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| 00:14.40 | Mango45 | waves to everyone. |
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| 02:12.24 | carrar | cupcakes? |
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| 03:53.22 | ChannelZ | my milkshakes bring all the boys to the yard |
| 03:53.47 | WIMPy | You shake milk? |
| 03:53.59 | apb1963 | runs into the house and peeks out the window into the yard. |
| 03:54.46 | apb1963 | what da heck was that? |
| 03:55.36 | apb1963 | shudders and contemplates pulling out his eyeballs |
| 03:58.50 | ChannelZ | http://www.youtube.com/watch?v=jLaxmMDrGmo |
| 03:59.29 | WIMPy | Stanton Warriors - Where's the beef? |
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| 04:34.36 | floren | evening all :) |
| 04:34.46 | floren | i have a quick question about voicemail |
| 04:35.08 | floren | to stop the message info, i use envelope=no |
| 04:35.24 | floren | what option is related to skipping the next message directly? |
| 04:36.00 | floren | right now, if i listen to a message, at the end will list the options, i press 7 to delete it and i would like to have the next message automatically played |
| 04:36.26 | floren | instead, it is presenting the voicemail options |
| 04:36.49 | floren | i have to press 3 (i think) to listen to next message |
| 04:37.36 | floren | the flow i'm interested in is: listen to message > 7/9 to delete/save > next message starts playing |
| 04:43.27 | floren | i got it, is nextaftercmd |
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| 04:53.28 | WIMPy | likes people who answer their own questions :-) |
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| 05:14.18 | floren | :) |
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| 05:16.31 | floren | i compiled astrisk with mp3 support |
| 05:17.27 | floren | i'm wondering if is worth the effort to send voicemails as mp3, instead of wav |
| 05:17.57 | floren | however, i want to use google speech to have the audio file transcribed into text on email body |
| 05:18.01 | floren | it will be a neat hack |
| 05:18.31 | WIMPy | I prefer free stuff. |
| 05:18.50 | floren | free as wav? :) |
| 05:19.00 | floren | or you reffer to google speech |
| 05:19.05 | WIMPy | For example. |
| 05:23.13 | ChannelZ | format_mp3 or whatever it is isn't an encoder AFAIK |
| 05:24.28 | WIMPy | It's not only about encoding. |
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| 05:29.09 | WIMPy | Time for some sleep |
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| 05:50.25 | floren | nite WIMPy |
| 05:50.30 | floren | ChannelZ: thanks |
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| 05:50.50 | [TK]D-Fender | floren: * will not save VM's as MP3 |
| 05:50.51 | floren | i'm setting moh to play g722 files by default |
| 05:51.18 | floren | [TK]D-Fender: hello :) |
| 05:51.41 | floren | i set the vm to wav it works great |
| 05:52.14 | floren | i dont care if it takes some space, i get a raw format with great sound quality |
| 05:52.34 | floren | i asked previously the mp3 question just to learn |
| 05:52.52 | floren | which brings me to the netxt question related to moh |
| 05:53.21 | floren | i dont see nothing into ATDG book related to moh playing one specific file only |
| 05:53.25 | floren | is this possible? |
| 05:55.18 | [TK]D-Fender | make a class that points to a folder.... with just 1 file in it |
| 05:56.12 | floren | i see, thanks [TK]D-Fender |
| 05:57.02 | floren | i thought about it, but i was sure it would be some hidden option that allowed me to define a specific sound per class |
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| 06:07.15 | floren | what is the replacement of deprecated SetMusicOnHold? |
| 06:13.44 | [TK]D-Fender | "core show function CHANNEL" |
| 06:13.55 | floren | thanks [TK]D-Fender |
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| 06:28.51 | v0lZy | Hi |
| 06:29.03 | v0lZy | I have a question related to voicemail |
| 06:30.07 | v0lZy | I have configured my setup so that pressing *XX calls the internal voicemail of any registered account. |
| 06:30.29 | v0lZy | That way I can leave reminders to specific people |
| 06:31.28 | v0lZy | Now I want to integrate this with an intercom system... the intercom I am doing through confbridge, but the point is that the intercom doesnt join people who are already talking on the phone. I want to leave the same message I announce to the intercom in these peoples voicemail boxes. |
| 06:32.20 | v0lZy | Now I suppose I could originate *XX, but the act of doing so triggers the prompt message I otherwise have configured for voicemail messages. |
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| 06:32.55 | v0lZy | Short of deleting the prompt message from each voicemail box, does anyone have an idea how to 'skip' over it? |
| 06:36.31 | v0lZy | I know that sending # skips the message, but how can i send a keypress from the dialplan? I was thinking of creating an alternative context for voicemail where dialing *XX# would pass the # to the voicemail system |
| 06:40.56 | v0lZy | ah, the D flag to dial... |
| 06:45.24 | floren | ok, another question. Set(CHANNEL()) should be before or after Dial() |
| 06:45.42 | floren | i see it logically as before |
| 06:46.26 | apb1963 | v0lZy: This may help. Or not. :) http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail |
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| 06:52.12 | apb1963 | v0lZy: If it were me, I'd check my condition (people on the phone, whatever) and then run VoiceMail([flags]boxnumber[@context][&boxnumber2[@context]][&boxnumber3]) in particular, VoiceMail(s1234) seems to be what you're looking for or perhaps even VoiceMail(s${EXTEN}) |
| 06:52.38 | apb1963 | but I'm still pretty noobish so, caveat emptor! |
| 06:54.31 | v0lZy | the s silences the prompt? |
| 06:54.58 | v0lZy | thanks apb1963 |
| 06:55.02 | v0lZy | that seems to be what i want |
| 06:55.03 | v0lZy | thank you |
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| 06:56.40 | apb1963 | skips it |
| 06:56.42 | apb1963 | same thing |
| 06:59.41 | v0lZy | -- Executing [SkipGreetingAndRecordVoiceMessage@AppVoiceMail:5] VoiceMail("SIP/16-00000484", "s44@default") in new stack |
| 06:59.41 | v0lZy | <PROTECTED> |
| 07:00.15 | v0lZy | because there is no s44... |
| 07:00.18 | v0lZy | just 44... |
| 07:00.47 | v0lZy | ah |
| 07:00.53 | v0lZy | as of 1.6 flags go at the end, ok |
| 07:00.54 | v0lZy | . |
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| 07:22.43 | ChannelZ | "as of 2008..." |
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| 09:06.32 | v0lZy | Any ideas, can i originate a call from voicemail? |
| 09:07.10 | v0lZy | as in, using intercom, those people that are currently on the phone and cant be interrupted should get the announcement in their voicemail |
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| 09:45.41 | ThothCastel | after rebooting asterisk ubuntu server machine |
| 09:46.13 | ThothCastel | the asterisk is running but when a sip client tries to register, it doesn't get registerd as it used to before the reboot |
| 09:46.37 | ThothCastel | why? is there any other services that shoud be executed before trying to registyer? |
| 09:49.57 | ThothCastel | help pls |
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| 09:54.08 | Gugge | ThothCastel: restart apache :) |
| 09:54.10 | Gugge | asterisk |
| 09:54.27 | Gugge | i bet asterisk started before the network |
| 09:56.32 | ThothCastel | Gugge: apache? I don't have apache running |
| 09:58.39 | Gugge | i guess that was why i wrote "asterisk" 2 seconds later? |
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| 10:08.56 | ThothCastel | Gugge: thanks but I have already tried that too... weird as it was working fine yesterday |
| 10:09.28 | Gugge | then you turn on sip debugging, and check what happens |
| 10:16.53 | ThothCastel | how to? |
| 10:17.30 | ThothCastel | gugge: I tried asterisk -vvvvvr |
| 10:17.47 | Faustov | sip set debug on |
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| 10:20.06 | ThothCastel | Faustov: debugging enabled. Now when I click on register on the softphone, then nothing happens and phone is not registered |
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| 10:49.43 | ThothCastel | sip clients aren't registering on my asterisk server, it was working just fine yesterday |
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| 12:06.22 | puzzled | hi |
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| 13:33.12 | Greenlight | Would I be right in thinking that using sox to append two wavs together, is a fairly light operation ? |
| 13:35.15 | WIMPy | Depends on the contents. If you mix heavy metall it isn't light. |
| 13:35.28 | Greenlight | ^^ |
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| 13:35.58 | WIMPy | YOu should try easy listening. |
| 13:37.06 | Greenlight | I've agreed to implent functionality whereby users can "pause" recording of one side of the conversation. I had imagined it to be easier. Seems I need to stop MixMonitor, then start a Monitor and them append the half from monitor to the orignal file. |
| 13:37.56 | Greenlight | For some reason I'd initially thought MixMonitor took an argument to just record one side ;/ |
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| 13:43.40 | mjordan | Greenlight: it does in Asterisk 11 |
| 13:43.41 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_MixMonitor |
| 13:44.28 | Greenlight | Ahh wonderful! Thought I'd seen it somewhere, I must have googled the old docs earleir! Thanks! |
| 13:45.49 | Greenlight | Yea, wow makes things so much more straightforward |
| 13:46.30 | mjordan | good! |
| 13:47.46 | file | throws a duck at mjordan |
| 13:47.56 | mjordan | ducks the duck |
| 13:51.54 | jmetro | the first step to converting any client is to remove their old phone system, that way they cant call with problems. |
| 13:54.14 | Greenlight | Hmm quick on with those arguments to MixMonitor - I read it as it will record the normal file *and* a second file with the receive audio only -- is that correct or have I misunderstood ? |
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| 13:56.49 | mjordan | that appears to be the case |
| 13:57.56 | Greenlight | Ideally I just want the one file. I'll play around with it and see... perhaps if I omit the second file name. Worst case I can delete the second file after |
| 13:57.59 | Greenlight | Thanks again |
| 13:58.25 | mjordan | no problem - good luck :-) |
| 13:58.44 | WIMPy | Or jusr record to /dev/null? |
| 13:58.49 | WIMPy | just |
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| 14:02.56 | Greenlight | oo yea that works |
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| 14:53.27 | pensmit | What is the difference between asterisk version 1.8 and 11? |
| 14:53.35 | pensmit | Why are there two? |
| 14:53.48 | _Corey_ | pensmit: 9.20 |
| 14:54.04 | pensmit | Thanks |
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| 14:54.11 | pensmit | That's all I needed |
| 14:54.13 | pensmit | :) |
| 14:54.22 | WIMPy | Two? There are a lot more versions. |
| 14:54.34 | _Corey_ | pensmit: http://www.asterisk.org/downloads/asterisk/all-asterisk-versions |
| 14:54.58 | _Corey_ | that should get you started |
| 14:55.00 | _Corey_ | :) |
| 14:55.22 | igcewieling | pensmit: Major changes between versions is documented in the UPGRADE*.txt files. |
| 14:55.22 | pensmit | I'm reading that now. |
| 14:55.45 | pensmit | If you were to start a new server, would you go 11 or 1.8? |
| 14:55.53 | pensmit | thanks igcew |
| 14:56.05 | WIMPy | 11 off course. |
| 14:56.08 | _Corey_ | 11 is fine for production |
| 14:56.20 | WIMPy | Why would you want to miss out on features and bug fixes? |
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| 14:56.37 | jmetro | 11 |
| 14:58.00 | mjordan | FYI: the reason why there are two LTS releases overlapping is to give people plenty of time to move from one LTS to the next. |
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| 15:05.30 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 15:17.22 | igcewieling | mjordan: and people STILL refuse to upgrade. |
| 15:36.03 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 15:37.26 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 15:38.46 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 15:57.14 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 15:57.20 | Ice_Strike | I've installed Asterisk on the new server |
| 15:57.22 | Ice_Strike | I am getting |
| 15:57.23 | Ice_Strike | asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory |
| 16:00.31 | jmetro | did you mess up? |
| 16:00.35 | Katty | have you tried taco salad? |
| 16:01.35 | jmetro | I have, i dont like olives thoug. |
| 16:01.55 | Katty | yeah, not a big olive fan either :< |
| 16:02.14 | Katty | we had stuff in the garden to use, tho. |
| 16:02.21 | Katty | green pepper, jalapeno, onion, tomato |
| 16:02.23 | Katty | lettuces |
| 16:03.08 | igcewieling | Ice_Strike: run ldconfig -vvv then start Asterisk again |
| 16:03.29 | Katty | hi ice weasel. |
| 16:04.09 | igcewieling | Katty: hey squirrel grrl! |
| 16:04.27 | Ice_Strike | igcewieling Ah much better, thanks. |
| 16:04.39 | igcewieling | Ice_Strike: file a report in Jira |
| 16:05.37 | mjordan | Ice_Strike: what distro? |
| 16:05.56 | igcewieling | mjordan: CentOS 6.x at least |
| 16:05.59 | jmetro | Katty: as logn as its got cheese, beans, tomatoe, sour cream, and maybe some meats, im okay with it. |
| 16:06.41 | Katty | jmetro: you ever tried stirring sriracha into sour cream? |
| 16:07.11 | jmetro | No, i've never wanted to spend 8$ on hot sauce |
| 16:07.16 | mjordan | k. I thought we had an issue in Jira for that. The build system should automatically update the dynamic linker run-time bindings itself. |
| 16:07.30 | Qwell | $8? Sriracha is like 47 cents for a 17 gallon bottle. |
| 16:08.11 | jmetro | Qwell: not around me apparently. Its literally 8$ for the half-liter here |
| 16:12.45 | Katty | jmetro: whatttt. i can buy a 1 liter bottle here for 3 bucks |
| 16:12.58 | Katty | jmetro: where are you at?! |
| 16:13.00 | igcewieling | sounds like jmetro has an idea for an import/export business 8-) |
| 16:13.21 | Katty | wonders if the postal server will ship sriracha |
| 16:13.41 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
| 16:14.00 | WIMPy | Sounds like that 80s story about packet switched beer. |
| 16:14.45 | jmetro | Katty: I'm just west of Chicago, i havent been able to find a store for cheap sriracha around me. |
| 16:14.52 | floren | hi all :) |
| 16:15.14 | Katty | jmetro: you poor thing. |
| 16:15.21 | Katty | jmetro: if i'm ever out that way, i'll bring you some |
| 16:15.40 | Katty | jmetro: also, you might check the import stores if you have any. like in china town |
| 16:15.51 | jmetro | Trust me, i love my hot sauces, even the overly hot ones |
| 16:16.09 | jmetro | There's a guy that makes hotsauce locally around here, and his is pretty good [is now drooling] |
| 16:16.17 | Qwell | jmetro: Go to Walmart. $2.64. |
| 16:17.00 | Katty | Qwell: did you see the photo of derik holding the sriracha chili sauce on fb? |
| 16:17.06 | Katty | Qwell: the Giant Tub of it? ^____^ |
| 16:25.00 | Kobaz | anyone use a cisco spa 122? |
| 16:25.44 | apb1963 | No but I have peanut butter |
| 16:26.51 | Kobaz | the ata is registered and can make outgoing calls |
| 16:27.03 | Kobaz | but you call the box and it returns 486 busy here |
| 16:28.37 | coppice | jmetro: what is an overly hot sauce? |
| 16:31.31 | *** join/#asterisk navaismo (~navaismo@189.191.233.140) |
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| 16:36.27 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-58-47-18.lsanca.fios.verizon.net) |
| 16:39.55 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 16:42.45 | *** join/#asterisk gtTuna (~Tuna@d149-67-253-75.try.wideopenwest.com) |
| 16:43.04 | gtTuna | when/where does /SIP/Registry/xxxxx get updated? |
| 16:43.25 | gtTuna | i moved a phone to a different network, but Asterisk is still trying to send replies to my old network |
| 16:45.17 | *** join/#asterisk psanders (~psanders@ks-184-2-115-125.dhcp.embarqhsd.net) |
| 16:46.57 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 16:47.59 | igcewieling | gtTuna: when the device registers, I assume. |
| 16:50.31 | sawgood | igcewieling: hey |
| 16:53.08 | [404] | does asterisk round up or down for half seconds for billsec |
| 16:53.32 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 16:54.02 | *** join/#asterisk Cuzner (~ccuzner@198.41.29.45) |
| 16:54.08 | pabelanger | [404]: check the code |
| 16:54.29 | psanders | Hey, check out my framework for development with asterisk & java http://astivetoolkit.com |
| 16:54.50 | jmetro | coppice: Dave's insanity sauce is the one i use comfortably |
| 16:55.24 | jmetro | coppice: Otherwise i have concentrated capsaicin ala "Da Bomb - The Final Answer" |
| 16:55.30 | psanders | I will be here for a little bit in case of questions =} #astivetoolkit |
| 16:55.59 | [404] | pabelanger: thanks for that great answer....truly helpful, pat yourself on the back |
| 16:56.30 | Qwell | psanders: it would help if the site was actually up |
| 16:56.53 | jmetro | ^ |
| 16:57.13 | leifmadsen | psanders: fix your link |
| 16:57.15 | coppice | jmetro: so, you have a notion of hot. its this "overly" concept I can't relate to |
| 16:57.18 | pabelanger | [404]: how so? You can actually check the code and see what is happening. |
| 16:57.23 | pabelanger | vs waiting for something to answer |
| 16:58.13 | psanders | Qwell: true, thats the wrong link, here is the right one-> http://astivetoolkit.org |
| 16:58.14 | [404] | and coding something else |
| 16:58.59 | *** join/#asterisk Osage_ (~Osage@97-80-32-7.dhcp.slid.la.charter.com) |
| 17:00.15 | [404] | a possible but probably unlikely 184-1(you) could benefit from the answer |
| 17:04.45 | jmetro | psanders: why do i have to download it? i want it web based so i dont have to download it. |
| 17:05.58 | pabelanger | [404]: Then if you don't want to look, read: |
| 17:06.06 | pabelanger | *CLI> core show function CDR |
| 17:06.18 | pabelanger | use the f flag and set billsec to a floating point |
| 17:10.40 | psanders | jmetro: For now it works just like any other framework or library(including web frameworks) |
| 17:11.08 | jmetro | psanders: whats a framework? I just want to open it and start making asterisk apps |
| 17:11.55 | pabelanger | psanders: why java? |
| 17:12.10 | pabelanger | not trolling, serious curious |
| 17:12.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.96) |
| 17:15.05 | psanders | jmetro: The vision of this software is so people can develop software using external resources(databases, API's). If kind of hard(no impossible) if you have as a cloud service. |
| 17:18.38 | jmetro | psanders: pabelanger has a really good question - why java? I would never install java on a production server |
| 17:19.07 | psanders | pabelanger: I believe there is many good tools in other programming language(especially in Ruby) to do same task. |
| 17:19.15 | *** join/#asterisk ybeddyj (~ybeddyj@66.54.118.124) |
| 17:19.37 | jmetro | psanders: does your website need an english translator? |
| 17:19.37 | psanders | jmetro: Hopefully thats only your case :) |
| 17:20.06 | psanders | jmetro: Maybe, you tell me. |
| 17:20.42 | psanders | jmetro: I'm hispanic so... |
| 17:21.26 | jmetro | psanders: If you know anyone, it could benefit from it. It reminds me of a lot of the SNOM webpages i've been to. Its not awful, but my inner reading ODC gets me. |
| 17:21.31 | jmetro | OCD |
| 17:23.17 | psanders | jmetro: As a right now I don't have much help. I'm doing everything myself, including web design and translations. |
| 17:23.29 | psanders | jmetro: I hope that change later. |
| 17:23.35 | jmetro | The web design is spot on, dont worry about that. |
| 17:24.23 | psanders | jmetro: I appreciate the feedback. |
| 17:25.06 | jmetro | psanders: your sourceforge has helloworld.jar but not helloword.tar.gz |
| 17:25.34 | psanders | Nice talking with you guys, see you later. |
| 17:25.44 | jmetro | i wanted to see the source =O |
| 17:25.53 | psanders | jmetro: I will take a look of that. |
| 17:26.17 | psanders | jmetro: https://github.com/psanders/astivetoolkit go to examples... |
| 17:27.25 | psanders | I got to go. Hit me with any questions at dev@astivetoolkit.org |
| 17:28.28 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 17:31.50 | jmetro | oh i see |
| 17:31.59 | jmetro | astive toolkit is basically java implementation of dialplan |
| 17:32.43 | igcewieling | jmetro: What could possibly go wrong?!? |
| 17:34.52 | jmetro | i truly dont allow java on production though \_o.o_/ |
| 17:35.38 | igcewieling | Others have tried similar stuff. PHP and Perl as the ones I know if. They did not appear to have succeeded. |
| 17:37.56 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.96) |
| 17:48.54 | *** join/#asterisk peetaur2 (~peter@x2f16c82.dyn.telefonica.de) |
| 17:57.21 | *** join/#asterisk ooggetuigen (~m@120-163-ftth.onsbrabantnet.nl) |
| 17:57.28 | ooggetuigen | hi all |
| 17:57.55 | ooggetuigen | can someone tell me how i can forward the ringgroup to a mobile phone when no one picks up in the ring group? |
| 17:58.26 | WIMPy | Could it be you wanted to as in #freepbx? |
| 17:58.41 | ooggetuigen | i use freepbx ;) |
| 17:58.48 | ooggetuigen | should i ask it there? |
| 17:58.55 | WIMPy | Then ask there. |
| 17:58.59 | ooggetuigen | thx |
| 17:59.08 | jmetro | ooggetuigen: i could tell you how to do it in dialplan |
| 17:59.26 | jmetro | ooggetuigen: but freepbx mangles your extensions and makes that not worth your time. |
| 17:59.40 | ooggetuigen | hehe ok |
| 18:00.00 | ooggetuigen | i use it cause when i'm dead somone else could manage the asterisk |
| 18:00.01 | ooggetuigen | ;) |
| 18:00.57 | WIMPy | Not someone else, something else. |
| 18:01.00 | igcewieling | ooggetuigen: Most of us don't care why, just the fact you are using FreePBX and.... |
| 18:01.02 | igcewieling | ~freepbx |
| 18:01.02 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:02.48 | WIMPy | ... or just the most common bad word in #asterisk |
| 18:02.57 | ooggetuigen | lol :) |
| 18:08.27 | *** join/#asterisk imox (~imox@91-64-148-46-dynip.superkabel.de) |
| 18:15.14 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 18:19.00 | *** join/#asterisk colocate (~colocate@unaffiliated/colocate) |
| 18:21.14 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 18:22.20 | igcewieling | Customer: We are getting an error 21! Me: sounds like their BTN ported away. Sales: No, the number did not port away. Carrier: the number ported away on July 23rd. |
| 18:22.55 | igcewieling | If sales sent us correct information once a in while engineering might start believing what they say. |
| 18:23.38 | jmetro | <PROTECTED> |
| 18:23.52 | jmetro | im imagining those 4 people standing in a room together and sales just looking at everyone awkwardly |
| 18:24.55 | colocate | That depends on the sales staff. I've known several that wouldn't even feel awkward about it. |
| 18:25.33 | igcewieling | jmetro: this was all over e-mail. customers do NOT normally get to talk to me. |
| 18:25.52 | WIMPy | They're probably like: It's a technical thing. Not our business. |
| 18:26.02 | igcewieling | WIMPy: EXACTLY! |
| 18:26.28 | jmetro | igcewieling: as the tech, thats when i start playing "break my stride" and put my sunglasses on |
| 18:26.56 | igcewieling | It is kind of a catch-22. Most carriers want you to port your BTN first, but if the BTN ports out, the carriers will reject calls from any TNs with that BTN |
| 18:27.50 | igcewieling | The worst is when the customer ports away ALL their numbers, except for the Toll Frees. Since the TF no longer has a valid billing TN, calls fail. |
| 18:30.05 | jmetro | 188.138.111.5 ddosing with bad registrations. Wut |
| 18:33.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.96) |
| 18:34.23 | *** join/#asterisk asteriskmonkey (~Tardis@74-51-38-204.telnetcommunications.com) |
| 18:34.55 | navaismo | Hi seesm like i cant find in the wiki documentation to work with SIP MESSAGE within asterisk, is this supported? How can i by-pass the asterisk and just send peer to peer SIP messages? |
| 18:35.20 | asteriskmonkey | heya im running asterisk 11.4.0, and im getting failed invite and failed authentication errors, but its not showing me the source ip |
| 18:35.23 | asteriskmonkey | is there a setting to change that |
| 18:36.26 | WIMPy | You need to enable the security events. The log is known to be useless in that point. |
| 18:37.34 | igcewieling | jmetro: use fail2ban or allow the guest calls or registations and point them to a torture context |
| 18:38.31 | jmetro | nah, that costs minutes. |
| 18:38.45 | igcewieling | jmetro: no, it does not |
| 18:38.56 | igcewieling | they are not using your carrier, they come direct. |
| 18:39.01 | jmetro | i was about to say "wait a sec, no it doesnt" |
| 18:40.47 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 18:41.14 | asteriskmonkey | thanks |
| 18:41.19 | igcewieling | they will keep pounding your server until you ban them with iptables or start "accepting" the calls. |
| 18:42.14 | WIMPy | Where "acdept" doesn't have to involce answering the call. |
| 18:43.00 | WIMPy | Guess who made that mistake. |
| 18:44.14 | jmetro | already iptables dropped but thought i'd broadcast the IP for all |
| 18:44.19 | igcewieling | Anyone know what causes this or if it is important "dahdi: Detected time shift." |
| 18:45.27 | jmetro | i'm wagering on it has to do with timing in between servers, or time-slip via NTP |
| 18:46.15 | igcewieling | sincle server |
| 18:46.33 | igcewieling | and why does DAHDI care about clock time? |
| 18:47.03 | igcewieling | BTW, they are closing at 3pm |
| 18:47.03 | jmetro | im not sure how the timers work, but i know that time-slip on my domain servers can cause bad things. |
| 18:47.28 | jmetro | dont use dahdi so i'm not sure if it is the same principle |
| 18:48.15 | jmetro | http://lists.digium.com/pipermail/asterisk-users/2010-November/256456.html seems to suggest its system-clock related |
| 18:55.25 | igcewieling | Me: "What is the MAC for the two extensions." Customer: One is connected to a Dell, the other to a Mac. |
| 18:55.30 | *** part/#asterisk colocate (~colocate@unaffiliated/colocate) |
| 18:57.37 | navaismo | nevermind found the solution on Voip-Info.org yeah that site |
| 18:58.11 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:558f:1636:3ce:36f) |
| 19:27.35 | navaismo | hmm can't send or receive sip messages :S |
| 19:30.29 | igcewieling | navaismo: SIMPLE messages, correct? |
| 19:31.50 | WIMPy | Probably not the ones I had to deal with. |
| 19:33.35 | navaismo | igcewieling, not sure about the word "SIMPLE" I'm using MESSAGE function and App MessageSend |
| 19:34.44 | igcewieling | navaismo: SIP does not support the concept of sending text messages, additions to SIP, called "SIMPLE" handle that, which has nothing whatsoever to do with SMS |
| 19:35.10 | navaismo | but this is tricky, the SIP message came from a WEBRTC request using SIPML5, asterisk receive it correctly but when i try to send to destination nothing happens |
| 19:36.11 | igcewieling | eagerly awaits information about "the destination" |
| 19:36.22 | navaismo | so sipml5<IM reach asterisk>asterisk<IM from ast never reach>sipml5 |
| 19:36.50 | igcewieling | IM? Is that some kind of client? |
| 19:38.05 | igcewieling | Anyway, I wish you the best of luck in using these bleeding edge protocols. You might try making your set up somewhat less complicated. Might help with troubleshooting. |
| 19:40.01 | navaismo | Instan message |
| 19:40.38 | igcewieling | navaismo: I always thought of you as a non-idiot. |
| 19:41.06 | igcewieling | IIRC the only text protocol Asterisk supports is XMPP, so I'll assume that is what you are using to talk "IM" |
| 19:41.20 | navaismo | well i have logs on both sides but when asterisk "send" the message the other webrtc client didn't receive it, and the log show no activity |
| 19:41.50 | navaismo | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_MessageSend |
| 19:41.52 | igcewieling | have you tried simply sending a message to your XMPP client without all the webrtc sillyness? |
| 19:42.14 | navaismo | is not an xmpp client, it is a sip client |
| 19:42.57 | igcewieling | Does wireshark see the message going out? |
| 19:43.40 | navaismo | no wireshark log here...yet |
| 19:44.09 | navaismo | oh i noticed that now im an idiot for you hehehe |
| 19:45.10 | *** join/#asterisk ybeddyj (~ybeddyj@66.54.118.124) |
| 19:45.50 | WIMPy | Idiots live happier lives. |
| 19:45.52 | *** join/#asterisk cmendes0101 (~cmendes01@pool-173-58-47-18.lsanca.fios.verizon.net) |
| 19:46.43 | navaismo | Hmm I guess this is caused by no path between asterisk and webrtc2sip gateway hmmmm |
| 19:47.53 | navaismo | time to ask in other forum |
| 19:51.59 | *** join/#asterisk sarobat (ccd1d181@gateway/web/freenode/ip.204.209.209.129) |
| 20:05.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.54) |
| 20:07.27 | *** join/#asterisk MrPockets (~MevinKitt@unaffiliated/mrpockets) |
| 20:07.29 | MrPockets | ohhi |
| 20:08.49 | MrPockets | So I have this damned AstLinux appliance, and I'm trying to figure out how to forward a phone to a cellphone. Which config are the feature codes likely laid out in? |
| 20:09.05 | Qwell | MrPockets: #astlinux |
| 20:09.49 | jmetro | MrPockets: does astlinux butcher your files or do you know where it is in dialplan? |
| 20:09.50 | MrPockets | Qwell, oh man, that channel should have died years ago |
| 20:10.25 | MrPockets | jmetro, I don't *believe* it hacks the config files up as bas as some of like, the whitebox firewalls butchure IPtables routes |
| 20:10.42 | MrPockets | but admittingly, I've played iwth FreePBX a bit, but beyond that I'm kind of an ast noob |
| 20:11.01 | jmetro | In my configs, i have each company separated out into their own separate company folder with files, and "features" go into features.conf for that company |
| 20:11.18 | [TK]D-Fender | MrPockets: Telephony is not "faith-based" |
| 20:11.18 | jmetro | that maybe sets a status flag to "forward" |
| 20:11.18 | MrPockets | so the dial plan should define the features config? |
| 20:11.47 | jmetro | its all part of dialplan, so somewhere there is an exten => [buttonpress],1,Set(status to forward) or something similar. |
| 20:11.50 | MrPockets | [TK]D-Fender, good, because I don't have much faith in this system at all. |
| 20:11.56 | MrPockets | got it. |
| 20:11.56 | MrPockets | Thanks |
| 20:13.31 | jmetro | most likely in a features.conf or macros |
| 20:13.37 | jmetro | if they had any sense of organization |
| 20:14.05 | MrPockets | find / -name features.conf -print |
| 20:14.13 | MrPockets | 'erm, dirsregard. |
| 20:14.17 | [TK]D-Fender | "forwarding" would not be features.conf |
| 20:14.23 | [TK]D-Fender | that would be DIALPLAN : extensions.conf |
| 20:14.55 | jmetro | negative, the routing that checks the status and therefore forwards it should be extensions, the feature button you press to turn your phones to forward would be features. |
| 20:15.00 | [TK]D-Fender | . |
| 20:17.34 | *** join/#asterisk jsarrel (~jsarrel@24-158-61-198.static.hckr.nc.charter.com) |
| 20:21.34 | *** join/#asterisk sawgood1 (~sawgood@unaffiliated/sawgood) |
| 20:23.55 | MrPockets | Hm. |
| 20:25.17 | leifmadsen | anyone use video end points a fair amount? I'm doing some testing and getting one-way in a certain scenario. If asterisk is transcoding audio between legs, if the far end enables video, it kills their audio stream, but the near end is able to enable video without issue |
| 20:25.34 | leifmadsen | when no transcoding is involved it's fine, and the audio works until the video is enabled |
| 20:25.44 | *** join/#asterisk Tim_Toady (~fuzzy@snf-33276.vm.okeanos.grnet.gr) |
| 20:29.11 | MrPockets | Is it possible Followme is used instead of basic "forwarding" ? |
| 20:29.53 | *** part/#asterisk himura_x_ (~himurax@pc-34-179-160-190.cm.vtr.net) |
| 20:31.44 | igcewieling | I just use a Dial |
| 20:33.15 | MrPockets | dumb quesiton |
| 20:33.27 | igcewieling | MrPockets: |
| 20:33.28 | igcewieling | primary-care*CLI> core show application followme |
| 20:33.28 | igcewieling | Your application(s) is (are) not registered |
| 20:33.28 | igcewieling | Command 'core show application followme' failed. |
| 20:33.30 | MrPockets | but is /etc/init.d/asterisk the ony thing i need to reset to make changes in a config file take effect? |
| 20:33.36 | igcewieling | What application were you asking about again? |
| 20:34.15 | igcewieling | MrPockets: what about "core reload" |
| 20:34.20 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 20:35.29 | MrPockets | thanks |
| 20:38.18 | *** join/#asterisk mokmeister (~mokmeiste@86-40-247-195-dynamic.b-ras2.lmk.limerick.eircom.net) |
| 20:45.10 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 20:50.49 | jmetro | "core reload on my birthday" reloads on your birthday and gives you a cake |
| 20:51.19 | jmetro | [to make up for all teh calls it just dropped] |
| 20:52.43 | Qwell | reload doesn't drop calls |
| 20:53.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:00.42 | *** part/#asterisk aelliott22 (~aelliott@198.41.29.45) |
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| 21:21.01 | jmetro | hm |
| 21:21.16 | jmetro | can you set parking timeout from within features.conf |
| 21:21.18 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 21:22.56 | teloniusz | hi guys. Is it possible to setup SIP user so that more than one client can connect to it, client does not need to register, but has to authenticate? |
| 21:23.40 | igcewieling | teloniusz: yes. registration has nothing to do with allowing the client to make a call. |
| 21:24.02 | jmetro | any calls going to it would go to whatever registered last from what i remember |
| 21:24.39 | igcewieling | we have about 60 "clients" which use the same auth info against the same peer |
| 21:24.43 | WIMPy | If its a account you can register to. |
| 21:24.48 | igcewieling | jmetro: yes, but he said he is not registering |
| 21:25.56 | teloniusz | igcewieling: from different hosts? If i don't specify a host in my users.conf, I get "No matching peer" and instead of returning 407, asterisk tries to execute default context... |
| 21:26.38 | WIMPy | Set the host to dynamic, even if you don't register. |
| 21:27.04 | WIMPy | wonders if you can have an aacount without a host line. |
| 21:27.09 | igcewieling | teloniusz: you need a permit and deny settings for the peer |
| 21:27.26 | [TK]D-Fender | teloniusz: First, nobody uses "users.conf", next... we'd have to see what you created and what the call looks like coming in. |
| 21:27.37 | igcewieling | WIMPy: I believe host= is required to be set, but it isn't used |
| 21:28.20 | [TK]D-Fender | teloniusz: Next.. the only way to permit calls from MULTIPLE sources.. it to be using type=user |
| 21:28.24 | igcewieling | all which was required for us was to make sure the incoming usename matched and the address the request came from was in the allow= |
| 21:28.28 | *** join/#asterisk jkroon (~jkroon@196.37.70.13) |
| 21:29.04 | jkroon | hi guys, does anybody know of any issues in chan_sip that's been addressed from 11.3.0 to 11.5.0 specifically to improve handling of the incoming udp port 5060 frames? |
| 21:29.21 | file | nope |
| 21:29.22 | jkroon | according to netstat -nuap asterisk simply can't keep up with the incoming stream of SIP frames |
| 21:30.02 | igcewieling | [TK]D-Fender: this accepts calls from any permit'd network http://pastebin.com/5F6FLP9Y |
| 21:30.09 | jkroon | I (even late at night when there isn't supposed to be much load) see Recv-Q values of >200k at times, but usually around 40-80k (periodically it's drops down to zero) |
| 21:30.12 | teloniusz | wait, I'm preparing pastie... |
| 21:30.31 | WIMPy | Ouch |
| 21:30.32 | igcewieling | jkroon: what does lspci show for the network chip? |
| 21:30.33 | [TK]D-Fender | igcewieling: Friend might do as well as it's both |
| 21:30.44 | teloniusz | [TK]D-Fender: I have type=user, host=dynamic and specified username and secret. It doesn't seem to help. |
| 21:30.59 | jkroon | igcewieling, netstat reports stuff that's already off the NIC into the kernel, but it's an Intel e1000 either which way |
| 21:30.59 | igcewieling | teloniusz: what part of "you need a permit and deny" did you miss? |
| 21:31.13 | jkroon | it's *only* on port 5060 that I'm seeing this. |
| 21:31.32 | [TK]D-Fender | igcewieling: no reason to require permit/deny |
| 21:31.53 | igcewieling | jkroon: e1000 http://blog.krisk.org/2013/02/packets-of-death.html |
| 21:32.03 | jkroon | igcewieling, NIC keeps working |
| 21:32.14 | jkroon | i've read that article before. it's not what's causing the problem. |
| 21:32.15 | igcewieling | jkroon: something is either screwed up on your Asterisk or you run a LOT of calls. |
| 21:32.30 | jkroon | igcewieling, 30k calls/day ... |
| 21:32.37 | teloniusz | igcewieling: all of them ;> I've quite a problem with stereo processing. A moment... |
| 21:32.42 | jkroon | at the moment call concurrency is zero, and I'm still seeing it happen |
| 21:33.02 | WIMPy | Or too chatty clients. |
| 21:33.23 | WIMPy | Too short registration periods? |
| 21:33.29 | file | there is one thread handling SIP UDP traffic, if anything in that process blocks for a long period of time then stuff will back up |
| 21:33.30 | jkroon | 828 sip peers [Monitored: 154 online, 270 offline Unmonitored: 8 online, 396 offline] |
| 21:33.56 | jkroon | minexpiry=900 |
| 21:34.07 | WIMPy | Do you need to qualify them? |
| 21:34.12 | igcewieling | file: such as DNS lookups? |
| 21:34.19 | jkroon | WIMPy, ideally yes |
| 21:34.19 | file | igcewieling, yes. |
| 21:34.26 | jmetro | WIMPy: Too short registers could do it. I once ddosed my own aastras by setting the reg time to 0. |
| 21:34.43 | igcewieling | jkroon: try enabling dnsmgr |
| 21:34.50 | WIMPy | 0 is really short. |
| 21:34.58 | jkroon | igcewieling, already is |
| 21:35.32 | jkroon | hmm, could a single sip trunk trying to do DNS stuff cause this ... yes, from previous experience ... |
| 21:35.47 | file | yes. |
| 21:36.12 | jkroon | ok, none is using DNS, but some idiot has set host=10.0.3.2 which is DEFINITELY not reachable from my systems. |
| 21:36.44 | igcewieling | jkroon: this has been an issue since the beginning of Asterisk. jkroon: asteirsk does a reverse dns lookup on the ip of the system interfaces. |
| 21:36.55 | WIMPy | If you don;t allow it, it should find out rather quickly. |
| 21:36.55 | jkroon | WHAT?! |
| 21:37.05 | jkroon | igcewieling, is there any way to disable that? |
| 21:37.21 | teloniusz | igcewieling: OK, permit/deny didn't help, I'm back at preparing pastie. |
| 21:37.24 | jkroon | i've got clients coming in off 3G, and Vodacom specifically that'll time out (30s+ for a failure) |
| 21:38.11 | jkroon | ok, would it help to implement a dns-wall where I send all reverse lookups that simply returns names like a-b-c-d.fakerev.mydomain.co.za ? |
| 21:40.16 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 21:43.18 | jkroon | igcewieling, looks like the reverse DNS could make sense based on tailing and correlation of watch netstat -nuap and the DNS log, but also not completely, sometimes the buildup happens without a DNS query being received by the dns cache. |
| 21:49.31 | teloniusz | [TK]D-Fender: here: http://pastie.org/8172234 is excerpt of my configuration and logs during a call. |
| 21:52.02 | [TK]D-Fender | [17:22]teloniuszhi guys. Is it possible to setup SIP user so that more than one client can connect to it, client does not need to register, but has to authenticate? |
| 21:52.16 | [TK]D-Fender | teloniusz: You specified a SINGLE IP there... how does that permit multiple? |
| 21:52.50 | teloniusz | [TK]D-Fender: wait, where did I specify single IP? |
| 21:52.55 | [TK]D-Fender | teloniusz: You've also redacted everything to the point we can't verify if you made a typo somewhere, nor do we see the entire call. |
| 21:53.09 | [TK]D-Fender | permit=(client IP)/255.255.255.255 |
| 21:54.01 | teloniusz | [TK]D-Fender: yes, but this is irrelevant, if Asterisk reports "no matching peer", these settings aren't used at all |
| 21:54.28 | teloniusz | [TK]D-Fender: and still, I have a problem even with with single client |
| 21:54.48 | teloniusz | it works in exact the same way without permit/deny |
| 21:55.06 | [TK]D-Fender | From: "(caller number)" <sip:(caller number)@(client IP)>;tag=as04230893 <--- I'm betting this doesn't look anything like the USERNAME..... |
| 21:55.26 | jkroon | igcewieling, it's not the DNS issue... just implemented a reverse-wall and it made no difference. |
| 21:55.26 | [TK]D-Fender | So you should be setting "fromuser=theusername]" on your sender |
| 21:56.30 | teloniusz | [TK]D-Fender: oh, I see... This call is redirected by the client from PRI and callerID seems to be autmatically used as From: - so i should overwrite it with fromuser... |
| 22:01.12 | jkroon | igcewieling, even with qualify=no on all sip peers ... same problem. |
| 22:01.47 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:01.47 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 22:02.42 | WIMPy | How do you get that many packets without qualify? |
| 22:03.29 | jkroon | trying to tcpdump that now ... |
| 22:04.27 | jkroon | Correct auth, but based on stale nonce received from <-- seeing LOADS of that in the logs |
| 22:05.09 | teloniusz | [TK]D-Fender: that was it! Thanks a lot. |
| 22:06.08 | jkroon | on the other hand ... 1200 entries in three days really isn't that much, just a LOT of those in the last few minutes. |
| 22:06.56 | floren | anyone using srtp with a voip provider? i contacted voip.ms and i got them scared when i pronounced "srtp" |
| 22:06.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.190) |
| 22:07.55 | WIMPy | Sure. Only terrorists would encrypt their communication. |
| 22:08.01 | jkroon | :p |
| 22:08.03 | floren | heh WIMPy |
| 22:08.12 | floren | they are canadians :D |
| 22:08.17 | floren | so am i :) |
| 22:08.49 | pabelanger | why would you use a ITSP with srtp |
| 22:08.57 | pabelanger | since you would have to share your keys with them |
| 22:09.06 | jkroon | uhrm, PKI? |
| 22:10.01 | floren | pabelanger: not familiar at this extent with asterisk, thought it is possible to use a voip provider and have encrypted communications |
| 22:10.11 | WIMPy | Not only that. Frome there on it will not only be unencrypted, there will also be various agencied sniffing that (unencrypted) traffic at that point as well. |
| 22:11.18 | jkroon | yea, and in SA it's now law that telcos must be able to re-route calls to "intercept centres" (holds for all IP traffic) |
| 22:11.28 | floren | in other words, all my efforts to get srtp compiled properly into asterisk were a nice waste of time? :D |
| 22:11.48 | pabelanger | well, your encrypted calls are only secure as the networks you trust. Since you need to share your keys with them to route |
| 22:11.49 | WIMPy | They don;t just intercept everything? |
| 22:12.11 | pabelanger | So, basically voip.ms can still listen to your calls |
| 22:12.19 | floren | pabelanger: i see :) |
| 22:12.32 | jkroon | pabelanger, yes :). but even if you set up srtp to them they still can :) |
| 22:12.45 | pabelanger | correct |
| 22:13.06 | jkroon | WIMPy, igcewieling, is there any chance that sip/tcp might be causing issues with blocking still? |
| 22:13.37 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 22:13.54 | floren | thank you guys. my next asterisk endaveor is call parking :) |
| 22:16.28 | jkroon | if i'm eyeballing this tcpdump correctly every time before a stall I get a sequence where there is some SIP/TCP interaction, with the last bit before the stall being an incoming SIP block to asterisk, the ACK goes out, and then there is no more response packets to incoming UDP frames for a short period. |
| 22:21.26 | jkroon | ok, disabled tcp temporarily, issue remains :( |
| 22:26.09 | WIMPy | Any ote traffic at that point? Like DNS? |
| 22:26.43 | ChannelZ | OT: anyone have an ipad? |
| 22:28.53 | jkroon | no, no dns |
| 22:29.36 | *** join/#asterisk kikohnl (~keith@udp277730uds.hawaiiantel.net) |
| 22:30.35 | WIMPy | Any other i/o? |
| 22:30.54 | jkroon | some background disk i/o according to iotop |
| 22:31.37 | WIMPy | New kernel or something? |
| 22:31.50 | jkroon | but since asterisk should not be going to disk (except for monitoring and logging) i don't see how that should be a problem |
| 22:32.06 | jkroon | old enough that some would say they can replace it remotely (3.7.3) |
| 22:32.09 | jkroon | vanilla |
| 22:32.47 | WIMPy | i/o scheduler can do interesting things. |
| 22:33.16 | Kobaz | iotop is quite sexy |
| 22:34.11 | jkroon | WIMPy, cfq at the moment, but it's been that way for quite a while (used to use deadline before cfq) |
| 22:34.19 | jkroon | Kobaz, that it is. |
| 22:34.55 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 22:35.03 | jkroon | WIMPy, the only thing that makes sense for me is that something inside of chan_sip is blocking on some internal lock |
| 22:35.19 | jkroon | iax and dundi processing is continuing as normal even during those times. |
| 22:36.19 | jkroon | so i should rephrase as "only thing that makes sense for me so far is that ..." |
| 22:36.49 | WIMPy | makes sense |
| 22:36.54 | WIMPy | :-) |
| 22:37.59 | jkroon | ok, i've just skimmed the asterisk changelog - seems additional locking was introduced going into 11.4/5 - but this is stil on 11.3 |
| 22:38.15 | jkroon | or perhaps those locks fixes this crap? |
| 22:52.56 | *** join/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
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| 22:57.05 | jkroon | ok, what's the possibility around a patch around handle_incoming that logs the SIP packet if/when it takes longer than X number of milliseconds to process? and would that even help pin-point the issue ... |
| 22:58.27 | jkroon | since the only user is handle_request_do() this sounds possible IMHO. |
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| 23:56.42 | navaismo | 0 sip messages 3 score at the end of the day o/ |