| 00:00.57 | *** join/#asterisk nighty- (~nighty@69-165-220-105.dsl.teksavvy.com) |
| 00:01.38 | Freeaqingme | nvm. got it (someone is going to update the docs...) |
| 00:06.59 | *** join/#asterisk afink (~user@wsip-184-187-15-226.om.om.cox.net) |
| 00:22.32 | Freeaqingme | if someone wants to contribute to the manual, where should he/she start? |
| 00:27.04 | *** join/#asterisk pcpower (~bparker@fiveforty.net) |
| 00:28.12 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 00:28.26 | pcpower | I have a polycom phone setup as type=friend in sip.conf, I do not have an extension setup where it can called from, but somehow I am getting lots of random calls to this phone from all over the world |
| 00:28.31 | pcpower | how can I stop this? |
| 00:32.12 | [TK]D-Fender | pcpower: it is clearly getting direct calls from the outside, not your PBX then. Don't leave your phones directly open on the internet |
| 00:39.32 | *** join/#asterisk jgautier (18201997@gateway/web/freenode/ip.24.32.25.151) |
| 00:39.38 | jercos | ... Well shoot, it's making an unauthenticated forward of the INVITE instead of proxying it. Um. |
| 00:39.41 | jercos | stares at his configsd |
| 00:39.53 | jgautier | any landline experts out there? or any idea where if theres any irc channels? |
| 00:40.33 | jercos | jgautier: how do you mean, and whatall do you need? |
| 00:41.08 | jgautier | well i want to capture the caller id of a landline via the internet with something like tropo or twilio |
| 00:41.21 | jgautier | not sure if its possible |
| 00:41.28 | jgautier | i talked to some guys from there and we havent found anything |
| 00:41.33 | jgautier | just wanted to see if anyone had ideas here |
| 00:43.55 | [TK]D-Fender | ? |
| 00:44.20 | jgautier | ? |
| 00:45.01 | [TK]D-Fender | "capture the caller id of a landline via the internet" |
| 00:45.16 | [TK]D-Fender | How does "the internet" capture callerid? |
| 00:45.21 | jercos | with SCIENCE |
| 00:46.01 | jercos | I think he means using a cloud VoIP service, capturing the caller ID. |
| 00:46.06 | jgautier | yeah |
| 00:46.13 | jgautier | this is what i have but doesnt quite work all the way |
| 00:46.40 | jgautier | i use *73 to forward the landline to the number of the my VOIP based app i can then get the caller id |
| 00:46.48 | jgautier | but i dont know how to get back to the landline, so the call goes through all the way |
| 00:47.04 | jgautier | im thinking its not possible |
| 00:47.22 | [TK]D-Fender | You've just stated you hard-forwarded your telco line |
| 00:47.27 | [TK]D-Fender | That's it. Game over |
| 00:47.29 | [TK]D-Fender | No going back |
| 00:47.46 | [TK]D-Fender | impossible strategy |
| 00:47.55 | jgautier | yeah im just wondering if there could be some other hacky way using 3 way calling or something else |
| 00:48.01 | [TK]D-Fender | You want to grab callerid... put equipment on-site |
| 00:48.23 | jercos | DERP DERP HERP |
| 00:48.28 | jercos | well there's *my* problem solved. |
| 00:48.45 | jgautier | yeah would be a lot cooler to not have any extra hardware |
| 00:49.03 | jercos | For the record, setting fromuser and fromdomain can *occasionally* be helpful in making consistant calling through a sip trunk. |
| 00:49.06 | jercos | >.> |
| 00:52.19 | jgautier | is there some sort small hardware with asterisk, wifi, and a phoneline that i could buy? |
| 00:53.28 | [TK]D-Fender | Buy a netbook & Linksys SPA-3102 |
| 00:55.04 | jgautier | well the problem is i was trying to make an application for restaraunts to use to make reservations, if they have to get all this hardware and setup all this stuff they probably wont want to use that application |
| 00:55.43 | [TK]D-Fender | And maybe they shouldn't |
| 00:55.55 | jgautier | i wouldnt do it if i had to get stuff |
| 00:56.02 | jgautier | but if i could just go to my phone and type some codes i might |
| 00:56.13 | *** join/#asterisk god-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 00:56.47 | [TK]D-Fender | Feel free to complain to your telco that they don't offer internet based CID for their POTS service |
| 00:57.16 | [TK]D-Fender | [20:52]jgautieris there some sort small hardware with asterisk, wifi, and a phoneline that i could buy? |
| 00:57.22 | [TK]D-Fender | [20:55]jgautieri wouldnt do it if i had to get stuff |
| 00:57.27 | [TK]D-Fender | Immediate contradiction |
| 00:57.51 | jgautier | if the hardware was something small that would go inline on the phone line and had wifi i might do it |
| 00:57.55 | [TK]D-Fender | What can I buy for this? I DON'T WANT TO HAVE TO BUY ANYTHING! |
| 00:58.19 | [TK]D-Fender | jgautier: There hasn't been a market demanding what you're asking for. |
| 00:58.49 | [TK]D-Fender | jgautier: So you've got what's out there. Use it or not. Your call |
| 00:59.21 | jgautier | I am just trying to make sure I know about everything is out there |
| 00:59.32 | jgautier | that is out there |
| 01:00.27 | [TK]D-Fender | You might get away with using an embedded system like ALIX, but it's going to end up costing possibly even more and be a dead-end |
| 01:00.44 | [TK]D-Fender | Either way.. hardly seems worth it |
| 01:00.56 | jgautier | yeah thats what it seems like |
| 01:01.45 | [TK]D-Fender | Ok, I'm off for a while... |
| 01:01.55 | jgautier | see ya |
| 01:21.36 | *** join/#asterisk Carko (BloW@84-255-241-35.static.t-2.net) |
| 01:21.53 | Carko | hello |
| 01:24.36 | Freeaqingme | after having run 'make menuselect', where are the options I selected saved? Would like to reuse them later. |
| 01:24.51 | Carko | can anyone tell me why would asterisk "say" back the called number when the call is coming via SIP from an unknown peer? |
| 01:24.56 | Carko | here's an example: http://pastebin.com/raw.php?i=EJ4QBP62 |
| 01:38.55 | Freeaqingme | nvm, got my answer |
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| 02:46.06 | p3nguin | saxa: You should not use DMZ for anything. Forward only the ports you need. |
| 03:09.24 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 03:18.56 | *** join/#asterisk dandate2 (~dan@124.6.157.210) |
| 03:19.13 | dandate2 | is dynamic agent queue weight penalties functioning in 1.4? |
| 03:36.16 | *** part/#asterisk pcpower (~bparker@fiveforty.net) |
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| 04:23.14 | dlynes | Does anyone happen to know if there's any iax2 security fixes coming up for asterisk 1.6.2.21? |
| 04:25.35 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 04:33.18 | *** join/#asterisk OldMonk (~raju@182.68.197.191) |
| 04:34.41 | OldMonk | hi. my queue problem is now an iax2/pstn problem |
| 04:35.54 | OldMonk | fsck |
| 04:41.04 | dlynes | If I can log in to https://issues.asterisk.org/jira/, I don't need to reset my password? |
| 04:41.24 | dlynes | Also, with this new Jira, does it mean we have to resubmit our license agreement? |
| 04:52.31 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279681716.dsl.bell.ca) |
| 04:54.25 | p3nguin | agreement for what? |
| 04:55.05 | dijib | for guiness |
| 04:55.24 | dijib | i still dont get how to use the reseller portal for a cheap rate account |
| 05:08.08 | *** join/#asterisk god-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 05:18.04 | p3nguin | Your rate will not get any lower. |
| 05:18.36 | SeRi | dijib, LOL stop trying to cheat the system. :P |
| 05:19.24 | SeRi | like p3nguin said the rates wont get lower. you can raise them though.. |
| 05:19.52 | SeRi | out to work. cya! |
| 05:19.54 | p3nguin | The reseller area allows you to set rates to resell. |
| 05:20.12 | dijib | im not, im trying to use the premium outgoing route as my primary and have a secondary cheaper rate for calls i dont care about |
| 05:20.59 | SeRi | dijib, It does not work that way. any who I have to go. p3nguin can take this one :) |
| 05:21.02 | dijib | <PROTECTED> |
| 05:21.11 | dijib | p3nguin, doesnt want it |
| 05:22.01 | p3nguin | You could use two accounts and set two routes. |
| 05:22.38 | p3nguin | But I'm headed out, so you'll do it by yourself or wait 10 hours. |
| 05:22.47 | dijib | but then i need different deposits in each account? |
| 05:23.04 | dijib | eff it |
| 05:25.02 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
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| 06:12.34 | Naikrovek | awww shit. polycom ucs 4.0 supports sip reason header. which is TEH AWESOME |
| 06:13.35 | Naikrovek | WHOA: "Added support for User Profiles. Users may log into and out of the phone using a serverindependent, configuration file-based, authentication method." |
| 06:13.52 | Naikrovek | sa-weeet |
| 06:15.27 | Naikrovek | wow. deploying this on a test phone monday |
| 06:16.11 | Naikrovek | "Productivity features such as LDAP, Local Call Recording, and Visual Conference Management |
| 06:16.11 | Naikrovek | are enabled without the requirement of a license file." holy shit |
| 06:16.25 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:17.57 | Naikrovek | oh wait i can't download it yet. pewp. |
| 06:18.41 | ChannelZ | jizzes in his pands |
| 06:18.46 | ChannelZ | s/pands/pants/ |
| 06:25.54 | *** join/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 06:26.01 | jmls | morning all |
| 06:26.33 | jmls | any views on which text-to-speech engine is the best these days for asterisk 10 ? |
| 06:27.09 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
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| 07:05.11 | Naikrovek | festival? |
| 07:05.12 | Naikrovek | no idea |
| 07:20.39 | *** join/#asterisk SparFux (~raoul@e182019147.adsl.alicedsl.de) |
| 07:20.48 | SparFux | Yo. |
| 07:26.17 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 07:29.16 | ChannelZ | Oy |
| 07:33.58 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 07:37.57 | SparFux | Well, my dahdi modification at least survived the night without crashing :-P |
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| 08:05.55 | krotos | hi :) |
| 08:08.43 | ChannelZ | yarr |
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| 08:44.41 | saxa | p3nguin: yes will do that always. |
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| 09:02.01 | jmls | AAARRRRGGGGGGHHHHHHHHHHH |
| 09:02.14 | jmls | res_calendar is driving me mad |
| 09:02.30 | jmls | anyone got a working connection with a googleapps calendar ? |
| 09:02.38 | jmls | (not gmail) |
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| 09:55.50 | Carko | hello |
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| 15:25.33 | autofsckk | hello everybody, is there a rss feed with important security issues about asterisk ? |
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| 16:20.17 | Naikrovek | [TK]D-Fender: did you see all the new features in UCS 4.0.0? whoas |
| 16:20.35 | [TK]D-Fender | Nope... |
| 16:20.40 | [TK]D-Fender | Somewhat behind on those |
| 16:21.44 | Naikrovek | http://downloads.polycom.com/voice/voip/uc/UC_Software_Release_Notes_4_0_0.pdf |
| 16:22.35 | Naikrovek | lots of the productivity license is now free, SIP REASON is supported, hot desking supported by the phone |
| 16:22.48 | Naikrovek | 321 and up |
| 16:22.53 | Naikrovek | well, 321 and later |
| 16:23.02 | Naikrovek | 331 as well of course |
| 16:24.01 | [TK]D-Fender | Hrm, wanna see the hot-desking. And SIP Reason offers an 'Answered Elswhere" option IIRC which has impact with Queues for missed calls. |
| 16:24.09 | [TK]D-Fender | (if they did it right) |
| 16:25.11 | Naikrovek | yes |
| 16:25.30 | Naikrovek | new webadmin as well. |
| 16:25.30 | [TK]D-Fender | LOL, major "bricking" warning on bootrom changes between incompatible versions... |
| 16:25.33 | [TK]D-Fender | This is a big one |
| 16:25.46 | Naikrovek | yah i read that |
| 16:25.50 | [TK]D-Fender | I was hoping they'd jsut pull the web side out to make room for features we care about ;) |
| 16:25.57 | p3nguin | How do I hangup a call that is in state Ringing? I can usually use channel request hangup to kill one that is Up, but it isn't affecting this channel that is Ringing. |
| 16:26.01 | Naikrovek | well the web thing is MUCH nicer now |
| 16:26.18 | [TK]D-Fender | p3nguin: Tried a redirect? |
| 16:26.26 | [TK]D-Fender | p3nguin: Toss it off a cliff |
| 16:26.32 | p3nguin | No, but I will try now. |
| 16:27.37 | p3nguin | Redirect it to any arbitrary extension's hangup should do it? |
| 16:29.09 | p3nguin | It says it was successfully redirected, but the channel is still stuck in state Ringing. |
| 16:30.20 | [TK]D-Fender | :/ |
| 16:30.25 | p3nguin | It doesn't seem to matter where I redirect it, it says successfully redirected but nothing changes. |
| 16:30.32 | [TK]D-Fender | What kind of channel? |
| 16:30.35 | p3nguin | local |
| 16:30.56 | [TK]D-Fender | Is it bridged to anything? |
| 16:31.06 | p3nguin | I tried redirecting to an Answer(), to a Hangup(), to something random... |
| 16:31.18 | p3nguin | No, this is the only active channel. |
| 16:31.38 | [TK]D-Fender | Ok, launch another channel and use Bridge() to try to hook it. |
| 16:31.57 | [TK]D-Fender | If it succeeds them you may be able to kill it either through the dead end or your new one |
| 16:32.48 | p3nguin | I'm not sure where to start. I need to write dial plan to do that? |
| 16:33.34 | p3nguin | Maybe I can originate something on the cli. |
| 16:35.39 | [TK]D-Fender | yup, have another local channel just loop on long delay nd then try to bridge them via AMI |
| 16:35.43 | [TK]D-Fender | Or use the dialplan app |
| 16:35.57 | p3nguin | I don't do AMI, so dial plan it is. |
| 16:39.49 | p3nguin | 0 active channels |
| 16:39.51 | p3nguin | Got it. |
| 16:40.13 | [TK]D-Fender | \o/ |
| 16:40.17 | p3nguin | I think asterisk destroyed it due to an error, though. |
| 16:40.23 | p3nguin | [Oct 29 11:39:32] WARNING[2572]: chan_local.c:640 local_fixup: You can not bridge a Local channel to itself! |
| 16:40.30 | [TK]D-Fender | The best way to die. Violently :) |
| 16:41.09 | p3nguin | If that will make it die every time it happens, I'll be satisfied with that. |
| 16:44.23 | p3nguin | level 1: billsec=88896 |
| 16:44.33 | p3nguin | Good thing it was local. |
| 16:50.00 | *** join/#asterisk brunner (~chris@pdpc/supporter/gold/brunner) |
| 16:51.18 | brunner | I need to hire an asterisk guru for a series of small projects. Any takers? |
| 16:51.35 | p3nguin | Care to describe the tasks? |
| 16:52.15 | brunner | There's no end to the list. Ranges from configuring basic asterisk boxes to writing full blown telephony applications. |
| 16:53.26 | p3nguin | I could manage the asterisk part of it, but I don't do programming. Are you splitting it apart, or does one person need to be able to do it all? |
| 16:53.29 | [TK]D-Fender | Endless list.. niiicce |
| 16:54.02 | brunner | p3nguin: no, if one person can do just asterisk work, I'd be more than happy to hire them. |
| 16:54.20 | brunner | [TK]D-Fender: not an endless list for one project, but an endless list of projects |
| 16:54.31 | p3nguin | I'm up for asterisk deployment and configuration. |
| 16:54.34 | brunner | [TK]D-Fender: I own a company that relies on asterisk |
| 16:54.47 | brunner | p3nguin: fantastic! would you msg me? |
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| 17:54.53 | ChannelZ | OK here's one. I want to do something like Pickup() but rather than actually bridge the channels immediately, I want it to just redirect the call to the device I'm running the app from, so I can see who the caller is and decide if I actually want to answer it or not |
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| 17:56.02 | [TK]D-Fender | "see" how? |
| 17:57.09 | WIMPy | Do you want the call to stay on its original destination as well? Or just redirect it to you and either take or bin it? |
| 17:57.43 | ChannelZ | Because the phone would ring. Say mythical extension 100 is ringing. I'm in another room and hear it. I pick up the nearest phone and dial *100 (or whatever I come up with) which runs this mythical "Redirect" app, which causes the phone I'm on to ring as if the original call to 100 had been sent to whatever phone I'm on instead. |
| 17:59.25 | ChannelZ | WIMPy: Either way I suppose; Although I guess it would be nice if the original phone/extension in the dialplan was still "boss" in the sense that if I didn't pick it up on my secondary phone, it would continue on in the dialplan on the original exten and do whatever would normally happen for it (go to voicemail, etc.) rather than redirecting the call entirely |
| 18:00.25 | WIMPy | That would be quite tricky I guess. |
| 18:00.35 | WIMPy | But otherwise you could just do a redirect. |
| 18:00.56 | ChannelZ | How do you mean |
| 18:01.25 | WIMPy | Either 'channel redirect' or AMI 'redirect'. |
| 18:01.53 | ChannelZ | Trying to think if I could cobble it together out of existing apps; Like an AGI script that would fetch the CID from the desired ringing exten, and then place a new dummy call to my 'current' exten setting the CID.. then I could even do a Google Voice type thing where it asks me to press 1 to connect or 2 to ignore |
| 18:02.30 | ChannelZ | hmmm looking at 'channel redirect', thanks.. |
| 18:02.45 | WIMPy | Yes, but I think that would be a little complicated without AMI. |
| 18:03.30 | ChannelZ | I could do AMI |
| 18:04.49 | WIMPy | But if you've got the other extension on a button, you already get the callerID, don't you? |
| 18:05.14 | [TK]D-Fender | ChannelZ: in your *100 exten : use AMI to lookup the ringing channel ID. Lauch a time-delayed Originate and hangup. In the originate set the CID of the person calling 100 |
| 18:05.56 | [TK]D-Fender | on pickup you can call "Pickup()\ |
| 18:16.15 | ChannelZ | Yeah I think something like that should work |
| 18:16.24 | ChannelZ | Thanks |
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| 18:40.17 | SparFux | WIMPy: the dahdi-zaphfc rocks more and more :-) |
| 18:41.12 | SparFux | WIMPy: have I told you about the OSLEC? :-) |
| 18:41.25 | WIMPy | yes |
| 18:41.30 | SparFux | heh. |
| 18:41.41 | SparFux | WIMPy: and the zaphfc -> dahdi_hfcs renaming? |
| 18:42.13 | SparFux | which is btw complete. |
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| 18:46.25 | WIMPy | So you're adding another bit of complexity? |
| 18:47.54 | SparFux | complexity? no. |
| 18:48.13 | WIMPy | another new name? |
| 18:49.16 | SparFux | eh, yeah. zaphfc is a crap name. zap is from zaptel which is obsolete and discontinued. zaphfc is incomplete as hfc is even used in xpp in dahdi and other places. the sane complete name would be dahdi_hfcs or something, but definitely not zaphfc. |
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| 18:50.27 | SparFux | And having a module zaphfc is annoying. with zaphfc -> dahdi_hfcs you do lsmod|grep dahdi to list all your dahdi modules, including dahdi_hfcs. |
| 18:51.13 | WIMPy | Absolutely true. |
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| 18:52.46 | SparFux | WIMPy: and I want to really dig into the zaphfc.c code which is now names dahdi_hfcs.c :-P Therefore I want sane variable names which there are now. |
| 18:54.07 | SparFux | problem is, I have to rename the sf.net project :-( |
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| 20:36.43 | kchehab | i am using elastix 2.0 the recording of calls ( Monotor list) files been deleted every 24 hours ,any one know why |
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| 20:39.52 | [TK]D-Fender | kchehab: That is a question for #elastix . Its their scripts, nothing to do with Asterisk or FreePBX as that also has no stock process for this |
| 20:41.02 | kchehab | [TK]D-Fender you are right i was just testing it, |
| 20:41.51 | kchehab | using asterisk 1.6.2 how to send out of and early media (183 sip message ) after recieveing an invite from a user in order to reduce the PDD |
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| 20:42.53 | SparFux | Project renamed: https://sourceforge.net/projects/dahdi-hfcs/ |
| 20:48.24 | [TK]D-Fender | kchehab: * does not support this AFAIK |
| 20:51.20 | BlackBishop | [Oct 29 23:52:31] DEBUG[8080]: chan_gtalk.c:2221 load_module: Address family is not AF_INET |
| 20:51.23 | BlackBishop | [Oct 29 23:52:31] DEBUG[8080]: chan_gtalk.c:819 gtalk_get_local_ip: Address family is not AF_INET |
| 20:51.36 | BlackBishop | anyone here using chan_gtalk have any idea what is asterisk complaining about here ? |
| 20:51.52 | BlackBishop | externip and bindaddr are set .. ( tried without 'em too .. same thing ) |
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| 21:39.11 | dlynes | Are all the security issues surrounding iax2 mitigated by using keys, instead of passwords? Or are the keys just as easily hackable as say a 20 character random password? |
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| 22:57.04 | justdave | is there something you can set to cause an activity timeout on an IAX2 channel? |
| 22:57.48 | justdave | I seem to have a lot of ghost sessions left in meetme conferences from people who came in on an IAX2 trunk and had their internet drop before they hung up |
| 22:59.17 | justdave | I know SIP has an RTP timeout you can set |
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| 23:25.08 | Freeaqingme | leifmadsen, mind joining #asterisk-dev? |
| 23:26.37 | Freeaqingme | I've got some questions about documentation |
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| 23:36.17 | Naikrovek | why can't you just talk to him here |
| 23:37.04 | Naikrovek | justdave: kick them from the meetme. 'meetme kick ?' to see the options |
| 23:37.04 | Freeaqingme | Naikrovek, I could, it's not a problem at all. But if he didn't join that channel on purpose, I can imagine he doesnt want to discuss dev stuff right now |
| 23:37.30 | Naikrovek | he's not a dev; he works for/at/with digium, but he's not a dev (that I know of) |
| 23:38.02 | Freeaqingme | Naikrovek, IIRC he's been a lot in #asterisk-dev |
| 23:38.09 | Naikrovek | okay but he's not a dev |
| 23:38.20 | Naikrovek | i can go in there, too. doesn't make me a dev, or able to comment on dev issues |
| 23:38.23 | Freeaqingme | it's for development-related stuff. Not for devvers only |
| 23:38.25 | Naikrovek | nevermind i'm cranky and you're making little sense |
| 23:38.31 | Freeaqingme | ;) |
| 23:38.33 | Naikrovek | waddles away |
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