tho. |
| 09:55.14 | Brooklyn | The volume is too soft. Ofcourse we set the phones to the max volume but it is still to soft. |
| 09:55.24 | Brooklyn | Is there a way to adjust the volume gain for SIP channels? |
| 09:55.59 | Brooklyn | I know how to do this for ISDN connections but for SIP, is it possible with Asterisk? |
| 09:58.28 | trogs | not sure that it is possible for sip trunks |
| 09:58.47 | Brooklyn | Yes I am afraid its not possible. |
| 09:59.09 | Brooklyn | The Aastra phones are fine but the Philips wireless phones are so hard to use in a bit noisy environment. |
| 09:59.32 | Brooklyn | Its almost not possible to have a normal conversation. |
| 09:59.33 | trogs | you might be able to get a headset for the phones |
| 09:59.53 | Brooklyn | Ok but there is no option on the Asterisk box itself to gain volume for a few specific extensions? |
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| 10:00.49 | trogs | looks like there may be a new option in 1.56 |
| 10:00.50 | trogs | er |
| 10:00.51 | trogs | 1.6 |
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| 10:01.13 | kaldemar | Brooklyn: look at func VOLUME |
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| 10:01.37 | trogs | http://blog.dontcallmyboss.com/2008/09/asterisk-volume-function-dialplan.html |
| 10:02.57 | Brooklyn | Ok, I will have a look. Thank you! |
| 10:03.15 | trogs | think i will have a play with this tomorrow too. |
| 10:03.33 | trogs | might help make the speakerphone on some of my crappy phones work better ha. |
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| 10:12.32 | pisco_sour | Did chan mobile get integrated into Asterisk? If so, where can I find docs on it? |
| 10:14.00 | kaldemar | it's in the addons. take a look at doc/chan_mobile.txt in the addon source package. |
| 10:25.14 | pisco_sour | kaldemar: thx |
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| 11:45.36 | dlynes | Brooklyn: btw...you know that the Aastra phones have an input gain and an output gain setting, configurable via tftp, right? |
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| 11:46.22 | dlynes | Brooklyn: The Philips might have something similar, too |
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| 12:00.01 | Brooklyn | Thanks dlynes |
| 12:00.26 | Brooklyn | The problem is with the Philips phones, I will check the config options once again. Pretty sure its all set to the max now tho. |
| 12:00.34 | thisismyname | hi there i got some questions on the digest algorithm |
| 12:00.34 | thisismyname | let me describe it that way: |
| 12:00.34 | thisismyname | i am calling from A to B, with a switch in the middle |
| 12:00.35 | thisismyname | <PROTECTED> |
| 12:00.35 | thisismyname | A <- Switch (Auth Challange) |
| 12:00.35 | thisismyname | A -> Switch (INVITE with auth header) |
| 12:00.37 | thisismyname | A <- Switch (Ringing) |
| 12:00.39 | thisismyname | b side doesnt matter |
| 12:00.41 | thisismyname | noch A cancels the call |
| 12:00.44 | thisismyname | A -> Switch (CANCEL) |
| 12:00.46 | thisismyname | Now my question: Are the authorization header from the invite and the cancel the same? |
| 12:00.48 | thisismyname | or contains the digest algorithm a hash over the Method, too? |
| 12:02.01 | ManxPower-work | thisismyname: That sort of thing should be specified in the SIP RFCs. Most of use are not SIP protocol gururs here. |
| 12:05.18 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 12:05.21 | thisismyname | yeh.. it'll cost me some hoers to read into that... thought anyone perhaps has an idea |
| 12:05.27 | thisismyname | worth a try |
| 12:06.21 | thisismyname | but.. i got another question :).. regarding asterisk directly |
| 12:06.50 | thisismyname | i heard the 1.5er asterisk has a tool to convert wav files to g722... what is it called an wheere can i find it?? |
| 12:06.56 | thisismyname | 1.6er |
| 12:09.08 | kaldemar | thisismyname: in cli, "help file convert" |
| 12:09.49 | thisismyname | thx a lot.. il give a try |
| 12:12.07 | thisismyname | do g722 file have the ending .g722?? |
| 12:13.04 | *** join/#asterisk telnettech (n=bsimpson@office.callcopy.com) |
| 12:13.58 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 12:18.20 | *** join/#asterisk jojo^ (i=jol@ph4.se) |
| 12:18.40 | jojo^ | When on a SIP channel, is there another way to do a 302 redirect besides "Transfer" ? |
| 12:19.16 | jojo^ | I am looking for a way to pass custom SIP headers along with the 302 redirect |
| 12:22.12 | *** join/#asterisk [netman] (n=netman@249.Red-88-8-164.dynamicIP.rima-tde.net) |
| 12:24.43 | ManxPower-work | jojo^: that sounds more like you are looking for a SIP proxy (Asterisk is not a SIP proxy) |
| 12:25.29 | ManxPower-work | jojo^: A Transfer from the phone should not do a 302 |
| 12:26.32 | ManxPower-work | I don't know about dialplan Transfer. Very few people use that. |
| 12:26.58 | jojo^ | ManxPower-work, Well I've got a kamailio in front of the asterisk. But I'm doing some logic in an asterisk AGI (which are much more flexible).. But in the case of a call forward, I want to do a 302 redirect back to kamailio instead of creating a new leg with dial |
| 12:27.51 | Chainsaw | <PROTECTED> |
| 12:27.51 | Chainsaw | <PROTECTED> |
| 12:27.55 | jojo^ | The SIP headers would essentially be a way to tell kamailio what to do after the 302 |
| 12:28.02 | Chainsaw | ^ Last I checked 0 is not equal to 20, so why isn't it jumping to 512 instead of 650? |
| 12:28.29 | ManxPower-work | Chainsaw: you have a SYNTAX error. The CLI should display 0 or 1, not the formual |
| 12:28.49 | ManxPower-work | did you forget the $ in front of the [ ? |
| 12:28.55 | Chainsaw | has a look |
| 12:29.32 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 12:29.54 | Chainsaw | ManxPower-work: I did indeed. Thank you :) |
| 12:30.28 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:32.20 | ManxPower-work | <-- smrter than he looks |
| 12:34.11 | [TK]D-Fender | must ... resist ... cheap .. SHOT! |
| 12:34.17 | [TK]D-Fender | GGGAAAAAAAAAAAAAAHHHHHHHHHH!!!!!! |
| 12:34.21 | [TK]D-Fender | runs in circles |
| 12:34.37 | [TK]D-Fender | :d |
| 12:34.50 | beek | Morning [TK]D-Fender , ManxPower-work |
| 12:35.05 | [TK]D-Fender | beek: Mornin' |
| 12:35.28 | [TK]D-Fender | beek: 8-Ball says "Sure, Fine, Whatever" :) |
| 12:35.53 | beek | :D |
| 12:39.21 | *** join/#asterisk frk2 (n=faraz@zivios/member/fkhan) |
| 12:46.56 | *** join/#asterisk garymc (i=garymc@host81-131-239-245.range81-131.btcentralplus.com) |
| 12:49.17 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:54.11 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 12:55.54 | *** join/#asterisk mintos (n=mvaliyav@115.240.70.55) |
| 12:56.08 | beek | Mornin' jaytee |
| 12:56.27 | jaytee | morning beek |
| 13:07.14 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 13:09.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:10.56 | jaytee | morning Katty |
| 13:12.05 | *** join/#asterisk QaDeS (n=mklaus@213.157.13.70) |
| 13:13.15 | *** join/#asterisk glam (n=glam@58.254.68.109) |
| 13:16.33 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 13:16.38 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
| 13:16.41 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:17.26 | beek | morning Katty |
| 13:17.57 | riddlebox | hey all |
| 13:18.09 | beek | hey |
| 13:20.02 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 13:20.22 | tzafrir | ManxPower-work, so I guess your homepage looks like http://xkcd.com/ |
| 13:20.47 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 13:22.09 | ManxPower-work | tzafrir: This is my homepage: http://www.fnords.org/ |
| 13:22.37 | tzafrir | ManxPower-work, I know. Just an excuse to post the link |
| 13:22.54 | Chainsaw | tzafrir: Could you change the affected version on https://issues.asterisk.org/bug_view_page.php?bug_id=14163 to 1.6.1.6 please. |
| 13:23.10 | Chainsaw | tzafrir: (Just to give it a better chance of actually being looked at) |
| 13:23.22 | ManxPower-work | To me having a homepage is like having a pony. Sounds cool at first, but you eventually realize just how much of a time sink it is. |
| 13:23.30 | Katty | good morning jaytee (= |
| 13:23.32 | Katty | hugs jaytee |
| 13:23.36 | Katty | good morning beek! :> |
| 13:23.38 | Katty | hugs beek |
| 13:23.50 | jaytee | hugs Katty back |
| 13:24.11 | leifmadsen | Chainsaw: I don't think changing the version number will have any effect |
| 13:24.33 | leifmadsen | Chainsaw: it's already assigned, and marked as Ready for Review |
| 13:24.48 | tzafrir | Chainsaw, the version of asterisk on this bug is "SVN" |
| 13:24.58 | Chainsaw | leifmadsen: Since July, yes. |
| 13:25.12 | leifmadsen | Chainsaw: yes, and there are over 500 currently open issues |
| 13:25.17 | tzafrir | Are you sure you want to change it to 1.6.1.6 ? |
| 13:25.24 | Chainsaw | tzafrir: leifmadsen says no. |
| 13:25.32 | leifmadsen | tzafrir: it makes no difference |
| 13:25.47 | *** join/#asterisk retentiveboy (n=pdugas@72.54.144.26) |
| 13:25.53 | tzafrir | leifmadsen, however it has just 'SVN' without even a branch |
| 13:25.55 | Chainsaw | sighs and applies it downstream |
| 13:25.58 | leifmadsen | it's assigned and marked as Ready for Review, so will be dealt with when it is |
| 13:26.16 | ManxPower-work | Chainsaw: welcome to the Asterisk dev process. |
| 13:26.52 | jaytee | see, I never got a friggin pony cuz that fat bastard at the North Pole was too busy eating jelly donuts to look at the damn wishlist. At least if he had then I would have HAD the opportunity to learn about hard work by taking care of the pony but nooooooo! so it's all Santa Claus's fault that I'm an underachiever. |
| 13:27.32 | leifmadsen | ManxPower-work: help is always welcome to move issues forward, but it is currently marked as minor; block, crash, and major are higher priority |
| 13:27.44 | Chainsaw | ManxPower-work: It's better then the last bug, which was answered with "it's Gentoo's fault, go away" until I made a scene on the IRC channel. |
| 13:28.47 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 13:28.48 | ManxPower-work | Chainsaw: I gave up years ago. |
| 13:29.27 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 13:29.40 | leifmadsen | just bites his tongue |
| 13:30.18 | beek | jaytee: I always thought that the "jolly old elf" was a pedophile. |
| 13:30.51 | tzafrir | ELF is something you should execute |
| 13:31.11 | ManxPower-work | Chainsaw: You can fight Asterisk's oddities and live a miserable, pointless life or you can accept Asterisk's oddities and find nirvana. |
| 13:31.41 | voipmonk | amen |
| 13:33.15 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 13:47.26 | *** join/#asterisk Gnoll (n=Gnoll@host242-31-dynamic.1-87-r.retail.telecomitalia.it) |
| 13:47.27 | Gnoll | hi |
| 13:47.58 | plundra | Does the language-variable change how ifTime()-behaves? (Change the timezone for instance) |
| 13:49.03 | [TK]D-Fender | plundra: What does "language' have to do with "time zones" ? |
| 13:50.01 | plundra | [TK]D-Fender: Nothing I would hope, just checking, because a few people in our customer service says calls are geting through on weird hours. |
| 13:50.57 | [TK]D-Fender | plundra: feel free to show us something we can debug, because that was a pretty baseless stab in the dark... |
| 13:51.14 | Chainsaw | plundra: And it's not people forgetting to log out? That's been such an issue here that we automated a 10pm "log everyone out". |
| 13:52.18 | Gugge | i have an asterisk using 100% cpu, is there any cli commands that can show whats using the cpu? |
| 13:52.18 | [TK]D-Fender | Chainsaw: He's talking about 1 dialplan app... not a Queue |
| 13:52.25 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 13:52.30 | plundra | Chainsaw: I'm using an ifTime(...) { Queue(...);} else { Playback(openhours); }; |
| 13:53.28 | *** join/#asterisk Skeeter- (i=Skeeter-@190-141.cgocable.ca) |
| 13:53.44 | Skeeter- | anyone knows how to customize Polycom hardkeys |
| 13:54.07 | plundra | I really can't recreate the problem, and that is pretty anoying :) I think ifTime behaves JUST as I except, all the time. |
| 13:54.37 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 13:54.50 | Gugge | plundra: set asterisk to log, and then tell people to tell you when it happens |
| 13:54.51 | plundra | But just to be sure, ifTime() always compares against localtime? |
| 13:54.53 | Gugge | then check the log |
| 13:55.44 | plundra | Gugge: That sounds like a plan yes :) I'm at home sick right now, but I'll do that later. |
| 13:57.03 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 13:57.13 | plundra | Again a shoot in the dark, but could it be the Date-header in SIP that is set improperly? |
| 13:59.22 | Gugge | stop guessing, its a waste of your time :) |
| 13:59.49 | plundra | Yeah I know, but it's so frustrating... |
| 13:59.53 | Skeeter- | when i press my polycom message button, it dials my ext. but i would like it to dial *97 |
| 14:01.28 | *** join/#asterisk DavidR2008 (n=chatzill@208.34.240.2) |
| 14:02.02 | *** join/#asterisk freezey (n=freezey@static-64-61-84-174.isp.broadviewnet.net) |
| 14:03.20 | *** join/#asterisk k4mi1 (n=qmpelkam@xdsl-1545.wroclaw.dialog.net.pl) |
| 14:06.56 | Gugge | Am i just blind, or is there no link to asterisk-addons on asterisk.org anymore? |
| 14:07.44 | k4mi1 | hey all, do You know if it is possible to configure asterisk to play music instead of dialing tone? :-) |
| 14:09.21 | jaytee | blind as a bat |
| 14:09.28 | Gugge | k4mi1, my guess is that background(file,n) would work |
| 14:09.38 | Gugge | jaytee, i thought so ... wanna give me a clue? :) |
| 14:09.40 | [TK]D-Fender | Gugge: Yeah, they screwed stuff up bad... |
| 14:10.04 | *** join/#asterisk djdefi (n=rtrauntv@mta.novacoast.com) |
| 14:10.18 | Gugge | k4mi1, no not background ... i dont know then :) |
| 14:10.40 | k4mi1 | :) |
| 14:10.44 | [TK]D-Fender | k4mi1: clarify "dialing tone" |
| 14:10.50 | k4mi1 | ok |
| 14:11.12 | k4mi1 | I am a person who make a call to other person |
| 14:11.15 | jaytee | Gugge, ok you're not blind. Looks like they frigged up the download page on the site |
| 14:11.43 | [TK]D-Fender | m4core show application dial" <------ read the instructions |
| 14:11.54 | k4mi1 | when I wait for the answer I usually hear a dialing tone/ring tone like beeeeep ........ beeeeeep ....... :-) |
| 14:11.59 | [TK]D-Fender | jaytee: No direct link, and they killed the "base HTTP' one as well. |
| 14:12.04 | [TK]D-Fender | jaytee: Big piss-off |
| 14:12.07 | k4mi1 | instead of this I want to hear music |
| 14:12.55 | [TK]D-Fender | k4mi1: core show application dial" <------ read the instructions |
| 14:13.05 | k4mi1 | let me see... |
| 14:13.34 | jaytee | well, there's still this to download the tarball http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 14:14.43 | *** join/#asterisk cyberfab007 (n=cyberfab@CPE001b11cf4f69-CM0014f85c3ada.cpe.net.cable.rogers.com) |
| 14:17.21 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:20.13 | *** join/#asterisk Chesther (n=cam2@cam2-mac.cit.cornell.edu) |
| 14:20.44 | ManxPower-work | Skeeter-: I'm sure most anyone that has read the Polycom Admin guide. |
| 14:24.48 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:25.17 | [TK]D-Fender | would tend to agree |
| 14:26.01 | *** join/#asterisk gardo (n=gardo@121.96.137.100) |
| 14:27.32 | *** join/#asterisk DMeloUK (n=Dominic_@64.129.94.138) |
| 14:35.11 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:41.49 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:41.50 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:48.37 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:49.31 | *** join/#asterisk torrancew (n=torrance@btr.macconsultinggroup.com) |
| 14:50.01 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 14:50.05 | torrancew | with asterisk voice mail, what's the biggest advantage of using imap for storage? |
| 14:50.06 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:50.23 | *** join/#asterisk ZeRo_____CoOoL (n=ZeRoCoOo@196.203.72.98) |
| 14:50.27 | leifmadsen | centralized voicemail? |
| 14:50.51 | torrancew | i'm leery of putting our company's imap passwords all in voicemail.conf |
| 14:50.55 | ZeRo_____CoOoL | hello |
| 14:51.08 | torrancew | or any text file, for that matter |
| 14:51.38 | ZeRo_____CoOoL | I have a small question, please I'm trying to compile asterisk, I have some problems |
| 14:51.39 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 14:51.52 | leifmadsen | ~ask |
| 14:51.53 | 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. |
| 14:52.06 | tuxx- | :) |
| 14:52.18 | tuxx- | hehe |
| 14:52.21 | tuxx- | against our will xD |
| 14:53.12 | [TK]D-Fender | torrancew: then don't |
| 14:53.31 | ZeRo_____CoOoL | Os: OpenSuse |
| 14:53.47 | ZeRo_____CoOoL | asterisk 1.6 |
| 14:53.50 | torrancew | [TK]D-Fender, it's not optional? |
| 14:53.51 | ZeRo_____CoOoL | error |
| 14:54.03 | ZeRo_____CoOoL | usr/lib/gcc/i586-suse-linux/4.3/../../../../i586-suse-linux/bin/ld: Dwarf Error: Offset (6800) greater than or equal to (null) size (1076084689). |
| 14:54.13 | [TK]D-Fender | torrancew: Feel free to configure voicemail via any means you want |
| 14:54.31 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:54.33 | ZeRo_____CoOoL | reinstalled binutils |
| 14:54.38 | torrancew | [TK]D-Fender: i meant the imap password, is it optional? |
| 14:54.47 | ZeRo_____CoOoL | but no way |
| 14:57.29 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:57.42 | [TK]D-Fender | torrancew: extconfig your voicemail |
| 14:57.48 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 14:58.12 | [TK]D-Fender | torrancew: But then THAT will have a plain text PW in a file too |
| 14:58.18 | torrancew | right |
| 14:58.23 | *** part/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:58.23 | [TK]D-Fender | torrancew: So you're damned if you do, damned if you dont.... |
| 14:58.29 | torrancew | thanks |
| 14:58.33 | [TK]D-Fender | torrancew: So fix the premissions on it |
| 14:58.41 | *** join/#asterisk moy (n=moy@74.12.131.148) |
| 14:58.51 | [TK]D-Fender | torrancew: and secure your sever in other ways |
| 14:59.17 | torrancew | [TK]D-Fender: not an option here - i'm happy with that solution, but my employer is not so much |
| 14:59.19 | torrancew | kk |
| 14:59.28 | torrancew | oops* wrong window |
| 14:59.44 | torrancew | he runs the email server like a madman |
| 14:59.47 | [TK]D-Fender | torrancew: short answer : TFB <- |
| 15:00.04 | torrancew | TFB? |
| 15:00.08 | [TK]D-Fender | ~tfb |
| 15:00.09 | infobot | it has been said that tfb is Too #&^$ing bad.... |
| 15:00.17 | torrancew | ah |
| 15:00.20 | torrancew | fair enough |
| 15:00.24 | torrancew | thanks all the same |
| 15:00.41 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:01.37 | [TK]D-Fender | torrancew: Feel free to not use IMPA for storage then :) |
| 15:01.41 | [TK]D-Fender | IMAP* |
| 15:01.45 | Skeeter- | is there any voice recogniztion direcotory free out there |
| 15:01.48 | torrancew | point taken |
| 15:02.32 | [TK]D-Fender | Skeeter-: Sphinx |
| 15:02.50 | Skeeter- | [TK]D-Fender: thanks |
| 15:03.01 | ZeRo_____CoOoL | any one can help me please ? |
| 15:07.25 | ManxPower-work | ~ask |
| 15:07.26 | 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. |
| 15:07.44 | torrancew | ManxPower-work: he already did |
| 15:07.52 | [TK]D-Fender | yup |
| 15:08.29 | ZeRo_____CoOoL | Os: OpenSuse |
| 15:08.39 | ZeRo_____CoOoL | asterisk version: 1.6.1.6 |
| 15:08.43 | *** part/#asterisk dlewis (n=dlewis@about/security/staff/dlewis) |
| 15:08.44 | ZeRo_____CoOoL | and the error: |
| 15:08.47 | ZeRo_____CoOoL | usr/lib/gcc/i586-suse-linux/4.3/../../../../i586-suse-linux/bin/ld: Dwarf Error: Offset (6800) greater than or equal to (null) size (1076084689). |
| 15:08.59 | ZeRo_____CoOoL | a problem with "ld" |
| 15:09.08 | ZeRo_____CoOoL | I reinstalled binutils |
| 15:09.13 | ManxPower-work | ZeRo_____CoOoL: I would call that a problem with your distro |
| 15:09.14 | ZeRo_____CoOoL | binutils-devel |
| 15:09.26 | Qwell | ZeRo_____CoOoL: Why did you reinstall binutils? |
| 15:09.41 | ZeRo_____CoOoL | because I have a problm with "ld" |
| 15:09.52 | ZeRo_____CoOoL | the linkers |
| 15:09.59 | *** join/#asterisk dmz (n=dmz@h96-61-156-43.mtjltn.dsl.dynamic.tds.net) |
| 15:10.02 | Qwell | well, you did something to hose your toolchain pretty nicely |
| 15:12.46 | ZeRo_____CoOoL | ManxPower-work: I have to change the version of OpenSuse ? |
| 15:13.09 | ZeRo_____CoOoL | or to use another distro ? |
| 15:13.28 | p3nguin_ | Is openSUSE broken? |
| 15:13.35 | [TK]D-Fender | p3nguin_: His is. |
| 15:13.42 | [TK]D-Fender | (apparently) |
| 15:15.27 | ZeRo_____CoOoL | I tried to installed on 2 box |
| 15:15.34 | ZeRo_____CoOoL | I'll change the distro |
| 15:15.40 | ZeRo_____CoOoL | thanks :) |
| 15:20.46 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:24.01 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 15:24.44 | *** join/#asterisk TheDavidFactor (n=chatzill@208.34.240.2) |
| 15:26.55 | tzafrir | Actually, he provided very few details. And as "OpenSUSE" was the "unusual" detail, it was blamed |
| 15:28.01 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 15:31.05 | Qwell | tzafrir: as much as I would have liked to blame SUSE, I refrained. If he's willing to just "reinstall binutils", I suspect unknowledgable tampering. |
| 15:31.18 | Qwell | ie; he screwed something up |
| 15:34.11 | Naikrovek | yeah |
| 15:34.44 | *** join/#asterisk KavanS (n=KavanS@173.50.141.22) |
| 15:34.57 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:37.44 | dlynes | Has anyone tried using didx.org for buying dids? |
| 15:38.44 | ManxPower-work | dlynes: Once they get your e-mail address, you'll NEVER get them to stop spamming you |
| 15:39.12 | dlynes | ManxPower-work: i c |
| 15:39.23 | dlynes | ManxPower-work: and i'm guessing their dids don't work either, eh? |
| 15:39.38 | dlynes | ManxPower-work: I've tried registering to their stupid server, and it always times out |
| 15:40.21 | Katty | jblack: the results are in! |
| 15:40.49 | ManxPower-work | dlynes: they don't have any DIDs, They are just a broker for other people's DIDs |
| 15:41.12 | Katty | jblack: over the course of 4 days, on average, i need more Vitamin D, E, B1, B2, B6, B5, Iron, Potassium, and copper |
| 15:42.05 | ManxPower-work | hands Katty a multi-vitamin |
| 15:42.49 | dlynes | ManxPower-work: ok, thanks |
| 15:43.05 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:43.10 | dlynes | Anyone happen to know anyone with some Indianapolis DIDs, then? |
| 15:43.24 | jaytee | I'm in Indianapolis |
| 15:43.41 | ManxPower-work | dlynes: any ITSP |
| 15:43.47 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 15:43.48 | Katty | ManxPower-work: hmm. yeah. i'm thinking that may be the only way i can get all of my vitamins. |
| 15:43.52 | Katty | ManxPower-work: i'm eating very healthy :/ |
| 15:44.00 | Katty | ManxPower-work: not sure if i can eat more. |
| 15:44.18 | Naikrovek | Katty: you almost need to eat some dirt to get all those nutrients |
| 15:44.21 | dlynes | ManxPower-work: Nah...not any itsp...all the ones i've checked don't have any |
| 15:45.34 | ManxPower-work | dlynes: vitelity.net |
| 15:46.07 | ManxPower-work | they have a bunch of rate centers in IN. Do you have an NPA-NXX? |
| 15:46.10 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 15:46.36 | raden_work | Naikrovek, morning long time no chat |
| 15:46.41 | Naikrovek | yeah |
| 15:46.42 | Naikrovek | busy |
| 15:47.21 | Katty | ManxPower-work: hrm :< |
| 15:47.46 | Katty | ManxPower-work: but i don't want to eat dirt. how about a nice multi b vitamin cookie instead? :< |
| 15:49.19 | dlynes | ManxPower-work: nah...just care that it's local to indianapolis...thanks |
| 15:50.12 | ManxPower-work | dlynes: Level 3 covers that area and most ITSPs use Level 3 |
| 15:50.58 | jaytee | dlynes, call Axia Technologies or ProNet Indy. they can probably point you in the right direction for this city |
| 15:51.47 | *** join/#asterisk gandhijee (i=akp@66.202.34.165) |
| 15:52.02 | *** join/#asterisk Tim_Toady (n=moi@adsl57-46.kln.forthnet.gr) |
| 15:52.28 | dlynes | jaytee: thanks |
| 15:52.32 | Katty | ugah. Liver. |
| 15:52.41 | Katty | sighs |
| 15:52.45 | Katty | guess i'm going to try liver. |
| 15:52.53 | jaytee | double ugah, Spleen |
| 15:53.09 | Katty | yeah not sure if i'm going to be able to stomach that stuff. |
| 15:53.15 | *** join/#asterisk wcselby (n=wcselby@216.110.88.194) |
| 15:53.29 | *** join/#asterisk slinksh0t_ (n=slinksh0@adsl-9-177-182.mia.bellsouth.net) |
| 15:54.05 | *** join/#asterisk psilikon (n=psilikon@static-173-65-4-24.tampfl.fios.verizon.net) |
| 15:54.17 | Katty | pouts a bit |
| 15:55.20 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 15:55.20 | eppigy | :< |
| 15:56.15 | wcselby | o/ |
| 15:56.24 | Katty | hi wcselby |
| 15:58.23 | torrancew | another question from my higher-ups: they don't like the idea of the parking lot, and want to know if it's possible to grab someone from the lot without knowing the number |
| 15:58.37 | raden_work | Naikrovek, same here service most this week take it easy bro on the road again |
| 15:58.49 | raden_work | hugs katty . Hi bye see you later |
| 15:58.50 | torrancew | or to bypass it all together, as our ip phones have their own little hold feature that is local to the phone |
| 15:58.52 | Naikrovek | later, raden_work. have fun |
| 15:59.03 | TSM2 | torrancew: yup make a script that shows them all the people waiting in the parking lot |
| 15:59.18 | torrancew | not his happy solution |
| 15:59.23 | torrancew | is there any way to bypass the lot? |
| 15:59.29 | torrancew | my understanding is that there's not |
| 15:59.29 | TSM2 | yup turn it off |
| 15:59.30 | ManxPower-work | torrancew: It may be possible, but it would be a lot of work. |
| 15:59.45 | ManxPower-work | torrancew: Asterisk does not support picking up on-hold calls. |
| 15:59.48 | TSM2 | features.conf, just change the code to somthing complicated |
| 15:59.53 | [TK]D-Fender | torrancew: "the lot"? there could be MANY |
| 15:59.59 | torrancew | right |
| 16:00.07 | torrancew | we're a small office, we use 1 lot |
| 16:00.16 | [TK]D-Fender | torrancew: then you only have 1 # to dial |
| 16:00.18 | torrancew | we just transitioned this weekend |
| 16:00.20 | ManxPower-work | torrancew: then only define 1 lot |
| 16:00.21 | [TK]D-Fender | torrancew: so where is the "guessing"? |
| 16:00.26 | torrancew | well, it's 1 lot with 20 seats |
| 16:00.38 | ManxPower-work | torrancew: Now you are making no sense. |
| 16:00.49 | ManxPower-work | 1 lot is 1 lot. lots don't have seats |
| 16:00.54 | torrancew | ok, we have 1 parking lot, that reserves extens 701-720 |
| 16:01.01 | torrancew | or is that a misunderstanding on my part? |
| 16:01.06 | TSM2 | what do you want to do... |
| 16:01.12 | [TK]D-Fender | torrancew: go see the first slot thats taken then |
| 16:01.12 | ManxPower-work | torrancew: NO! You have 19 parking lots |
| 16:01.19 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 16:01.19 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:01.26 | torrancew | ManxPower-work: ok, sorry about that |
| 16:01.27 | [TK]D-Fender | ManxPower-work: No, 1.6 implements multiple lots <- |
| 16:01.48 | torrancew | i am happy with it as is |
| 16:01.54 | [TK]D-Fender | torrancew: But I've already given you the suggestion on what to do. |
| 16:01.54 | ManxPower-work | [TK]D-Fender: so do my 1.4.x scripts, but that doesn't help this guy. |
| 16:01.57 | TSM2 | so the problem is... |
| 16:02.21 | torrancew | my employer seems to want a plain old, non pbx set up, but through asterisk - he wants a pushbutton way to pick up calls on hold, or interfere with existing calls |
| 16:02.23 | ManxPower-work | TSM2: so the problem is he wants his PBX to act like a key system. |
| 16:02.29 | torrancew | exactly |
| 16:02.41 | [TK]D-Fender | torrancew: "not happening". Get over it |
| 16:02.59 | ManxPower-work | The other problem is that the original SIP specifications do not support this. |
| 16:03.00 | torrancew | thank you. i just needed to have that come from someone with more * experience |
| 16:03.09 | Qwell | If you're removing the only benefit of parking...why use it? |
| 16:03.13 | torrancew | agreed |
| 16:03.14 | TSM2 | ManxPower: ahhhhh |
| 16:03.21 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 16:03.23 | Qwell | use hold/transfer properly. |
| 16:03.25 | torrancew | again, he wasn't happy with that answer coming from me |
| 16:04.12 | *** join/#asterisk ttl- (n=patrick@94.224.78.192) |
| 16:04.44 | TSM2 | well there is a way to make it work as a key system (i think) but its a mess |
| 16:04.54 | TSM2 | so dont go there |
| 16:05.00 | Chesther | Just because you can do it, doesn't mean you should do it. |
| 16:05.23 | torrancew | thank you all |
| 16:05.54 | torrancew | last question, is there way to have asterisk email off a VM without storing it in its own VM system? |
| 16:05.58 | TSM2 | i can see he has one of thoes directors that says, "if you cant do it, find someone who can" :) |
| 16:06.00 | Qwell | delete=yes |
| 16:06.01 | *** join/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 16:06.03 | TSM2 | iknow i have one of them |
| 16:06.33 | torrancew | actually, i have a boss who just thinks "i've seen companies that can do it. so it can be done. i want it." |
| 16:06.52 | ManxPower-work | torrancew: then he should have used one of those Key Systems |
| 16:06.55 | TSM2 | same thing :) |
| 16:07.00 | torrancew | Qwell: excellent. how does it happen? |
| 16:07.03 | torrancew | TSM2: fair enough |
| 16:07.09 | torrancew | TSM2: small, 4 man company |
| 16:07.28 | TSM2 | ffs just transfer |
| 16:07.30 | TSM2 | :) |
| 16:07.53 | torrancew | ffs? |
| 16:07.55 | torrancew | ~ff |
| 16:07.56 | infobot | methinks ff is even in dutch |
| 16:07.56 | ryduh | torrancew: sorry for coming in late. what does your boss want done? |
| 16:08.23 | torrancew | ryduh: he wants asterisk to act like a key system, not a pbs |
| 16:08.24 | torrancew | pbx* |
| 16:08.35 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:953e:cce:5f9b:82e7) |
| 16:09.09 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:09.28 | torrancew | ryduh: he wants to not deal with call parking, and have traditional key system hold |
| 16:09.58 | torrancew | because he is inconvenienced by remembering and dialing 3 numbers to retrieve calls |
| 16:10.08 | ryduh | interesting. looks like a lot of work for a 4 person office |
| 16:10.29 | torrancew | well, we're expanding pretty quickly, but agreed |
| 16:10.45 | torrancew | this all started because he wants to be able to call from his cell, but have the cid look like it's the office |
| 16:10.50 | KavanS | SIP provders are TCP? |
| 16:10.52 | ryduh | torrancew: what does your company do? if you don't mind me asking |
| 16:11.02 | TSM2 | torrancew: look at DISA |
| 16:11.03 | torrancew | (we're a Mac consulting firm, and i'm their only linux guy) |
| 16:11.38 | TSM2 | or get a nokia E71, its got SIP capability |
| 16:12.06 | ryduh | very nice. once I started messing with a mac at work and realized they are built on top of unix I definitely bought one and have loved it ever since |
| 16:12.16 | Chesther | If the boss is a Mac guy, an E71 isn't going to cut it. :-/ |
| 16:12.23 | TSM2 | groan.... |
| 16:12.33 | torrancew | TSM2: he's an iphone kind of guy |
| 16:12.45 | ManxPower-work | sounds to me like you need a new boxx |
| 16:12.46 | Chesther | I nice click-to-dial app on the iphone might work. |
| 16:12.48 | ManxPower-work | and boss too |
| 16:12.55 | Chesther | s/I/A/; |
| 16:12.57 | TSM2 | DISA |
| 16:13.18 | torrancew | Chesther: that's what i'm also looking for |
| 16:13.57 | torrancew | TSM2: he actually wants me using a company iphone, but when the nokia n900 hits ground here, he can kiss off on that part |
| 16:14.08 | TSM2 | yup i want one of thoes too |
| 16:14.16 | torrancew | i've had an n800 for years, and i love it, it's just not a phone |
| 16:15.06 | torrancew | though, i'm something of a fundamentalist FOSS guy, so if it runs linux, that's a big plus in my book |
| 16:15.27 | TSM2 | i used to have win phones, until oneday i got annoyed and stopped using it, now have a simple sony one, i want a phone with good enterpise email stuff, unfortunatly my requirements puts me in Blackberry terretory, grrr |
| 16:15.57 | ryduh | TSM2: why the grrr? You don't like BB? |
| 16:16.17 | TSM2 | they are ok, but i wish they had a BES server for people with linux boxes |
| 16:16.24 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:16.28 | torrancew | ryduh: i personally hated the BB perl, back when i worked an on-call job relying on one |
| 16:16.32 | TSM2 | they are also until recently missing major stuff |
| 16:17.40 | ryduh | I've had a Storm for a while now.. and coming from an LG enV, I love it. I haven't had any experience with the Pearl. TSM2: Missing what? |
| 16:18.33 | TSM2 | up until recently there was no 3G and WiFi |
| 16:19.24 | torrancew | anyway, how can i go about deleting internal VMS so that people just have to deal with the emailed copies? |
| 16:20.20 | ryduh | I think there is an option to delete after emailing |
| 16:20.22 | TSM2 | torrancew: do you use freepbx? |
| 16:20.53 | torrancew | not atm, i didn't particularly care for the bulk it added to extensions.conf |
| 16:20.58 | TSM2 | torrancew: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf |
| 16:21.17 | TSM2 | torrancew: it may be bulk, but far easier to do everything you want to do without the hassle |
| 16:21.31 | TSM2 | talk about bulk and you only have 4 people |
| 16:23.05 | torrancew | TSM2: good point |
| 16:23.38 | torrancew | TSM2: my main qualm is this - I have a decent grasp on dialplans and the like, and can troubleshoot the system as is |
| 16:24.31 | torrancew | TSM2: FreePBX seemed to want ZAP, not DAHDI (yes, i know it's just the new incarnation, but the way I understand, you have to have ZAP/g1 or DAHDI/g1 with correct syntax |
| 16:24.47 | TSM2 | you can still do custom stuff with freepbx, just all the main chores are done for you, good thing i like is at all the major points, freepbx includes in a custom function, you can then write your own thing and ovveride fpbx |
| 16:24.58 | TSM2 | freePBX will take dahdi |
| 16:25.09 | torrancew | TSM2: which version do i need? |
| 16:25.11 | TSM2 | its still called ZAP though |
| 16:25.18 | torrancew | TSM2: ah ok |
| 16:25.28 | torrancew | TSM2: i saw that and was very concerned |
| 16:25.35 | TSM2 | why not just use a disto like elastix or PIAF etc... |
| 16:25.46 | TSM2 | all the hard work done for you :) |
| 16:26.02 | TSM2 | i would like to ground up everything myself, just dont have the time |
| 16:26.07 | torrancew | TSM2: this box is in production for some other things, and we would rather not have the downtime on those (mainly nagios for server monitoring for our clients) |
| 16:26.37 | TSM2 | should realy have it split off for the sake of voice quality |
| 16:26.42 | torrancew | TSM2: also, it didn't help that when i first looked @ freepbx, i didn't have a good grasp on asterisk's internal mechanisms |
| 16:26.47 | Qwell | torrancew: You don't have a dev or backup box? |
| 16:26.48 | torrancew | TSM2: tell that to the Man |
| 16:26.57 | torrancew | Qwell: i've been begging him for one |
| 16:27.14 | TSM2 | ok, put a artificial load on the box and let call quality suffer, then say i told you so :) |
| 16:27.34 | torrancew | TSM2: if i hold out another week or 2, we'll have enough real load on it |
| 16:27.38 | TSM2 | for 4 call system you can get by with even a old P4 box |
| 16:28.10 | torrancew | TSM2: P4, 3.0 Ghz dell box, running LAMPA stack, and a VirtualBox VM with windows server 08 on it for quickbooks |
| 16:29.29 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 16:29.30 | torrancew | as i speak, it's got about 700MB free ram |
| 16:29.34 | torrancew | out of 4GB |
| 16:29.49 | torrancew | no calls on the line, no real activity whatsoever |
| 16:29.52 | torrancew | pretty much idling |
| 16:30.08 | TSM2 | i have built ours on a dc7900 (core duo) box + Sangoma A101 + Sangoma A200 + 2port HP NIC, testing shows its fine for our load of 40 users, i have in our budget a proper server though if needed |
| 16:30.27 | torrancew | TSM2: i don't _have_ a budget |
| 16:30.40 | TSM2 | iknow but its gona sting you soon |
| 16:30.42 | Qwell | ...you're running Asterisk on a box that is running Windows in a VM? |
| 16:30.55 | torrancew | TSM2: i would love to be told "here's XXXX dollars, and these are our needs" |
| 16:30.58 | torrancew | Qwell: not my idea |
| 16:30.59 | TSM2 | as long as you have it in writing from him then you have covered ya arse |
| 16:31.03 | Qwell | get a budget. |
| 16:31.06 | Qwell | Now. |
| 16:31.42 | torrancew | Qwell: until he learns the hard way that we need one, it won't happen. he has to learn the hard way. |
| 16:31.44 | Qwell | Tell them you made a mistake allowingit. |
| 16:31.53 | Qwell | torrancew: It's your mistake. |
| 16:32.06 | TSM2 | unfortunaly i know your delema too well, much better now here |
| 16:32.12 | torrancew | TSM2: congrats |
| 16:32.55 | torrancew | Qwell: i'm not sure what mistake you're referring to. I have covered my arse every step here, but in a 4 man company, with little real corporate infrastructure, the owner makes the calls. |
| 16:33.13 | Qwell | torrancew: The mistake of allowing it to happen. |
| 16:33.35 | torrancew | Qwell: there is no "allow" on my end. there is Do What I'm Told. |
| 16:33.43 | torrancew | and stress the insanity of it |
| 16:33.58 | Qwell | There is also responsibility. It's your responsibility as an admin to "Just Say No". |
| 16:34.08 | superbeef | it's alwasy frustarting when you your forced to make something wrong anyway |
| 16:34.10 | superbeef | wrong=work |
| 16:34.11 | wcselby | Qwell - haha, and lose his job |
| 16:34.22 | torrancew | wcselby: exactly |
| 16:34.27 | Qwell | wcselby: when the system goes down, who do you think is going to be blamed? |
| 16:34.34 | TSM2 | now for my question if anyone knows, the polycom phones have a setting 'tcpIpApp.sntp.daylightSavings.enable', now UK has just changed this weekend but the phones did not change, anyone had this issue |
| 16:34.53 | Qwell | When all the data gets destroyed... Do you really think they'll just say "yeah, you're right. we should have gotten a second server. Oh well!" |
| 16:34.58 | torrancew | Qwell: that is when i show him the emails i sent long ago telling him the best way to do things |
| 16:34.59 | wcselby | Qwell - have you ever worked in a small company like that? it's always going to be torrancew fault, no matter what |
| 16:35.14 | Qwell | wcselby: of course it is. that's why you say no. |
| 16:35.26 | wcselby | Qwell - and then the owner finds someone that'll say yes |
| 16:35.30 | Qwell | If you get fired for that.. You're better off. Let's be honest here. |
| 16:35.30 | wcselby | i've worked in too many jobs like that |
| 16:35.50 | wcselby | easy to say, harder to live with |
| 16:36.26 | wcselby | for now, his best option is to "make it work", and constantly complain about getting his own server |
| 16:36.37 | torrancew | Qwell: if i had a degree, a nice savings account, etc, i would be better off, but i'm 22, no degree, just lots of closet unix experience, and a laughable amount of $ in the bank. i have to play the cards i've been dealt, know when to hold & fold, and try to advance myself |
| 16:37.00 | wcselby | complain in a way that won't get him fired, of course :) |
| 16:37.08 | torrancew | wcselby: of course |
| 16:37.31 | torrancew | also, we're hiring an accountant soon (finally), and then i may be able to get a real budget |
| 16:37.34 | wcselby | anyways, I've been in your situation torrancew - i'm glad I'm freelance now |
| 16:37.41 | torrancew | wcselby: that's my goal |
| 16:37.55 | superbeef | anybody good at troubleshooting pri-issues? My D-Channels keep ong dropping periodcally... seems to happen more with load.. some sort of timer issue, but isitll cant figure out what to try even after intense debug: http://pastebin.ca/1643911 |
| 16:38.18 | torrancew | it's actually one of the main reasons i agreed to do this asterisk project - i recognized that it's a good thing to have experience in, and something i didn't have the capital to put together myself, so it's resume' fluff |
| 16:38.26 | torrancew | though don't get me wrong, i do enjoy it |
| 16:38.42 | torrancew | that's just how i justify doing this ridiculous infrastructure for him |
| 16:40.41 | Katty | mmm lunch was good. |
| 16:41.23 | superbeef | bite sized 3muskateer microwaved on top of a dorrito? |
| 16:41.57 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 16:42.17 | Katty | no |
| 16:42.25 | Katty | salmon and spinach salad, with baked sweet potato |
| 16:43.43 | torrancew | Katty: mmm sweet potatos |
| 16:46.43 | ManxPower-work | superbeef: that sounds tasty |
| 16:46.55 | torrancew | what can i use for a 2 second timeout between sending Voicemail(xxxx) completing, and HangUp()? |
| 16:47.18 | ManxPower-work | torrancew: "core show applications" |
| 16:47.56 | ariel_ | afternoon folks |
| 16:48.21 | torrancew | ManxPower-work: that's one of those little commands that knowing a while back would have been nice |
| 16:48.26 | torrancew | ManxPower-work: thanks |
| 16:49.01 | ryduh | ManxPower-work: Ditto. thanks |
| 16:49.52 | superbeef | ManxPower-work: yeah it was a stoner snack i came up with in the glory days |
| 16:50.11 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 16:50.13 | wcselby | anyone know which versions of asterisk support cdr_tds? |
| 16:50.37 | TSM2 | is there a way for me to make it so that the number that is said when you park a call is just repeated again and again? do i need to modify the code and recompile? |
| 16:51.24 | ManxPower-work | superbeef: Any stoner that can't bake is a sad stoner indeed. |
| 16:53.51 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:53.51 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:55.46 | drcarumas | Hi guys, Hmm I'm here thinking for how long will V.1.4.X will be updated and suported by asterisk? I have to make some decisions on asterisk production implementation. Any documentations you can advice me reading? i cant find much. thanks. |
| 16:56.23 | wcselby | they talked about that at astricon's closing Q&A |
| 16:56.37 | wcselby | I think 1.4.x will be supported through like, July 2010? |
| 16:56.38 | Nivex | drcarumas: http://lists.digium.com/pipermail/asterisk-dev/2009-October/040082.html |
| 16:56.48 | drcarumas | thx! :) |
| 16:57.04 | wcselby | sorry, Dec 2010... :) |
| 16:57.23 | drcarumas | that's cool eheh |
| 16:58.46 | drcarumas | i'm still trying to define if i go to 1.6 or stick with 1.4, since 1.4 can give me all i want. And i'm thinking on stability also. 1.4 is rock solid. |
| 16:59.21 | drcarumas | Nivex, will read that thank you. |
| 16:59.22 | wcselby | Well, by Dec 2010, 1.8 should be out, which will be the next 4 year release |
| 16:59.45 | wcselby | or at least, that was the stated goal |
| 17:00.10 | drcarumas | i'm reading that now |
| 17:00.39 | Chainsaw | What I have found is that upgrading across more then a single release can be a bit painful, config-wise. (1.2 -> 1.6 was not easy) |
| 17:00.49 | drcarumas | that the version will stay in 1.6.3 and them they will release 1.8.X since they have been puting major code changes on each new 1.6v release. |
| 17:01.05 | TSM2 | im trying to make it so the park number is repeated several times, any ideas?> |
| 17:01.25 | drcarumas | Chainsaw, hmm that's something i have to consider. |
| 17:01.30 | wcselby | remember, 1.6 is not a release, it's a set of releases....i.e 1.6.0 is different from 1.6.1, and 1.6.2 will be a new release also. like the difference between 1.2 and 1.4 |
| 17:01.42 | *** join/#asterisk ecrane (n=ecrane@o1-69-19-166-10.static.o1.com) |
| 17:01.46 | *** part/#asterisk ecrane (n=ecrane@o1-69-19-166-10.static.o1.com) |
| 17:01.59 | drcarumas | hmmm k. |
| 17:02.07 | drcarumas | i'm realy confused in what's better |
| 17:02.26 | drcarumas | i have to read more to make sense of all this. |
| 17:02.27 | wcselby | so going from 1.4 to 1.8 would essentially be jumping forward by 4 releases |
| 17:02.41 | torrancew | does anyone have experience with FOP2? |
| 17:02.43 | wcselby | thus, you may want to look at 1.6.1 or 1.6.0 |
| 17:02.49 | drcarumas | wcselby, and do you think that will be a pain in the ass? |
| 17:02.50 | torrancew | The Man wants something pretty to look at |
| 17:02.51 | Chainsaw | drcarumas: If you're doing a new deployment now, I would personally go for 1.6.1(.6) |
| 17:02.51 | *** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com) |
| 17:03.02 | ryduh | If I dial my DID, I can send DTMF tones and asterisk can determine what is pressed. When I dial out from asterisk (.call file), DTMF tones aren't sent or are sent all at one time so I hear one long tone if I send 3 tones. Does this sound like problem with my provider or my setup? |
| 17:03.15 | Chainsaw | drcarumas: Just to minimise the gap you'll have to jump later. |
| 17:03.19 | *** join/#asterisk ecrane (n=ecrane@69.19.166.10) |
| 17:03.24 | drcarumas | Chainsaw, yes it's a new dep. |
| 17:03.39 | drcarumas | Chainsaw, hmm you got a good point tthere. |
| 17:03.42 | wcselby | they also said they were going to try and stop depricating stuff, so the jumps shouldn't be as painful....but there was deprication between the 1.4 -> 1.6.0 -> 1.6.1 -> 1.6.2 |
| 17:04.12 | wcselby | drcarumas - you really need to find the version that has the features you want, and then set it up in a test environment (if possible) to check the stability |
| 17:04.15 | drcarumas | i've read that 1.2 to 1.4 was painfull. i bet 1.4 to 1.6 can also be. |
| 17:04.35 | drcarumas | wcselby, i was doing that now eheh. with VM |
| 17:04.39 | ryduh | Is there a place listing the changes & deprecations made between versions? |
| 17:04.49 | Qwell | ryduh: UPGRADE.txt |
| 17:04.50 | Chainsaw | drcarumas: AgentCallbackLogin disappearance, for one. If you start on 1.6.1, you have the worst problems behind you. |
| 17:04.54 | ryduh | :) |
| 17:05.02 | Chainsaw | drcarumas: Because I can guarantee you that you'll end up using something deprecated if you start on 1.4 |
| 17:05.14 | wcselby | the UPGRADE files from the source document what's changed |
| 17:05.22 | Chainsaw | drcarumas: Be it dial plan notation or commands like AgentCallbacklogin. |
| 17:05.29 | drcarumas | Chainsaw, i got pretty much developd this application on 1.4 |
| 17:05.32 | ManxPower-work | Qwell: not JUST upgrade.txt but also UPGRADE14.txt and UPGRADE12.txt |
| 17:05.59 | ManxPower-work | If something was deprecated in 1.4 and removed in 1.6, then the info will be in the 1.4 upgrade file, not the 1.6 one. |
| 17:06.08 | drcarumas | Chainsaw, ok. Now i'm worried . I'm going to start compiling 1.6.X and give it a try with the code i've developed and see how it goes |
| 17:06.34 | Chainsaw | drcarumas: Stick with 'core set verbose 10' on 1.6 and see how it does, yes. |
| 17:06.40 | drcarumas | if all goes well 1.6.X will go do production |
| 17:06.40 | drcarumas | :) |
| 17:06.45 | Nugget | macros in 1.6.1 are flaky. just goin gfrom 1.6.0 to 1.6.1 was a pain for me |
| 17:07.12 | Nugget | I had to rip out all the macros and replace them with gosubs |
| 17:07.14 | Chainsaw | Hm, no, 1.6.0 -> 1.6.1 was a walk in the park for me. What worked on 1.6.0 kept working. |
| 17:07.43 | Nugget | (which then broke all my fancy cdr stuff) |
| 17:08.18 | drcarumas | Who boy...better get hand on the job |
| 17:09.24 | drcarumas | so, you all agree i should start with 1.6.1.6 ? |
| 17:09.28 | drcarumas | to give itr a try. |
| 17:09.42 | wcselby | it'd be a good place to start testing, yes |
| 17:09.46 | ManxPower-work | drcarumas: when does this project go into production? |
| 17:09.49 | torrancew | nobody with FOP2? |
| 17:10.04 | drcarumas | ManxPower-work, 3 or 4 months |
| 17:10.04 | wcselby | torrancew - sorry never used it |
| 17:10.18 | ryduh | nobody for my DTMF question? |
| 17:10.21 | ManxPower-work | drcarumas: I would go with the latest 1.6.1.x then |
| 17:10.48 | drcarumas | ok, thank you all for your help. :) |
| 17:11.12 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 17:13.36 | *** join/#asterisk Great_Anta_baka (n=Great_An@196.33.159.83) |
| 17:13.40 | *** join/#asterisk TimToady_ (n=moi@adsl55-109.kln.forthnet.gr) |
| 17:14.35 | torrancew | ryduh: no clue for you man |
| 17:15.17 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:15.22 | ryduh | FOP2 looks nice. I'm watching switchvox's demo video right now have you looked at that? |
| 17:15.27 | ryduh | http://www.digium.com/en/products/switchvox/ |
| 17:16.42 | ryduh | torrancew: what does he want to look at? |
| 17:17.10 | *** join/#asterisk hfb (n=hfb@pool-98-112-210-252.lsanca.dsl-w.verizon.net) |
| 17:17.29 | torrancew | ryduh: looked @ switchvox, went this way to cut costs. i'm not certain what he wants to see, and i don't rightly think he is, either |
| 17:18.07 | ryduh | lol. why not question until he backs down and just stick with *? |
| 17:18.13 | torrancew | ryduh: i was asking him how he wants his voicemail sent, and he said "until i can see something, i don't know. i'll just plug in the old system (2 line analog handsets) if i have to" |
| 17:19.12 | torrancew | for the longest time, he wanted to split the POTS line into * and the analog handsets, and wanted to be able to transfer held calls b/w the two |
| 17:21.02 | *** join/#asterisk bluregard (n=matt@OSH-248-60.onshore.com) |
| 17:21.10 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:21.34 | ryduh | torrancew: did you switch over to VoIP? |
| 17:21.36 | torrancew | alot of the questions i've asked here have mainly been going through the motions so someone with more experience officially shoots them down |
| 17:21.46 | torrancew | nope, still using Cox |
| 17:22.01 | torrancew | 4 lines on a circular hunt |
| 17:22.26 | ryduh | I'll 'officially' shoot down anything you want |
| 17:22.41 | torrancew | lol |
| 17:22.42 | torrancew | thanks |
| 17:24.46 | ryduh | shoots down FOP@ |
| 17:24.54 | ryduh | s/@/2/ |
| 17:25.51 | [TK]D-Fender | torrancew: How does switchvox "cut costs"? |
| 17:26.19 | torrancew | [TK]D-Fender: no, we went with * from scratch to cut costs |
| 17:26.24 | ryduh | [TK]D-Fender: it doesn't. It costs. He went w/ * to cut costs |
| 17:29.43 | wcselby | isn't there a switchvox lite out there right now? |
| 17:30.14 | Great_Anta_baka | only a free version |
| 17:30.19 | Great_Anta_baka | which only allows for 15 peers |
| 17:30.21 | Great_Anta_baka | kinda lame |
| 17:30.23 | wcselby | https://www.digium.com/en/forms/swvx_demo.php |
| 17:30.30 | wcselby | but he's only got 4 users |
| 17:30.39 | wcselby | so I would think he'd be good for a little while |
| 17:30.53 | Great_Anta_baka | guess so. I havent used it |
| 17:30.55 | Great_Anta_baka | so cant comment |
| 17:31.15 | wcselby | neither have i, but i"ve met a lot of the switchvox people - seems to be a pretty good solution |
| 17:31.45 | Great_Anta_baka | what about elastix? |
| 17:31.52 | [TK]D-Fender | So is an * you have absolute control over... |
| 17:32.01 | Great_Anta_baka | havent used that either but also heard its pretty good |
| 17:32.16 | wcselby | [TK]D-Fender - I agree |
| 17:32.20 | Great_Anta_baka | yeah.. prefer doing the dial plan myself |
| 17:32.25 | wcselby | [TK]D-Fender - just giving him options |
| 17:32.33 | *** part/#asterisk ManxPower-work (n=ewieling@24.42.221.26) |
| 17:33.15 | ryduh | I think he was looking for a system that his boss could "SEE" |
| 17:33.24 | [TK]D-Fender | Elastix gets very little love in #freepbx . too bogged down with extra crap and is comparable to trixbox... or so i hear |
| 17:33.30 | Great_Anta_baka | think i cut my ties with user interfaces to asterisk after working with freepbx |
| 17:33.52 | TSM2 | elastix is good, if you dont want the other stuff it just does not run |
| 17:34.07 | Kobaz | how would i make sure that music on hold always starts from the beginning |
| 17:34.29 | wcselby | Kobaz - mine just does |
| 17:34.35 | wcselby | Kobaz - what are you noticing? |
| 17:34.36 | Great_Anta_baka | is music on hold kinda like a multicast |
| 17:34.45 | Kobaz | Great_Anta_baka: it depends on the backend |
| 17:35.09 | Kobaz | wcselby: well, sometimes it doesn't start over... but it might have been another thing i was monkeying with |
| 17:35.20 | Great_Anta_baka | i saw in the new snom firmware (v8) they support multicasting |
| 17:35.46 | Kobaz | if you stream music from a program like madplay, it will broadcast the same music to everyone |
| 17:36.01 | *** join/#asterisk rene- (n=fft@200.34.66.137) |
| 17:37.47 | rene- | hello guys, what is your experience deploying asterisk behind firewalls like fortinet or sonicwall? do they give you good voice quality? i wonder if the packet per sec count goes south with all those ALG features |
| 17:38.34 | angryuser | rene-, i tryes only pfsense with traffis shaper, works perfectly |
| 17:38.45 | angryuser | *i've tryed |
| 17:38.49 | ian6 | any difference should be negligible. |
| 17:38.52 | wcselby | rene- - never used those firewalls |
| 17:40.05 | rene- | wcselby: what do u usually work with? |
| 17:40.36 | wcselby | rene- - cisco asa / pix firewalls, and also I have one client using juniper |
| 17:40.46 | wcselby | and none of them have caused any issues to really speak of |
| 17:41.00 | jplank | rene - turn off an SIP ALG |
| 17:47.01 | Great_Anta_baka | yeah i had to turn off my SIP features in my Mikrotik |
| 17:47.38 | cusco_ | hi |
| 17:49.57 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 17:50.12 | Katty | sticks nose into the puffs with vicks box and breathes deeply |
| 17:51.01 | TSM2 | im modding res_features.c so it says the park number several times, i would like to put a pause between it repeating, dont know how |
| 17:53.36 | ryduh | I'm going to try to rephrase my situation and add some new information in hopes that maybe it will trigger someone's response. I call my DID and transfer to Voip.ms' DTMF test. When I press the digits on my phone the test will tell me which keys I pressed. Now when I use the same configuration with X-Lite instead of a regular phone, the DTMF test does not respond to anything. Does anyone have any ideas how I can go about debugging this problem? I've tried w |
| 17:53.36 | ryduh | dtmf modes |
| 17:55.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:55.43 | p3nguin_ | Does DTMF actually work on your X-Lite? |
| 17:55.44 | Katty | ATTENTION. |
| 17:55.45 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 17:55.46 | Katty | I HAVE CHOCOLATE |
| 17:55.48 | Katty | that is all. |
| 17:56.18 | *** join/#asterisk ketema (n=ketema@ketema.net) |
| 17:57.15 | guax | Katty, what was that? |
| 17:57.25 | Katty | that was me overcome with pure joy. |
| 17:58.00 | Chainsaw | Chocolate. The world is complete. (For a few blissful minutes, at least) |
| 17:58.03 | guax | is a man, cant understand that behavior with chocolate, but is comparing to receive a new computer |
| 17:58.24 | Chainsaw | guax: Chocolate. Not some mars bar. |
| 17:58.32 | Katty | ugah. |
| 17:58.37 | Katty | no, real chocolate. 70% cocoa chocolate. |
| 17:58.40 | Katty | omnomnomnomnom chocolate. |
| 17:58.43 | Chainsaw | The real deal. |
| 17:58.46 | Katty | nodsnods |
| 17:59.26 | guax | ATTENTION, I HAVE COFFE. thats all |
| 17:59.30 | cusco_ | could somebody take a look at http://paste.debian.net/50007/ ?? - [Oct 26 17:33:31] WARNING[18628] pbx.c: Channel 'DAHDI/7-1' sent into invalid extension 's' in context 'FAX', but no invalid handler |
| 17:59.32 | rene- | hehehe |
| 17:59.34 | guax | oh, now i understand |
| 18:00.20 | ryduh | p3nguin_: It seems a have another problem. I don't have sound in XLite. I'll be back after I figure this out |
| 18:00.23 | jblack | hello, world |
| 18:00.33 | [TK]D-Fender | cusco_: What shows us that file is used by anything? |
| 18:00.33 | guax | cusco_, did you include it and dialplan reloaded? |
| 18:01.04 | cusco_ | ok, hold |
| 18:01.40 | TheDavidFactor | jblack, congratulations your basic program worked ;-) |
| 18:02.13 | jblack | TheDavidFactor: huh? |
| 18:02.28 | ryduh | lol @ TheDavidFactor |
| 18:02.42 | jblack | oh, heh. nope. i wrote that in ReSt |
| 18:02.43 | TheDavidFactor | "hello, world" typically used in the most basic of programming |
| 18:03.25 | [TK]D-Fender | senses great grammatical failure |
| 18:04.23 | jblack | i cant come up with a sufficiently sarcastic reply. please imagine something witty |
| 18:06.24 | [TK]D-Fender | jblack: Deference .... you have chosen ... wisely :) |
| 18:06.55 | *** join/#asterisk mutex (n=mutex@67-23-13-54.static.slicehost.net) |
| 18:06.55 | jblack | no, I have chosen exaustedly. :) |
| 18:06.58 | mutex | so I have an account with sonovoip |
| 18:07.09 | mutex | they tell me their registration is by IP only, not username/password |
| 18:07.25 | mutex | if I leve the password off of my register => command, asterisk crashes :-( |
| 18:07.26 | [TK]D-Fender | mutex: sOUNDS LIKE THERE IS NO REGISTRATION AT ALL |
| 18:07.32 | mutex | am I doint something wrong ? |
| 18:07.47 | mutex | ummm ok |
| 18:08.02 | mutex | as in, I just send my calls, no registraion ? |
| 18:08.03 | [TK]D-Fender | mutex: Don't register. Place a call. Does it come in? |
| 18:08.11 | mutex | haven't tried, but I suppose I should |
| 18:08.17 | [TK]D-Fender | mutex: Did you enter your IP on their server somewhere? |
| 18:08.23 | mutex | yes |
| 18:08.56 | [TK]D-Fender | mutex: Odds are you aren't supposed to register then |
| 18:09.02 | mutex | ok |
| 18:09.02 | [TK]D-Fender | mutex: go test a call |
| 18:09.05 | mutex | my misunderstanding |
| 18:09.58 | *** join/#asterisk garymc (n=garymc@host81-131-239-245.range81-131.btcentralplus.com) |
| 18:12.18 | *** join/#asterisk buttons840 (n=buttons8@63.230.20.246) |
| 18:12.49 | buttons840 | Am I correct in my belief that the OriginateResponse reasons are poorly documented? |
| 18:15.49 | ryduh | I have a basic Asterisk configuration and a X-Lite softphone. When I set up xlite to register with voip.ms the DTMF tones work correctly but when I register with my asterisk server, I don't hear any sound. Where should I start to debug> |
| 18:15.59 | buttons840 | one phone returns a originateresponse, failure, reason 0, and another returns reason 3, both are just ring no answers |
| 18:16.33 | ryduh | * sees the call and bridges it: Packet2Packet bridging SIP/100-00dbb580 and SIP/voipms-00dc1320 |
| 18:16.40 | [TK]D-Fender | ryduh: There are 2 things involved. X-Lite to *. then * to voip.ms. |
| 18:16.47 | [TK]D-Fender | ryduh: Test them INDEPENDENTLY |
| 18:21.13 | *** join/#asterisk QaDeS (n=mklaus@p4FC72DC4.dip0.t-ipconnect.de) |
| 18:21.33 | *** join/#asterisk ManxPower-work (n=ewieling@24.42.221.26) |
| 18:21.44 | ManxPower-work | GAWDDAMNEDCUSTOMER! |
| 18:21.48 | ManxPower-work | S! |
| 18:23.15 | p3nguin_ | ryduh: Do you expect registration to make sounds? |
| 18:23.46 | ryduh | p3nguin_: no? I should hear the answering machine on destination |
| 18:24.37 | p3nguin_ | ryduh: Are you able to place calls using xlite? |
| 18:24.46 | cusco_ | hi, sorry guax [TK]D-Fender, I was away. |
| 18:24.55 | cusco_ | Yea so i included it, it didn't seem to be included |
| 18:25.09 | ryduh | That is what I'm trying to test right now.. and not succeeding in. |
| 18:25.57 | cusco_ | after an ael reload: http://paste.debian.net/50015/ |
| 18:26.04 | p3nguin_ | ryduh: Can you receive calls on xlite? |
| 18:27.48 | cusco_ | what would be: [Oct 26 18:23:00] WARNING[11415] app_fax.c: WARNING T.30 Cannot open source TIFF file '/tmp/fax/27837461.output.tif' |
| 18:30.37 | ryduh | p3nguin_: I can call in from a landline and get Xlite to ring. I accept the answer but do not hear anything. |
| 18:30.44 | ManxPower-work | registration has NOTHING with being able to place calls from a phone to Asterisk |
| 18:31.01 | ManxPower-work | ryduh: disallow=all and allow=ulaw never ever allow=all |
| 18:31.50 | ryduh | ManxPower-work: I have both of those set in all 3 of my sip peers |
| 18:32.48 | p3nguin_ | ryduh: Do you hear audio even in one direction? |
| 18:33.06 | ryduh | p3nguin_: nope. no audio |
| 18:34.23 | *** join/#asterisk Skeeter- (i=Skeeter-@216.218.2.65) |
| 18:34.44 | Skeeter- | anyone successfully got astassistant to work??? |
| 18:36.11 | cusco_ | doesn't even knows what that is |
| 18:37.36 | Skeeter- | ASTassistant is a FOP |
| 18:37.36 | Skeeter- | software |
| 18:38.34 | ryduh | p3nguin_: I'm going to try a different softphone |
| 18:39.10 | ManxPower-work | ry |
| 18:39.23 | ManxPower-work | ryduh: is there NAT or firewall involved on the server side? |
| 18:39.46 | p3nguin_ | ryduh: zoiper usually Just Works. |
| 18:40.07 | ryduh | ManxPower-work: yes there is. let me check what I have setup with iptables |
| 18:40.14 | ryduh | p3nguin_: is that for mac? |
| 18:40.20 | ManxPower-work | ryduh: no, check the NAT server doc |
| 18:40.22 | ManxPower-work | ~answers |
| 18:40.23 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 18:40.31 | p3nguin_ | ryduh: Windows, Mac, and Linux. |
| 18:40.41 | ManxPower-work | you need the ports forwarded as well as localnet and externip set or you should not expect to get audio |
| 18:41.35 | p3nguin_ | Oh, you didn't forward the RTP ports? |
| 18:41.46 | ManxPower-work | make sure you have any NAT features disabled in the client and make sure you don't forward ports on the client router |
| 18:41.57 | *** join/#asterisk hugorebelo (n=hugorebe@200.171.132.124) |
| 18:42.12 | buttons840 | ^ about OriginateReponse events, is there any reliability to the reasons, as they seem random, and I can't determin what 0, 1, and 3 means, which are the reasons i am investigating. |
| 18:42.12 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 18:42.43 | buttons840 | i can't find any reliable documentation on this |
| 18:46.06 | *** join/#asterisk coppice (n=chatzill@host86-132-15-123.range86-132.btcentralplus.com) |
| 18:47.49 | *** join/#asterisk ecret (n=ecret@99.246.116.189) |
| 18:50.26 | ryduh | ManxPower-work: Let me explain the situation real quick. I have a server with a static IP. The client is behind a firewall which I know nothing about and cannot configure. |
| 18:52.27 | [TK]D-Fender | ryduh: Go follow the guide |
| 18:52.56 | ryduh | walks away with tail between legs. |
| 18:57.06 | garymc | [TK]D-Fender : My polycom is setup and registers now, why is there a BIG delay before both parties hear any sound? Would that be a NAT issue only? |
| 18:58.02 | hardwire | ryduh: aww |
| 18:58.11 | hardwire | also.. eww. |
| 19:04.51 | *** join/#asterisk GameGamer43 (n=GameGame@69.129.142.83) |
| 19:11.22 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:21.02 | rene- | buttons840: use the source little padawan |
| 19:21.32 | rene- | buttons840: have you used adhearsion? |
| 19:23.03 | *** join/#asterisk securevoip (n=securevo@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 19:27.15 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 19:28.40 | *** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 19:30.56 | IBC_jkenney | Hello i am stuck i have a problem i updated to 1.6.6 on fri evening because i was having problems with ram usage and thought it was asterisk. We run 64 Bit Ubuntu with 8 gigs of ram with 8 total processor cores. I am now getting a error in my kern log stating this warning: `asterisk' uses 32-bit capabilities (legacy support in use) |
| 19:31.18 | IBC_jkenney | Can anyone tell me why? |
| 19:31.47 | jblack | sounds like you built asterisk as 32 bit, rather than 64. |
| 19:31.59 | jblack | but I'm guessing, tbh |
| 19:33.08 | IBC_jkenney | how would it build 32 automatically on a 64 bit OS |
| 19:34.30 | jblack | rephrase the question, please |
| 19:35.19 | IBC_jkenney | why would configure compile a 32bit revision of asterisk on a native 64 bit os |
| 19:35.51 | jblack | at any rate, you can check by running: file `which asterisk` |
| 19:36.31 | wcselby | IBC_jkenney - did you build from source or install from apt-get? |
| 19:36.52 | jblack | It's that "would" word. If you mean "could", by various variables that could be in your environment, by having a 32 libraries while lacking 64 bit libraries, etc. |
| 19:37.07 | IBC_jkenney | source |
| 19:37.14 | jblack | It's not hard at all to convince gcc to target 32 bit. |
| 19:37.20 | jblack | what did file tell you? |
| 19:37.49 | wcselby | http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=492620 |
| 19:38.10 | IBC_jkenney | /usr/sbin/asterisk: ELF 64-bit LSB executable, x86-64, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.15, not stripped |
| 19:38.50 | wcselby | I think it was built with a 32 bit lib somewhere? |
| 19:39.00 | jblack | wcselby probably found it for you, but how about ldd `which asterisk`, and look for 32 bit libraries. |
| 19:39.31 | leifmadsen | likely you have a 32-bit library installed instead of just the 64-bit versions -- this happens on CentOS/RHEL as well when you don't specifically say, "yum install openssl-devel.x86_64" |
| 19:40.09 | leifmadsen | after installing CentOS on 64-bit hardware, you should run something like, "yum remove *.i386" |
| 19:40.17 | leifmadsen | now, whether debian has this same issue, I'm not sure |
| 19:40.27 | jblack | apparently it does. |
| 19:40.49 | jblack | no doubt he's filing all the libraries * is linked to on his system as we speak, and probably will come up with libpcap. |
| 19:41.45 | jblack | pardon, libcap. |
| 19:44.50 | *** join/#asterisk ecret (n=ecret@99.246.116.189) |
| 19:44.59 | *** join/#asterisk fainsys (n=fainsys@74.7.247.152) |
| 19:46.11 | *** join/#asterisk WHYS (n=drumm@137.28.94.209) |
| 19:49.26 | garymc | So anyone know why im getting big length of silence when I call the office from home with my remotley connected Polycom phone? |
| 19:49.45 | ManxPower-work | garymc: try canreinvite=off |
| 19:50.02 | ManxPower-work | ,,er,, canreinvite=no (chances are "off" will work, but it'll just look weird) |
| 19:50.21 | garymc | where is that setting in the polycom or extension settings ? |
| 19:50.44 | *** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net) |
| 19:51.08 | TheDavidFactor | Is there a place to ask about why things are done the way they are in asterisk? specifically I have a question about how ast_gethostbyname works |
| 19:51.16 | ManxPower-work | garymc: is the call coming in on SIP or PSTN? |
| 19:51.22 | garymc | SIP |
| 19:51.28 | garymc | reinvite is set to NO |
| 19:51.29 | ManxPower-work | TheDavidFactor: #asterisk-dev |
| 19:51.42 | ManxPower-work | garymc: you have it set to know in the sip.conf entry for your ITSP? |
| 19:51.43 | IBC_jkenney | sorry gues i had to walk away from my desk |
| 19:51.48 | ManxPower-work | know? "no", of course. |
| 19:52.00 | garymc | NAT is set to yes |
| 19:52.14 | garymc | Not sure on that one Manx |
| 19:52.21 | ManxPower-work | garymc: is your asterisk server behind nat or firewall? |
| 19:52.28 | garymc | firewall |
| 19:52.43 | ManxPower-work | garymc: what ports do you have allowed? |
| 19:52.48 | garymc | 5060 |
| 19:52.53 | ManxPower-work | that's all? |
| 19:52.55 | IBC_jkenney | http://pastebin.com/ |
| 19:52.55 | garymc | 10000-20000 |
| 19:53.00 | ManxPower-work | all UDP? |
| 19:53.09 | IBC_jkenney | plus i have a problem where ram is being eaten up on that box |
| 19:53.12 | IBC_jkenney | but one at a time |
| 19:53.14 | IBC_jkenney | :) |
| 19:53.15 | garymc | yes |
| 19:53.44 | ManxPower-work | garymc: Asterisk is on a public IP? |
| 19:53.50 | IBC_jkenney | http://pastebin.com/d1cba6aed |
| 19:53.52 | IBC_jkenney | sorry |
| 19:54.05 | garymc | its a static IP address |
| 19:54.14 | ManxPower-work | that was not my question. |
| 19:54.56 | garymc | .......... like a ip address I can access from anywhere |
| 19:55.10 | p3nguin_ | You're getting closer. |
| 19:55.22 | ryduh | static doesn't change. public is about access |
| 19:55.41 | garymc | well i can access it with a password yes |
| 19:55.46 | p3nguin_ | Getting colder. |
| 19:55.59 | ryduh | could I ping the IP if I wanted to? |
| 19:56.04 | p3nguin_ | IP doesn't have passwords. |
| 19:56.04 | ManxPower-work | Half the time people "without NAT but with a firewall" really have NAT going on. |
| 19:56.23 | garymc | ok |
| 19:56.30 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 19:56.33 | Nugget | heh |
| 19:56.35 | garymc | you could ping the IP if you wanted too |
| 19:56.40 | p3nguin_ | Let's make this easier. What is the IP address of the server? |
| 19:56.53 | garymc | 81.134.0.102 |
| 19:57.03 | p3nguin_ | yay! It's public! |
| 19:57.10 | Nugget | if you have the password! :) |
| 19:57.10 | garymc | yes |
| 19:57.16 | garymc | ;) |
| 19:57.28 | wcselby | is that the IP address on the server itself? or it's the IP address of your firewall? |
| 19:57.42 | garymc | id say the firewall |
| 19:57.45 | wcselby | hell, not even really relevant right now |
| 19:57.47 | garymc | well the router |
| 19:57.47 | Nugget | then it's not public. |
| 19:57.54 | Nugget | i.e. there's NAT |
| 19:57.57 | wcselby | you're having a long pause before audio is heard on a connected call? |
| 19:58.03 | garymc | yes |
| 19:58.14 | wcselby | how do you know the call is actually connected? |
| 19:58.18 | ManxPower-work | garymc: do an "ifconfig" on the server and then pastebin the results |
| 19:58.31 | wcselby | i.e can you show us the SIP debug and the CLI output of a recently connected call? |
| 19:58.34 | garymc | it is connected |
| 19:58.42 | wcselby | and how long is the pause? |
| 19:58.44 | *** part/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 19:58.51 | *** join/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 19:59.02 | garymc | about 10 seconds |
| 19:59.15 | garymc | testing now, |
| 19:59.27 | wcselby | this is only with your polycom at your house? i.e not with a softphone from somewhere else? |
| 19:59.43 | garymc | seems to be |
| 19:59.51 | wcselby | how aobut a softphone from your house? |
| 20:00.39 | [TK]D-Fender | [15:57]<Nugget>i.e. there's NAT <- not necessarily... but his description IS worth rat-shit |
| 20:01.50 | IBC_jkenney | would the 32bit library's cause my memory hole issue |
| 20:02.00 | garymc | Rat shit is worth £100 a ton! |
| 20:02.58 | garymc | I know that cos I used to work for a farmer and rat shit ended up in the grain in the silos. I asked him how he got the rat shit out before he sold it on, hes said "I dont Rat Shit's Worth £100 per ton" ;) |
| 20:03.10 | p3nguin_ | res_musiconhold.c:555 monmp3thread: Request to schedule in the past?!?! What exactly does this mean? |
| 20:04.40 | ManxPower-work | p3nguin_: means your computer is slow |
| 20:05.10 | p3nguin_ | Seriously? |
| 20:05.19 | ManxPower-work | p3nguin_: yes. |
| 20:05.37 | p3nguin_ | That is outputted on the console when the MoH stream has a song change. |
| 20:05.39 | jblack | Are you running * on xen? |
| 20:05.41 | ManxPower-work | It's a harmless message. Might cause minor audio artifacts, but that's about it. |
| 20:06.01 | ManxPower-work | p3nguin_: mp3 decoding takes large amounts of CPU, that's why 1.4+ doesn't use it by default. |
| 20:06.10 | p3nguin_ | I'm running it on a PIII 933MHz / 512MB box. |
| 20:06.16 | ManxPower-work | yup, that would do it. |
| 20:06.23 | p3nguin_ | I'm using an ogg stream. |
| 20:06.29 | ManxPower-work | AND you're STREAMING? |
| 20:06.32 | jblack | Yeah, that's a slow box. |
| 20:06.38 | ManxPower-work | I wish you the best of luck, p3nguin_ |
| 20:07.02 | jblack | My netbook is faster than that |
| 20:07.49 | ManxPower-work | jblack: it's not too slow as long as you don't try too much transcoding (and playing MoH OGG stream IS transcoding) |
| 20:07.57 | p3nguin_ | ogg123 uses no more than 3% CPU and sox no more than 8% CPU. |
| 20:08.55 | jblack | sure, if he sticks to passthrough, and keeps the timing light, he'd be fine |
| 20:08.59 | *** join/#asterisk ibercom (i=96d6cd10@gateway/web/freenode/x-fqflaqrjgxidotyk) |
| 20:09.31 | ManxPower-work | jblack: GSM and G726 transcoding should work. They don't use up too much CPU. |
| 20:09.42 | jblack | Ok. |
| 20:10.02 | jblack | I'd thnk he'd have trouble transcoding more than 1 or 2 channels at a time (i.e. one call) |
| 20:10.03 | garymc | OK heres my SIP debug http://pastebin.ca/1644196 |
| 20:10.21 | garymc | Vary length of paused sound between calls |
| 20:10.24 | p3nguin_ | What's the absolute best quality codec, even if it uses the most bandwidth? |
| 20:10.39 | ibercom | hello, what means this warning chan_dahdi.c: Unable to disable echo cancellation on channel 2: Invalid argument |
| 20:10.48 | jblack | ulaw is lossless, I believe. |
| 20:10.56 | jblack | but I'll double check |
| 20:10.59 | ManxPower-work | To: <sip:200@192.168.1.68;user=phone> |
| 20:11.14 | ManxPower-work | looks like you don't have nat=yes in [200] in sip.conf |
| 20:11.34 | p3nguin_ | I use ulaw with no problems. I should switch to gsm and see if that message goes away during streaming. |
| 20:11.36 | jblack | p3nguin_: ulaw. |
| 20:11.53 | ManxPower-work | p3nguin_: I don't see how that is going to help at all. |
| 20:12.06 | ManxPower-work | unless you are going to receive a GSM stream |
| 20:12.17 | torrancew | Has anyone used fring? Or any iPhone SIP client, for that matter. |
| 20:12.19 | jblack | p3nguin_: If you have gobs of space, convert your oggs to wav. |
| 20:12.26 | torrancew | with asterisk, of course |
| 20:12.29 | p3nguin_ | Oh, I see. |
| 20:12.37 | jblack | I think the book recommends that for older machines. |
| 20:12.53 | p3nguin_ | I think I am already converting the oggs to wav and then playing back with sox. |
| 20:12.55 | *** join/#asterisk securevoip (n=securevo@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 20:13.13 | jblack | if they're in wav, then no need to use sox at all. |
| 20:13.14 | ManxPower-work | First he says he's streaming, now he's saying he batch converts. |
| 20:13.35 | p3nguin_ | /usr/bin/ogg123 -q -b 128 -p 32 -d wav -f - http://mystream/ |
| 20:13.37 | jblack | maybe he's streaming wavs. :) |
| 20:13.48 | p3nguin_ | /usr/bin/sox -r 44000 -t wav - -r 8000 -c 1 -t raw - vol 0.12 |
| 20:14.03 | p3nguin_ | I assure you it's a stream. |
| 20:14.31 | ManxPower-work | That's too scary even for me. |
| 20:14.31 | jblack | Ok, so, when I said convert, I meant "save the oggs as wav format, and use the builtin MOH to play the WAVs" |
| 20:14.36 | *** part/#asterisk ManxPower-work (n=ewieling@24.42.221.26) |
| 20:15.39 | jblack | Good going. You killed manxpower. |
| 20:15.48 | p3nguin_ | oops |
| 20:15.50 | garymc | ManxPower-work Nat is set to YES in ext 200 |
| 20:16.20 | p3nguin_ | I didn't see my solution as a problem. It works very well, except it outputs that one notice message when the song changes. |
| 20:16.36 | jblack | Ok, if it's not a problem, it's not a problem. |
| 20:16.44 | jblack | That's your call. :) |
| 20:16.49 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:17.08 | jblack | You're not asking "How do I get rid of this warning without changing anything", are you? |
| 20:17.18 | p3nguin_ | I'm always open to suggestions. If there's a better way to stream this ogg stream and play it for MoH, I'm willing to consider it. |
| 20:17.46 | jblack | batch convert them before hand to wav, and play the ways, is the standard way for cpu starved boxes. |
| 20:17.59 | jblack | play the wavs, that is. |
| 20:18.23 | p3nguin_ | Right now, I am outputting the wavs to stdout, then using sox to play from stdin. |
| 20:18.29 | jblack | If you're happy enough with the way things are and can tolerate the warning, you can keep things as they are. |
| 20:18.39 | garymc | anyone wanna tes an extension from there end for me, I can then know if its my Routers ports messing up |
| 20:18.49 | torrancew | garymc: sure |
| 20:18.59 | p3nguin_ | I wouldn't want to fill up the hard drive with wav audio that I have strum into my box. |
| 20:19.07 | jblack | p3nguin_: You don't have enough cpu to transcode codecs, handle calls, _and_ convert ogg to wav format at the same time. |
| 20:19.51 | p3nguin_ | jblack: And yet I do it. Odd, huh? |
| 20:20.02 | jblack | that warning is saying you're close enough to the line that it can tell. It's up to you to decide if the problem is bad enough that it shows in meatspace. If not, awesome. If so, either change the work done, or change the thing doing the work. |
| 20:20.33 | jblack | p3nguin_: I don't think so, no. You're just on the line, is all. |
| 20:21.55 | p3nguin_ | jblack: But on the line of what? I'm continuously monitoring CPU usage and load average, and it never goes up to anything even similar to a heavy load. |
| 20:22.31 | p3nguin_ | I've never had like 100 calls, all on hold with streaming music playing, but you get the idea. |
| 20:22.34 | jblack | On the line that your ogg stream isn't keeping up with asterisk, which is trying to do musiconhold. |
| 20:22.34 | jblack | Are you just fucking with me? |
| 20:22.53 | jblack | If you're happy, leave it. Problem solve.d |
| 20:23.05 | wcselby | torrancew - I use iSip and really like it |
| 20:23.11 | wcselby | had it setup in like 5 minutes from the time I downloaded it |
| 20:23.49 | *** join/#asterisk ska (n=ska@cpe-70-124-88-58.austin.res.rr.com) |
| 20:23.56 | ska | Anhyone use askozia here? |
| 20:24.29 | ska | My inbound calls wont forward to the phone. I get "extension not found" errors. |
| 20:24.58 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 20:25.51 | p3nguin_ | Maybe you don't have that extension. Show a debug from a call where there is a problem. |
| 20:27.53 | p3nguin_ | Oh, that notice doesn't even appear with every song change. Maybe it's not really my issue at all, but is a bump in the stream. |
| 20:28.20 | jblack | sigh |
| 20:30.13 | [TK]D-Fender | checkout time, BBL |
| 20:30.22 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:38.28 | *** part/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:38.30 | Kobaz | sdfs,dmf,samdfns,adf |
| 20:39.01 | Kobaz | any time i do anything new, asterisk is crashing |
| 20:39.31 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 20:39.31 | *** mode/#asterisk [+o denon] by ChanServ |
| 20:42.25 | garymc | anyone got time to test a extension for me? |
| 20:42.38 | garymc | im getting big delays, with my Phone connected remotley |
| 20:42.49 | garymc | need to know its my NAT not Office NAT |
| 20:43.28 | Kobaz | asterisk is causing a kernel panic |
| 20:45.49 | jblack | maybe your dns is horked up. |
| 20:46.13 | jblack | it's just the start of the call? |
| 20:46.13 | Kobaz | me? |
| 20:49.57 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:50.15 | *** join/#asterisk diatonic1 (n=chillman@mail.clearwater-research.com) |
| 20:50.31 | p3nguin_ | garymc: What do you need to be done? |
| 20:50.45 | *** join/#asterisk Micc_ (n=dotirc@c-98-225-59-171.hsd1.wa.comcast.net) |
| 20:51.13 | torrancew | does anyone use the gizmo client with an asterisk box? |
| 20:53.26 | *** join/#asterisk mltlnx (n=mltlnx@66.117.118.248) |
| 20:54.42 | *** join/#asterisk killfill (n=killfill@200.63.96.244) |
| 20:54.46 | killfill | hi |
| 20:55.03 | killfill | if i park (i.e. transfer to 700) |
| 20:55.07 | killfill | how do i pick the call up? |
| 20:55.26 | Kobaz | killfill: look at your dialplan |
| 20:55.27 | riddlebox | dial whatever parking slot it tells you, 701 |
| 20:55.33 | riddlebox | ... |
| 20:55.35 | outtolunc | torrancew, yes people have, note.. it does go through gizmo's servers, even if the end point is back on your own network (or at least used too) |
| 20:55.53 | *** part/#asterisk securevoip (n=securevo@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 20:55.59 | killfill | Kobaz: well i should have asked "how to i enable picking up parked calls from dialplan"?.. :) |
| 20:57.18 | p3nguin_ | The dialplan shouldn't have extens for parking. |
| 20:57.34 | Kobaz | killfill: you need to include parkedcalls from the context you want to pick up the calls using |
| 20:57.43 | p3nguin_ | When you transfer to 700, you should be told which slot it was put in. |
| 20:58.03 | Kobaz | or you can do something custom, and do a goto(parkedcalls, exten, 1) |
| 20:58.22 | torrancew | outtolunc: basic question/desire is that we be able to route any calls possible through our asterisk box as sip |
| 20:58.32 | torrancew | outtolunc: can gizmo do that? |
| 20:58.39 | killfill | hm.. and its in features.conf.. |
| 20:59.41 | outtolunc | torrancew, there are (where) configurations for setting up your asterisk box with your gizmo creds.. |
| 20:59.41 | p3nguin_ | killfill: parkext and parkpos. |
| 21:00.08 | torrancew | outtolunc: not even a remote desire for us |
| 21:01.12 | torrancew | The Man likes dialectic |
| 21:01.13 | killfill | -- Added extension '701' priority 1 to parkedcalls |
| 21:01.24 | torrancew | i don't like opening 5038 to the outside world |
| 21:01.31 | killfill | Executing [701@numberplan-custom-1:1] Playback("SIP/31-087c5000", "cannot-complete-as-dialed") :( |
| 21:01.40 | p3nguin_ | killfill: For example, if you have parkext set to 700, you transfer calls to 700 to park them. When you do that, that parking attendant will tell you which slot it has been parked in, as configured in parkpos. If it says you are parking it in 704, you would then dial 704 to pick it up. |
| 21:02.06 | killfill | hm.. |
| 21:02.18 | outtolunc | torrancew: 5038 is for the manager, it has nothing to do with sip |
| 21:02.21 | p3nguin_ | killfill: Do not add parking settings into extensions.conf. |
| 21:02.25 | torrancew | right |
| 21:02.34 | torrancew | outtolunc: dialectic needs to access the management console |
| 21:03.28 | *** join/#asterisk voxter (n=hardcore@76.77.91.250) |
| 21:04.00 | killfill | how do i tell, 701 is a parked extension?.. i.e. exten = _700,1,WHatApp? |
| 21:04.19 | p3nguin_ | (1602.20) <p3nguin_> killfill: Do not add parking settings into extensions.conf. |
| 21:04.24 | p3nguin_ | killfill: this means NONE. |
| 21:04.53 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 21:05.25 | p3nguin_ | killfill: I'll go over it one more time. Configure the parking extension and the parking positions in features.conf. |
| 21:05.25 | Kobaz | p3nguin_: i wouldn't totally agree with that... it depends on how you want parking to work |
| 21:05.25 | *** join/#asterisk mltlnx_ (n=mltlnx@66.117.120.3) |
| 21:05.37 | p3nguin_ | kobaz: Why? Use it the way it was built to work and go on with something else. |
| 21:05.38 | killfill | ahá |
| 21:05.40 | Kobaz | for simplicity, yes... configure everything in features... and include parkedcalls |
| 21:05.44 | killfill | got it thanks! |
| 21:06.02 | Kobaz | p3nguin_: it's up to the user how they want to use it |
| 21:06.30 | ryduh | ~answers |
| 21:06.31 | infobot | extra, extra, read all about it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 21:06.31 | Kobaz | p3nguin_: i actually do have a specific range of extensions confgured for call park pickup, since i run some other code before doing a pickup |
| 21:09.18 | Katty | :> |
| 21:09.23 | Katty | allos. |
| 21:10.45 | ibercom | Why I get this warning "chan_dahdi.c: Unable to disable echo cancellation on channel 2: Invalid argument" when the primary channel go down ? |
| 21:10.48 | Kobaz | kernel panics, asterisk crashes, the usual |
| 21:13.14 | Katty | kernel panics make me all sad inside. |
| 21:13.45 | Kobaz | aye |
| 21:13.51 | Kobaz | especially when they are on a production box |
| 21:17.07 | jblack | katty: That's because a baby penguin dies every time a kernel panics |
| 21:17.15 | torrancew | anyone help debug this: http://pastebin.ca/1644307 ? |
| 21:18.25 | torrancew | using dialectic, trying to make a phone call, that was the output |
| 21:18.32 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 21:18.46 | Kobaz | fourth time's a charm? |
| 21:18.52 | Kobaz | i downgraded the kernel, and wanpipe drivers |
| 21:22.34 | ibercom | leifmadsen: this is the problem, i think that the config is good. |
| 21:23.09 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-yugvtsxuqpycuunt) |
| 21:23.49 | *** part/#asterisk fish-bulb (n=cstewart@nat/digium/x-yugvtsxuqpycuunt) |
| 21:24.00 | Kobaz | it was crazy... it's like as soon as the pri had four-five calls, bam, kernel panic |
| 21:24.38 | *** part/#asterisk cyberfab007 (n=cyberfab@CPE001b11cf4f69-CM0014f85c3ada.cpe.net.cable.rogers.com) |
| 21:24.49 | *** join/#asterisk xmitter (n=xmitter@c-24-20-192-165.hsd1.or.comcast.net) |
| 21:25.53 | xmitter | howdy. Anyone have an easy walkthrough to get asterisk up and running under a VPS? |
| 21:26.29 | p3nguin_ | It's the same as if it weren't on a VPS, I'm quite sure. |
| 21:26.35 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:26.39 | torrancew | xmitter: depends on what you're doing, and the VPS shouldn't affect anything |
| 21:26.56 | xmitter | hmm ok, I tried running through the FreePBX install and kept running into problems |
| 21:27.17 | p3nguin_ | That's to be expected. |
| 21:27.22 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 21:27.26 | p3nguin_ | Try Asterisk by itself. |
| 21:27.29 | Kobaz | 285 calls processed |
| 21:27.32 | Kobaz | so far so good |
| 21:27.42 | manxpower | Can anyone recommend a good, reliable source for cheap VoIP stuff like ATAs? |
| 21:27.51 | xmitter | OK yeah I'll start over and just do a base astersik install then. Thanks |
| 21:28.01 | torrancew | xmitter: freepbx nearly drove me up the wall |
| 21:28.16 | Kobaz | manxpower: voipsupply, variant distribution |
| 21:28.18 | [TK]D-Fender | and "running into problems" says nothing at all |
| 21:28.24 | jblack | I've been happy with treatment at voipsupply. |
| 21:28.25 | torrancew | again, can anyone help debug these issues? they're manager related |
| 21:28.26 | torrancew | http://pastebin.ca/1644307 |
| 21:28.38 | Kobaz | i wish i could find a stable asterisk setup |
| 21:28.43 | jblack | but i haven't ordered from them in a year. |
| 21:29.04 | Kobaz | [TK]D-Fender: what kernel do you use for axeterisk? |
| 21:29.08 | outtolunc | torrancew: it says no such host |
| 21:29.18 | [TK]D-Fender | Kobaz: Linux |
| 21:29.29 | Kobaz | kernel version? |
| 21:29.50 | torrancew | outtolunc: oddly enough, it's not trying to find a host |
| 21:29.53 | torrancew | should it be dialed as sip? |
| 21:29.54 | [TK]D-Fender | Kobaz: relatively modern 2.6 |
| 21:30.00 | Kobaz | anything in particular? |
| 21:30.01 | torrancew | it was a POTS number being dials |
| 21:30.04 | torrancew | dialed* |
| 21:30.14 | Kobaz | i'm on 2.6.27.38 now |
| 21:31.06 | Kobaz | 2.6.30 was panicing like nuts |
| 21:31.32 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-ooqovjgbqdziaola) |
| 21:32.33 | Nugget | I use 7.2 :) |
| 21:32.53 | Nugget | man uname |
| 21:33.55 | Kobaz | kernel version, damnit |
| 21:34.02 | Kobaz | he |
| 21:34.03 | Kobaz | heh |
| 21:34.03 | Nugget | yes. |
| 21:34.09 | Nugget | 7.2 |
| 21:34.11 | TSM | uname -a comon |
| 21:34.18 | Nugget | s nugget@carrera:~>uname -r |
| 21:34.18 | Nugget | 7.2-STABLE |
| 21:34.39 | ryduh | that's not a linux kernel version |
| 21:34.45 | Nugget | that's correct |
| 21:34.48 | TSM | that is FreeBSD |
| 21:34.50 | [TK]D-Fender | ryduh: DUH |
| 21:34.56 | [TK]D-Fender | :p |
| 21:35.02 | Kobaz | ryduh: i know |
| 21:35.32 | p3nguin_ | nugget: uname -irs |
| 21:36.09 | ryduh | whoops |
| 21:36.26 | p3nguin_ | kobaz: What did you run to show calls processed? |
| 21:37.27 | torrancew | mac friendly, asterisk compatible softphone, anyone? |
| 21:37.39 | Kobaz | p3nguin_: core show channels |
| 21:37.39 | ryduh | Are the only ways to send DTMF tones through SendDTMF and the D options on Dial() ? |
| 21:37.47 | Qwell | torrancew: zoiper |
| 21:37.55 | Kobaz | p3nguin_: it's new in 1.6.x |
| 21:38.10 | p3nguin_ | oh |
| 21:38.28 | manxpower | Thanks everyone! |
| 21:38.30 | TSM | torrancew: fring too |
| 21:38.30 | torrancew | i wish ekiga had a mac version easy to deploy to these people |
| 21:38.34 | torrancew | it KISS |
| 21:38.35 | p3nguin_ | I thought I saw a count of calls processed on 1.4 at one time, but I cannot figure out what command I would have used to show it. |
| 21:43.16 | Kobaz | dunno |
| 21:44.27 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 21:47.52 | *** join/#asterisk Chodorenko (n=chodoren@ext.one.by) |
| 21:49.58 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) |
| 21:54.05 | manxpower | show channels |
| 21:54.26 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 21:56.49 | Kobaz | 1050 calls processed |
| 21:56.52 | Kobaz | mmm |
| 21:56.55 | Kobaz | good so far |
| 22:04.13 | cusco_ | 5920 calls processed |
| 22:04.43 | cusco_ | woa.. |
| 22:04.50 | cusco_ | asterisk was printings loads of stuff... |
| 22:04.56 | cusco_ | was pressing tab all the time |
| 22:05.07 | cusco_ | and then I just kept my finger down... |
| 22:05.20 | cusco_ | perfpbxr*CLI> core show |
| 22:05.20 | cusco_ | Segmentation fault |
| 22:05.42 | ryduh | that doesn't sound good |
| 22:06.01 | outtolunc | 'core show Segmentataion fault' is not a valid command <G> |
| 22:07.05 | Kobaz | heh |
| 22:07.17 | Kobaz | cusco_: hopefully not a production system? |
| 22:07.24 | cusco_ | in a production system! |
| 22:07.31 | cusco_ | it was not asterisk that went down |
| 22:07.36 | Kobaz | got the backtrace? |
| 22:07.37 | Kobaz | oh |
| 22:07.38 | cusco_ | only the clint attached to the daemon |
| 22:07.40 | Kobaz | just the console client? |
| 22:07.42 | Kobaz | oh |
| 22:07.42 | cusco_ | client |
| 22:07.59 | cusco_ | so, its ok. I got scared it was the real asterisk service.. heh |
| 22:08.12 | cusco_ | where does it dump it to? |
| 22:08.39 | outtolunc | wherever it was started from, or /tmp (if you have ulimit set high enough for it to core) |
| 22:08.42 | cusco_ | shouldn't it be at $PWD |
| 22:08.44 | cusco_ | ah |
| 22:09.04 | cusco_ | init.d/asterisk starts it, dunno where from |
| 22:09.16 | Kobaz | so...i was asked before from katty "what's up" |
| 22:09.31 | Kobaz | and i said... you know.. the usual... kernel panics, asterisk crashes |
| 22:09.41 | Kobaz | cusco_: we're all in the same boat it seems |
| 22:10.23 | cusco_ | hm... yes |
| 22:10.37 | cusco_ | calls dropping with some dahdi "reject frame error" |
| 22:10.46 | cusco_ | happening a lot.. |
| 22:11.31 | cusco_ | or some.. "no more room in schedule" |
| 22:11.44 | cusco_ | lol |
| 22:12.00 | cusco_ | anyway, we don't like to put up with client's problems anyways so.. |
| 22:12.03 | *** join/#asterisk erieslab233 (n=joshua@74.5.28.151) |
| 22:12.09 | cusco_ | it has a bright side |
| 22:12.20 | cusco_ | everybody gets happy whenever asterisk goes down :P |
| 22:12.25 | manxpower | cusco_: make sure you have the latest dahdi and the latest libpri. I seem to remember seeing a PRI issue on heavy loaded systems that was fixed recently. |
| 22:12.46 | cusco_ | manxpower: that is good to know, migrating is being a proble latelly |
| 22:13.02 | cusco_ | we have another machine up to date configure etc... |
| 22:13.23 | manxpower | cusco_: upgrading to the latest of those two should not be terrible. |
| 22:13.24 | cusco_ | but its digium card is not a dual span like the one in production |
| 22:13.37 | erieslab233 | Excuse me, is there an order in this room or should I feel free to rudely shout out my questions? :) |
| 22:13.41 | cusco_ | hmm |
| 22:13.45 | TSM | yup |
| 22:13.47 | manxpower | ~ask |
| 22:13.48 | 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. |
| 22:14.36 | cusco_ | who is here agains his will? (besides me?) |
| 22:14.42 | erieslab233 | I love you all, first. I am new to asterisk for only a few weeks and thanks to all the great documentation have deployed my own server and have 20 SIP phones running in multiple locations... |
| 22:14.46 | erieslab233 | Thank you |
| 22:15.07 | erieslab233 | My question is basic, I have like 20 DID's, but don't think I need so many |
| 22:15.36 | cusco_ | manxpower: thaks for the hints, I will see that it gets upgraded tonight |
| 22:15.47 | p3nguin_ | erieslab233: And the question part of this is... |
| 22:16.10 | cusco_ | manxpower: by the way, what issue whas that with libpri? |
| 22:16.25 | erieslab233 | that is coming, just wanted to say thanks first... stalling trying to find my paper |
| 22:16.26 | p3nguin_ | erieslab233: With 20 DIDs and 20 users, you can give everyone his own DID. |
| 22:16.57 | p3nguin_ | erieslab233: And if you have a block of 20, that will be even better. |
| 22:17.01 | erieslab233 | so everyone having their own DID IS a good thing?? |
| 22:17.16 | p3nguin_ | erieslab233: It's nice for people trying to call other people. |
| 22:17.16 | erieslab233 | Vitelity :) many are in blocks |
| 22:17.22 | TSM | not always, but if you have them and you dont mind paying for them then all ok |
| 22:17.36 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 22:17.45 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 22:18.21 | erieslab233 | that is my question TSM... I don't mind paying it is cheap, but I wonder if most others use one DID per extension? |
| 22:18.37 | p3nguin_ | erieslab233: For example, if you have 6185453000-6185453999, you could have 1000 users and everyone could have their own phone number. |
| 22:18.58 | erieslab233 | I mean the question is what is the point of extensions if they each have their own DID other than internal calling? |
| 22:19.02 | TSM | i ususaly do, it all depends on size, usualy as the company gets bigger the less DIDs you requite |
| 22:19.26 | TSM | also if you run callcenters then there is no need for DIDs for everyone |
| 22:20.00 | p3nguin_ | You could have just one DID and an IVR to allow dialing a user's extension internally. |
| 22:20.13 | erieslab233 | exactly! We only publish a handfull of numbers to the public and all outbound calls show our main toll-free on the recieving end's caller id |
| 22:20.54 | erieslab233 | so 20 DID's is overkill if we do not publish half these numbers? |
| 22:20.59 | p3nguin_ | erieslab233: Sounds like you could safely discontinue those other DIDs if you don't want to use them for anything specific. |
| 22:21.07 | Kobaz | manxpower: yeah i upgraded to latest libpri, i had the latest dahdi already |
| 22:21.23 | Kobaz | manxpower: and i downgraded my kernel, and downgraded wanpipe to the latest feature freeze version |
| 22:21.23 | p3nguin_ | erieslab233: I would probably reserve 2 or 3 for special purposes. |
| 22:21.32 | erieslab233 | then just use DISA to call direct you suggest from my mobile for example? |
| 22:21.41 | rene- | erieslab233 but it you have one DID for each user, when somebody receives a call from your pbx and tries to return the call, it will get to the exact person that dialed him and not an IVR |
| 22:22.22 | erieslab233 | rene all outbound caller id displays the same number ;) |
| 22:22.30 | *** part/#asterisk ibercom (i=96d6cd10@gateway/web/freenode/x-fqflaqrjgxidotyk) |
| 22:22.43 | manxpower | We used to get 100 DIDs for like $3/month. |
| 22:22.48 | p3nguin_ | erieslab233: Maybe each person could have his own DID, though. Direct to his phone. |
| 22:22.50 | erieslab233 | WOW! |
| 22:23.00 | manxpower | Then you might as well get a DID for everyone and have less work for an operator to do. |
| 22:23.08 | erieslab233 | yes p3nguin that was what we initially thought too |
| 22:23.23 | erieslab233 | just don't want to be creating a mess in the server |
| 22:23.41 | p3nguin_ | It's not much work to do. |
| 22:23.44 | rene- | erieslab233: and having all outbound caller id displaying the same number is a good thing because? |
| 22:23.55 | ryduh | I've found my DTMF problem. The old Panasonic system isn't responding to the DTMF tones I send. |
| 22:24.03 | rene- | scrap it |
| 22:24.06 | erieslab233 | I am a noob so I feel uncomfortable with alot of what I have done so far, but it seems to somehow automagically work |
| 22:24.29 | *** join/#asterisk pirulo (n=andres@70.56.223.76) |
| 22:24.30 | p3nguin_ | erieslab233: Walmart, for example, has a DID for each department, and each dept has his own caller ID instead of the main store number. |
| 22:24.49 | manxpower | ryduh: tonelength option in the zap config. It should be in the sample config |
| 22:24.58 | erieslab233 | because at this point in time I don't trust my employees to deal with every issue ;) We work in healthcare and much is confidential, as in not for staff to deal with |
| 22:25.13 | p3nguin_ | erieslab233: So when the pharmacy calls you, you see Walmart Pharmacy on your CID unit instead of just Walmart and the main number. |
| 22:25.15 | erieslab233 | I am not a control freak ;) |
| 22:25.31 | ryduh | manxpower: that's exactly what I was looking up :) |
| 22:25.53 | p3nguin_ | erieslab233: It's really your choice as the administrator. |
| 22:26.23 | p3nguin_ | erieslab233: If I had 20 users and 21 DIDs, I would have a main number with an IVR and 20 direct-to-phone extensions. |
| 22:26.35 | erieslab233 | My next question is a bit more complex... I want another facility to connect to the same server. If I worry latency might be a problem would VPN'ing another box in really do anything vs. a straight SIP phone connecting? |
| 22:26.45 | manxpower | toneduration= is the correct option. |
| 22:26.46 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 22:26.48 | rene- | VPN would increase the latency |
| 22:26.51 | erieslab233 | ok |
| 22:26.57 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 22:27.03 | p3nguin_ | erieslab233: use IAX2 to interconnect the two Asterisk servers. |
| 22:27.10 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 22:27.11 | erieslab233 | what are the advantages of adding another server at a remote location, any? |
| 22:27.44 | rene- | well if it has its own telephone lines and internet goes down then the remote site gets to have phone service |
| 22:28.00 | erieslab233 | 100% Voip |
| 22:28.50 | rene- | if main node dies, then you still have voice service at your location |
| 22:28.53 | rene- | (remote) |
| 22:29.12 | erieslab233 | It is remote both in networking and literal terms, a maintenance garage in a building out in the country, I don't see the need in paying $50 per month for a POTS line |
| 22:29.12 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 22:29.33 | erieslab233 | so connectivity relying on a PSTN line is the only real advantage? |
| 22:29.44 | rene- | also if you have an itsp that is closer geographically to your remote location than yourself then you save alot of hoops |
| 22:30.20 | erieslab233 | just check the dns vs geographical rene-? |
| 22:30.54 | TSM | if your plan is just to have more people at another site then there are some advantages to have all calls running over IAX2 vs multiple SIP channels, if reinvite is turned on then you dont have long latancy and bw issues when there is same office calls at the remote site |
| 22:31.01 | rene- | yes |
| 22:31.08 | rene- | less than three extensions it isnt worth it |
| 22:31.21 | rene- | more than 10 you should have it |
| 22:31.21 | erieslab233 | perfect answers thank you |
| 22:31.48 | erieslab233 | just one is needed I guess I need to get a real router for my static IP's then lol |
| 22:32.00 | TSM | also take advantage of local outbound call rates, problem may be with the outbound CID |
| 22:32.21 | TSM | look at your ISP for static IP |
| 22:32.35 | TSM | all routers will support it |
| 22:33.22 | erieslab233 | my $30 belkin walmart special won't because ATT gave me 5 statics and the router will not function as PPPoe |
| 22:33.38 | erieslab233 | need to upgrade to the $40 router haha |
| 22:33.43 | TSM | ahh yeh you guys use PPPoE out there... |
| 22:33.47 | *** join/#asterisk wastrel (n=wastrel@nylug/member/wastrel) |
| 22:33.50 | TSM | i personaly like sonicwall firewalls, they support PPPoE |
| 22:34.07 | TSM | depends on if you have ethernet presentation though as they only have that |
| 22:34.12 | erieslab233 | Everyone keeps telling me about this sonicwall thing, are they expensive? |
| 22:34.59 | wastrel | hi. every night when i go home i forward my desk line to my cell phone. I'd like to automate this with some cron tomfoolery on the asterisk server - is there a relatively straightforward way to do this? |
| 22:35.04 | *** part/#asterisk DMeloUK (n=Dominic_@64.129.94.138) |
| 22:35.49 | ryduh | wastrel: you could just set up a GoToIfTime based on time in your dialplan |
| 22:36.07 | TSM | erieslab233: depends on the model you want, the NSA240 units are quite cheep, the total secure units full with 3yr warranty etc, all in i think is like under $2k |
| 22:36.40 | TSM | you can also have two of them in HA mode, the 2nd box comes in way way cheeper for that |
| 22:36.53 | erieslab233 | whoa! no way, I will go for the $50 walmart before that! |
| 22:36.59 | TSM | i use PRO2040 units which has now been superceeded by the NSA2400 unit |
| 22:37.42 | TSM | yeh diffrence is we have 25m/25m in our UK office and nearly 100m in our NY offices, so we need good units |
| 22:38.00 | TSM | you pay for good stuff, depends on if you value your internet |
| 22:38.21 | wastrel | ryduh: thanks. i don't know how i'd go about that - I didn't set up this box though i now own it. I saw this as a relatively low-impact project to get to know the system better |
| 22:38.33 | erieslab233 | no value just a easy cheap tool that is cheaper than paying ATT to give me a pbx |
| 22:38.44 | wastrel | i'm looking at the exten => line for my number in extensions.conf |
| 22:39.02 | *** join/#asterisk Trepinon (n=Lance@150.74.176.69.static.ip.tnltd.net) |
| 22:39.04 | ryduh | wastrel: I have our system set up so after 9pm and before 9am, calls get forwarded to voicemail. |
| 22:39.06 | erieslab233 | ok another question, my wife wants to call her sister in Thailand everynight, so I bought a huge amount of minutes with Teliax. I want to configure one of my DID's to forward directly to the international number for her alone. Easy dialing. Is this just done in inbound routes based on her caller id? |
| 22:39.08 | TSM | wastrel: why not use the ring group function, make a virtual extention |
| 22:39.10 | wastrel | i should start there and learn to write a dialplan? |
| 22:39.32 | Trepinon | from a sip trunk to an iax2 trunk what would cause asterisk to convert the call from ulaw to gsm ? |
| 22:39.32 | wastrel | TSM: as explained above - i'm clueless is why :] |
| 22:39.34 | ryduh | wastrel: yes |
| 22:39.38 | *** join/#asterisk Katty (n=Angela@adsl-70-253-172-140.dsl.stlsmo.swbell.net) |
| 22:39.45 | Katty | :> |
| 22:39.57 | TSM | erieslab233: look at DISA, or create custom extention and fill in the dial command |
| 22:40.13 | Katty | ground bison is good. |
| 22:40.16 | Katty | in case anyone was wondering |
| 22:40.17 | erieslab233 | I liked DISA better too for the sake of a PIN ;) |
| 22:40.22 | TSM | wastrel: you got FreePBX? |
| 22:40.36 | wastrel | TSM: no it's asterisk on a centos box the former admin has left the company |
| 22:40.48 | TSM | haaa |
| 22:41.08 | TSM | wastrel: look at getting yourself FreePBX if you are lacking * knowhow |
| 22:41.23 | TSM | problem is you will have to fairly much start agian |
| 22:41.26 | TSM | well that poo |
| 22:41.32 | wastrel | i have some cargo cult procedures for adding and removing extensions to handle onboarding and such but i don't know how it works |
| 22:41.53 | erieslab233 | Last Question, In an attempt to save money on an overhead paging system I cracked the case of a $20 ebay sip phone, wired the speakerphone speaker to the auxilary input of our amplifier and registered the SIP phone with an extension and set to auto-answer... |
| 22:42.08 | erieslab233 | I have done this in all 3 of my buildings, but all have different extensions |
| 22:42.25 | erieslab233 | I can easily page any building, but how do I page all 3 at the same time? |
| 22:42.46 | erieslab233 | it seems that the first SIP line to answer gets the call and that's that? |
| 22:42.49 | Qwell | erieslab233: use the page application |
| 22:42.53 | loather-work | Dial(SIP/123&SIP/124&SIP/125) |
| 22:42.54 | Qwell | Page() application rather |
| 22:43.05 | loather-work | err, Page(), yeah. |
| 22:43.29 | erieslab233 | they will all have the opportunity to answer then? |
| 22:43.47 | erieslab233 | is that similar to group ring function except more than one line can get the call? |
| 22:44.28 | Katty | Dear Universe, thank you for avocados. Love, Katty |
| 22:44.43 | loather-work | Page() sets up a fake MeetMe conference bridge and transfers all the calls into it. |
| 22:44.49 | erieslab233 | PERFECT |
| 22:44.52 | erieslab233 | thank you all |
| 22:45.36 | loather-work | welcome :) |
| 22:47.39 | Trepinon | does anyone here know much about asterisk and codex? |
| 22:47.53 | Trepinon | for somereason I cannot figure out my cals are gettings converted from ulaw to gsm |
| 22:48.05 | Nugget | do you care how much a person knows, or do you just care if they know the answer to whatever question you have in mind? |
| 22:48.37 | Nugget | It's not like you're going to turn down the right answer because the person doesn't know much else, right? |
| 22:49.04 | Trepinon | I welcome any sugestions |
| 22:49.05 | Nugget | if you were choking on a piece of steak would you gesture wildly and pantomime "does anyone here know a lot about medicine?" |
| 22:49.13 | Trepinon | lol |
| 22:49.48 | Trepinon | your right about that, how then should I state the question? |
| 22:50.11 | Nugget | a bit more info about what 'calls' means would be a great place to start. |
| 22:50.13 | [TK]D-Fender | Trepinon: its getting transcoded because of the codecs you set for your devices and outside peers |
| 22:51.40 | Trepinon | i have a inbound sip trunk which I dail out on iax2 |
| 22:51.44 | Trepinon | allow=ulaw |
| 22:51.51 | Trepinon | is set on all of my trunk listings |
| 22:53.00 | [TK]D-Fender | Trepinon: Feel press to pastebin your configs an the complete trace of a call with SIP & IAX2 debug enabled |
| 22:53.27 | Trepinon | okay one minute thank you so much for your help |
| 22:56.14 | *** join/#asterisk bn-7bc (n=bjarne-i@mac.lan.noare-1.holmedal.net) |
| 22:59.42 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 23:06.09 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-136-66.dsl.stlsmo.sbcglobal.net) |
| 23:08.33 | Trepinon | http://pastebin.com/m18784e42 |
| 23:08.38 | Trepinon | I think i've fixed the issue |
| 23:08.51 | Trepinon | one of the iax trunks was set to allow=ulaw,alaw |
| 23:09.04 | Trepinon | but for some reason its still being turned into alaw. |
| 23:09.23 | Katty | mmm, taco salad is so dreamy. |
| 23:09.35 | Trepinon | I love taco salad |
| 23:09.48 | Katty | Dear Universe, thank you for Taco Salad. Love, Katty |
| 23:11.28 | loather-work | mmmm tacos |
| 23:12.24 | Trepinon | nothing better |
| 23:12.40 | *** join/#asterisk dysinger (n=tim@cpe-75-85-135-191.hawaii.res.rr.com) |
| 23:13.21 | *** part/#asterisk Trepinon (n=Lance@150.74.176.69.static.ip.tnltd.net) |
| 23:14.34 | jblack | heh |
| 23:14.44 | Katty | hello mister black |
| 23:14.50 | rene- | I GOT COFFEE! |
| 23:14.58 | jblack | rotf |
| 23:15.13 | jblack | Hi Katty. Taco salad teh mana of teh gods? |
| 23:15.17 | leifmadsen | Mr. Black! |
| 23:15.18 | Katty | something like that |
| 23:15.32 | jblack | That's right, and don't you forget it |
| 23:15.39 | jblack | Mr Foliage. |
| 23:15.39 | leifmadsen | has that "Mister Black" sound from the Simpsons on the Kamp Krusty episode |
| 23:15.55 | jblack | cracks a big grin |
| 23:17.58 | jblack | katty: 7 day: 137. 30 Day: 142 90 day: 160 365 day: probably around 450. |
| 23:18.32 | Katty | ^_- |
| 23:18.40 | Katty | i'm lost. |
| 23:19.07 | jblack | bg. in simpler, yet more confusing words, I'm becoming consistantly less sweet. |
| 23:19.17 | Katty | OH!, that. |
| 23:19.21 | Katty | hmm. interesting. |
| 23:19.28 | jblack | I have to tell someone. |
| 23:20.25 | jblack | shrugs |
| 23:21.14 | Katty | this is very good |
| 23:21.20 | Katty | you keep this up, you're going to be downright bitter. |
| 23:21.29 | leifmadsen | is already a bitter old man! |
| 23:21.33 | leifmadsen | and I'm only 28! :) |
| 23:21.36 | Katty | leifmadsen: we still love you. |
| 23:21.36 | jblack | Well, um.. as it happens... |
| 23:21.55 | jblack | I'm going from sweet-n-sour to just plain sour. |
| 23:22.01 | jblack | Yeah. lmadsen rocks. |
| 23:23.49 | *** join/#asterisk SirFoxey (n=SirFoxey@unaffiliated/sirfoxey) |
| 23:24.50 | Katty | jeebus, stuffed. |
| 23:24.56 | SirFoxey | I have a quick question about asterix now prior to trying it. Can anyone tell me if I install the asterix now installation and I want to update the asterix version down the track for new features, it is a easy update or am i better just getting the source from the start. |
| 23:24.57 | *** join/#asterisk voipmonk (n=voipmonk@66.49.238.52) |
| 23:25.15 | Katty | SirFoxey: it's very easy to upgrade your version. |
| 23:25.41 | SirFoxey | ok awesome, we have been caught out with elastix and I am now wanting to move to a real version of asterix. |
| 23:25.47 | SirFoxey | thanks muchly for your help Katty |
| 23:27.27 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 23:28.18 | Katty | well. time to do laundry and clean up and all that jazz. |
| 23:28.28 | Katty | back in about 30 |
| 23:29.14 | LemensTS | How do i round a number in the dialplan? Say i have 1.22, how do i round it to 1 or 2 |
| 23:29.23 | *** join/#asterisk djabbour (n=djabbour@ool-457be7e8.dyn.optonline.net) |
| 23:29.47 | djabbour | The CentOS/RHEL yum repo that Asterisk provides... is it compatible with a Xen CentOS installation? |
| 23:29.55 | manxpower | LemensTS: "core show functions" didn't have anything useful? |
| 23:30.03 | djabbour | I immagine different packages would be required for the kernel module |
| 23:30.16 | manxpower | djabbour: we recommend compiling from source |
| 23:30.23 | *** join/#asterisk VooDooNOFX (n=Joe@rrcs-24-43-123-93.west.biz.rr.com) |
| 23:30.30 | djabbour | manxpower, I figured |
| 23:30.34 | djabbour | manxpower, not a big deal |
| 23:31.00 | djabbour | manxpower, Is there anything specific I should look out for with Xen? Is there a document I could read on any specifics? |
| 23:31.07 | VooDooNOFX | I've got a couple used cisco 7940 phones here that aren't responding to what Cisco says should be the factory reset guidelines. has anyone here flashed these things back to sip before? |
| 23:31.11 | VooDooNOFX | they're using MGCP now. |
| 23:31.40 | ian6 | don't they just load whatever firmware the tftp server tells them to load? |
| 23:31.53 | manxpower | djabbour: What I did was take an Asterisk src rpm, built it to my preferences, then used that on all my servers as part of a kickstart autoinstall |
| 23:32.43 | djabbour | manxpower, ah yes that would be nice for a managed repo. As much as I've never liked RPM, I do really like kickstart/cobbler |
| 23:33.10 | cusco_ | <PROTECTED> |
| 23:33.41 | djabbour | manxpower, what do you do for any required libs? Or is everything else in the core yum repos? |
| 23:34.41 | manxpower | djabbour: I don't think I needed anything that was not as part of CentOS 5.3 except for things like libpri, zaptel/dahdi, spandsp (for fax), etc. |
| 23:34.58 | manxpower | you won't need any of those if you're going to do this in a VM |
| 23:35.32 | djabbour | manxpower, Well zaptel is required, no? |
| 23:35.40 | manxpower | djabbour: why? |
| 23:35.50 | *** join/#asterisk wonderworld (n=ww@62.143.22.226) |
| 23:36.52 | LemensTS | manxpower: http://pastebin.com/m21e250fe this is what i get when i try ROUND...it doesnt show in core show functions...is this an addon? |
| 23:36.56 | LemensTS | im on 1.6 |
| 23:37.10 | manxpower | LemensTS: it it's not in the show functions then it does not exist. |
| 23:37.53 | manxpower | Grab a programming book and figure out how to round the old fashioned way. |
| 23:38.39 | LemensTS | Old fashioned way? |
| 23:38.43 | LemensTS | From the dialplan? |
| 23:39.05 | jblack | How did you get a floating point number in your dialplan in the first place? |
| 23:39.39 | LemensTS | jblack: diving duration by 60 |
| 23:39.49 | LemensTS | dividing i mean |
| 23:40.00 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.8 (2009/10/26), 1.6.0.15 (2009/09/03), 1.4.26.2 (2009/09/03), *-Addons 1.6.1.1 (2009/07/24), 1.6.0.3 (2009/07/24), 1.4.9 (2009/07/24), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.1 (2009/07/02) -=- Related channels: #asterisknow #switchvox #asterisk-bugs |
| 23:40.50 | leifmadsen | Asterisk 1.6.1.8 has been released due to a security vulnerability in chan_sip. For more information, please see the security advisory at http://downloads.asterisk.org/pub/security/AST-2009-007.pdf and the release announcement at http://www.asterisk.org/node/49855 |
| 23:41.05 | jblack | I don't know of many (any?) math functions in the dailplan. I'd pass your work off to an agi. |
| 23:41.21 | [TK]D-Fender | jblack: Expressions 101 |
| 23:41.31 | [TK]D-Fender | jblack: Go read the CHANNELVARIABLES doc |
| 23:41.55 | [TK]D-Fender | is off for a bit.... |
| 23:42.02 | LemensTS | http://www.voip-info.org/wiki/view/Asterisk+Expressions thats where i was reading |
| 23:42.03 | loather-work | there are a surprising amount of math functions in the dialplan. |
| 23:42.10 | LemensTS | thats where i got ROUND(x) |
| 23:42.17 | loather-work | i haven't had a need for any of them, but it's pretty cool it supports it, anyways |
| 23:42.40 | jblack | Oh, duh me. |
| 23:43.21 | jblack | try wrapping it in $[ ] |
| 23:44.14 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 23:44.47 | leifmadsen | yay |
| 23:44.50 | jblack | cmon... arc tan? who needs the arc tangent in a dialplan? |
| 23:45.05 | *** join/#asterisk tzafrir__laptop (n=tzafrir@212.179.75.202) |
| 23:45.31 | leifmadsen | why is it I think there is a MATH() function? |
| 23:46.30 | *** join/#asterisk KingDavidNYC (n=Chris123@pool-96-224-162-219.nycmny.east.verizon.net) |
| 23:46.46 | KingDavidNYC | hello everybody!!! |
| 23:46.48 | jblack | No idea. I see neither an application nor a function called as such. The aforementioned page wraps math in $[ ] |
| 23:48.17 | jblack | http://www.voip-info.org/wiki/view/Asterisk+cmd+Math |
| 23:50.22 | jblack | "show function MATH" is more useful than "show function math" |
| 23:51.02 | leifmadsen | yes :) |
| 23:51.14 | leifmadsen | uppercase is important -- if you tab complete, it'd have worked |
| 23:52.54 | jblack | I try to get more exercise these days. :) |
| 23:53.10 | jblack | bah. I don't even buy that |
| 23:53.56 | SaiSoma | http://www.collegehumor.com/video:1921846 |
| 23:54.12 | KingDavidNYC | Can anybody please tel me a bit more about "no reply to our critical packets" error? |
| 23:54.48 | KingDavidNYC | actually, if the router has 5060 open? why then I am getting that error? |
| 23:56.55 | KingDavidNYC | the network admin is asking me what other ports to open, and I dont know which ones to tell him |
| 23:57.19 | leifmadsen | KingDavidNYC: just 5060 for SIP messages, any RTP ports for media which is defined in rtp.conf |
| 23:57.27 | manxpower | KingDavidNYC: SIP runs on port 5060 for Signaling (call setup, teardown, etc), audio defaults to ports 10,000 - 20,000 all UDP. |
| 23:57.45 | manxpower | you can configure the local port range in /etc/asterisk/rtp.conf assume two ports for every active call. |
| 23:58.11 | KingDavidNYC | manxpower: should I tell the sysad to open from 10,000 to 20,000 to UDP? |
| 23:58.54 | manxpower | KingDavidNYC: how many calls do you need to have at the same time? |
| 23:59.06 | KingDavidNYC | maybe 40 |
| 23:59.07 | manxpower | KingDavidNYC: I assume you have an Asterisk server behind a firewall? |
| 23:59.15 | KingDavidNYC | yes |
| 23:59.23 | manxpower | KingDavidNYC: set rtp.conf to use 10000 - 10100 |
| 23:59.44 | LemensTS | exten => s,n,Set(i=${MATH(2.6+0,int)}) i got that to work, but it rounds it down to 2 hmmm |
| 23:59.52 | manxpower | you'll need to forward those ports if there's NAT involved, in addition to setting up localnet and externip. Is there NAT involved? |