| 00:00.00 | Greenlight | In the CLI it just says "Putting chan <X> into PBX again" |
| 00:00.33 | Greenlight | Just don;t know why on earth you'd want or need that behaviour |
| 00:00.43 | Greenlight | Or how to disable it |
| 00:01.39 | SeRi | sip set debug on and core set verbose 3 |
| 00:02.49 | Greenlight | Ok got sip debug on away to try it again |
| 00:03.10 | SeRi | pb the results |
| 00:03.41 | SeRi | bb in a min |
| 00:09.46 | Greenlight | http://pastebin.com/WDtYUcNK |
| 00:10.19 | Greenlight | SIP stuff all looks okay. The extension hanging up is local and that seems to be fine, but the remote side is getting chucked back into the dialplan |
| 00:14.48 | Greenlight | I guess I can just write some code to detect the situation and fire a hangup on the other channel via the AMI, but that kinda feels like a hacky fix |
| 00:18.38 | p3nguin | seri: I just got my genuine Apple OEM headphones/mic. |
| 00:18.53 | SeRi | p3nguin, And results? |
| 00:18.54 | p3nguin | The mic works perfectly on my iPod touch 3rd gen. |
| 00:18.59 | SeRi | sweet! |
| 00:19.18 | SeRi | I am glad its working! |
| 00:19.20 | p3nguin | It was apparently a quirk with those Chinese knock-offs. |
| 00:19.27 | SeRi | lmao. |
| 00:19.41 | SeRi | does not buy Chinese knock offs :P |
| 00:19.44 | p3nguin | I have the $30 Apple headphones/mic now. |
| 00:20.04 | SeRi | lol |
| 00:20.08 | SeRi | damn expensive apple |
| 00:20.24 | SeRi | I am sure is something they change so you have to by the 3rd gen shit |
| 00:20.29 | ChannelZ | I forget, is white the new brushed aluminum? |
| 00:21.07 | p3nguin | They have another set that is $79, but I have the $29 model. |
| 00:21.44 | p3nguin | Apple says they are compatible with 2nd gen and up. |
| 00:22.08 | p3nguin | But the knock-off is only working on 2nd gen. |
| 00:22.21 | p3nguin | I still don't understand that, but I don't really care anymore! |
| 00:22.37 | p3nguin | Tonight I will be testing the SIP phones on the iPod. |
| 00:23.17 | SeRi | p3nguin, nice! |
| 00:23.29 | SeRi | conf in if you can |
| 00:24.01 | p3nguin | TinyUmbrella reports that the iPhone has 4.0 on it. |
| 00:24.10 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 00:24.18 | SeRi | p3nguin, yours? |
| 00:24.25 | p3nguin | Yes. |
| 00:25.00 | SeRi | ouch... well it can be unlock for sure... though once you do consider downgrading... 4.0 on 3G is very slow unless you tweak the hell out of it |
| 00:25.26 | p3nguin | It has baseband 05.13.04 |
| 00:26.00 | p3nguin | I prefer to have iOS 4 on it. 3 doesn't have some features I want. |
| 00:26.16 | SeRi | p3nguin, make sure you turn of spot light |
| 00:26.20 | SeRi | off* |
| 00:26.24 | p3nguin | What does it do? |
| 00:26.40 | SeRi | makes your iOS crawl to hell |
| 00:27.06 | SeRi | 3G/3GS do not have the hardware requirement to run spot light on the background all the time |
| 00:27.25 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v028-178.mobile.uci.edu) |
| 00:27.25 | p3nguin | Do I just deselect all the options in spotlight? |
| 00:27.42 | p3nguin | There is no toggle for on/off. |
| 00:28.24 | SeRi | yes deselect them |
| 00:29.54 | SeRi | p3nguin, so you are in? |
| 00:30.34 | p3nguin | Not yet. Taking things one step at a time. |
| 00:30.58 | SeRi | got it. |
| 00:31.19 | SeRi | p3nguin, from CC: |
| 00:31.21 | SeRi | Hello, |
| 00:31.21 | SeRi | Our managers are going over your suggestion within our labs specifically testing Asterisk 1.8 as well as going over any differences between version 1.6 and 1.8 and make any changes to our guide to reflect those differences. If you have any other questions, please let us know. |
| 00:32.24 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 00:43.04 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v028-178.mobile.uci.edu) |
| 00:55.02 | *** join/#asterisk adolfomaltez (~taro@190.62.201.140) |
| 00:55.23 | *** join/#asterisk openstandards (~openstand@87.127.196.56) |
| 00:58.15 | openstandards | evening all, i've configured a polycom before but won a avaya 6930 on ebay on looking to configure it to talk to asterisk....asks about a callsv i'm assuming that would be asterisk or am i completely won. I've tried googling for a solution but still unsure has anyone configured a avaya device before? |
| 00:59.51 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 01:00.45 | SeRi | p3nguin, where you at with your iphone? |
| 01:00.52 | SeRi | haz popcorn |
| 01:04.32 | *** join/#asterisk phiscribe (~phiscribe@98.157.3.94) |
| 01:05.10 | leifmadsen | s/pco// |
| 01:06.18 | SeRi | I need to try that |
| 01:06.33 | SeRi | s/need/wtf/ |
| 01:06.39 | SeRi | rofl1 |
| 01:07.02 | SeRi | nice lol |
| 01:07.09 | *** part/#asterisk jmwpc (~jmwpc@c-24-5-58-60.hsd1.ca.comcast.net) |
| 01:07.36 | phiscribe | opinions on distro's based around asterisk? Beneficial or about the same as just adding to existing distro? |
| 01:15.20 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:15.20 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 01:18.07 | openstandards | phiscribe: what do you mean? do you mean things such as PIAF? |
| 01:19.47 | phiscribe | astlinux afonality asterisknow, ect... do these have value above just installing asterisk so to a debian based distor? |
| 01:19.57 | openstandards | personally I've used pbx in a flash wasn't bad however i prefer to install asterisk/freeswitch on my own you'll learn more doing so |
| 01:21.11 | phiscribe | ok, i share the human train of resisting the friendly manual and seeking shortcuts, (but know eventualy you gotta get cozy with that manaul), |
| 01:22.03 | phiscribe | trait that is |
| 01:22.22 | openstandards | i'm a bit anal when it comes to configuring things, I prefer to know how something is done when it comes to fixing things it pays off thats the biggest key point in setting it up yourself |
| 01:22.27 | blitzrage | web interfaces are great until you need to do something that it doesn't handle |
| 01:22.28 | blitzrage | then you're fscked |
| 01:24.03 | blitzrage | indeed |
| 01:24.12 | blitzrage | that's why documentation is useful :) |
| 01:24.48 | phiscribe | its always useful, i always use it last hehe like i dont ask for directions |
| 01:24.52 | phiscribe | when driving lol |
| 01:26.26 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:26.34 | SeRi | phiscribe, astlinux is not gui base. at least nit like trixbox or freepbx |
| 01:26.51 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 01:27.10 | SeRi | astlinux only gives you a webui for status, editing conf files, and managing astdb. |
| 01:27.47 | phiscribe | i guess ill just fire up some virtual machines and dig in |
| 01:27.52 | SeRi | I use astlinux and I only use the webui to check the status of things |
| 01:28.49 | SeRi | the rest I do from cli |
| 01:29.41 | SeRi | phiscribe, my recommendation.... stay away from gui versions of asterisk.... |
| 01:29.57 | SeRi | if you *MUST* have one than try freepbx |
| 01:30.10 | SeRi | thats about my only recommendation |
| 01:31.25 | blitzrage | I use astlinux for my dhcp server :0 |
| 01:31.30 | SeRi | leifmadsen, by the way I just check the version on my samsung and is running 2.3.7. though is CM7. |
| 01:31.32 | phiscribe | well a gui helps me initialy, just to grook the thing, but i got some old boxes that cli would suit, and im intersted in small office environments |
| 01:31.35 | blitzrage | (no asterisk functionality actually :)) |
| 01:31.48 | blitzrage | SeRi: coolio I'll have to check out CM7 |
| 01:32.00 | p3nguin | Using a GUI will not help you understand anything about Asterisk. |
| 01:32.06 | SeRi | ^^ |
| 01:32.12 | SeRi | BOOM HEAD SHOOT! |
| 01:33.09 | blitzrage | who wants to understand anything about Asterisk? |
| 01:33.22 | SeRi | blitzrage, astlinux is well polished. I like the functionality and the fact is so light weight |
| 01:33.52 | phiscribe | well, i can use cli just fine, but people i have to work with cant, so gui is king to that set |
| 01:33.59 | *** join/#asterisk kikohnl (~kotis@ext-dip-171.hnl.cdsinc.com) |
| 01:34.26 | openstandards | blitzrage: i've got a quick question i had freeswitch set up on a openvz box and recently moved over to asterisk and basically i'm having sip registation problems with n900 now, what would you suggestion i do to debug and resolve the problem |
| 01:34.57 | blitzrage | that's a pretty vague question... I'd look at the authentication methods and follow the SIP debug and find out why things aren't working |
| 01:35.24 | p3nguin | So far, VaxPhone is not my favorite soft phone. I don't like the interface that much. |
| 01:39.02 | SeRi | makes note: VaxPhone sucks. noted. |
| 01:40.13 | p3nguin | I have to select a line to use before I can dial. After I dial, when I am done, I press hangup. I go to dial a new number... but wait! the old number is still on the dial line. Have to delete the old number first, then dial the new number. |
| 01:40.14 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:40.53 | SeRi | now that sucks donkey balls :/ |
| 01:41.07 | p3nguin | Some people may not have a problem with it. |
| 01:41.12 | p3nguin | I don't like it. |
| 01:41.25 | p3nguin | Call quality is good, though. |
| 01:44.46 | SeRi | meh.... that sucks though..... |
| 01:45.11 | SeRi | so far siphon is what I have been stuck with... |
| 01:45.48 | SeRi | though now my android has a whole new meaning. the native sip just kills the purpose of having my iphone around.... |
| 01:48.52 | *** join/#asterisk jmwpc (~jmwpc@c-24-5-58-60.hsd1.ca.comcast.net) |
| 01:53.26 | SeRi | p3nguin, http://futurist.se/gldt/wp-content/uploads/11.10/gldt1110.png |
| 01:56.10 | p3nguin | Groundwire/Acrobits is difficult to configure. It has a lot of settings. And I'm not able to get it registered to my asterisk on the first attempt. Now I'll have to go back through all the settings and figure out where the problem is. Also not my favorite. |
| 01:56.17 | p3nguin | It does look nice, though. |
| 01:56.52 | SeRi | noted. |
| 01:58.29 | SeRi | p3nguin, is your fail2ban working properly now or does it still have issues? |
| 01:59.51 | p3nguin | I'm sure it still has the same problem that it had before... but I haven't tested it. |
| 02:00.04 | p3nguin | Oh, wait, I changed the subnet mask. Maybe it does work now. |
| 02:00.25 | SeRi | yes you did. I told you to do try that :) |
| 02:00.47 | SeRi | s/do try/try/ |
| 02:06.00 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-jrphoojbcpwqhszh) |
| 02:11.44 | p3nguin | iPico is really nice, and it's easy to configure, but there's something wrong with the audio. |
| 02:16.45 | SeRi | noted. iPico has issues. |
| 02:17.06 | p3nguin | But wait... |
| 02:17.24 | p3nguin | Media5-fone is doing the same thing. |
| 02:17.33 | p3nguin | Maybe it is my battery, which is almost dead. |
| 02:17.48 | SeRi | I had issues with media5-fone :( |
| 02:18.02 | p3nguin | Media5-fone (Pro) is really nice, too. |
| 02:18.22 | p3nguin | It's easy to configure, and it even has built in SIP trace. |
| 02:18.24 | SeRi | I think ill try the pro when ill have the extras cash to spend. |
| 02:18.42 | p3nguin | You don't have a jailbroken idevice? |
| 02:18.48 | SeRi | Yes I do |
| 02:18.59 | LiuYan | ~spandsp |
| 02:18.59 | infobot | extra, extra, read all about it, spandsp is cool : http://www.soft-switch.org/installing-spandsp.html, or a softmodem that comes with app_rxfax and app_txfax for asterisk, or the shizzznit |
| 02:19.04 | p3nguin | Then you can always steal it for free. |
| 02:19.13 | SeRi | Thats true :) |
| 02:20.44 | *** join/#asterisk F2Knight (~ben@c-24-22-60-186.hsd1.or.comcast.net) |
| 02:21.22 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-eeikjgqxbwcktfad) |
| 02:21.29 | F2Knight | [TK]D-Fender, p3nguin, what do you think is the best format to store audio prompts in? ulaw? wav? other? |
| 02:21.48 | p3nguin | I have them in ulaw, wav, and sln. |
| 02:21.58 | SeRi | ulaw and wav here |
| 02:22.28 | F2Knight | most of mine are already in ulaw but was wondering if there was an advantage to transcoding them to another format... |
| 02:22.40 | p3nguin | Sure there is. |
| 02:23.03 | p3nguin | If you have files in other codecs, they won't be transcoded during the call. |
| 02:23.18 | phiscribe | later telphonyites |
| 02:23.23 | p3nguin | If you have a ulaw file and your call is g729, it'll transcode. |
| 02:23.27 | SeRi | damn callcentric does not respond to pings... bastards |
| 02:23.32 | *** part/#asterisk phiscribe (~phiscribe@98.157.3.94) |
| 02:23.47 | p3nguin | But if you have files in g729, the files will play g729 when possible, and no transcoding needs to occur. |
| 02:24.14 | F2Knight | is transcoding really that expensive? |
| 02:24.33 | p3nguin | That's a relative term. |
| 02:24.54 | p3nguin | If you have high call volume and/or high CPU load, yes it is. |
| 02:25.10 | p3nguin | If your box is basically idling, you'll never feel it. |
| 02:25.15 | *** join/#asterisk andresmujica (~andres@ubuntu/member/andresmujica) |
| 02:25.51 | F2Knight | what is considered ..'high' call volume, or CPU Load |
| 02:26.26 | p3nguin | We're still dealing with relative terms. High for me is 100% usage. |
| 02:26.44 | p3nguin | Also, if your CPU is very low-powered, transcoding is a bigger hit. |
| 02:27.06 | F2Knight | yea that would be high by any standard :) this box is a dual core pentium D 3.2Ghz w/4GB ram |
| 02:27.55 | F2Knight | and almost nothing load wise. but I am moving clients over to it and was wondering ... |
| 02:28.19 | F2Knight | current box is a 2.4ghz dual w/1gb ram. we run out of ram more then anything on it. |
| 02:28.45 | SeRi | F2Knight, Than I think you answer your own question |
| 02:29.06 | SeRi | Your new box should do ok |
| 02:29.07 | F2Knight | sure enough... don't worry about it :) |
| 02:29.24 | SeRi | Than dont make it public ;) |
| 02:29.31 | p3nguin | The Acrobits phone that is labeled as Softphone and says SIP on the icon behaves the same as the other Acrobits phone: does not register after prelim config, hard to figure out, etc. |
| 02:29.42 | F2Knight | but in the future .. is there any format that would be best to record prompts in .. one that lets say would make it easier to convert to other codecs if the need arrised. |
| 02:29.49 | SeRi | p3nguin, so far... any winners? |
| 02:30.04 | p3nguin | slin (sln) is the native format, if that helps at all. |
| 02:30.24 | p3nguin | iSip and Bria are both okay, too. |
| 02:30.42 | F2Knight | um... I suppose it does.. I could record off the phones as slin, and keep recording as wav from a microphone setup. |
| 02:30.50 | p3nguin | I like iPico and Media5-fone, but I need to make sure the choppy audio isn't related to the phones. |
| 02:32.01 | p3nguin | iSip has really loud audio. |
| 02:32.04 | F2Knight | Softphone, on my android and iphone I use 3CX, it has been more stable and easier to config then any others I have looked at. |
| 02:32.08 | SeRi | I liked Media5-Phone if I can only get pass the crashing... |
| 02:32.12 | F2Knight | on PC I use blink |
| 02:32.29 | SeRi | F2Knight, what version of android? |
| 02:33.29 | p3nguin | iSip doesn't chop, but those other two chop. It's probably not my battery causing it. |
| 02:33.34 | F2Knight | I run 2.3.4 cyanogen |
| 02:33.49 | SeRi | I am on 2.3.7 cyanogen 7.1.0.1 and it has native SIP. |
| 02:34.04 | SeRi | 2.4 should also have native sip |
| 02:34.10 | SeRi | I mean 2.3.4 |
| 02:34.12 | F2Knight | yea so does the whole 2.3 tree. |
| 02:34.27 | SeRi | why dont you use that instead? |
| 02:34.28 | p3nguin | hahahha |
| 02:34.34 | p3nguin | WiFone just pwned me. |
| 02:34.43 | SeRi | p3nguin, ? |
| 02:34.45 | F2Knight | but I like the 'app' because I can set 2 IPs, one for 'internal' one for 'external' |
| 02:34.48 | p3nguin | PIRACY WARNING |
| 02:34.53 | SeRi | LOL |
| 02:34.55 | SeRi | hahahah |
| 02:35.01 | p3nguin | This is a pirated version of the app and nothing will work. |
| 02:35.22 | SeRi | F2Knight, and you cant with native? |
| 02:35.47 | F2Knight | its also free, and is the same app for Android and Apple. Meaning I can create a simplier instruction set and not care what device people use. |
| 02:35.54 | SeRi | wonders how is it that he uses two the same way.... |
| 02:36.07 | F2Knight | Native is not as easy to configure for most end users. |
| 02:36.38 | SeRi | lol... sure. ok. |
| 02:36.45 | p3nguin | SIP Client doesn't even run for me right now, so I may have to pass on testing it. |
| 02:36.47 | SeRi | p3nguin, thats halarious |
| 02:41.02 | SeRi | Nice. to monitor up time of CC servers you have to use ping.callcentric.com |
| 02:41.19 | SeRi | which is a WHOLE different server than what I NEED to ping for.... |
| 02:41.21 | SeRi | bah... |
| 02:41.29 | p3nguin | vphone is okay, too, but not very featureful. |
| 02:41.54 | SeRi | p3nguin, have you try sipbri? |
| 02:42.09 | SeRi | I mean siphone |
| 02:42.11 | SeRi | lol |
| 02:42.13 | p3nguin | Can't. |
| 02:42.27 | SeRi | y? |
| 02:42.41 | p3nguin | It requires a lesser iOS than I currently have. Remember? |
| 02:43.07 | SeRi | o yes. :/ |
| 02:47.46 | p3nguin | SIAX is easy to set up, but introduces a lot of delay, and also is a little choppy. |
| 02:49.55 | SeRi | SIAX is under heavy development right? |
| 02:50.03 | p3nguin | Looks like a toss-up between iSip and Bria. |
| 02:50.12 | p3nguin | I'm not too sure about that. I never looked into it. |
| 02:50.14 | SeRi | ah I see |
| 02:50.24 | p3nguin | I just grabbed them all to test and review. |
| 02:50.35 | SeRi | cool |
| 02:50.44 | p3nguin | I think iSip will be my go-to soft phone. |
| 02:56.16 | LiuYan | installed asterisk via yum, compiled spandsp manually, but SendFax failed with: res_fax.c: Could not locate a FAX technology module with capabilities (SEND) |
| 02:56.38 | SeRi | Mhhhhh going to check it out. |
| 02:56.48 | SeRi | where did you grab it from? |
| 02:56.57 | SeRi | app store? |
| 02:58.11 | SeRi | p3nguin, did you try 3cx? |
| 02:58.31 | p3nguin | I don't remember finding it. |
| 02:58.36 | SeRi | 3CX did not work out for me..... I had lots of issues. |
| 03:03.32 | *** join/#asterisk kaushal (~kaushal@59.161.17.10) |
| 03:03.36 | kaushal | Hi |
| 03:09.20 | p3nguin | Hmm. I think I just jailbroke my iPhone from a Windows XP virtual machine. |
| 03:09.32 | F2Knight | Got a logic error here I think... its a multi-tenet issue |
| 03:09.41 | *** join/#asterisk luckman212_phone (~luckman21@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 03:09.46 | p3nguin | I didn't know it could work from a virtual machine. |
| 03:10.12 | F2Knight | All device are part of context=local-dids |
| 03:10.24 | SeRi | p3nguin, why wouldnt you be able? |
| 03:10.37 | p3nguin | redsn0w = bad ass |
| 03:10.44 | SeRi | Nice! |
| 03:10.53 | SeRi | you raped it? |
| 03:11.08 | F2Knight | local-dids has a lot of includes for other dialpaterns. |
| 03:11.09 | p3nguin | It popped up for just a few seconds with the emergency lock screen telling me to connect to iTunes. |
| 03:11.15 | p3nguin | I thought it was going to stop there. |
| 03:11.30 | p3nguin | But then it showed me the battery and said slide to unlock! |
| 03:11.38 | p3nguin | So I slid, and now I'm sitting on the home screen. |
| 03:11.38 | SeRi | sweet! |
| 03:11.53 | SeRi | nice :) |
| 03:12.01 | F2Knight | issue is that when client A has extension 102, and client B has a 102, how can I include another context.. only dynamicly... |
| 03:12.49 | F2Knight | I want to do something like includes { ael-mt-extensions-${ACCOUNTCODE} ; }; |
| 03:14.42 | F2Knight | this way when a caller dials 102 on there network they route to where they wanted.. idealy the dial command is Dial(local/${exten}@${accountcode}/n) as everyone in the multi-tenet should have there own context.. i think. |
| 03:14.57 | *** part/#asterisk andresmujica (~andres@ubuntu/member/andresmujica) |
| 03:15.28 | SeRi | is going to buy XO laptops for the kids. Christmas presents. |
| 03:16.24 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 03:18.02 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 03:19.04 | SeRi | http://www.rowetel.com/blog/?p=135 <---------- I want one! |
| 03:19.37 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176002135.dsl.bell.ca) |
| 03:19.56 | SeRi | waz up dijib |
| 03:20.00 | dijib | :D |
| 03:20.02 | dijib | hey man |
| 03:20.04 | dijib | tired |
| 03:20.05 | dijib | as usual |
| 03:20.17 | dijib | kindof unsatisfied from dinner whih was pissa |
| 03:20.18 | SeRi | lol |
| 03:20.19 | dijib | pizza |
| 03:20.23 | dijib | man im pooped |
| 03:20.29 | SeRi | in the face? |
| 03:20.30 | dijib | need a smoke |
| 03:20.36 | SeRi | lol |
| 03:20.41 | dijib | in what face? |
| 03:20.43 | SeRi | is hungry |
| 03:21.02 | p3nguin | I can't friggin' believe this. That was too easy. |
| 03:21.04 | SeRi | digiv, lol dont know... been drinking :) |
| 03:21.19 | SeRi | p3nguin, told you :) |
| 03:21.23 | SeRi | and no sim!!!!!!!!!!!!! |
| 03:21.31 | SeRi | You could have done this like 1month ago! |
| 03:22.15 | dijib | whats happenin? |
| 03:22.15 | p3nguin | I should take out the SIM and see if it cares. |
| 03:22.30 | dijib | it does |
| 03:22.36 | dijib | did you jailbreak it? |
| 03:22.42 | p3nguin | Yes. |
| 03:22.50 | dijib | still needs sim.. but yeah neat things eh |
| 03:22.53 | dijib | cydia |
| 03:22.57 | dijib | installeous |
| 03:23.12 | SeRi | I doubt it needs a sim. |
| 03:23.14 | p3nguin | I haven't installed installous yet, but I'll get there. |
| 03:23.21 | dijib | im fairly certain it does |
| 03:23.30 | dijib | SBsettings |
| 03:23.30 | p3nguin | I doubt it does. |
| 03:23.42 | dijib | think thinking it does |
| 03:24.02 | dijib | ive been trying to jailbrake this ipod 2g but im having no luck getting into dfu |
| 03:24.10 | dijib | or my cable is effed |
| 03:24.13 | dijib | well both of them |
| 03:24.32 | SeRi | dijib, I am sure it does not need the sim |
| 03:24.52 | dijib | ok then whatever |
| 03:25.00 | dijib | did you ever get that package? |
| 03:25.02 | dijib | p3nguin, ? |
| 03:25.41 | dijib | and would anybody be able to point me in the right direction of any Transmission Control examples pertaining to sip iax? |
| 03:26.39 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.166.29.160) |
| 03:26.57 | SeRi | dijib, I use IAX2 what transmission controls you are looking for? |
| 03:27.27 | dijib | sorry not transmission traffic control. |
| 03:27.41 | SeRi | like QoS? |
| 03:27.50 | dijib | i just dont know how your doing all this event based switching of the bandwidth plans |
| 03:28.02 | p3nguin | It's in the same category, but it's not QoS. |
| 03:28.04 | dijib | and p3nguin did you ever get to the bottom of that sync issue? |
| 03:28.35 | SeRi | dijib, I sont use QoS. Is traffic shapper |
| 03:28.46 | SeRi | s/is/use/ |
| 03:29.04 | dijib | but your traffic shape in what suite? |
| 03:29.26 | SeRi | pfsense |
| 03:29.28 | SeRi | bsd base |
| 03:29.54 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 03:30.01 | dijib | would something like this work for my scenario? |
| 03:30.02 | dijib | http://www.howtoforge.com/voip_qos_traffic_shaping_iproute2_asterisk |
| 03:30.13 | F2Knight | DTMF ... RFC vs INFO which is better? |
| 03:31.14 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 03:31.21 | ChannelZ | RFC |
| 03:32.19 | dijib | ok i think i need to filter the traffic rather than just allocating bandwidth like im doing now |
| 03:36.16 | SeRi | p3nguin, I got my TRIP ITAD :) |
| 03:39.53 | SeRi | dijib, what made you decide on that? |
| 03:40.46 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:42.04 | SeRi | now I am waiting for my freenum info :) |
| 03:43.52 | p3nguin | I see you at 1503. |
| 03:44.40 | p3nguin | Would you like to have my isn-outbound context? |
| 03:47.13 | *** join/#asterisk shmaltz (~chatzilla@ool-addc98d2.static.optonline.net) |
| 03:47.32 | *** join/#asterisk master_of_master (~master_of@p57B559B2.dip.t-dialin.net) |
| 03:48.56 | shmaltz | anyone here that knows elastix? |
| 03:49.24 | p3nguin | We've heard of it, but we don't support it. |
| 03:49.39 | shmaltz | p3nguin, thanks |
| 03:49.50 | shmaltz | anyhow, are you familiar with it? |
| 03:50.27 | shmaltz | is there a channel for that? |
| 03:50.37 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 03:50.38 | p3nguin | ~elastix |
| 03:50.38 | infobot | well, elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 03:51.19 | shmaltz | ty p3nguin |
| 03:51.25 | shmaltz | I knew I can rely on you :P |
| 03:53.36 | shmaltz | ~p3nguin |
| 03:53.36 | infobot | p3nguin strives to maintain his elitist reputation by refusing to use GUIs where they aren't beneficial. See: FreePBX. |
| 03:53.48 | shmaltz | lol |
| 03:53.55 | shmaltz | ~shmaltz |
| 03:57.24 | SeRi | p3nguin, yes. can I have it? |
| 03:57.30 | p3nguin | http://pastebin.com/gR5Ejtuy |
| 03:58.03 | SeRi | p3nguin, sweet! thanks! :) |
| 03:58.13 | SeRi | is ready for copypasta |
| 03:58.14 | SeRi | nice |
| 03:59.49 | *** join/#asterisk BuenGenio (~Gene@203.145.92.179) |
| 04:09.54 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 04:14.39 | p3nguin | siphon |
| 04:14.50 | SeRi | yes |
| 04:15.27 | SeRi | p3nguin, you are 1505? |
| 04:15.46 | SeRi | ops no.. lol |
| 04:15.48 | SeRi | never mind |
| 04:16.02 | *** join/#asterisk irroot (~gregory@41.51.171.129) |
| 04:16.17 | SeRi | p3nguin, you tring out siphon? |
| 04:16.24 | SeRi | trying |
| 04:16.49 | p3nguin | No, scroll up like 65 lines. |
| 04:17.08 | p3nguin | I'm going to try siphon in a minute. |
| 04:17.35 | p3nguin | I was looking for the 3CX phone, and I see something about Acrobits. I tried two Acrobits phones earlier, and they didn't register. |
| 04:17.35 | SeRi | you took out the sim? |
| 04:17.38 | p3nguin | no |
| 04:18.00 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 04:18.09 | SeRi | To me 3CX has issues. |
| 04:18.31 | SeRi | works ok for my brother in windows apparently... |
| 04:19.13 | SeRi | and F2Knight swears by it.... so I am probably doing something wrong or just plane does not like my firmware |
| 04:19.24 | p3nguin | Okay, I tried this one earlier. |
| 04:20.14 | p3nguin | (2029.31) <p3nguin> The Acrobits phone that is labeled as Softphone and says SIP on the icon behaves the same as the other Acrobits phone: does not register after prelim config, hard to figure out, etc. |
| 04:20.49 | SeRi | yes. I remember |
| 04:21.13 | p3nguin | And I don't know how to configure accounts for siphon. I started it, and it says to configure the SIP account in the siphon settings menu. I went to Settings, Siphon... and the page is completely blank. |
| 04:21.28 | SeRi | ah the nice bug |
| 04:21.42 | SeRi | you need sbssettings to get rid of the blank page bug |
| 04:22.49 | SeRi | http://code.google.com/p/siphon/issues/detail?id=406 |
| 04:23.39 | SeRi | download there version |
| 04:23.44 | SeRi | their* |
| 04:23.51 | p3nguin | This thing is a lot slower than my iPod. |
| 04:24.05 | p3nguin | I've already installed the sbsettings from bigboss. |
| 04:24.14 | p3nguin | Do I need to remove it and get a different one? |
| 04:24.32 | SeRi | http://code.google.com/p/siphon/issues/detail?id=406 <------ install the one from that place |
| 04:24.52 | p3nguin | Is that a yes? |
| 04:24.56 | SeRi | yes |
| 04:25.35 | SeRi | you need ssh enabled. |
| 04:25.49 | SeRi | once enabled scp the file to a temp folder. |
| 04:25.55 | SeRi | than dpkg -i file.deb |
| 04:26.04 | SeRi | reboot phone |
| 04:26.29 | SeRi | default root ssh password is "alpine" |
| 04:29.38 | *** join/#asterisk Akuma (~Akuma@modemcable131.103-179-173.mc.videotron.ca) |
| 04:30.08 | Akuma | hello, I have a couple of questions regarding asterisk and voip |
| 04:30.16 | p3nguin | ~ask |
| 04:30.16 | 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. |
| 04:30.27 | Akuma | I am looking to set it up for my business |
| 04:30.48 | Akuma | and I'm looking to have a single VOIP line, would I be able to do call queueing? |
| 04:31.09 | p3nguin | Let's start out by understanding that there are no lines in VoIP. |
| 04:31.36 | p3nguin | Other than that, you can do call queueing. |
| 04:32.29 | Akuma | Ok, here is what I want to do: receive the calls through my DID number over my voip service and be able to handle multiple calls at the same time that have been placed to that external number |
| 04:32.33 | Akuma | is that possible? |
| 04:32.45 | p3nguin | Sure. |
| 04:33.03 | p3nguin | The only limit is on how many channels your provider allocates to your DID. |
| 04:33.26 | Akuma | Oh ok, so it all depends on the number of channels for the DID |
| 04:33.49 | Akuma | what should I be looking for in a voip service provider in order to get this? |
| 04:33.50 | p3nguin | Some origination plans only provide two channels, where others are unlimited channels. |
| 04:34.39 | Akuma | I'm fairly new to this, so the information is quite overwhelming at this point and I've been getting lost on what is what and what is required |
| 04:35.19 | p3nguin | ~origination |
| 04:35.29 | p3nguin | ~termination |
| 04:35.36 | p3nguin | ~itsp |
| 04:35.36 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 04:36.20 | Akuma | ~itsplist-ca |
| 04:36.20 | infobot | methinks itsplist-ca is Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 04:36.28 | SeRi | p3nguin, you got it? |
| 04:36.52 | p3nguin | Not yet. This is slow. |
| 04:37.48 | p3nguin | Maybe an iPhone 4 is faster. I like their look better, anyway. |
| 04:38.22 | SeRi | I told you.... Now if you downgrade to 3.x it will be faster. |
| 04:38.27 | p3nguin | After I install openssh, how do I start it? |
| 04:38.36 | SeRi | it starts on its own |
| 04:38.40 | p3nguin | okay |
| 04:38.48 | SeRi | but you could use sbs toggle |
| 04:39.25 | p3nguin | Is that separate from sbsettings? |
| 04:39.29 | SeRi | I am caught on the same dilemma with my 3GS |
| 04:39.45 | SeRi | No. Enable the sbs hot key |
| 04:39.51 | SeRi | Its included by default |
| 04:40.03 | p3nguin | I don't know how. |
| 04:40.17 | SeRi | go to the settings scroll down to sbs settings |
| 04:40.30 | p3nguin | I removed sbsettings. |
| 04:40.32 | SeRi | and enable your toggle to activate the sbs menus. |
| 04:40.57 | SeRi | I have mine set to appear when slide the finger across the top |
| 04:41.04 | SeRi | ah. ok |
| 04:41.33 | SeRi | well than if you want to enable or disable at will you will need ether sbs or just a toggle ssh switch from cydia |
| 04:41.36 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-088-066-130-242.pools.arcor-ip.net) |
| 04:42.28 | SeRi | I liek sbs because it gives me /reboot/poweroff/ssh&wifi toggle/wifi IP/3G IP/etc.... |
| 04:44.07 | p3nguin | I'm on that code.google.com link... |
| 04:44.15 | p3nguin | What am I supposed to get from there? |
| 04:44.33 | Akuma | I'm looking at voip.ms site, but can't seem to find any details on the number of channels for the DID |
| 04:44.45 | SeRi | p3nguin, this: http://code.google.com/p/siphon/downloads/detail?name=Siphon-2.3rc4.deb&can=2&q=#makechanges |
| 04:45.03 | p3nguin | akuma: If you use the per minute DID, you'll get like 25 channels. |
| 04:45.17 | p3nguin | If you use the flat rate monthly rate, you get two channels. |
| 04:45.30 | Akuma | that means I'd be able to handle 25 calls simultanously? |
| 04:45.39 | p3nguin | inbound calls, yes. |
| 04:45.45 | p3nguin | Outbound calls will be separate. |
| 04:46.10 | Akuma | so I need a different kind of service for outbound calls? |
| 04:46.14 | p3nguin | seri: You told me to get sb settings from that link. This is a link to siphon. |
| 04:46.29 | p3nguin | akuma: You'll have termination service as soon as you activate your account with them. |
| 04:46.33 | SeRi | ooo no sorry I meant from cydia |
| 04:46.54 | p3nguin | I installed sb settings from cydia. Then you told me I need to remove it and install it from the link. |
| 04:46.57 | Akuma | Thank you for the info |
| 04:46.58 | p3nguin | So I removed it. |
| 04:47.03 | p3nguin | Then I couldn't find it on the link |
| 04:47.15 | Akuma | you've been very helpful, greatly appreciate it! |
| 04:47.16 | SeRi | hu? ok we got crossed in some bad wording.... |
| 04:47.23 | p3nguin | So can I install sb settings again via cydia from bigboss? |
| 04:47.25 | SeRi | I meant siphon. siphon is bad in cydia |
| 04:47.28 | SeRi | Yes |
| 04:47.51 | SeRi | sorry about that?!? :/ |
| 04:49.58 | p3nguin | Did you click on the download link? |
| 04:50.03 | p3nguin | for siphon |
| 04:50.24 | SeRi | p3nguin, yes. |
| 04:50.24 | p3nguin | It says the file has been deprecated and blah blah blah. Do you wish to continue? |
| 04:50.31 | SeRi | the problem is that there is no other verssion |
| 04:50.44 | SeRi | so I used that one. it works. |
| 04:51.24 | SeRi | just hit continue. |
| 04:52.21 | SeRi | I can tell that there is a bunch of kids hanging around #cyanogenmod.... Gezzzz... |
| 04:53.39 | *** part/#asterisk cbwest (~cbwest@nat/cisco/x-eeikjgqxbwcktfad) |
| 04:53.46 | Naikrovek | go hang out in #minecraft, lol. put sunglasses on, because the stupidity in there will blind you |
| 04:54.30 | Naikrovek | you'll get I-don't-want-to-live-on-this-planet-anymore syndrome in just a few minutes |
| 04:54.49 | SeRi | ROFL! |
| 04:56.36 | SeRi | d00d this kids have dick and ass stuck in there mouth.... You say something and you get a reply "you will get ass raped for that" like wtf? I am just idling looking at this idots talk shit to each other |
| 04:56.44 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 04:59.27 | SeRi | wtf |
| 04:59.49 | SeRi | am i still in or did I split? |
| 04:59.56 | p3nguin | You're here. |
| 05:00.14 | SeRi | d00 my modem went haiwire for a sec |
| 05:06.03 | p3nguin | Siphon doesn't want to register either. |
| 05:06.47 | SeRi | to your system? |
| 05:07.03 | p3nguin | It never touches asterisk. |
| 05:07.03 | SeRi | works perfect for me |
| 05:07.24 | SeRi | weird.... |
| 05:07.27 | SeRi | works ok here |
| 05:11.57 | p3nguin | I figured it out. |
| 05:12.22 | p3nguin | It doesn't like a DNS host name for some stupid reason. I put in the IP address of asterisk and it works. |
| 05:13.14 | SeRi | ah I see. |
| 05:13.25 | SeRi | I am about had it with damn ast linux |
| 05:13.31 | SeRi | astlinux* |
| 05:19.33 | SeRi | p3nguin, It worked for me when registering to voip.ms |
| 05:19.39 | SeRi | maybe something internal? |
| 05:19.46 | p3nguin | It doesn't like my internal DNS name for asterisk. |
| 05:19.49 | SeRi | Than again I tried it away from home. |
| 05:19.56 | SeRi | I see |
| 05:20.47 | SeRi | how do you like it? |
| 05:21.29 | SeRi | by the way what sim are you using? |
| 05:21.49 | p3nguin | I put an AT&T SIM in it. |
| 05:21.56 | SeRi | Ah I see. |
| 05:22.06 | SeRi | Try taking it out when you get a chance |
| 05:23.46 | SeRi | d00d shit is going crazy around here... :/ |
| 05:23.54 | SeRi | my cell just rebooted |
| 05:24.31 | SeRi | weird shit :/ |
| 05:33.04 | p3nguin | iSip is suffering from the same no DNS issue that Siphon did. |
| 05:33.17 | p3nguin | I couldn't get iSip to work, either, so I put in the IP address and it worked. |
| 05:33.30 | p3nguin | Not sure what the DNS problem is. |
| 05:36.11 | SeRi | weird... |
| 05:36.21 | SeRi | internal dns? |
| 05:36.27 | p3nguin | yes |
| 05:37.03 | SeRi | Mhhhhhhhhhh...... I use unbound. Let me try and replicate it |
| 05:37.37 | p3nguin | Keep in mind, I used iSip on my iPod and host name works fine. |
| 05:37.58 | SeRi | p3nguin, did you had to do anything with dns to get ITAD numbers? |
| 05:38.14 | p3nguin | You get your ITAD domain from IANA. |
| 05:38.20 | SeRi | I am reading their instrcutions |
| 05:38.41 | p3nguin | Then you can configure your DNS in freenum.org. |
| 05:38.51 | p3nguin | Then you configure your ISN in your asterisk. |
| 05:39.43 | p3nguin | I don't see the openssh toggle in sbsettings. Do I need to install something else to get it? |
| 05:39.46 | SeRi | ok so I got the ITAD number, now waiting on freenum password |
| 05:40.46 | SeRi | p3nguin, "set windows toggles" |
| 05:41.19 | p3nguin | I have switched SSH from off to on. |
| 05:41.21 | p3nguin | Now what? |
| 05:41.32 | SeRi | now ssh to the ip of the phone as root |
| 05:41.38 | SeRi | it will take longer tahn usual |
| 05:41.47 | SeRi | because is generating the keys as you log in |
| 05:41.48 | p3nguin | Then what? |
| 05:42.17 | SeRi | now you ahve access to install .deb files |
| 05:42.24 | p3nguin | You're sending me down the wrong road again. I'm just asking how to turn on and off ssh. |
| 05:42.33 | SeRi | o shit |
| 05:42.37 | SeRi | sorry |
| 05:43.09 | SeRi | ok now set your activation method |
| 05:43.30 | SeRi | default should be fine |
| 05:44.08 | SeRi | but se what you would like |
| 05:44.14 | p3nguin | Just pressing the green ssh icon to make it red switches it off? |
| 05:44.40 | SeRi | switche the toggle option off |
| 05:44.47 | SeRi | meaning you wont see it on your toolbar |
| 05:45.08 | p3nguin | I don't want that. I want to be able to turn on and off the ssh service. |
| 05:45.40 | SeRi | yes Ia m talling you how. |
| 05:45.54 | p3nguin | I think you're sending me down the wrong road again. |
| 05:45.59 | SeRi | d00d call me |
| 05:46.01 | SeRi | please |
| 05:46.04 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-zkipsjolnmyrsuxg) |
| 05:46.23 | SeRi | you got my cc id |
| 05:46.27 | SeRi | just hit option 1 |
| 05:46.33 | p3nguin | I'm going to be very clear and specific so there isn't more confusion. |
| 05:46.52 | p3nguin | I enabled the toggle for ssh. It was off; I turned it on. |
| 05:47.39 | p3nguin | I used the default activation method to bring up the settings thing at the top with respring, power, etc. |
| 05:47.46 | SeRi | ok good. now go to "Activation methods" and choose an activation method to bring the sbs settings forward from any screen |
| 05:47.52 | p3nguin | In the box was only a big green SSH icon. |
| 05:47.54 | SeRi | ok good |
| 05:47.56 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-wrdlkcumfjlignbf) |
| 05:47.59 | SeRi | Yes! |
| 05:48.04 | SeRi | That means ssh is enable :) |
| 05:48.06 | p3nguin | I pressed it, and the big green SSH icon became a big red SSH icon. |
| 05:48.11 | SeRi | you hit that and it turns red |
| 05:48.17 | p3nguin | (2344.14) <p3nguin> Just pressing the green ssh icon to make it red switches it off? |
| 05:48.20 | SeRi | that means disable |
| 05:48.22 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-zlutwiekxddrxaxv) |
| 05:48.46 | p3nguin | That's what I asked nearly five minutes ago. |
| 05:49.01 | SeRi | well I thought you where still under the toggle options |
| 05:49.25 | SeRi | sorry |
| 05:52.13 | *** join/#asterisk timahvo1 (~rogue@197.179.38.171) |
| 05:52.45 | p3nguin | Now I just have to figure out why the iPhone doesn't like my DNS. |
| 05:53.37 | ChannelZ | Steve does not approve of your DNS |
| 05:54.15 | SeRi | lol |
| 05:55.29 | SeRi | p3nguin, I just try mine with the dns of my server internally and it worked |
| 05:55.34 | SeRi | I run unbound dns. |
| 05:55.56 | SeRi | dont just use the hostname. add the domain as well |
| 05:56.03 | SeRi | in my case pbx.redes |
| 05:56.31 | p3nguin | When I said hostname, I meant the FQDN. |
| 05:56.52 | SeRi | I see. |
| 05:57.51 | SeRi | freaky video http://www.youtube.com/watch?v=K_2Z32qCYBg <----- |
| 06:01.44 | SeRi | ok g/n guys. p3nguin hit me up if you need help... I am sure I have caused you more headache than help.... |
| 06:01.48 | ChannelZ | Sluts on poles |
| 06:01.53 | SeRi | lmao |
| 06:02.10 | ChannelZ | This makes me want to stab my ear holes out |
| 06:02.15 | SeRi | lol |
| 06:02.33 | SeRi | I take you dont like dubstep |
| 06:02.36 | SeRi | :) |
| 06:02.41 | SeRi | well I am out guys. |
| 06:02.43 | SeRi | g/n |
| 06:04.21 | ChannelZ | I guess there is something worse than dance music |
| 06:05.41 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:10.33 | p3nguin | Well wtf. It's only a problem on the iPhone. |
| 06:11.55 | p3nguin | The DNS server, the iPod, the D-Link SIP phone, and the laptop all know asterisk.local. |
| 06:19.03 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 06:26.00 | *** join/#asterisk AmirBehzad (~behzad@31.184.187.1) |
| 06:43.36 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za) |
| 06:49.56 | *** join/#asterisk neurosys (~neurosys@c-67-191-66-234.hsd1.fl.comcast.net) |
| 06:50.08 | neurosys | is the meetme module gone from 1.8? |
| 07:00.08 | kaldemar | neurosys: no |
| 07:01.34 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 07:01.41 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 07:05.11 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 07:13.02 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 07:22.58 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:23.31 | *** join/#asterisk BuenGenio (~Gene@203.145.92.22) |
| 07:24.39 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 07:25.49 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 07:25.50 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 07:31.22 | *** join/#asterisk irroot (~gregory@196-215-57-105.dynamic.isadsl.co.za) |
| 07:34.18 | *** join/#asterisk nicola_pav (~chatzilla@62.219.238.196) |
| 07:35.34 | nicola_pav | hello. anyone familiar with Fax for asterisk? I have purchased 3 fax channels. I need to enable fax detection on a sip trunk. any hints please? |
| 07:36.54 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 07:39.49 | p3nguin | sip.conf : faxdetect=yes |
| 07:40.29 | nicola_pav | p3nguin: enabled it already. but no fax detection |
| 07:40.43 | p3nguin | That's all there is to fax detection. |
| 07:40.57 | nicola_pav | faxdetect is for fax detection over PRI |
| 07:41.05 | nicola_pav | what about fax detection over a sip trunk? |
| 07:41.08 | p3nguin | That's a negative. |
| 07:41.18 | nicola_pav | what do u mean? |
| 07:41.22 | p3nguin | (0139.49) <p3nguin> sip.conf : faxdetect=yes <---- this |
| 07:51.16 | nicola_pav | what about NVFax? |
| 07:51.37 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 07:54.35 | irroot | nicola_pav fax whats that ?? |
| 07:54.56 | ChannelZ | fax detect works fine over SIP for me, ulaw/alaw |
| 07:55.16 | nicola_pav | irroot: digium has FFA. ie.e fax for asterisk |
| 07:55.25 | irroot | there is in trunk a faxdetect framehook that will work with transcoding |
| 07:55.30 | nicola_pav | ChannelZ: i need to detect fax over a sip trunk |
| 07:55.42 | ChannelZ | yeah.. it works |
| 07:56.13 | ChannelZ | again only if you're using ulaw/alaw, if you're using gsm or g729 it ain't going to work (detect OR the fax its self) |
| 07:56.24 | *** join/#asterisk Namiq (~Namiq@217.25.22.67) |
| 07:57.25 | Namiq | hi, could anyone suggest best softphone for asterisk? |
| 07:58.06 | ChannelZ | What OS? |
| 07:58.06 | p3nguin | Windows, Mac, Linux, or other? |
| 07:58.15 | ChannelZ | uses Zoiper and is reasonably pleased with it |
| 07:58.30 | irroot | nicola_pav detecting fax and using fax are 2 seperate animals over a sip trunk you should only be using T.38 |
| 07:58.33 | Namiq | for windows, for mobile phones |
| 07:58.41 | IsUp | Namiq: eyeBeam, X-Lite, Zoiper for Win |
| 07:58.49 | Namiq | thanks |
| 07:59.07 | p3nguin | I prefer Zoiper (classic) on Windows PC. Never looked into Windows Mobile. |
| 07:59.08 | irroot | likes blink on ubuntu lightweight not rich in features |
| 07:59.47 | irroot | for windows mobile load android and take your pick 2.3 comes with sip built in |
| 08:01.40 | ChannelZ | it's not particularly good but it works |
| 08:06.49 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:07.56 | LiuYan | Namiq: try PhonerLite: http://www.phonerlite.de/download_en.htm |
| 08:15.37 | *** join/#asterisk Cadey (5a980315@gateway/web/freenode/ip.90.152.3.21) |
| 08:16.18 | Cadey | Hi guys, I cant remember a feture that im almost certain I used with asterisk. I have a memory of being able to upload in a text file a command like, Extension : 1001, Dial : 91111 etc |
| 08:16.33 | Cadey | and then asterisk would pick it up (file monitor, cron job etc) and action the command |
| 08:16.43 | Cadey | but for the life of me cant remember the name of the feature |
| 08:16.58 | bulkorok | that are call files |
| 08:17.11 | *** join/#asterisk tehrabbitt-droid (~tehrabbit@unaffiliated/tehrabbitt) |
| 08:17.22 | bulkorok | http://www.the-asterisk-book.com/unstable/call-file.html |
| 08:17.34 | Cadey | thats the one :) |
| 08:17.35 | Cadey | thank you |
| 08:17.38 | bulkorok | :) |
| 08:17.54 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 08:18.06 | tehrabbitt-droid | hey quick question, what would cause this error when trying to dial out: -- Locally bridging SIP/1100-0000006a and SIP/FlowroutePersonal-0000006b == Spawn extension (PersonalLines, 201XXXXXXX, 2) exited non-zero on 'SIP/1100-0000006a' |
| 08:18.07 | Cadey | do you happen to know if there is a way to puch config changes like that, incase for some stupid reason I cannot get onto AMI or SSH |
| 08:18.16 | tehrabbitt-droid | it rings, and then hangs up |
| 08:18.17 | Cadey | *push |
| 08:18.48 | Cadey | other that building a config changer into the dial plan :) |
| 08:18.57 | bulkorok | not really no |
| 08:19.05 | Cadey | kk :) |
| 08:19.30 | bulkorok | ami, dialplan, ssh... maybe the internal http-server... |
| 08:19.42 | ChannelZ | tehrabbitt-droid: you'd have to look at a SIP debug and see what is going on, if your local device is rejecting for some reason, etc. |
| 08:20.46 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 08:21.13 | tehrabbitt-droid | ChannelZ: what would it look like if it was being rejected |
| 08:21.50 | ChannelZ | ?? Could be any number of things |
| 08:22.07 | ChannelZ | pastebin the output of a test call if you want us to take a look. |
| 08:22.37 | Cadey | I do forget that AMI has a http/javascript version |
| 08:24.16 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 08:24.37 | tehrabbitt-droid | hm now theres no audio when it connects :-\ |
| 08:24.40 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:24.49 | tehrabbitt-droid | it must be NAT related :-\ |
| 08:25.58 | ChannelZ | Did you setup externip and localnet properly in sip.conf? |
| 08:28.31 | tehrabbitt-droid | i should check that |
| 08:30.38 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:31.36 | tehrabbitt-droid | externip fixed it! |
| 08:31.37 | tehrabbitt-droid | :-D |
| 08:31.41 | tehrabbitt-droid | thanks ChannelZ |
| 08:31.44 | tehrabbitt-droid | overlooked it X_X |
| 08:31.57 | ChannelZ | Glad you have it running |
| 08:33.58 | *** part/#asterisk tehrabbitt-droid (~tehrabbit@unaffiliated/tehrabbitt) |
| 08:34.08 | *** join/#asterisk tehrabbitt-droid (~tehrabbit@unaffiliated/tehrabbitt) |
| 08:34.17 | tehrabbitt-droid | newho thanks again ChannelZ |
| 08:34.19 | tehrabbitt-droid | night! |
| 08:34.19 | tehrabbitt-droid | :-D |
| 08:34.24 | *** part/#asterisk tehrabbitt-droid (~tehrabbit@unaffiliated/tehrabbitt) |
| 08:34.38 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 08:36.15 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 08:36.37 | joobie | hey boys.. anyone know a model for the cheaper cisco switches with 48port poe? |
| 08:37.01 | Cadey | 3648 ? |
| 08:37.37 | LiuYan | hi, is there any reason that AsteriskGUI http://asterisk-server:8088/static/config/index.html keep reloading? |
| 08:37.50 | Cadey | Cisco 3448 even |
| 08:38.17 | Cadey | we have used 3448 POE's in teh past |
| 08:38.32 | Cadey | youcan get them qutie cheap as refurbs which is the route we took |
| 08:38.50 | Cadey | pick them up from £500 to £750 |
| 08:39.11 | Cadey | joobie ^^^ |
| 08:39.24 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:40.08 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:40.19 | *** join/#asterisk Nasga (~Nasga@110.113.117.78.rev.sfr.net) |
| 08:41.17 | Cadey | unless you mean the "linksys" range when you say cheap cisco |
| 08:43.34 | *** join/#asterisk freetown2 (~freetown2@ipvpn098241.netvigator.com) |
| 08:44.21 | freetown2 | hi all, i have a problem with Dial not honoring the r option...i get silence |
| 08:44.31 | ChannelZ | Yeah I guess it depends on your definition of 'cheap' |
| 08:44.39 | freetown2 | it worked before though... |
| 08:44.51 | ChannelZ | a 48 port PoE I don't think is cheap from anyone |
| 08:45.38 | freetown2 | and one that does not need an extra power supply to make sure all 48 ports get juice |
| 08:45.40 | ChannelZ | freetown2: perhaps your channel is answering |
| 08:45.40 | Cadey | well you cna pick up 48 port POE linksys for a couple of hundread |
| 08:45.57 | ChannelZ | Cadey: with all 48 having power? |
| 08:46.09 | ChannelZ | And not crashing with them all using it? |
| 08:46.11 | Cadey | freetown2 : as long as your putting 5watt phones in most 48 port POE's will be fine |
| 08:46.39 | freetown2 | ChannelZ, channel is answering? which one? the called one? |
| 08:46.39 | Cadey | but by default cisco push 14.5 watts (i think) when the POE power negotiation fails, which it tends to do when you plug in a non-cisco phone |
| 08:46.46 | ChannelZ | yeah |
| 08:47.08 | Cadey | so you have to change the power rating for each port on the switch, takes 2 seconds but annoying when you first experience it :) |
| 08:47.38 | freetown2 | ChannelZ, can't be...the phone is still ringing but the calling phone just get silence |
| 08:48.02 | ChannelZ | What does the console say about the call progress? What kind of channels are these? |
| 08:48.28 | freetown2 | incoming sip -> sip trunk to ucm |
| 08:48.35 | ChannelZ | Having to use 'r' in the first place is usually due to broken call progress indications anyway. |
| 08:50.52 | freetown2 | well, i had to Answer the call to get the sip line identifier to tie it to a particular ext on the ucm... |
| 08:51.11 | ChannelZ | so? |
| 08:52.10 | freetown2 | so...the call progress indicator from the ucm side things is broken...is that what you are saying? |
| 08:52.37 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 08:53.46 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:6110:ec21:d6ac:2537) |
| 08:54.44 | ChannelZ | well lets say besides a bug, the only reason I can think of it's not working is because the remote end is reporting an answer when it hasn't. |
| 08:55.15 | freetown2 | well, it used to work until the ucm system got reboot due to power failure... |
| 08:55.31 | freetown2 | so it is highly likely something on that side of things... |
| 08:55.38 | freetown2 | i'll try digging there |
| 08:55.51 | ChannelZ | re: Have you looked at the console and the call progress it reports vs. reality? |
| 08:56.47 | ChannelZ | what's on the other side of this UCM? (what kind of devices) |
| 08:58.06 | freetown2 | cisco phones |
| 08:58.29 | freetown2 | <PROTECTED> |
| 08:59.46 | ChannelZ | Well check the console. If the UCM is reporting the channel as answered via SIP to Asterisk though it's really still ringing the device on the other side, there's not a lot you can do about it. But you've got to see if that's whats happening first |
| 09:00.50 | *** join/#asterisk BuenGenio (~Gene@203.145.92.22) |
| 09:01.51 | *** join/#asterisk happylife (~happylife@212.92.145.7) |
| 09:02.53 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
| 09:06.26 | freetown2 | ChannelZ, console does not report anything from ucm apart from an INVITE and later a BYE when i hung up |
| 09:08.33 | ChannelZ | with verbose up a little, you don't see anything like 'SIP/xxxx is making progress....' and 'SIP/xxx answered SIP/yyyy' and such? |
| 09:09.55 | freetown2 | only SIP/180 Ringing from asterisk <-> ucm. I do see stuff like SIP/2.0 200 OK for asterisk <-> sip provider though |
| 09:11.19 | freetown2 | the next asterisk <-> ucm message is a CANCEL... |
| 09:20.17 | *** join/#asterisk eicto (~morgoth@eicto.broker.freenet6.net) |
| 09:24.30 | *** join/#asterisk F2Knight (~ben@c-24-22-60-186.hsd1.or.comcast.net) |
| 09:24.51 | F2Knight | Is anyone using or has used red phone, foneBRIDGE? |
| 09:26.34 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 09:29.50 | joobie | Cadey, i coudlnt find the 3448 |
| 09:29.57 | joobie | they may be discontinued? |
| 09:30.01 | joobie | is there a current model?? |
| 09:30.02 | Cadey | possibly :S |
| 09:30.06 | joobie | ps dont mind the cheaper "linksys" |
| 09:30.55 | joobie | hmm can u drop the power rating to 5w? |
| 09:31.12 | joobie | i was reading into the 802.af standard and it was saying it's up to 14.5w |
| 09:32.31 | Cadey | try looking at the 2948 |
| 09:33.07 | Cadey | joobie, yes you can set the power on the ports on cisco switches |
| 09:33.21 | Cadey | we had to set them to 5watts for our aastra phones |
| 09:33.37 | joobie | looks like 2948 is EOL |
| 09:34.19 | WiretapSeven | lol wut..... yes |
| 09:34.23 | WiretapSeven | the 2948 is ancient |
| 09:34.36 | WiretapSeven | the 3448 is also ancient |
| 09:34.46 | WiretapSeven | joobie, you're after a Cisco switch that does what exactly? |
| 09:35.19 | WiretapSeven | if you want 'linksys' type, look at perhaps the Cisco SG300 series PoE switches |
| 09:35.36 | joobie | i want 48 poe ports |
| 09:35.43 | joobie | the SG range would be good |
| 09:35.48 | joobie | but i think they all are 24port |
| 09:35.49 | joobie | poe |
| 09:35.52 | WiretapSeven | ok, one sec |
| 09:36.28 | WiretapSeven | 2960S-48FPD-L for full-power PoE on all ports (Layer 2 switch) |
| 09:36.46 | WiretapSeven | or 2960S-48LPD-L for standard power PoE on all ports (also Layer 2) |
| 09:37.01 | WiretapSeven | those are both brand new devices, currently in production |
| 09:37.10 | WiretapSeven | oh, wait |
| 09:37.15 | WiretapSeven | those are the ones with tengig uplinks |
| 09:37.33 | WiretapSeven | you want either 2960S-48FPS-L or 2960S-48LPS-L |
| 09:38.07 | joobie | I wanted to run SPA502G's |
| 09:38.12 | joobie | would they require full-power? |
| 09:38.25 | joobie | like what is "full power" vs "standard power" ? 14.5 vs 5w? |
| 09:38.29 | WiretapSeven | 2960-48PST-L <-- this is the 100mbit version, standard power |
| 09:38.32 | WiretapSeven | joobie, oen sec |
| 09:39.01 | WiretapSeven | the difference is that the full power ones can deliver 15.4w to all ports |
| 09:39.17 | joobie | fark |
| 09:39.19 | joobie | they are not cheap |
| 09:39.29 | joobie | 2700$AUD is the cheapest for the 2960-48PST-L |
| 09:39.31 | WiretapSeven | and the standard power can provide it to some ports |
| 09:39.36 | WiretapSeven | yep, Cisco stuff is not cheap |
| 09:39.45 | joobie | what about the small business range? |
| 09:39.54 | joobie | do they offer the full 48 port poe in that range? |
| 09:40.06 | WiretapSeven | that doesn't include licensing or anything |
| 09:40.07 | WiretapSeven | hang a tick |
| 09:40.11 | joobie | thanks man |
| 09:40.17 | joobie | appreciate the hand btw |
| 09:40.32 | joobie | i have about 30 cisco tabs open atm with all my research :P |
| 09:41.44 | WiretapSeven | ESW-520-48P-K9 <-- standard power smallbusiness Cisco (not linksys) solution |
| 09:41.54 | Cadey | WiretapSeven : you missed the part he requested cheap cisco, hence the "old" switches I suggested |
| 09:41.58 | WiretapSeven | only 100mbit |
| 09:42.07 | WiretapSeven | Cadey, he also requested still in production :P |
| 09:42.23 | joobie | WiretapSeven, what about a linksys solution? |
| 09:42.27 | joobie | ESW-520-48P-K9 is $2k |
| 09:42.34 | WiretapSeven | the SG series (linksys) do not include 48 port PoE |
| 09:42.45 | WiretapSeven | your other option is, hang fire a moment |
| 09:42.47 | joobie | .. but when you say "standard power", do you mean i can only get 24 of those ports to run POE? |
| 09:42.54 | joobie | or I can run all 48 but with lower W |
| 09:43.05 | joobie | thanks |
| 09:43.09 | WiretapSeven | all 48 can |
| 09:43.13 | WiretapSeven | just not full power |
| 09:43.23 | *** join/#asterisk defswork (~andy@mx2.3gcomms.co.uk) |
| 09:43.31 | WiretapSeven | the 3560-48PS and 3560G-48PS |
| 09:43.41 | WiretapSeven | are now out of production, but near-new models |
| 09:43.46 | WiretapSeven | that meet your requirements |
| 09:43.49 | WiretapSeven | and should be sub 1k |
| 09:44.29 | WiretapSeven | you'll have to pick them up refurb or off ebay though |
| 09:45.03 | joobie | im seeing 3k and 5k for those respectively |
| 09:45.08 | joobie | ahh |
| 09:45.21 | joobie | this is probably not the best reference but - http://www.ebay.com/itm/NEW-Cisco-SPA-502G-1-Line-IP-Phone-Display-PoE-and-/400236500950 |
| 09:45.25 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 09:45.32 | joobie | this says the SPA502G drives by 5W |
| 09:45.50 | joobie | given this is true and i can go for a standard power |
| 09:45.50 | freetown2 | 3560 out of production?! |
| 09:45.53 | joobie | which would u recommend? |
| 09:45.57 | WiretapMac | don't look at the 3560-X or 3560S, those are in-production switches that cost a bomb |
| 09:46.12 | WiretapMac | freetown2: I believe the X and S are the current production models |
| 09:46.31 | freetown2 | so..what's a 3560V2? |
| 09:46.36 | WiretapMac | a V2 |
| 09:46.49 | WiretapMac | a model I missed when noting current production models :P |
| 09:47.06 | freetown2 | oh...phew... |
| 09:47.06 | WiretapMac | joobie: I would recommend on that front that you pick a standard-power one lol |
| 09:48.01 | joobie | WiretapMac, if i want cheap as chips CISCO or Linksys to drive those phones in a 48port poe standard-power, which would you recommend? preferrably if it had QoS so i can prioritize voip traffic? the spa502G has a LAN port so the LAN traffic would route down the same wire |
| 09:48.15 | WiretapMac | they all have QoS |
| 09:48.21 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 09:48.23 | schmidts | good morning |
| 09:48.29 | WiretapMac | but if you want 'cheap as chips' you're going to have ot pass up the Cisco :P |
| 09:48.42 | joobie | that's OK |
| 09:48.51 | Cadey | LinkSyS is cheap as chips however |
| 09:48.58 | WiretapMac | Cadey: not quite, Linksys is now |
| 09:49.02 | WiretapMac | 'Cisco Small Business' |
| 09:49.04 | WiretapMac | and still costs a bit |
| 09:49.08 | joobie | what would do standard poe over 48p in the linksys or cisco smallbusiness? |
| 09:49.11 | Cadey | and under writen by cisco now and performace has made massive gains |
| 09:49.12 | WiretapMac | you might have to look at something like netgear |
| 09:49.19 | Cadey | yeah but they are still really chea[ |
| 09:49.27 | Cadey | ouch, netgear :( |
| 09:49.29 | WiretapMac | not in the southern hemisphere |
| 09:49.33 | joobie | what would the netgear alternative be? |
| 09:49.34 | joobie | why ouch? :P |
| 09:49.36 | Cadey | they crash like a bitch in my experience |
| 09:49.41 | joobie | ahh |
| 09:49.45 | WiretapMac | yep |
| 09:49.46 | WiretapMac | they do |
| 09:49.49 | Cadey | dont like constant throughput |
| 09:49.50 | WiretapMac | have funky spanning-tree issues |
| 09:49.55 | joobie | i use the FS118/6 (8port poe) and it's stable |
| 09:49.57 | WiretapMac | which is fine, if you only have one or two switches |
| 09:50.34 | joobie | so in the cisco small business (linksys) range.. what would the cheapest option be for 48p standard |
| 09:50.42 | WiretapMac | they don't do any |
| 09:50.53 | WiretapMac | you'd need 2x 24p |
| 09:50.59 | WiretapMac | which will probably be cheaper than 1x 48p |
| 09:51.20 | joobie | which model Wiretap? |
| 09:51.23 | WiretapMac | for soem reason, 48 port switches always cost a bomb |
| 09:51.36 | joobie | btw.. is this one good for the 48p full poe?? http://www.ebay.com.au/itm/Cisco-WS-C3560-48PS-S-L3-PoE-switch-/230705489908?pt=AU_Networking&hash=item35b71e67f4 |
| 09:51.44 | WiretapMac | joobie: anything that does PoE across 24 ports can do 'full power' PoE generally |
| 09:52.09 | WiretapMac | joobie: it isn't a 'full power' swithc, but it'll deliver 7.5w across all ports |
| 09:52.18 | WiretapMac | or 15.4 across 24 |
| 09:53.45 | joobie | hey Wiretap |
| 09:54.02 | joobie | what do you think about that POE swithc on ebay |
| 09:54.08 | joobie | decent price? that's full poe right? |
| 09:54.23 | WiretapMac | it is standard PoE, but yes does have PoE on all 48 ports |
| 09:54.28 | WiretapMac | and yes, seems to be decent price |
| 09:54.35 | WiretapMac | the 24PS sells for around that here at times |
| 09:54.44 | joobie | ahh |
| 09:54.50 | WiretapMac | (I'm in NZ) |
| 09:56.10 | joobie | ahh |
| 09:56.12 | joobie | NZ'er |
| 09:56.27 | joobie | i went to a NZ'ers christening last weekend |
| 09:56.32 | joobie | there were some big ass NZ'ers there |
| 09:56.34 | irroot | WiretapMac we have used the new "Cisco" branded Linksys switch its pretty decent |
| 09:56.42 | WiretapMac | yep, they're not crap |
| 09:56.45 | WiretapMac | the UI is annoying |
| 09:56.56 | joobie | WiretapMac, do u reacon i should i buy that switch? |
| 09:57.04 | joobie | is it a "mad" price or just an average one |
| 09:57.12 | WiretapMac | joobie: I don't know the aussie cisco market |
| 09:57.20 | joobie | it must be like NZ |
| 09:57.22 | WiretapMac | other than that things are generally cheaper in NZ |
| 09:57.30 | joobie | serious? |
| 09:57.35 | WiretapMac | yep, don't know why |
| 09:57.37 | joobie | how much could u get that switch for in NZ |
| 09:57.49 | WiretapMac | not that cheap :P |
| 09:57.53 | joobie | heh |
| 09:57.58 | joobie | that is a refurb tho |
| 09:58.24 | WiretapMac | to give you some idea: http://www.ebay.com.au/itm/Cisco-2811-Integrated-Service-Router-/200679437817?pt=AU_Networking&hash=item2eb96d35f9#ht_500wt_1156 I bought one of these for NZ$350 a couple weeks back |
| 09:59.26 | WiretapMac | in my opinion, that is a pretty good price for that switch |
| 09:59.34 | WiretapMac | but it isn't my money and I'm not about to tell you how to spend it :P |
| 10:02.43 | Greenlight | Does anyone know what would cause a Local/ channel to not correctly hangup. I'm using it to originate a call between Chanspy and an application in the dialplan, and when the dialplan side hangs up, i'm left with just one half of the local channel (the ";1" part) which I'd have expected to hangup as well |
| 10:03.09 | joobie | ahh |
| 10:03.13 | joobie | i prefer linux as a router |
| 10:03.17 | joobie | rather than cisco |
| 10:03.19 | joobie | it's more flexible |
| 10:03.22 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 10:03.37 | joobie | so wire, i dropped that dood an email |
| 10:03.41 | joobie | asking him to bring it down to $750 |
| 10:03.51 | joobie | but if he doesnt do this.. what would u recommend given i need 48p and 5w |
| 10:03.54 | joobie | cheap as chips |
| 10:04.02 | joobie | i need a fallback solution:P |
| 10:04.32 | WiretapMac | joobie: start looking at other brands I guess :P |
| 10:04.47 | WiretapMac | and linux more flexible? I don't really see how tbh |
| 10:06.26 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 10:06.52 | dym | Hey all - just a quick opinion: Asterisk and Video Conferencing: doable? much pain? dont even consider it? |
| 10:07.11 | joobie | WiretapMac, what brand would u suggest is OK? I woulda went towards netgear but now im turned off... |
| 10:07.20 | defswork | dym, no one wants to see your face |
| 10:07.27 | WiretapMac | joobie: I'm not a 'cheap brands' guy, sorry |
| 10:07.32 | WiretapMac | I'm a juniper/cisco engineer |
| 10:07.38 | joobie | linux is heaps more fleixble, with all the modules u can add to it.. it opens up the optiosn beyond just access-lists |
| 10:07.48 | joobie | ahh |
| 10:07.54 | joobie | junipers are gay |
| 10:08.01 | defswork | joobie, hes paid 3 times as much for delivering half the system ;) |
| 10:08.01 | joobie | we had to support one of them |
| 10:08.05 | WiretapMac | joobie: I'm gay :P |
| 10:08.06 | joobie | it's so obscure |
| 10:08.14 | WiretapMac | and access lists? wut? |
| 10:08.19 | WiretapMac | that isn't all there is to it |
| 10:08.24 | joobie | i mean obscure to the point that u think they are making it obscure to make ti nicshe |
| 10:08.31 | WiretapMac | lol |
| 10:08.44 | WiretapMac | my main complaint with Juniper is the inability to bloody copy-paste a config |
| 10:08.45 | joobie | see, u kno wwat i mean ;P |
| 10:10.01 | joobie | heh |
| 10:11.59 | joobie | WiretapMac, is there nothing in the cisco small business range that does the 48p standard? |
| 10:12.24 | WiretapMac | nothing in the cisco small business range other than the ESW series that even does 48p PoE |
| 10:15.07 | joobie | how much is the ESW? |
| 10:16.21 | dym | defswork: excuse me? |
| 10:17.06 | dym | For someone IRCing from a company mailserver, you certainly have a big mouth. |
| 10:18.11 | defswork | dym, would you like me to change my reverse dns to something else perhaps ? |
| 10:20.30 | joobie | defswork, given you let your LAN traffic out the same IP of your MX, you need to change more than your reverse DNS |
| 10:20.42 | joobie | BAM |
| 10:20.49 | defswork | joobie, help yourself |
| 10:21.18 | joobie | I had to |
| 10:21.30 | joobie | it was like one of those annoying hairs that linger around after u shave |
| 10:23.43 | joobie | i was just thinken |
| 10:23.47 | joobie | how can people not believe in god |
| 10:24.06 | joobie | when we have teeth that spawn |
| 10:24.10 | joobie | like how can u put that down to evolution |
| 10:24.30 | joobie | teeth in the exact spot where we eat |
| 10:25.12 | defswork | and feet in the exact place where we touch the floor |
| 10:25.15 | defswork | amazing isnt it |
| 10:25.41 | joobie | ye |
| 10:25.42 | joobie | a |
| 10:26.03 | joobie | like hwo could u logically think that teeth, feet, eats, eyes, nose are all evolution |
| 10:26.21 | joobie | it's more far fetched than believing in a creator |
| 10:32.49 | *** join/#asterisk F2Knight (~ben@c-24-22-60-186.hsd1.or.comcast.net) |
| 10:33.15 | F2Knight | Q: is there a way to match inbound DIDs with a kind of 'or' statement? |
| 10:33.52 | F2Knight | something like exten => 1|5555551212 which would respond to 15555551212 or 5555551212 |
| 10:35.24 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 10:36.14 | kaldemar | F2Knight: _[15]5555551212 |
| 10:36.56 | F2Knight | kaldemar, in side the [] braces is an optional ? |
| 10:37.22 | kaldemar | F2Knight: i misread, what's inside braces is not optional, so that won't work. |
| 10:37.41 | F2Knight | yea. I thought the [] was a range. |
| 10:38.17 | kaldemar | [15] would be 1 or 5. |
| 10:38.37 | F2Knight | right ... unless it could do [ 1] |
| 10:38.46 | F2Knight | :) |
| 10:39.23 | F2Knight | what about [1X] |
| 10:39.29 | F2Knight | do you think that would work? |
| 10:40.31 | WIMPy | [1-5] is a range. |
| 10:40.48 | kaldemar | F2Knight: no. X inside [] is a literal "X". |
| 10:42.57 | *** join/#asterisk fazool (~ada_has@203.130.22.202) |
| 10:43.42 | fazool | any one having experience of using Asterisk.net lib |
| 10:43.45 | fazool | ? |
| 10:44.32 | F2Knight | WIMPy, trying to find a way to handle DID's that may or may not have a 1 before them |
| 10:44.53 | F2Knight | that is with out having to put 2 entries in the extenensions |
| 10:45.06 | fazool | i want to capture call transfer-event |
| 10:46.17 | fazool | unable to find bindling |
| 10:46.28 | fazool | can any1 help |
| 10:46.52 | fazool | which handler should i use? |
| 10:47.02 | kaldemar | F2Knight: use two extensions. |
| 10:47.04 | fazool | for call transfer |
| 10:47.41 | fazool | i am talking about source code |
| 10:47.51 | fazool | asterisk.net |
| 10:48.34 | F2Knight | fazool, I think everyone here is pretty much using linux |
| 10:49.11 | *** join/#asterisk BuenGenio (~Gene@203.145.92.14) |
| 10:50.54 | fazool | yeah |
| 10:56.47 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:09.43 | Namiq | any one could help with chan_sip.c:21648 handle_request_register no matching peer found? I'm trying to register username for softphone |
| 11:10.44 | *** join/#asterisk black187 (~black187@93-103-22-42.static.t-2.net) |
| 11:26.55 | IsUp | Namiq: did you do 'sip reload' after you made changes? |
| 11:56.22 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 12:12.26 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 12:16.07 | *** join/#asterisk TheHaven (~haven@nostromo.thehavennet.org.uk) |
| 12:20.11 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.132.191) |
| 12:24.43 | *** join/#asterisk ubsafder (~ubsafder@2001:5c0:1400:a::55f) |
| 12:25.43 | ubsafder | hello i am writting a perl AGI how can i find in the script the codec of the incoming call ? |
| 12:25.59 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 12:27.21 | ubsafder | i can't find it in the AGI <enviroment dump . |
| 12:29.17 | tompaw | Hey guys, is there a way to automatically time out meetme conferences if there's only 1 person in them for some time? |
| 12:29.24 | tompaw | Or do I have to do it manually? |
| 12:29.40 | tompaw | (by manually I mean scripting and cron'ing AGI + meetme list) |
| 12:29.45 | IsUp | ubsafder: i dont think theres a variable stores codec info |
| 12:30.46 | IsUp | ubsafder: you can see codec manually of channel by 'show channel <channel>' on CLI. it says 'NativeFormats', 'WriteFormat', 'ReadFormat'. maybe you can execute asterisk -rx 'show channel <channel>' and parse this info. |
| 12:34.31 | IsUp | tompaw: probably you need scripting or cronjob for this. also check 'A' and 'x' options but its not right for your case i think |
| 12:34.54 | ubsafder | are you sugesting the AGI script restart an asterisk to get the info ? or there is a AGI cmd to do that ? |
| 12:35.11 | IsUp | tompaw: also you can use MeetMeCount for your script |
| 12:36.04 | IsUp | ubsafder: you can use 'System', but i am not sure where its sending output to |
| 12:38.37 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 12:39.01 | IsUp | ubsafder: and may i ask why you need this info? :p |
| 12:39.13 | leifmadsen | you can't get the output from the script using System() |
| 12:39.16 | leifmadsen | only with SHELL() |
| 12:41.04 | IsUp | leifmadsen: yes right, i forget it. its only for Asterisk 1.8, right? |
| 12:41.25 | leifmadsen | 1.6.x something |
| 12:42.12 | IsUp | okay |
| 12:42.34 | IsUp | ubsafder: also you can use 'DumpChan' too |
| 12:53.48 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-ooolrbowuokokjpr) |
| 12:57.19 | anonymouz666 | what's the oldest issue still valid in asterisk? when the internet goes down, chan_sip locks? this one is a big candidate |
| 12:57.20 | *** join/#asterisk pietro1 (~pietro@2-230-42-19.ip200.fastwebnet.it) |
| 12:57.35 | *** part/#asterisk cbwest (~cbwest@nat/cisco/x-ooolrbowuokokjpr) |
| 13:16.56 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:28.19 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mjfrogphhsepnbwj) |
| 13:31.14 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:33.45 | leifmadsen | anonymouz666: can you rephrase the question? |
| 13:34.56 | schmidts | anonymouz666 you mean the dns problem, i am not sure if this issue still exists in 1.8 |
| 13:34.57 | tompaw | IsUp: thanks for the tip, will give it a try. |
| 13:35.57 | tompaw | IsUp: I was actually thiking about parsing the output of "meetme list" every minute and for 3 consecutive matches with 1 participant - kill the conference. |
| 13:36.27 | tompaw | If i kill a meetme conf, will all the channels connected to it be hung up? |
| 13:37.45 | anonymouz666 | schmidts: what is different in 1.8 regarding this? |
| 13:37.57 | anonymouz666 | dnsmgr on register? |
| 13:38.34 | schmidts | anonymouz666 IMHO netsock was completly rewritten to support IPv6 |
| 13:39.19 | anonymouz666 | is this just a guess, or you are saying that is already fixed? |
| 13:39.43 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-rovlzraqpxsvuegr) |
| 13:41.44 | [TK]D-Fender | tompaw, Why not just kill the participant? |
| 13:42.42 | tuxxie | i have several users that are going to need to use softphones that will be placing calls from untrusted nated networks. Would IAX or SIP be the best choice? |
| 13:43.39 | tompaw | [TK]D-Fender: it might be considered illegal in some countries. |
| 13:43.54 | *** join/#asterisk bchia (~Adium@user-24-236-94-155.knology.net) |
| 13:44.00 | *** part/#asterisk bchia (~Adium@user-24-236-94-155.knology.net) |
| 13:44.13 | [TK]D-Fender | tompaw, We are talking about disconnecting a call right? to avoid repeat connections? |
| 13:45.20 | tompaw | [TK]D-Fender: yep ;-) Actually, to avoid hung meetme channels with only 1 participant who didn't hang up properly. |
| 13:47.46 | [TK]D-Fender | tompaw, Ok, I presume possibly not the same person in each right? Just when you see a single participant? Wouldn't be healthy if you hav an empty conference then multiple people would have to join in < 1 minute not to get killed off |
| 13:47.48 | schmidts | anonymouz666 its a guess ;) |
| 13:48.25 | [TK]D-Fender | tompaw, I'd grab the channel list, dump them for the CDR details for time, and use that to judge. |
| 13:49.37 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:49.52 | anonymouz666 | schmidts: but a lot of DNS code was modified, at least the comments "this could block for a long time" does not exist anymore on chan_sip |
| 13:51.26 | [TK]D-Fender | tompaw, just to be sure they're stale. |
| 13:56.12 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 13:58.14 | puzzled | hi |
| 14:02.44 | anonymouz666 | schmidts: good information on asterisk-users thread, it is good to know the impact of debug_threads in production, my setup is also > 50k/day and this makes things very hard to debug in case of problem |
| 14:03.56 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 14:04.18 | *** part/#asterisk pietro1 (~pietro@2-230-42-19.ip200.fastwebnet.it) |
| 14:08.33 | *** join/#asterisk VPeralta (~VPeralta@99-157-188-60.lightspeed.miamfl.sbcglobal.net) |
| 14:09.43 | VPeralta | stange dahdi problems with new server can anyone help? |
| 14:10.06 | schmidts | anonymouz666 i havent tried it if it was only the dont_optimize flag or together with debug_threads, i had both turned on and it was evil. both turned off and it runs really smooth |
| 14:10.38 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 14:11.00 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 14:11.43 | VPeralta | AEX800P works for 4-8 hours then card is unresponsive. try to unload driver and get this message "wctdm24xxp 0000:02:08.0: Unable to read I/O registers." |
| 14:12.24 | MrTelephone | Can someone enlighten me on the variables of CallerPres(). Is there a page that defines them? I'm not sure the difference between 'not screened, passed screen, failed screen. |
| 14:13.27 | MrTelephone | vperalta, did you try a different pci slot? |
| 14:13.49 | VPeralta | its a 1u dell ;) |
| 14:15.59 | VPeralta | Card is in a Dell r210 II dell 1u rack server it only has one slot. |
| 14:16.22 | VPeralta | only way to get it back up is to reboot |
| 14:17.05 | VPeralta | i switched to trunk dahdi drivers and it seems to stay up longer but i can quite figure out where its hanging |
| 14:18.06 | VPeralta | i also enabled debug on the driver but dont see any messages when it fails |
| 14:19.21 | MrTelephone | I never used that hardware but I had trouble with sangoma t1 cards and the pci bus. I did end up getting one of the cards RMA'd. It would randomly crash with PCI bus errors every couple weeks. A little longer than your 4-8 hours. |
| 14:20.54 | VPeralta | MrTelephone thanks for the advice |
| 14:21.22 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-xvpvyxqaausqlgsh) |
| 14:21.38 | MrTelephone | It's interesting that different drivers changed your problem. I feel your pain. |
| 14:21.43 | VPeralta | did you see errors when it would crash? |
| 14:22.03 | VPeralta | thats the weird problem i dont see any errors until i try to relaod the driver |
| 14:22.16 | MrTelephone | It was a long time ago I don't remember exactly but it had something to do with the pci bus. I could reload the card drivers but it wouldn't work until I rebooted. |
| 14:22.26 | VPeralta | at least if i could see the error i could figure out where the problem was stemming from |
| 14:22.35 | MrTelephone | yeah I would see some errors in the syslog |
| 14:25.07 | VPeralta | MrTelephone thanks again, i will go back to banging my head againt the wall now |
| 14:27.12 | MrTelephone | if you can try it in another pc? |
| 14:28.56 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 14:30.51 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 14:31.46 | *** join/#asterisk VPeralta (~VPeralta@199.27.100.5) |
| 14:32.57 | VPeralta | Its installed in the clients office out of the country |
| 14:39.31 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 14:40.40 | *** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 14:43.04 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:50.26 | *** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj) |
| 14:52.19 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 14:53.37 | ilj | hi, I've this testbox with asterisk 1.6.2.9 and when I make an incoming call Echo application is used to test the channel. the problem is that when asterisk tries to execute Echo it prints a WARNING[3286]: pbx.c:3695 pbx_extension_helper: No application 'Echo' for extension (incoming, s, 3). ... |
| 14:53.42 | ilj | ... However, the actual application seems to be available in the filesystem path: http://pastebin.com/kqCry00H Well what's going on? :) |
| 14:55.16 | wdoekes2 | module load app_echo |
| 14:56.26 | wdoekes2 | (that's btw not the modules dir you're looking at.. look in /usr/lib/asterisk/modules) |
| 14:56.45 | [TK]D-Fender | Funny we see /root and /src in there ... and not the actual modules folder <- |
| 14:56.54 | ilj | yeah, now I know since module load ... said that there is no such app in /usr/lib... |
| 14:57.15 | ilj | hey [TK]D-Fender, great to see you around |
| 14:57.20 | [Outcast] | what is the standard amount of calls you can get through a quad core box these days. |
| 14:57.34 | [TK]D-Fender | [Outcast], many :) |
| 14:57.38 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 14:58.10 | ilj | wdoekes2, is thinking of environment path, like in bash, applicable when we talk about asterisk? |
| 14:58.35 | ilj | or should I just simlink a .so file from /usr/share to /usr/lib ? |
| 14:58.52 | [Outcast] | are we talking 500, 1000, 5000? I am trying write plan and need to know I should add another server |
| 14:59.21 | [TK]D-Fender | [Outcast], Depends what they're doing |
| 14:59.37 | [Outcast] | just standard calls 1-to-1 |
| 14:59.45 | [Outcast] | g711 |
| 14:59.49 | [TK]D-Fender | [Outcast], Yuo should already know that you asked a blanket question with no details that are needed for any sense of scalability |
| 15:00.23 | schmidts | [Outcast] it depends on your asterisk version, for example with 10 i guess you will hit the network limit before asterisk will go down |
| 15:00.29 | [TK]D-Fender | [Outcast], How much do you need to account for? |
| 15:01.00 | [TK]D-Fender | [Outcast], What is the calling rate? |
| 15:02.11 | IsUp | after some load, my DAHDI channels are not working on outdial. when i do 'dahdi show channel xx' it says 'PRI Flags: Resetting' |
| 15:02.16 | [Outcast] | 20/sec with 10000 sim sessions |
| 15:03.16 | IsUp | if i see 'Resetting', channel is not working until next restart. sometimes its fixed after some time. im having this issue since 2 months. ive tried resetinterval= and so on |
| 15:03.41 | schmidts | [Outcast] you know this would be around 840 Mbit of rtp traffic with g711 |
| 15:04.15 | [Outcast] | yes |
| 15:04.26 | schmidts | and you want to run this on a single server? |
| 15:04.32 | [Outcast] | schmidts, no |
| 15:04.39 | schmidts | ok ;) |
| 15:04.58 | [Outcast] | schmidts, it would be a cluster with opensips steering the traffic |
| 15:05.18 | schmidts | my latest load tests showed me asterisk 10 could handle 1750 calls per second up to 13000 calls but this was just testing, not real calls |
| 15:05.29 | schmidts | [Outcast] good idea ;) |
| 15:06.08 | [Outcast] | schmidts, that at least gives me idea to go from. |
| 15:06.40 | [Outcast] | what processor and how much mem |
| 15:06.57 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:07.04 | schmidts | [Outcast] but this was with 10, normal 1.8 can handle normally only 6 times less than this amount |
| 15:07.16 | [Outcast] | agreed |
| 15:07.26 | [Outcast] | 10 is my target |
| 15:08.42 | schmidts | [Outcast] i would recommend you my development branch unleash-the-beast cause it has most of the performance backported from 10 into 1.8 and i use it allready in production and it works just fine |
| 15:11.02 | wdoekes2 | ilj: you shouldn't have to symlink |
| 15:11.07 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:11.13 | [Outcast] | schmidts: I will try it out. what type and speed of processor? Also how much memory |
| 15:11.35 | wdoekes2 | make install should do things correctly.. but it may conflict with your settings in your asterisk.conf. verify that |
| 15:13.56 | schmidts | [Outcast] i have also a kamailio sip proxy in front and virtual machines where asterisk itself is running and one of the host machines is a HP DL380G6 with dual Six Core Intel Xeon CPUs and 12 GB of Ram and thats enough for me |
| 15:14.29 | [Outcast] | great thanks |
| 15:15.03 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 15:15.22 | schmidts | [Outcast] and i have noticed you dont need a lot of ram anymore, for example one virtual machine now uses 1,2 gb ram but has allready processed 82k calls in nearly 9 days |
| 15:15.42 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 15:15.43 | [Outcast] | that is awesome |
| 15:19.55 | *** join/#asterisk Sorcier_FXK (~Sorcier_F@unaffiliated/sorcierfxk) |
| 15:23.06 | r0m|u | p3nguin, you got your dns issue with your iphone resolved? |
| 15:28.52 | *** join/#asterisk brainiac (~brainiac@24-183-217-180.dhcp.jcsn.tn.charter.com) |
| 15:29.23 | p3nguin | I never did figure it out. It's only on the iPhone, though. |
| 15:33.53 | r0m|u | ssh in and add it to the hosts file. |
| 15:36.16 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 15:37.15 | p3nguin | I'd rather just have working DNS on it. |
| 15:37.48 | p3nguin | By the way, it's only giving me a problem on the local domain. |
| 15:37.51 | *** join/#asterisk Wiretap7 (~Wiretap@unaffiliated/wiretap) |
| 15:39.16 | *** join/#asterisk liuyan1 (~LiuYan@222.125.132.191) |
| 15:39.30 | *** join/#asterisk bshipman (~bshipman@fw1.safedataisp.net) |
| 15:40.11 | *** join/#asterisk dobby156 (~joe@79.135.102.10) |
| 15:40.11 | dobby156 | hi |
| 15:40.12 | bshipman | good morning, looking for a spot of advice on DAHDI and outbound dialing |
| 15:40.27 | dobby156 | is there a way of negating in the asterisk pattern matching |
| 15:40.42 | dobby156 | I am using version 1.6 |
| 15:40.45 | dobby156 | Thank You |
| 15:41.05 | p3nguin | That's not a version. |
| 15:41.12 | p3nguin | That's not even a branch. |
| 15:41.44 | p3nguin | (not that it will matter for this particular question, but you should still know) |
| 15:41.53 | dobby156 | p3nguin: I mean that generation |
| 15:41.54 | dobby156 | sorry |
| 15:42.05 | dobby156 | *meant |
| 15:42.30 | p3nguin | Things that were in the 1.6.0 BRANCH were not necessarily in the 1.6.1 or 1.6.2 BRANCHES. |
| 15:42.59 | p3nguin | and vice versa |
| 15:43.16 | MrTelephone | I'll have to troll and say that iphone is a problem in itself |
| 15:43.50 | MrTelephone | ./nick applhtr |
| 15:44.22 | p3nguin | Maybe. |
| 15:44.50 | p3nguin | It's just weird how it does not do DNS for my local domain, but it works for other domains... and all other devices work on the local domain. |
| 15:44.55 | dobby156 | p3nguin: but do you know of negating pattern matching? |
| 15:44.57 | MrTelephone | that odbc dialplan stuff is the best invention ever made |
| 15:45.17 | MrTelephone | bows to the programmers of asterisk |
| 15:45.37 | p3nguin | dobby156: My best advice is: Don't make a pattern that matches. |
| 15:45.57 | p3nguin | For a better answer, give detailed information on what you're doing currently and what you want to be different. |
| 15:46.22 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-ytvcdgxijxnauamo) |
| 15:46.27 | MrTelephone | p3nguin, what do you know about call screening/presentation? |
| 15:46.37 | p3nguin | some |
| 15:46.45 | Qwell | throws a lurker-bomb at cbwest |
| 15:46.46 | p3nguin | What do you want to know about it? |
| 15:46.59 | MrTelephone | Why is there passed screeing and failed screening options. what does that mean |
| 15:47.07 | *** part/#asterisk cbwest (~cbwest@nat/cisco/x-ytvcdgxijxnauamo) |
| 15:47.11 | Qwell | heh |
| 15:47.15 | p3nguin | hahaha |
| 15:47.19 | Qwell | oh, Cisco. |
| 15:47.32 | Qwell | I bet he's a suit. |
| 15:47.36 | _Corey_ | That was effective |
| 15:47.50 | Qwell | _Corey_: I've been trying to ask him what brings him here, being with Cisco and all. |
| 15:47.57 | Qwell | he always leaves when I talk to him :p |
| 15:48.06 | Qwell | shady |
| 15:48.07 | MrTelephone | cisco west? |
| 15:48.40 | p3nguin | Cisco B. West |
| 15:48.42 | MrTelephone | I have 4 cisco appliances and if I could I would ditch them |
| 15:48.58 | bshipman | could i pick someone's brain about a rather abrupt loss of outbound dialing using ast 1.6 / dahdi ? |
| 15:49.08 | MrTelephone | bshipman, give it a try |
| 15:49.54 | r0m|u | p3nguin, https://discussions.apple.com/thread/2785131?start=0&tstart=0 |
| 15:49.55 | bshipman | nothing has changed in my confs - i can receive inbound overs POTS using a sangoma card |
| 15:50.06 | bshipman | i get a dial tone when i jack in |
| 15:50.27 | MrTelephone | I thought apple went broke? |
| 15:50.38 | r0m|u | p3nguin, look at that link. looks like a bug in ios |
| 15:50.49 | bshipman | i see a nifty line DAHDI/8-1 answered SIP/h3018 .... |
| 15:50.50 | MrTelephone | ~troll~ |
| 15:51.10 | bshipman | but I don't think its actually completed the dial |
| 15:51.37 | bshipman | i'm calling one of our 800's so that it loops to the CO and comes back in on the 800 T-line |
| 15:51.58 | [TK]D-Fender | bshipman, Unless you did callprogress=yes for your channels then analog channesl are considered answered as soon as they are placed. This pays no regard to ever getting dialtne in the first place |
| 15:51.59 | bshipman | watching the T I never see the inbound call arrive |
| 15:55.26 | *** join/#asterisk andalou (~chatzilla@190.238.192.238) |
| 15:56.59 | bshipman | ok ... any thoughts - this is an abrupt loss of outbound - I've inherited these systems as a privilege of being a programmer not from any telephony experience |
| 15:57.26 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:57.37 | p3nguin | r0m|u: This report does not match my case. |
| 15:58.19 | MrTelephone | anyone want to battle it out on battlefield3 for pc tonight? |
| 15:58.42 | p3nguin | r0m|u: I'm using iOS 4.0 and any-host-name.local does not resolve. They are saying it will resolve when using any-host-name.local but not just any-host-name. |
| 15:58.49 | MrTelephone | bshipman, i think most of us started that way |
| 15:59.11 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:59.17 | MrTelephone | I went from mopping washroom floors to analyzing t1 circuits with an osciliscope |
| 15:59.54 | *** part/#asterisk andalou (~chatzilla@190.238.192.238) |
| 16:00.52 | bshipman | MrTelephone: thx |
| 16:01.31 | MrTelephone | DAHDI/8-1 is your service from the telco? |
| 16:01.56 | bshipman | 8-1 refers to the port on my analog card |
| 16:02.07 | [TK]D-Fender | bshipman, plug an analog phone in parallel with the line. Listen in on the call attempt. What do you hear? |
| 16:02.17 | bshipman | i have 8 ports - low order are inbound in the hunt |
| 16:02.23 | MrTelephone | Your sip line isn't passing the digits to dahdi? |
| 16:03.04 | MrTelephone | shouldn't it say DAHDI/8-1/19001111111 answered SIP/whatever |
| 16:03.36 | bshipman | one sec and i'll pastebin |
| 16:05.17 | bshipman | http://pastebin.com/48WsVmVA |
| 16:05.46 | bshipman | I've tried reversing the dialout order from 8 -> 3 to 3 -> 8 to no avail |
| 16:06.08 | bshipman | trying to eliminate a bad FXO module on the card or line trouble |
| 16:07.04 | bshipman | [TK]D-Fender I'll have to get a tech to wire me up another jack or find a pigtail it is straight from the punch down to a jack for the jard |
| 16:07.06 | bshipman | card* |
| 16:08.02 | MrTelephone | bshipman. do you have a phone test set to plug in parallel like fender suggested? |
| 16:08.32 | bshipman | i have a phone just no test jack i'll have to find a 1 to 2 jack |
| 16:09.18 | MrTelephone | wish i could teleport you some duplex connectors. I have a whole bag of em here :( |
| 16:10.27 | bshipman | lol in the Great Purge of our junk room apparently we have a small box off odds and ends |
| 16:10.31 | MrTelephone | it doesn't even report that the line is ringing |
| 16:10.38 | MrTelephone | ? |
| 16:10.49 | MrTelephone | did you try calling another number besides the 1800 |
| 16:10.57 | bshipman | cthulu be praise we have one |
| 16:11.20 | MrTelephone | maybe the telco is saying piss off for calling me from the same number your 1-800 number is directed to |
| 16:11.23 | bshipman | i'll have to hardcode the dial plan - right now it tests to see if its toll free |
| 16:11.37 | bshipman | IFF toll free dial over DAHDI out of the pbx |
| 16:11.46 | MrTelephone | oh I see |
| 16:12.00 | bshipman | ELSE SIP to Asterisk T machine and dial out there |
| 16:12.11 | MrTelephone | it just stopped working all of a sudden? |
| 16:12.26 | bshipman | more or lesse |
| 16:12.39 | bshipman | we have 2 clone PBX's |
| 16:12.50 | bshipman | PBX1 started giving nasty garbled static |
| 16:13.01 | bshipman | the primary line in the hunt group died from the Teleco |
| 16:13.21 | bshipman | our T teleco and our POTS teleco are different btw |
| 16:13.44 | bshipman | so i switched to PBX2 and manually moved all the POTS lines except the one that was crap |
| 16:13.48 | bshipman | no dice |
| 16:14.10 | bshipman | the card in PBX started w/o errors in sys log and does receive calls just fine |
| 16:14.19 | bshipman | PBX1 that is to say |
| 16:14.46 | r0m|u | p3nguin, mhhh I see |
| 16:15.45 | MrTelephone | are you sure its not dialing out a group port that doesn't have a telephone line plugged in? |
| 16:15.58 | MrTelephone | because you said you moved all the lines over except 1? |
| 16:16.21 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 16:16.39 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 16:17.04 | bshipman | the line is fixed now - i'm back on PBX1 |
| 16:17.32 | bshipman | [trunkgroups] |
| 16:17.32 | bshipman | [channels] |
| 16:17.33 | bshipman | context=default |
| 16:17.33 | bshipman | usecallerid=no |
| 16:17.33 | bshipman | echocancel=yes |
| 16:17.34 | bshipman | echocancelwhenbridged=yes |
| 16:17.34 | bshipman | group=1 |
| 16:17.35 | bshipman | callgroup=1 |
| 16:17.35 | bshipman | pickupgroup=1 |
| 16:17.36 | bshipman | immediate=no |
| 16:17.36 | bshipman | ;Sangoma AFT-A200 [slot:0 bus:7 span:1] <wanpipe1> |
| 16:17.36 | bshipman | context=inbound |
| 16:17.37 | bshipman | group=0 |
| 16:17.37 | bshipman | echocancel=yes |
| 16:17.41 | *** kick/#asterisk [bshipman!~north@pdpc/sponsor/digium/Qwell] by Qwell (pastebin) |
| 16:17.44 | [TK]D-Fender | ~pb |
| 16:17.44 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:17.47 | [TK]D-Fender | ^^^ |
| 16:17.53 | Qwell | he's dead, Jim |
| 16:17.54 | *** join/#asterisk bshipman (~bshipman@fw1.safedataisp.net) |
| 16:17.57 | p3nguin | bshipman: Don't ever do that. |
| 16:17.58 | [TK]D-Fender | ~pb |
| 16:17.59 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:18.00 | MrTelephone | lol |
| 16:18.02 | [TK]D-Fender | ^^^ |
| 16:18.04 | MrTelephone | relax |
| 16:18.21 | Qwell | MrTelephone: You are not Frankie. I do not have to listen to you. |
| 16:18.23 | bshipman | sorry it was a handful of lines, my bad |
| 16:18.57 | ubsafder | i have a incoming (ulaw|h263) what format can i use to record ? |
| 16:19.09 | pabelanger | Qwell: well played sir |
| 16:19.15 | Qwell | takes a bow |
| 16:19.43 | MrTelephone | bshipman, is it just me or you missing a 'channels=1-8' line or whatever ports they are |
| 16:20.20 | ubsafder | record FILE XXXX h263 gives me bunch of errors |
| 16:20.21 | [TK]D-Fender | MrTelephone, Qwell axed him before his flood got that far |
| 16:20.29 | MrTelephone | lol |
| 16:20.40 | Qwell | [TK]D-Fender: I just spent like 30 seconds trying to word that. |
| 16:20.54 | bshipman | channels 1-2 don't exist |
| 16:21.03 | [TK]D-Fender | Qwell, :) |
| 16:21.04 | bshipman | there is no fxo / fxs module there |
| 16:21.33 | MrTelephone | bshipman, you got kicked before the channels line got posted |
| 16:21.54 | bshipman | when a miracle of time happens I have fxs modules to stick in there for a fax |
| 16:22.06 | bshipman | ah |
| 16:22.10 | MrTelephone | those analog cards are junk really |
| 16:22.14 | bshipman | channel => 3-8 |
| 16:22.41 | bshipman | lol thats a complaint for my former co-worker and the boss man |
| 16:23.14 | bshipman | mine is to program python and deal w/ weirdness in the phone system |
| 16:23.33 | bshipman | like calls abruptly truncating w/ no log messages - they just randomly die |
| 16:23.37 | MrTelephone | if you want something 100% you have to spend big bucks |
| 16:23.57 | bshipman | i mean over all in 2 years of managing these systems I really can't complain too much |
| 16:24.23 | bshipman | the sangoma cards that manage our T lines are doing ~200k minutes / month with virtually no downtime |
| 16:24.29 | MrTelephone | i get the odd complaint but for the value you can't really complain like you said |
| 16:25.01 | MrTelephone | we only do 100k a year |
| 16:25.51 | MrTelephone | i'm mickey mouse compared to most. :( |
| 16:26.37 | *** join/#asterisk dlublink (~david@206-248-174-128.dsl.teksavvy.com) |
| 16:27.46 | dlublink | I am calling 3 local channels "Local/123@context1&Local/124@context1&Local/932@context2". I want to set a variable in the new channels, but I want a different value for each channel. So _VARIABLE and __VARIABLE don't work. Is there a way for me to set a variable differently on each of the three new local channels ? |
| 16:28.06 | *** join/#asterisk irroot (~gregory@41.49.35.38) |
| 16:28.36 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v028-178.mobile.uci.edu) |
| 16:29.40 | bshipman | we're running industrial data collections via IVR |
| 16:29.53 | bshipman | i think i may have a dead port on my gd card |
| 16:29.55 | bshipman | sigh |
| 16:30.15 | bshipman | ... that might be the port that got hit when the line went down originally |
| 16:30.47 | voipeng | any suggestions on which repo to get fora new asterisk server? Looking for something with the gui included |
| 16:31.05 | MrTelephone | maybe, phone lines are nasty for surges |
| 16:31.09 | *** join/#asterisk gandhijee (akp@ip67-152-15-148.z15-152-67.customer.algx.net) |
| 16:31.13 | Qwell | dlublink: _123SOMEVAR, _124SOMEVAR, then read it later with ${${EXTEN}SOMEVAR} |
| 16:31.28 | gandhijee | hey, how are the Aastra phone? are the speaker phones on them good? |
| 16:31.46 | bshipman | have you found any successful products that will suppress the surge sufficently? |
| 16:32.27 | MrTelephone | our local telephone company has these special surge punch down blocks with removable surge supression thingies. I don't know what they are called. |
| 16:32.28 | dlublink | Qwell, that was my fallback plan, I was hoping for something easier ;). Thanks :) |
| 16:32.50 | MrTelephone | Maybe someone here will know what they are called? |
| 16:34.35 | [TK]D-Fender | gandhijee, Not bad, but not great |
| 16:34.42 | [TK]D-Fender | gandhijee, Polycom > all |
| 16:35.03 | gandhijee | yeah i know the polys are awesome, i have them in all the other places i have |
| 16:35.19 | Naikrovek | he's not just being religious, he's right. aastra speakers suck in comparison |
| 16:35.28 | gandhijee | the place i am at now doesn't have as much money yet so they are a bit shoe string |
| 16:35.30 | voipeng | no one can recommend an iso to download for asterisk with a gui they have had success with? |
| 16:35.44 | gandhijee | has anyone tried those yealink titan phones? |
| 16:35.53 | p3nguin | voipeng: see AsteriskNOW |
| 16:36.05 | navaismo | voipeng: compile asterisk and dahdi then install freepbx |
| 16:36.13 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 16:36.31 | p3nguin | voipeng: AsteriskNOW is built on CentOS and comes with your choice of FreePBX, the Asterisk GUI, or no GUI. |
| 16:38.21 | MrTelephone | do you have to put the mysql password in odbc.ini AND asterisks res_odbc.conf? |
| 16:39.21 | Qwell | MrTelephone: I don't think it needs to be in odbc.ini |
| 16:42.06 | MrTelephone | thanks qwell, you saved me like 10 minutes to try it, send me a bill :P |
| 16:42.28 | *** join/#asterisk vpopov (~happylife@46.251.83.126) |
| 16:42.35 | Qwell | MrTelephone: Paypal, that'll be $399.90. |
| 16:42.40 | MrTelephone | I'm trying to do long distance blocking based on ODBC lookups instead of changing the sip users context |
| 16:42.50 | *** join/#asterisk tehrcon (~tercon@mail.imprinters.com) |
| 16:43.14 | [TK]D-Fender | gandhijee, I ahve never really seen any place where Aastra was really any cheaper than Polycom |
| 16:43.41 | MrTelephone | Do you think that's scary? I'm using local machine database for lookups in case |
| 16:43.57 | gandhijee | [TK]D-Fender: one of the guys here apparently can get a hold of some cheap from some friend of his or something |
| 16:44.19 | Qwell | gandhijee: Polycom pricing probably beats Aastra friend pricing. |
| 16:44.36 | MrTelephone | polycom 331's are not bad and affordable |
| 16:44.45 | Naikrovek | 321s are like $75 |
| 16:44.48 | gandhijee | ehh aastra friend pricing is like 50 bux a phone for i think 9480 or something like that |
| 16:45.04 | gandhijee | they are from some place that went under or something |
| 16:45.05 | MrTelephone | if you need 2 network ports make sure you get 331. I almost bought 321's by mistake |
| 16:45.08 | p3nguin | 550 is ~$200, 650 is ~$300 |
| 16:45.11 | Qwell | gandhijee: so, used? |
| 16:45.15 | gandhijee | yeah |
| 16:45.19 | Naikrovek | ... |
| 16:45.36 | gandhijee | i think it was like 2 years ago VoIP supply was selling 450's @ 150 for 2 or something liket hat |
| 16:45.42 | gandhijee | i should have bought a shit ton then |
| 16:45.46 | MrTelephone | I still have polycom 501's from 6 years ok and none of them died yet |
| 16:45.47 | p3nguin | I'm in the midst of a deployment, so I recently looked at prices. |
| 16:45.59 | [TK]D-Fender | gandhijee, Well that is honestly a hell of a deal |
| 16:46.05 | MrTelephone | that sounds like a firesale |
| 16:46.13 | [TK]D-Fender | gandhijee, In all fairness if they're in decent shape go for it |
| 16:46.32 | gandhijee | [TK]D-Fender: yea it was, bought up like 40 450 for 2 offices |
| 16:46.32 | MrTelephone | I recently bought 450's for 200 and marked them up to 235 |
| 16:46.59 | MrTelephone | someone is losing money on those and it isn't polycom |
| 16:47.08 | [TK]D-Fender | gandhijee, 450's are a hard model to justify |
| 16:47.23 | gandhijee | [TK]D-Fender: ?? whys that? |
| 16:47.26 | MrTelephone | they have shitty screens |
| 16:47.39 | gandhijee | ahh, well i only paid 75 bux for them, so it wasn;t that bad |
| 16:47.48 | MrTelephone | stolen? |
| 16:48.03 | gandhijee | but agreed, there is a microbrowser in it but since the screen is too small its kinda fail |
| 16:48.09 | MrTelephone | Do you think MyStats button on polycom will ever be implemented in asterisk? |
| 16:48.17 | [TK]D-Fender | gandhijee, standard retail price vs higher & lower models where functionality is concerned |
| 16:48.22 | gandhijee | MrTelephone: nah, voip supply or someone had a sale like 2 years back |
| 16:48.43 | gandhijee | now that i think about it, i think it was acutally polycom sponsered sale |
| 16:49.08 | MrTelephone | maybe the 450's are shit so they had to clear warehouse |
| 16:49.30 | MrTelephone | why would you spend +100 to get a 550 over a 450? |
| 16:49.34 | [TK]D-Fender | <gandhijee> but agreed, there is a microbrowser in it but since the screen is too small its kinda fail <- depends what you intend to use it for of course, but not really "spacious" |
| 16:50.19 | MrTelephone | who wants to do a video call on a vvx1500? |
| 16:50.34 | gandhijee | MrTelephone: like i said, i paid 75 bux a phone - cheap had HD voice and multiline - company was happy, and i made 25 bux a phone |
| 16:50.48 | gandhijee | and the 450 |
| 16:51.00 | gandhijee | s are fine, quality is just as good as my 601 and 501 |
| 16:51.08 | *** join/#asterisk jsarrel (~jsarrel@66-191-161-122.dhcp.gnvl.sc.charter.com) |
| 16:51.10 | MrTelephone | I don't even know what HD voice is. What is wrong with the non-hd voice? |
| 16:51.32 | [TK]D-Fender | 450 is a good phone, just priced a little odd to find the right reason to by at standard pricing |
| 16:51.56 | MrTelephone | your step down is a the 300 series though and those lack buttons and screen? |
| 16:52.04 | _Corey_ | MrTelephone: I've had the VVX on my desk for 6 months and have made 3 video calls ;) |
| 16:52.11 | [TK]D-Fender | http://www.telephonydepot.com/Catalog/Polycom-Phones |
| 16:52.43 | MrTelephone | vvx is nice |
| 16:52.47 | _Corey_ | eh |
| 16:53.00 | [TK]D-Fender | Cheap user : dedicated LAN wiring, IP 321 (no need for switch, pays for cabling) |
| 16:53.04 | _Corey_ | I'm not too wild about having to reach up to the touch screen all the time |
| 16:53.22 | _Corey_ | I like the new Polycom phone they had at Astricon |
| 16:53.23 | MrTelephone | hey why didn't polycom fix the handset yet. Even the newer 650/450/550 have the problem where the handset falls off the phone easily? |
| 16:53.28 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:53.30 | [TK]D-Fender | Medium user : IP 335 (backlight and passthrough) |
| 16:53.41 | gandhijee | holy hell, i'd never pay 175 for a 450 |
| 16:53.44 | [TK]D-Fender | MrTelephone, I use the hook. Works well |
| 16:53.45 | _Corey_ | MrTelephone: Pull the hook out if you find them sliding |
| 16:53.48 | Naikrovek | doesn't fall off my phone easily |
| 16:53.53 | _Corey_ | it pops out |
| 16:54.19 | MrTelephone | ok i didn't know I'll check it out |
| 16:54.37 | MrTelephone | well I have this plantronics headset and it lifts the handset up but then it falls on the floor |
| 16:54.39 | MrTelephone | lol |
| 16:55.00 | IsUp | is there any way to trigger 'resetinterval', i mean PRI resetting without restart Asterisk |
| 16:55.00 | _Corey_ | Just poke it from the front and you can spin it around from behind |
| 16:55.01 | Naikrovek | why are you using a handset lifter... push the handset button |
| 16:55.23 | _Corey_ | yeah, that's why they have the electronic lifter ;) |
| 16:55.24 | Naikrovek | push the headset button i mean |
| 16:55.27 | [TK]D-Fender | MrTelephone, OH. yeah, that gets messy. I've tied them to the lifter with elastics and in some cases put a screw on the bar of ht lifter to directly press the switch leaving the handset off entirely |
| 16:55.29 | MrTelephone | lol |
| 16:55.48 | MrTelephone | i have electrical tape holding the lifter in place. I need elastics now |
| 16:55.50 | Naikrovek | i can't believe that handset lifters even exist anymore. wtf kind of hack is that |
| 16:55.51 | _Corey_ | get a compatible headset and the cable |
| 16:56.06 | MrTelephone | lifter is universal I guess |
| 16:56.08 | [TK]D-Fender | Naikrovek, For people to cheap to change their phones. |
| 16:56.21 | Naikrovek | it's a hack from 40 years ago that no one thought was stupid except me, apparently |
| 16:56.45 | _Corey_ | I had a woman complain when we upgraded her from the mechanical lifter to the integrated polycom because she missed watching it go up/down all day |
| 16:56.51 | MrTelephone | What other options does a person have? |
| 16:56.56 | Naikrovek | ... that person needs to be shot before they breed |
| 16:56.58 | MrTelephone | lol |
| 16:57.07 | MrTelephone | I like the lifter. the sound it makes is cute |
| 16:57.10 | Qwell | _Corey_: there's no pleasing people like that |
| 16:57.23 | _Corey_ | I didn't mind, actually... she was really good looking |
| 16:57.25 | _Corey_ | hahaha |
| 16:57.30 | p3nguin | I can understand using a handset lift when the person is using a bluetooth headset with an answer button on it. |
| 16:57.34 | Qwell | crazy's crazy, man |
| 16:57.50 | Naikrovek | p3nguin: get a phone that doesn't need to have the handset lifted to answer the flippin call |
| 16:58.09 | Naikrovek | handset lifters are stop-gap hacks and need to be eliminated |
| 16:58.31 | p3nguin | If there's a device that allows for answering the call via remote headset, then that would eliminate the lift. |
| 16:58.49 | _Corey_ | Jabra and Plantronics both make compatible units for Polycom phones |
| 16:58.51 | Naikrovek | if there aren't devices like that i'd be shocked |
| 16:58.57 | _Corey_ | they're not cheap though |
| 16:59.02 | p3nguin | with remote answer? |
| 16:59.06 | _Corey_ | sure |
| 16:59.26 | _Corey_ | they use a little cable that plugs into that 6 pin (?) connector on the back |
| 16:59.40 | MrTelephone | I needed something that would work on my rotary phone |
| 16:59.42 | MrTelephone | don't make fun |
| 16:59.43 | Naikrovek | this is why bluetooth > all. |
| 16:59.57 | _Corey_ | then you go into the headset mode menu and set it to Plantronics or DHSG |
| 17:00.01 | Naikrovek | this is the kind of thing that bluetooth is for |
| 17:00.02 | MrTelephone | Lifter + Rotary phone |
| 17:00.21 | p3nguin | Okay, so you can answer a phone call... but how to you make one? |
| 17:00.22 | Naikrovek | i think MrTelephone is talking to us from 1955 or so. |
| 17:00.43 | _Corey_ | p3nguin: Well, you still need to dial your calls... ;) |
| 17:00.55 | bshipman | thx all for the advice - i have at least 1 dead analog port |
| 17:00.57 | [TK]D-Fender | MrTelephone, Rotary?! You spoiled little prissy! What am I supposed to do for my hand-crank model? And that wench keeps leaving her station! |
| 17:01.09 | MrTelephone | do they make a headset yet with a keypad on the side?? what movie was that on??? |
| 17:01.23 | p3nguin | _corey_: If you activate the lift, you could make clicking sounds into the mic to emulate the pulse dialing. |
| 17:01.43 | MrTelephone | tk-fender. they make lifters for the hand crank too. what about a crank turning addon by plantronics? |
| 17:02.13 | _Corey_ | p3nguin: Hmm, yeah probably on MrTelephone's rotary rig but not so much on the Polycom... ;) |
| 17:02.15 | [TK]D-Fender | BRILLIANT |
| 17:02.17 | MrTelephone | is away to the research and development department and then to the marketing department after that |
| 17:03.00 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:04.50 | MrTelephone | ./dcc send tk-fender patent_lifter_hand_crank_phone.pdf |
| 17:05.49 | *** join/#asterisk ccesario (~ccesario@189.29.62.245) |
| 17:06.35 | *** join/#asterisk aetaric (~dustins@74-130-83-20.dhcp.insightbb.com) |
| 17:08.00 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 17:08.20 | MrTelephone | exten => _1XXXXXXXXXX,n(blockld_message),PlayBack(/etc/asterisk/nolongdistance) is underscore allowed there? |
| 17:08.35 | Qwell | Why wouldn't it be? O.o |
| 17:08.44 | MrTelephone | special character or something |
| 17:08.50 | MrTelephone | what about a space :P |
| 17:08.57 | MrTelephone | that would probably be a disaster |
| 17:09.02 | Qwell | oh, in the label |
| 17:09.04 | Qwell | yeah that should be fine |
| 17:09.10 | Qwell | space probably wouldn't |
| 17:09.25 | MrTelephone | i'll quote the b4st3rd |
| 17:09.34 | MrTelephone | ;P |
| 17:09.36 | Qwell | I doubt quoting would work |
| 17:10.32 | MrTelephone | corey, what was the new phone at astricon by polycom? I'm pissed off at polycom because we spent 6,000 on an hdx video conference system a couple years ago and it doesn't even support sip? wtf? |
| 17:11.06 | [TK]D-Fender | MrTelephone, I've got a VSX 7000 that supposedly does but I could never get working with it.. |
| 17:11.42 | MrTelephone | ever sad |
| 17:11.56 | *** join/#asterisk hardwire (~spencersr@cl-36.anc-01.us.sixxs.net) |
| 17:11.58 | MrTelephone | smarten up polycom. why are they hanging onto that h232 or whatever crap that is |
| 17:12.37 | MrTelephone | mgcp and h232 gatekeeper crap needs to be tossed. |
| 17:13.10 | MrTelephone | who do you talk to or email about that? ceo@polycom.com |
| 17:13.47 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 17:15.18 | MrTelephone | I was testing out the hdx 6000 video conference at my home in my house coat and feet on the coffee table while these business people from japan were joining in to test there system out. Was that ever funny |
| 17:18.09 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:18.48 | MrTelephone | Would that be cool if you met your wife or husband on a polycom test site? |
| 17:20.48 | Naikrovek | i dunno, are they good looking |
| 17:21.23 | *** join/#asterisk afink (~afink@wsip-184-187-15-226.om.om.cox.net) |
| 17:22.11 | chuckf | If my wife were on the polycom test site I'd be wondering what the heck she was doing there |
| 17:22.26 | [TK]D-Fender | "Still having issues, the picture is all fuzzy." "No no no, she actually looks like that...." |
| 17:22.29 | Naikrovek | flirting with MrTelephone |
| 17:22.54 | chuckf | better MrTelephone than me I guess |
| 17:22.59 | Naikrovek | it'll be like omegle polycom-style |
| 17:23.05 | chuckf | :) |
| 17:24.38 | [TK]D-Fender | Chat Roulette <- |
| 17:24.53 | [TK]D-Fender | "Cannot be unseen" |
| 17:25.07 | Naikrovek | true enough |
| 17:25.26 | chuckf | is that still running? |
| 17:25.30 | Naikrovek | roulette is an appropriate word for that. lose 99/100 times |
| 17:25.34 | IsUp | sorry i am asking again but i cant find any solution. is there any way to trigger 'resetinterval', i mean PRI resetting without restart Asterisk? because after some load, channels are going to "PRI Flags: Resetting" and outgoing calls are not working. |
| 17:25.35 | Naikrovek | oh yeah |
| 17:26.09 | [TK]D-Fender | Naikrovek, Wesley Snipes said to always bet on black ...s o why is it so full of white trash? :) |
| 17:26.13 | MrTelephone | some test calls are good looking |
| 17:26.20 | Naikrovek | [TK]D-Fender: lol |
| 17:26.46 | MrTelephone | 10 year old kids here were watching old men flash their penis' on chat roulette. the cops were involved and everything |
| 17:26.54 | [TK]D-Fender | is pissed ... first day of snow. |
| 17:27.46 | MrTelephone | I had dns = > mysql-asterisk-local instead of dsn = > mysql-asterisk-local in my res_odbc. Another waste of 30 mins because of a typo. |
| 17:28.10 | afink | [TK]D-Fender: yuck |
| 17:28.37 | afink | btw, the issue the other day was a password problem. |
| 17:29.05 | MrTelephone | terrible afink |
| 17:29.35 | chuckf | [TK]D-Fender: is it much snow? |
| 17:30.25 | [TK]D-Fender | chuckf, Not a hug hit, but its HERE. I think its supposed to be around 5cm today, and then warming up enough to be gone a day or few |
| 17:31.09 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 17:33.47 | MrTelephone | people start dying here when it snows |
| 17:34.06 | MrTelephone | 13 year old kid from newfoundland just got killed by an 18 wheeler a couple days ago |
| 17:34.07 | *** join/#asterisk kylix (~trek@ppp-157-69.27-151.libero.it) |
| 17:34.31 | kylix | Hi ! Anybody using Jaber with Asterisk? |
| 17:34.45 | IsUp | hello kylix, not me. |
| 17:34.50 | kylix | :) |
| 17:35.02 | p3nguin | ~ask |
| 17:35.02 | 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. |
| 17:35.37 | kylix | Practically I don't know how to intercept an IM message from a buddy |
| 17:36.11 | kylix | I'd need when receiving an IM properly formatted, like a password, to give back that user (always IM) some options |
| 17:37.30 | kylix | or, for example, I'd give back the user some possible options after he gives me a "BUZZ" |
| 17:38.14 | kylix | I saw a lot of examples on the net but all of them were sending the options only after a call intercepted in the dialplan |
| 17:38.30 | kylix | Instead of that call, I need to monitor the IM messages |
| 17:38.58 | kylix | some sort of an IVR but built on the IM options |
| 17:40.06 | chuckf | sounds like you need a jabber bot kylix |
| 17:40.58 | kylix | I have asterisk installed on a modem/router based on OpenWRT |
| 17:41.13 | kylix | That's why I'd have to do it all on that router ... |
| 17:41.48 | kylix | Somebody told me it is possible to do it with Asterisk 10 but, for the moment, asterisk 10 is not available on OpenWRT repository |
| 17:42.35 | kylix | and who knows if my openwrt router manages to chew Asterisk 10 ... |
| 17:46.29 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v028-178.mobile.uci.edu) |
| 17:50.51 | MrTelephone | if you change odbc.ini what asterisk modules should I be reloading? I tried reloading res_odbc.so and it keeps saying res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified |
| 17:51.28 | MrTelephone | my original dsn works.. hmm |
| 17:53.03 | *** join/#asterisk Tond (~tond@bas3-toronto47-1242432806.dsl.bell.ca) |
| 17:53.05 | Tond | Hi, |
| 17:53.41 | Tond | If i set my rtp port range to a smallr range of 10000 - 15000 will I face one way audio? |
| 17:55.14 | p3nguin | No. |
| 17:55.39 | p3nguin | You just need enough ports in the range to support as many calls as you intend to make/receive. |
| 17:55.54 | p3nguin | Consider at least one port for each direction per call. |
| 17:56.11 | MrTelephone | I think i installed a non distribution odbc package and the config file is in a different location. dammit |
| 17:56.14 | p3nguin | So you'll be able to support at least 2000 calls with that range. |
| 17:56.37 | Tond | p3nguin> thanks. Because I am facing one way or no audio on some calls and trying to figure out where it is coming from |
| 17:56.44 | p3nguin | That's a NAT problem. |
| 17:56.45 | [TK]D-Fender | rfc2833 = 2 per direction |
| 17:56.53 | Tond | should i set rtpchecksums=yes ? |
| 17:57.03 | [TK]D-Fender | unrelaed |
| 17:57.07 | p3nguin | Oh, two... so that reduces to maybe 1000 calls. |
| 17:57.09 | Tond | I am not behind nat, neither is the other party. Public ip to public ip |
| 17:57.11 | [TK]D-Fender | unrelated* |
| 17:58.10 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za) |
| 17:58.23 | Tond | Hmm. Where do you think it can be coming from? |
| 17:58.35 | Tond | sometimes I get no audio and sometimes oneway |
| 17:58.43 | *** join/#asterisk timahvo1 (~rogue@41.90.63.215) |
| 17:58.46 | Tond | and I am not doing transcoding |
| 17:58.58 | [TK]D-Fender | Tond, Think you should be showing us a complete call with SIP debug enabled... |
| 17:59.00 | [TK]D-Fender | ~pb |
| 17:59.00 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 17:59.02 | [TK]D-Fender | ^^^^ |
| 17:59.53 | Tond | [TK]D-Fender> Sure |
| 18:05.34 | Tond | Here is the SIP debug: http://pastecode.com/fv |
| 18:07.47 | *** join/#asterisk owmyeyes (~hlau@fwny2.delivery.com) |
| 18:09.24 | [TK]D-Fender | Now without masking anything except passwords... |
| 18:09.35 | *** join/#asterisk voipeng (~tom@70.44.203.146.res-cmts.brd2.ptd.net) |
| 18:09.50 | [TK]D-Fender | And the complete call, not just half |
| 18:09.56 | voipeng | when installing asterisk now, is the asterisk-gui or freepbx prefered? |
| 18:10.02 | [TK]D-Fender | along with standard full verbose |
| 18:10.09 | [TK]D-Fender | voipeng, FreePBX |
| 18:10.12 | voipeng | ok thanks |
| 18:12.27 | voipeng | asterisk 1.6 or 1.4 with freepbx? 1.6 i imagine |
| 18:13.06 | [TK]D-Fender | voipeng, latest, then bump to 1.8 after bringing the relase up to date |
| 18:13.14 | voipeng | k |
| 18:13.25 | [TK]D-Fender | voipeng, max out the ISO options. Then update FreePBX to 2.9, then * to 1.8 |
| 18:13.43 | voipeng | k |
| 18:14.39 | [TK]D-Fender | voipeng, If this is a purely testing environment you can leave * behind at 1.6.x for a little bit. You will want the lastest stable FreePBX however for a fair evaluation of what the GUI offers you |
| 18:15.27 | voipeng | testing for now, would like to offload conference bridge to this box if i can get it working with our existing pbxes |
| 18:15.47 | MrTelephone | I thought ${ACCOUNTCODE} was a variable? |
| 18:16.05 | p3nguin | ${CDR(accountcode)} |
| 18:16.31 | MrTelephone | ahh right. thanks |
| 18:16.50 | p3nguin | I did the same thing recently and couldn't figure out why it was blank. |
| 18:17.07 | MrTelephone | I know and I just searched asterisk variables and it was referring to the old method |
| 18:17.29 | p3nguin | So it used to be valid? |
| 18:18.26 | *** join/#asterisk kikohnl (~kotis@ext-dip-171.hnl.cdsinc.com) |
| 18:23.01 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 18:24.08 | ilj | wdoekes2, I tried symlink anyway, but it wouldn't load the app because it was compiled with different version of the tools than that which was used to compile asterisk |
| 18:24.08 | MrTelephone | yeah in the old versions |
| 18:24.08 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 18:24.26 | p3nguin | Okay, so I didn't lose my mind... |
| 18:24.28 | p3nguin | this time. |
| 18:25.01 | MrTelephone | penguin are you good with regex? I have this line {IF(${REGEX("^807229" ${CALLERID(num)})}?${CALLERID(num):3}:${CALLERID(num)}) }) 4 times because I need to match 807229, 807826, 807822, 807825,807228. |
| 18:25.22 | MrTelephone | I want to consolidate into one line but I tried before with no luck |
| 18:26.08 | MrTelephone | ^807(229,826,822,825,228)? |
| 18:26.19 | p3nguin | 807[28]2[25689] maybe? |
| 18:27.44 | p3nguin | I guess that technically matches more than just your four numbers, though. |
| 18:28.06 | MrTelephone | can you do groups inside character classes? |
| 18:28.25 | [TK]D-Fender | - Tond has quit () \o/ |
| 18:29.04 | [TK]D-Fender | MrTelephone, If you have 4 lines to do that then your use of regex is wasteful |
| 18:29.36 | MrTelephone | I tried grouping and I don't think it worked |
| 18:29.56 | [TK]D-Fender | MrTelephone, then you should correct it |
| 18:30.07 | [TK]D-Fender | MrTelephone, Should only have been a minor "oops" |
| 18:30.46 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:31.07 | MrTelephone | ${IF(${REGEX("^807(229,826,822,825,228") ${CALLERID(num)})}?${CALLERID(num):3}:${CALLERID(num)}) }) |
| 18:31.13 | MrTelephone | shouldn't that work or am I stupid? |
| 18:31.27 | MrTelephone | " is in the wrong place |
| 18:34.44 | MrTelephone | this might work /^807[(229),(826),(822),(825),(228)]/ |
| 18:36.31 | *** join/#asterisk enoch (~carmine@unaffiliated/enoch) |
| 18:36.38 | enoch | hi all |
| 18:37.10 | navaismo | hi |
| 18:38.08 | enoch | i have to make a 8 isdn-channels pbx server |
| 18:38.20 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 18:38.25 | enoch | what is the best cost effective hardware to use? |
| 18:38.33 | enoch | digium sells it over 3k dollars :( |
| 18:39.23 | Naikrovek | why isdn |
| 18:39.47 | WIMPy | enoch: There are lots of alternatives. But the hardware is pretty identical. Always a HFC-4S chip. |
| 18:39.47 | Naikrovek | oh this is for some applications where the audio quality is important |
| 18:40.36 | enoch | i need an hw for 8 bri port (16 line channels) |
| 18:40.54 | WIMPy | Ok, then it's te HFC-8s |
| 18:40.57 | Naikrovek | so get the card you need |
| 18:41.06 | WIMPy | You said 8 channels. |
| 18:41.13 | enoch | 8 bri ports sorry :D |
| 18:41.35 | WIMPy | Junghanns, Beronet, etc. |
| 18:41.57 | WIMPy | But wouldn't a PRI be cheaper? |
| 18:43.58 | enoch | bri and pri difference? depends on isp? |
| 18:44.07 | [TK]D-Fender | enoch, http://www.voipango.de/ISDN-Interfaces/Digium/BRI-PRI-E1-T1-J1/Digium-B410PF-Quad-BRI-HW-EC.html |
| 18:44.11 | WIMPy | PRI=30 channels |
| 18:44.18 | [TK]D-Fender | enoch, Dunno where yuo got your $3K figure from |
| 18:44.21 | WIMPy | (or 24 in NA) |
| 18:44.44 | enoch | wwait |
| 18:45.23 | [TK]D-Fender | "digium sells it over 3k dollars :(" <- Umm.. what & where? |
| 18:45.32 | enoch | [TK]D-Fender: sorry it was 8 t1/e1 |
| 18:45.39 | *** join/#asterisk pembo13 (~chatzilla@mail.mediagistic.com) |
| 18:45.57 | WIMPy | What? |
| 18:45.59 | enoch | [TK]D-Fender: but digium sells max 4 bri board |
| 18:46.00 | [TK]D-Fender | enoch, Not at all the same. |
| 18:46.06 | WIMPy | What ecxactely do you have/need? |
| 18:46.09 | [TK]D-Fender | enoch, ... so you buy two of them |
| 18:46.12 | enoch | so i have to buy two 4 bri? |
| 18:46.28 | WIMPy | Or one 8 BRI. |
| 18:46.30 | [TK]D-Fender | enoch, nobody figured an 8 port BRI card was worth building it seems. |
| 18:46.32 | Qwell | with that many channels, why not just get a single PRI? |
| 18:46.54 | WIMPy | There are lots of 8 BRI cards. |
| 18:46.59 | [TK]D-Fender | enoch, 2 cards. Or some other external device (which will probably be rather expensive) |
| 18:47.11 | [TK]D-Fender | WIMPy, Link away. |
| 18:47.12 | WIMPy | But Indeed I'd suggest looking at a PRI instead. |
| 18:47.17 | [TK]D-Fender | As would I |
| 18:47.22 | enoch | [TK]D-Fender: im ok now... |
| 18:47.31 | enoch | and for the rest of hw? |
| 18:47.44 | enoch | i've never installed a 16 channels pbx |
| 18:47.54 | WIMPy | I already mentionde Junghanns and beronet. You fond them on voipango as well. |
| 18:48.08 | *** join/#asterisk staff_nowa (~staff@84.32.248.7) |
| 18:48.16 | *** join/#asterisk AmirBehzad (~behzad@86.57.4.72) |
| 18:48.23 | WIMPy | Sirrix may also have one. |
| 18:48.27 | enoch | what cpu do i need? approximately |
| 18:48.33 | WIMPy | As already said: They are all the same anuyway. |
| 18:48.33 | staff_nowa | hello, i have one question, can i send locally sms message to sip accounts? |
| 18:48.43 | staff_nowa | [Nov 23 20:44:22] WARNING[29715] channel.c: Unable to find a codec translation path from (nothing) to (gsm) |
| 18:48.44 | staff_nowa | [Nov 23 20:44:22] WARNING[29715] file.c: Unable to open invalid (format (nothing)): Function not implemented |
| 18:48.44 | staff_nowa | [Nov 23 20:44:22] WARNING[29715] app_playback.c: ast_streamfile failed on Message/ast_msg_queue for invalid |
| 18:48.52 | pembo13 | is it possible to use to asterisk installs to essentially tunnel from one phone network to another over the internet? |
| 18:48.56 | pembo13 | cellphone -- landline -- asterisk -[internet]- asterisk -- landline -- phone |
| 18:49.18 | WIMPy | pembo13: yes |
| 18:49.32 | [TK]D-Fender | WIMPy, Ah, I see it... missed the port splitters |
| 18:49.57 | pembo13 | WIMPy: what is your recommended resource? books or on-line documentation ? |
| 18:50.14 | WIMPy | ~book |
| 18:50.14 | infobot | rumour has it, thebook is Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or ~buybook |
| 18:50.21 | WIMPy | pembo13: ^^ |
| 18:50.59 | pembo13 | WIMPy: any Amazon affiliation numbers, etc that would benefit the community? |
| 18:52.01 | enoch | [TK]D-Fender: for 16channels, ivr and fax what hw do i need approximately |
| 18:52.29 | [TK]D-Fender | enoch, common desktop. Interfaces with enough ports. |
| 18:52.52 | enoch | [TK]D-Fender: like a phenom x2? or i5? |
| 18:52.57 | [TK]D-Fender | more than enough |
| 18:54.17 | pembo13 | WIMPy: my apologies, i've found the book linked to from the .org site |
| 18:54.21 | WIMPy | It should be a little more than the usual "the oldes machine from the attic" as you might want to run echo cancellation on the PC. |
| 18:54.54 | WIMPy | But seriousely: look at a PRI. |
| 18:55.16 | *** join/#asterisk lcat (~lcat@187.45.254.202) |
| 18:55.28 | WIMPy | Here they wouldn't give you 8 single lines. |
| 18:56.01 | enoch | thanks all |
| 18:56.03 | enoch | bye |
| 18:57.36 | MrTelephone | asterisk doesn't like this regex ^807(?:229|826|822|825|228) |
| 19:00.05 | [TK]D-Fender | MrTelephone, Stop pummeling yourself and stop using regex... not needed |
| 19:00.08 | staff_nowa | ff_nowa> hello, i have one question, can i send locally sms message to sip accounts? |
| 19:00.10 | staff_nowa | ? |
| 19:01.05 | staff_nowa | join #asterisk-ru |
| 19:01.29 | MrTelephone | Fender, do you stay that because you have another idea in mind?? |
| 19:02.04 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v028-178.mobile.uci.edu) |
| 19:03.26 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 19:06.24 | [TK]D-Fender | MrTelephone, | <----- |
| 19:06.40 | [TK]D-Fender | MrTelephone, or 4 cleaner looking lines |
| 19:14.34 | *** join/#asterisk SaadTalaat (c1e3293a@gateway/web/freenode/ip.193.227.41.58) |
| 19:15.08 | MrTelephone | It was flooding the console |
| 19:16.08 | MrTelephone | when you set setcallerpres(prohib) I know it changes your sip From: header to From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as018e3025. but does it also change a value in the Q931/ISDN messages? |
| 19:17.09 | WIMPy | MrTelephone: It should. |
| 19:17.15 | p3nguin | SetCallerPres() is deprecated. |
| 19:17.28 | WIMPy | yes |
| 19:18.00 | MrTelephone | I've had some instances where certain company systems wouldn't accept calls from certain caller information. Have you seen that before? I used to set callerid(num)="" to null and it would cause that |
| 19:18.04 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 19:18.12 | WIMPy | And yes, it works with chan_dahdi and chan_lcr. From my list, it didn't seem to have worked with chan_misdn. |
| 19:18.40 | p3nguin | I don't accept invalid caller id numbers. |
| 19:18.55 | JunK-Y | moo! |
| 19:18.58 | MrTelephone | so what is the actual callerid if you set callerpres prohib? |
| 19:19.11 | [TK]D-Fender | JunK-Y, Salut |
| 19:19.23 | WIMPy | MrTelephone: The same as without. |
| 19:19.37 | MrTelephone | is that equivalent to "" or NULL? |
| 19:19.51 | [TK]D-Fender | no |
| 19:20.06 | [TK]D-Fender | " <- literal char |
| 19:21.09 | MrTelephone | but i think if you set callerid(num) = "" in the dialplan the isdn message has DISPLAY IE BLANK instead of none at all. I would have to confirm |
| 19:21.31 | p3nguin | Quotes should never be used in caller id. |
| 19:21.36 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 19:21.43 | WIMPy | An empty CGPN might be rejected as invalid. |
| 19:22.14 | JunK-Y | at least, use CALLERID(num)=666 ! |
| 19:22.35 | WIMPy | And you must not send DISPLAY on TE interfaces, altough it seems to work on _some_ NI-2 switches. |
| 19:23.13 | MrTelephone | we are on an old switch so it sends as DISPLAY. found this out the hard way after spending 3,000 on a switch that doesn't support callerid name on DMS100 |
| 19:23.35 | p3nguin | In my experience, if you try to shove an invalid caller id number out the PRI, the call will default to your assigned callerid number. |
| 19:23.35 | MrTelephone | so a big middle finger to adtran |
| 19:24.21 | MrTelephone | penguin, that happens, it will replace with our main number |
| 19:24.27 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:24.37 | _Corey_ | p3nguin: Depends on the carrier... I have a bunch that will send along whatever nonsense we set |
| 19:25.14 | MrTelephone | So what is the proper way to hide a customers name/number when calling out ISDN or something similar? |
| 19:25.43 | WIMPy | MrTelephone: set CALLERID(pres) |
| 19:25.45 | MrTelephone | ok |
| 19:26.26 | WIMPy | =prohib it seems. |
| 19:26.28 | p3nguin | It should accept the same values as the old SetCallerPres() |
| 19:26.32 | MrTelephone | I asked before but I'll try asking again. failed_screening or screened. what is the point? |
| 19:26.53 | MrTelephone | prohib,unavailable,allowed seem to be the common options |
| 19:27.03 | _Corey_ | I've only used prohib |
| 19:27.15 | p3nguin | Set(CALLERID(num-pres)=prohib) |
| 19:27.33 | WIMPy | They are only useful for received numbers or in NT interfaces. |
| 19:28.51 | MrTelephone | Is it just a value in the isdn message then saying its screened. Then whatever equipment you use will either support that feature or not. |
| 19:29.14 | WIMPy | What kind of support? |
| 19:29.24 | MrTelephone | not sure |
| 19:29.31 | MrTelephone | seems to be a useless option |
| 19:30.04 | WIMPy | No, it's used for calls from lines with CLIP(no screening) enabled. |
| 19:30.16 | MrTelephone | So your customer dials *67 or whatever to hide their caller information on demand. You still have to set pres=prohib somehow? |
| 19:30.45 | WIMPy | You will receive two caller IDs: The unscreened one and a network provided one. |
| 19:30.49 | p3nguin | Unless you are passing *67 all the way to the provider and the provider acts on it, you're in charge of all vertical service codes. |
| 19:32.12 | MrTelephone | so I'll have to test CallerName and if it's anonymouse set pres to prohib. |
| 19:32.48 | p3nguin | How would it have gotten to anonymous in the first place? |
| 19:33.33 | MrTelephone | because the endpoints we use automatically change the headers to Anonymous if the user dials the proper * code. |
| 19:33.46 | *** join/#asterisk shadowapex (~William@adsl-70-232-210-174.dsl.irvnca.sbcglobal.net) |
| 19:33.50 | WIMPy | yes |
| 19:33.58 | *** join/#asterisk Dovid (~Dovid@host-78-158-94-199.wlan-guest.nycmny02.us.sargasso.net) |
| 19:34.00 | p3nguin | That's the point where you should be doing all the changes. |
| 19:35.35 | *** join/#asterisk dandate2 (~dan@124.6.157.210) |
| 19:35.45 | dandate2 | how can i disconnect a sip channel from cli/ |
| 19:35.50 | MrTelephone | When a client calls outside our network and the callee dials *69 it returns to our main megalink number. Is that a setting at the telco they can fix? |
| 19:37.03 | p3nguin | That's called callerid number. Send the call from another number, and the callee will do a call return on the other number. |
| 19:37.46 | MrTelephone | But if it's anonymous or unknown it will return to the main PRI number I guess. |
| 19:38.23 | p3nguin | If you mark your calls as anonymous, are the calls really arriving at the destination as anonymous? |
| 19:38.48 | p3nguin | My guess is that they are not. |
| 19:38.51 | dandate2 | one of my extentions went afk on an outbound call, anyway to terminate the call from CLI? |
| 19:38.55 | WIMPy | Only at the network termination. |
| 19:39.06 | p3nguin | dandate2: Extensions don't do that, devices do. |
| 19:39.10 | MrTelephone | It requires more testing on my part. |
| 19:39.25 | p3nguin | (1323.35) <p3nguin> In my experience, if you try to shove an invalid caller id number out the PRI, the call will default to your assigned callerid number. |
| 19:39.28 | p3nguin | (1324.37) <_Corey_> p3nguin: Depends on the carrier... I have a bunch that will send along whatever nonsense we set |
| 19:40.07 | dandate2 | yeah one of my agents with eyebeam is afk forever on an outbound calling racking up minutes. doesnt cli have an option to administratively disconnect it like fop does in freepbx? |
| 19:40.23 | p3nguin | channel request hangup ... |
| 19:40.34 | MrTelephone | I'm just wondering if I set prohib, call a person outside the network, they dial *69, should the call return to our default address? |
| 19:41.00 | WIMPy | *69 is a call back? |
| 19:41.03 | MrTelephone | yeah |
| 19:41.11 | WIMPy | That should go to the number you supplied as caller ID. |
| 19:41.15 | p3nguin | Verify that the call is delivered to the destination with an anonymous caller id number. |
| 19:41.21 | MrTelephone | what if you set prohib? |
| 19:41.31 | WIMPy | It shouldn't matter if it's allowed or prohibited. |
| 19:41.35 | p3nguin | My guess is that it is delivered with your default caller id number, not anonymous. |
| 19:42.11 | MrTelephone | so I should make sure that callerid(num) is set properly and then set prohib |
| 19:42.21 | WIMPy | yes |
| 19:42.22 | p3nguin | I would try that. |
| 19:42.45 | dandate2 | alright i found soft hangup but it says the extetnion # is not a valid channel |
| 19:42.52 | MrTelephone | I've been putting this off but the odd time you get kids fooling around pranking people. the people dial *69 and they get my office. lol |
| 19:43.00 | p3nguin | dandate2: That's because it isn't valid. |
| 19:43.15 | p3nguin | dandate2: Extensions don't make calls, devices using channels do. |
| 19:43.17 | p3nguin | dandate2: hangup the channel. |
| 19:43.43 | WIMPy | MrTelephone: So make sure you send callid caller IDs. |
| 19:43.44 | dandate2 | how do i find the channel ID? its not this callID under 'sip show channels' is it ? |
| 19:43.55 | WIMPy | s/call/val/ |
| 19:44.27 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 19:44.28 | MrTelephone | my mistake then. I was clearing callerid(num) with empty "". it does work for the display on the remote end but really it isn't the proper way |
| 19:44.34 | p3nguin | dandate2: You'll have to look at your channels, probably with core show channels concise, and determine which one you want to drop. |
| 19:44.48 | p3nguin | "" is not a valid caller ID number. |
| 19:44.55 | p3nguin | It's not even valid is a name. |
| 19:45.02 | p3nguin | Quotes should not be used in Caller ID. Period. |
| 19:45.11 | MrTelephone | I'm just saying because you set a variable you need quotes |
| 19:45.14 | dandate2 | so i would type soft hangup SIP/144-00007641 ? |
| 19:45.15 | MrTelephone | no? |
| 19:45.27 | p3nguin | no |
| 19:45.38 | p3nguin | Set(CALLERID(num)=) |
| 19:45.52 | p3nguin | null |
| 19:46.42 | p3nguin | Or use a blank char. Set(CALLERID(num)= ) |
| 19:46.45 | [TK]D-Fender | dandyes |
| 19:46.51 | [TK]D-Fender | dandate2, yes |
| 19:48.51 | MrTelephone | I was on crack I have no quotes in my dialplan. We don't always dial with 10 digits here so that makes it twice as confusing. When we dial long distance I set callernum to 10 digits and local calls I set to 7 digits to appear properly on the phone |
| 19:50.00 | p3nguin | I thought we were talking about outbound caller id settings. |
| 19:50.04 | _Corey_ | MrTelephone: You set your callerid to be 7 digits?? |
| 19:50.19 | WIMPy | The destination should not matter when setting the ID. |
| 19:50.43 | WIMPy | And if it does thare's probably somethign wrong. |
| 19:50.58 | p3nguin | I certainly wouldn't reduce a valid 10-digit caller id number to 7 just to display it on a phone. |
| 19:51.04 | MrTelephone | well if I send all 10 digits to sip channels then they will see all 10 |
| 19:51.36 | p3nguin | The real caller id number should be used unless there is a really good reason to not use it. |
| 19:55.15 | p3nguin | A lot of phones will even have a setting to strip off the area code for you to display only the local 7-digit number on the screen. |
| 19:55.26 | p3nguin | But you have to provide the real 10-digit number for that to work. |
| 19:55.42 | MrTelephone | The dms100 should strip the shit off |
| 19:56.08 | MrTelephone | I only send 7 digits to the telco as well for local calls |
| 19:57.04 | WIMPy | You can probably tell it to do so. |
| 19:57.47 | MrTelephone | incoming isdn calls are sent 7 digit only. |
| 19:58.10 | MrTelephone | I wish they would just adopt the 10 digit dialing everywhere |
| 19:58.33 | WIMPy | Oh, I thought you were saying it was the other way round. |
| 19:58.45 | MrTelephone | both ways |
| 19:58.45 | WIMPy | Can't you define AC in dahdi? |
| 19:58.55 | MrTelephone | remmber my regex? Set(CALLERID(num)=${IF(${REGEX("^807(229|826|822|825|228)" ${CALLERID(num)})}?${CALLERID(num):3}:${CALLERID(num)}) }) |
| 19:59.26 | WIMPy | Yes, you can. |
| 19:59.33 | WIMPy | See "localprefix" |
| 19:59.53 | MrTelephone | incoming callerid shows up 10 digits so I have to strip the area code off so that it displays as the number should be dialed |
| 20:00.02 | WIMPy | The screening features of chan_dahdi are limited, but not absent. |
| 20:00.21 | WIMPy | Err, ok, so it was the way I thought at first. |
| 20:01.00 | p3nguin | I would use the full 10-digit caller id numbers throughout, and if the telco only allows 7-digit for the local area code, I'd deal with that in dialplan for the outbound calls. |
| 20:01.06 | _Corey_ | MrTelephone: Does your carrier reject the calls if they're sent to them 10 digit? In my experience in these 7 digit areas I've never had to remove numbers |
| 20:01.07 | WIMPy | But as p3nguin said, altering the display of caller IDs should be done on the phone. |
| 20:01.10 | MrTelephone | the number dialed is 7 digit but callerid is 10 digit. does that make sense to you? |
| 20:01.17 | p3nguin | Yes. |
| 20:01.33 | MrTelephone | yeah they reject 10 digit callouts for local calls |
| 20:01.33 | WIMPy | Again: What p3nguin said. |
| 20:01.52 | MrTelephone | Do you understand my confusion though? |
| 20:02.08 | _Corey_ | MrTelephone: What carrier is this? (I need to re-rank my bad carrier list) |
| 20:02.10 | MrTelephone | I'll set all callerid to 10 digits and work with it from there |
| 20:02.18 | p3nguin | Every landline service I've known requires a call in the local area code to dial by 7 digits, but caller id is always the full NANP caller id number. |
| 20:03.01 | MrTelephone | bell canada |
| 20:03.27 | MrTelephone | they suck but there is only 1 dms 100 for northern ontario that I know of |
| 20:03.33 | p3nguin | Now in a geographic area where my next door neighbor can have a different area code, they may not even allow 7-digit dialing at all. |
| 20:04.10 | _Corey_ | p3nguin: Hmm, my experience with POTS lines in those markets is pretty much nil so maybe that explains it |
| 20:04.13 | MrTelephone | they were thinking about making 10 digit dialing across ontario.. hopefully they do. The urban areas already have to dial 10 |
| 20:04.46 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mjfrogphhsepnbwj) |
| 20:04.56 | _Corey_ | ouch |
| 20:05.05 | p3nguin | Standardize your shit to be 10-digits. If your telco requires 7 for local area code, strip only the local area code off in dial plan before the call is sent out. |
| 20:05.16 | MrTelephone | alright |
| 20:05.20 | p3nguin | It would be one or two lines of dialplan. |
| 20:05.23 | MrTelephone | I'll do that and I should be good to go |
| 20:05.32 | MrTelephone | You guys really helped me out a lot |
| 20:06.31 | MrTelephone | It explains some call return problems and not being able to get through to some answering systems |
| 20:07.38 | p3nguin | something like: exten => _314NXXXXXX,1,Dial(DAHDI/g0/${EXTEN:3}); where your area code is 314. |
| 20:10.16 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 20:22.27 | Naikrovek | http://www.youtube.com/watch?v=lTx3G6h2xyA |
| 20:23.22 | Naikrovek | also |
| 20:23.23 | Naikrovek | http://www.youtube.com/watch?v=IxPbgnO81sQ |
| 20:23.55 | Naikrovek | never has a song made me want to dance, not once. then this shit just pops up and now i'm too out of shape to dance |
| 20:24.36 | [TK]D-Fender | Round is a shape :) |
| 20:24.56 | p3nguin | haha |
| 20:25.01 | Naikrovek | lazy has a shape, and it is Naikrovekian in proportions |
| 20:27.36 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 20:28.14 | ChannelZ | That was one of the nerds from Freaks & Geeks |
| 20:28.24 | Naikrovek | yeah the dude with the radio |
| 20:28.37 | Naikrovek | martin starr? marvin starr i forget which |
| 20:29.19 | ChannelZ | Not even sure I just knew I recognized him |
| 20:30.02 | Naikrovek | the video gets better by a factor of about 4 when the woman joins in |
| 20:31.56 | Naikrovek | time for me to go visit my brother in the pokey |
| 20:32.00 | Naikrovek | (jail) |
| 20:35.35 | [sr] | hi people |
| 20:36.29 | [sr] | i need something, that can sniff an application that i use from my mobil operator, to see the form of the username that the software sends to the sip server |
| 20:37.02 | [sr] | i already have the server name, and can authenticate, but get an auth error, not sure if it's the useragent of the form of the username |
| 20:37.32 | p3nguin | Does the device use wifi? |
| 20:37.53 | [sr] | no, its a software for windows |
| 20:38.10 | [sr] | made by the own ISP |
| 20:38.17 | p3nguin | Oh. Then it's on your network, and you can easily sniff it with tcpdump or wireshark. |
| 20:40.48 | [sr] | hum, let me check the tools for win :) |
| 20:41.20 | *** part/#asterisk AmirBehzad (~behzad@86.57.4.72) |
| 20:41.57 | *** join/#asterisk AmirBehzad (~behzad@86.57.4.72) |
| 20:44.11 | [sr] | ISP's on this country always block stuff... they could win more money if this was open but ok |
| 20:44.34 | p3nguin | Port taxes! |
| 20:51.17 | [sr] | wow so much errors, this wierdo software connects to an HTTP service aswell, but all it gives is 404 not found, beatiful! |
| 20:51.42 | p3nguin | Consider that 404 is a valid response. |
| 20:51.50 | p3nguin | It might just be checking if the web server responds. |
| 20:52.32 | MrTelephone | probably downloading a config |
| 20:52.40 | MrTelephone | like the papsmear T's |
| 20:52.47 | [sr] | no, it's trying to download an avatar.jpg, maybe for profile |
| 20:52.51 | MrTelephone | or spaz2102's |
| 20:53.07 | MrTelephone | it's trying to cost you money |
| 20:54.02 | [sr] | ok useragent already have it |
| 20:58.50 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:00.11 | *** part/#asterisk AmirBehzad (~behzad@86.57.4.72) |
| 21:17.59 | [sr] | damn still have an " -- Got SIP response 503 "Service unavailable"" |
| 21:18.06 | [sr] | that's not the useragent |
| 21:18.18 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v028-178.mobile.uci.edu) |
| 21:38.34 | *** join/#asterisk pietro (~pietro@88-149-226-166.dynamic.ngi.it) |
| 21:48.59 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 22:05.55 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 22:25.39 | *** join/#asterisk Fubard (~brawr@vpn.bctconsulting.com) |
| 22:28.51 | Fubard | Upgraded from 1.6 branch to 1.8 (not sure if this is related or not) but now I am getting gsm files created in the /tmp directory. The files themselves are not really an issue, they take up little to no space but I am trying ot figure out what is causing them to be created. Anyone have any thoughts? |
| 22:29.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:29.42 | Fubard | they are all named after a DID, in the form of NPANXXXXXX.gsm, if that helps. |
| 22:32.43 | p3nguin | This is annoying as can be. I have two accounts on the same provider sending calls to the same asterisk system with two sip peer entries, but the incoming calls only match the first entry. |
| 22:33.06 | p3nguin | Even using type=friend and having defaultuser specified, calls still match the first entry. |
| 22:33.43 | p3nguin | Should I explicitly define a peer and a user for each account? |
| 22:34.23 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 22:36.37 | [TK]D-Fender | yup |
| 22:36.39 | Fubard | p3nguin: it sounds like a problem with your provider not passing the calls correctly. have you contacted them? |
| 22:37.03 | p3nguin | Could it be a result of my using trustrpid=yes? |
| 22:37.26 | [TK]D-Fender | nope |
| 22:37.59 | p3nguin | Let me split each into a peer and a user and see what we get. |
| 22:41.25 | p3nguin | Didn't help. Still matched the first peer. |
| 22:42.10 | p3nguin | I don't see anything in the sip debug that indicates which userid the call is intended for. |
| 22:43.10 | p3nguin | So that's going to be a problem. |
| 22:44.47 | [TK]D-Fender | p3nguin: Share up if you me to take a look |
| 22:50.02 | *** join/#asterisk phl4kx (~phl4kx@190.223.54.18) |
| 22:50.04 | phl4kx | hi friends |
| 22:50.06 | phl4kx | I have a quesiton |
| 22:50.08 | phl4kx | question |
| 22:50.17 | phl4kx | I have a TE122B digium Card |
| 22:51.03 | phl4kx | I like to use like E1, I have to put the jumper totaly?, or only the middle? |
| 22:51.04 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 22:51.52 | Fubard | exit |
| 22:53.07 | navaismo | any ideas where to look why the HPEC codec crahs the whole server when i try to call |
| 22:55.41 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 22:56.42 | *** join/#asterisk Russ (~russ@206.29.182.241) |
| 22:56.57 | phl4kx | please help |
| 22:57.36 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 22:57.43 | SeRi | phl4kx, 4 emergency call 911, 999, 900, or even 800. |
| 22:58.46 | phl4kx | <PROTECTED> |
| 22:59.04 | phl4kx | I like to use like E1, I have to put the jumper totaly?, or only the middle? |
| 22:59.13 | phl4kx | some help with this problem? |
| 22:59.28 | SeRi | phl4kx, I am sure some of the vets here will jump in soon. |
| 22:59.44 | SeRi | Most people are right now getting off working and on there way home... |
| 22:59.55 | phl4kx | ok, Im waiting thanks |
| 23:00.06 | SeRi | n/p and good luck. |
| 23:00.07 | SeRi | bbl |
| 23:04.14 | [TK]D-Fender | phl4kx: www.digium.com |
| 23:11.02 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 23:14.44 | *** join/#asterisk radic (~radic@dslb-178-002-216-111.pools.arcor-ip.net) |
| 23:17.43 | *** join/#asterisk CrashHD (CrashHD@209.65.244.26) |
| 23:22.49 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 23:25.03 | *** join/#asterisk S1lv3R (NoMail@178-83-38-168.dynamic.hispeed.ch) |
| 23:25.21 | S1lv3R | mrning |
| 23:30.28 | p3nguin | [tk]d-fender: I can't find anything in the packet to help it designate which entry asterisk should match on. http://pastebin.com/VySNEE5a |
| 23:30.48 | SeRi | waz up p3nguin |
| 23:31.43 | [TK]D-Fender | ok, it's IP matching for the peer so far... |
| 23:32.27 | p3nguin | ... because there's no username info in the packet. |
| 23:32.36 | p3nguin | At least that's my guess. |
| 23:32.56 | p3nguin | I defined the two "user" accounts before the "peer" accounts. |
| 23:33.39 | [TK]D-Fender | make sure your peer is "insecure=no" |
| 23:33.49 | [TK]D-Fender | that will stop any auto-accept based on host |
| 23:39.59 | p3nguin | When I set that, now I'm unavailable, and my calls go to the provider's voicemail system. |
| 23:43.45 | *** part/#asterisk umay (~chris@67-6-172-210.hlrn.qwest.net) |
| 23:44.27 | *** join/#asterisk kaushal (~kaushal@59.161.24.65) |
| 23:45.52 | [TK]D-Fender | Now see about tweaking up the users |
| 23:46.08 | S1lv3R | Gibt es hier auch deutschen Support ? |
| 23:46.32 | WIMPy | S1lv3R: #asterisk-de |
| 23:46.34 | [TK]D-Fender | of course none of this is a guarantee that it will work. They could simply be throwingun-authed calls at you left right and center.... |
| 23:46.54 | S1lv3R | WIMPy thanks |
| 23:52.39 | *** join/#asterisk neurosys (~neurosys@c-67-191-66-234.hsd1.fl.comcast.net) |
| 23:59.26 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 23:59.26 | *** mode/#asterisk [+o file] by ChanServ |
| 23:59.56 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
